From e01b4f624278d5efe5fb5da585ca371947b16680 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 14 Jun 2018 20:26:42 +0100 Subject: ASoC: dapm: Fix potential DAI widget pointer deref when linking DAIs Sometime a component or topology may configure a DAI widget with no private data leading to a dev_dbg() dereferencne of this data. Fix this to check for non NULL private data and let users know if widget is missing DAI. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 36a39ba30226..8ede773b1db8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4073,6 +4073,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) continue; } + /* let users know there is no DAI to link */ + if (!dai_w->priv) { + dev_dbg(card->dev, "dai widget %s has no DAI\n", + dai_w->name); + continue; + } + dai = dai_w->priv; /* ...find all widgets with the same stream and link them */ -- cgit v1.2.3 From c60b613a7097cff20fdd05e2891ce69542f0d5a3 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 14 Jun 2018 20:50:37 +0100 Subject: ASoC: topology: Give more data to clients via callbacks Give topology clients more access to the topology data by passing index, pcm, link_config and dai_driver to clients. This allows clients to fully instantiate and track topology objects. The SOF driver is the first user of these new APIs and needs them to build component topology driver and FW objects. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-topology.h | 23 ++++++++++++++--------- sound/soc/intel/skylake/skl-pcm.c | 7 ++++--- sound/soc/intel/skylake/skl-topology.c | 5 +++-- sound/soc/intel/skylake/skl-topology.h | 5 +++-- sound/soc/soc-topology.c | 31 ++++++++++++++++++------------- 5 files changed, 42 insertions(+), 29 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index f552c3f56368..e1f265e21ee1 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -30,6 +30,8 @@ struct snd_soc_dapm_context; struct snd_soc_card; struct snd_kcontrol_new; struct snd_soc_dai_link; +struct snd_soc_dai_driver; +struct snd_soc_dai; /* object scan be loaded and unloaded in groups with identfying indexes */ #define SND_SOC_TPLG_INDEX_ALL 0 /* ID that matches all FW objects */ @@ -109,35 +111,38 @@ struct snd_soc_tplg_widget_events { struct snd_soc_tplg_ops { /* external kcontrol init - used for any driver specific init */ - int (*control_load)(struct snd_soc_component *, + int (*control_load)(struct snd_soc_component *, int index, struct snd_kcontrol_new *, struct snd_soc_tplg_ctl_hdr *); int (*control_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* external widget init - used for any driver specific init */ - int (*widget_load)(struct snd_soc_component *, + int (*widget_load)(struct snd_soc_component *, int index, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); - int (*widget_ready)(struct snd_soc_component *, + int (*widget_ready)(struct snd_soc_component *, int index, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); int (*widget_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* FE DAI - used for any driver specific init */ - int (*dai_load)(struct snd_soc_component *, - struct snd_soc_dai_driver *dai_drv); + int (*dai_load)(struct snd_soc_component *, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); + int (*dai_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* DAI link - used for any driver specific init */ - int (*link_load)(struct snd_soc_component *, - struct snd_soc_dai_link *link); + int (*link_load)(struct snd_soc_component *, int index, + struct snd_soc_dai_link *link, + struct snd_soc_tplg_link_config *cfg); int (*link_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* callback to handle vendor bespoke data */ - int (*vendor_load)(struct snd_soc_component *, + int (*vendor_load)(struct snd_soc_component *, int index, struct snd_soc_tplg_hdr *); int (*vendor_unload)(struct snd_soc_component *, struct snd_soc_tplg_hdr *); @@ -146,7 +151,7 @@ struct snd_soc_tplg_ops { void (*complete)(struct snd_soc_component *); /* manifest - optional to inform component of manifest */ - int (*manifest)(struct snd_soc_component *, + int (*manifest)(struct snd_soc_component *, int index, struct snd_soc_tplg_manifest *); /* vendor specific kcontrol handlers available for binding */ diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index afa86b9e4dcf..1f4dd08d36c5 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1017,10 +1017,11 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }; -int skl_dai_load(struct snd_soc_component *cmp, - struct snd_soc_dai_driver *pcm_dai) +int skl_dai_load(struct snd_soc_component *cmp, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { - pcm_dai->ops = &skl_pcm_dai_ops; + dai_drv->ops = &skl_pcm_dai_ops; return 0; } diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index fcdc716754b6..647e52aecdc3 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3024,7 +3024,7 @@ void skl_cleanup_resources(struct skl *skl) * information to the driver about module and pipeline parameters which DSP * FW expects like ids, resource values, formats etc */ -static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, +static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, int index, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { @@ -3131,6 +3131,7 @@ static int skl_init_enum_data(struct device *dev, struct soc_enum *se, } static int skl_tplg_control_load(struct snd_soc_component *cmpnt, + int index, struct snd_kcontrol_new *kctl, struct snd_soc_tplg_ctl_hdr *hdr) { @@ -3619,7 +3620,7 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, return 0; } -static int skl_manifest_load(struct snd_soc_component *cmpnt, +static int skl_manifest_load(struct snd_soc_component *cmpnt, int index, struct snd_soc_tplg_manifest *manifest) { struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 6d7e0569695f..af198ea0379e 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -512,8 +512,9 @@ int skl_pcm_host_dma_prepare(struct device *dev, int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params); -int skl_dai_load(struct snd_soc_component *cmp, - struct snd_soc_dai_driver *pcm_dai); +int skl_dai_load(struct snd_soc_component *cmp, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); void skl_tplg_add_moduleid_in_bind_params(struct skl *skl, struct snd_soc_dapm_widget *w); #endif diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 53f121a50c97..9b33260fd537 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -259,7 +259,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg, int ret = 0; if (tplg->comp && tplg->ops && tplg->ops->vendor_load) - ret = tplg->ops->vendor_load(tplg->comp, hdr); + ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr); else { dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n", hdr->vendor_type); @@ -291,7 +291,8 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_load) - return tplg->ops->widget_load(tplg->comp, w, tplg_w); + return tplg->ops->widget_load(tplg->comp, tplg->index, w, + tplg_w); return 0; } @@ -302,27 +303,30 @@ static int soc_tplg_widget_ready(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_ready) - return tplg->ops->widget_ready(tplg->comp, w, tplg_w); + return tplg->ops->widget_ready(tplg->comp, tplg->index, w, + tplg_w); return 0; } /* pass DAI configurations to component driver for extra initialization */ static int soc_tplg_dai_load(struct soc_tplg *tplg, - struct snd_soc_dai_driver *dai_drv) + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { if (tplg->comp && tplg->ops && tplg->ops->dai_load) - return tplg->ops->dai_load(tplg->comp, dai_drv); + return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv, + pcm, dai); return 0; } /* pass link configurations to component driver for extra initialization */ static int soc_tplg_dai_link_load(struct soc_tplg *tplg, - struct snd_soc_dai_link *link) + struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg) { if (tplg->comp && tplg->ops && tplg->ops->link_load) - return tplg->ops->link_load(tplg->comp, link); + return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg); return 0; } @@ -643,7 +647,8 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr) { if (tplg->comp && tplg->ops && tplg->ops->control_load) - return tplg->ops->control_load(tplg->comp, k, hdr); + return tplg->ops->control_load(tplg->comp, tplg->index, k, + hdr); return 0; } @@ -1702,7 +1707,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, dai_drv->compress_new = snd_soc_new_compress; /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv); + ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); kfree(dai_drv); @@ -1772,7 +1777,7 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, set_link_flags(link, pcm->flag_mask, pcm->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link); + ret = soc_tplg_dai_link_load(tplg, link, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n"); kfree(link); @@ -2080,7 +2085,7 @@ static int soc_tplg_link_config(struct soc_tplg *tplg, set_link_flags(link, cfg->flag_mask, cfg->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link); + ret = soc_tplg_dai_link_load(tplg, link, cfg); if (ret < 0) { dev_err(tplg->dev, "ASoC: physical link loading failed\n"); return ret; @@ -2202,7 +2207,7 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, set_dai_flags(dai_drv, d->flag_mask, d->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv); + ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); return ret; @@ -2311,7 +2316,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, /* pass control to component driver for optional further init */ if (tplg->comp && tplg->ops && tplg->ops->manifest) - return tplg->ops->manifest(tplg->comp, _manifest); + return tplg->ops->manifest(tplg->comp, tplg->index, _manifest); if (!abi_match) /* free the duplicated one */ kfree(_manifest); -- cgit v1.2.3 From 503e79b793fea5de626db73accf8e8994bc4289d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 14 Jun 2018 20:53:59 +0100 Subject: ASoC: topology: Add callback for DAPM route load/unload Add a callback fro clients for notification about DAPM route loading and unloading. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-topology.h | 7 +++++++ sound/soc/soc-topology.c | 13 +++++++++++++ 2 files changed, 20 insertions(+) (limited to 'sound/soc') diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index e1f265e21ee1..401ef2c45d6c 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -32,6 +32,7 @@ struct snd_kcontrol_new; struct snd_soc_dai_link; struct snd_soc_dai_driver; struct snd_soc_dai; +struct snd_soc_dapm_route; /* object scan be loaded and unloaded in groups with identfying indexes */ #define SND_SOC_TPLG_INDEX_ALL 0 /* ID that matches all FW objects */ @@ -116,6 +117,12 @@ struct snd_soc_tplg_ops { int (*control_unload)(struct snd_soc_component *, struct snd_soc_dobj *); + /* DAPM graph route element loading and unloading */ + int (*dapm_route_load)(struct snd_soc_component *, int index, + struct snd_soc_dapm_route *route); + int (*dapm_route_unload)(struct snd_soc_component *, + struct snd_soc_dobj *); + /* external widget init - used for any driver specific init */ int (*widget_load)(struct snd_soc_component *, int index, struct snd_soc_dapm_widget *, diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 9b33260fd537..05d177d689e2 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1105,6 +1105,17 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, return 0; } +/* optionally pass new dynamic kcontrol to component driver. */ +static int soc_tplg_add_route(struct soc_tplg *tplg, + struct snd_soc_dapm_route *route) +{ + if (tplg->comp && tplg->ops && tplg->ops->dapm_route_load) + return tplg->ops->dapm_route_load(tplg->comp, tplg->index, + route); + + return 0; +} + static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { @@ -1153,6 +1164,8 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, else route.control = elem->control; + soc_tplg_add_route(tplg, &route); + /* add route, but keep going if some fail */ snd_soc_dapm_add_routes(dapm, &route, 1); } -- cgit v1.2.3 From 134c875bff58ac988c64a1ea2a337acba1711d4b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:56:51 +0000 Subject: ASoC: fsi: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3bae06dd121f..aa7e902f0c02 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1,16 +1,12 @@ -/* - * Fifo-attached Serial Interface (FSI) support for SH7724 - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto - * - * Based on ssi.c - * Copyright (c) 2007 Manuel Lauss - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Fifo-attached Serial Interface (FSI) support for SH7724 +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Kuninori Morimoto +// +// Based on ssi.c +// Copyright (c) 2007 Manuel Lauss #include #include -- cgit v1.2.3 From cb006e7b1712bb9507a218be7ed811ca8e65fc4d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:57:13 +0000 Subject: ASoC: hac: convert to SPDX identifiers Tidyup incoherence between MODULE_LICENSE and header license, too Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/hac.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 624aaf569fef..c2b496398e6b 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -1,13 +1,11 @@ -/* - * Hitachi Audio Controller (AC97) support for SH7760/SH7780 - * - * Copyright (c) 2007 Manuel Lauss - * licensed under the terms outlined in the file COPYING at the root - * of the linux kernel sources. - * - * dont forget to set IPSEL/OMSEL register bits (in your board code) to - * enable HAC output pins! - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Hitachi Audio Controller (AC97) support for SH7760/SH7780 +// +// Copyright (c) 2007 Manuel Lauss +// +// dont forget to set IPSEL/OMSEL register bits (in your board code) to +// enable HAC output pins! /* BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only * the FIRST can be used since ASoC does not pass any information to the @@ -343,6 +341,6 @@ static struct platform_driver hac_pcm_driver = { module_platform_driver(hac_pcm_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver"); MODULE_AUTHOR("Manuel Lauss "); -- cgit v1.2.3 From 217bc8c898b24fa2098a375a575f6170cfac45a9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:57:47 +0000 Subject: ASoC: ssi: convert to SPDX identifiers Tidyup incoherence between MODULE_LICENSE and header license, too Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/ssi.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 89ed1b107ac5..8125fa3840b6 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -1,14 +1,11 @@ -/* - * Serial Sound Interface (I2S) support for SH7760/SH7780 - * - * Copyright (c) 2007 Manuel Lauss - * - * licensed under the terms outlined in the file COPYING at the root - * of the linux kernel sources. - * - * dont forget to set IPSEL/OMSEL register bits (in your board code) to - * enable SSI output pins! - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Serial Sound Interface (I2S) support for SH7760/SH7780 +// +// Copyright (c) 2007 Manuel Lauss +// +// dont forget to set IPSEL/OMSEL register bits (in your board code) to +// enable SSI output pins! /* * LIMITATIONS: @@ -400,6 +397,6 @@ static struct platform_driver sh4_ssi_driver = { module_platform_driver(sh4_ssi_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver"); MODULE_AUTHOR("Manuel Lauss "); -- cgit v1.2.3 From 4e6fdaf1bd334e7e3ad9d6ef7aef4a433c770952 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:58:07 +0000 Subject: ASoC: siu: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/siu.h | 26 ++++++-------------------- sound/soc/sh/siu_dai.c | 26 ++++++-------------------- sound/soc/sh/siu_pcm.c | 27 +++++++-------------------- 3 files changed, 19 insertions(+), 60 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h index 6088d627c0e4..63a508fdfe78 100644 --- a/sound/soc/sh/siu.h +++ b/sound/soc/sh/siu.h @@ -1,23 +1,9 @@ -/* - * siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski - * Copyright (C) 2006 Carlos Munoz - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// siu.h - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski +// Copyright (C) 2006 Carlos Munoz #ifndef SIU_H #define SIU_H diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index ee2211635e92..f2a386fcd92e 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -1,23 +1,9 @@ -/* - * siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski - * Copyright (C) 2006 Carlos Munoz - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// siu_dai.c - ALSA SoC driver for Renesas SH7343, SH7722 SIU peripheral. +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski +// Copyright (C) 2006 Carlos Munoz #include #include diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index 172909570ed5..e263757e4a69 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -1,23 +1,10 @@ -/* - * siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral. - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski - * Copyright (C) 2006 Carlos Munoz - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// siu_pcm.c - ALSA driver for Renesas SH7343, SH7722 SIU peripheral. +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski +// Copyright (C) 2006 Carlos Munoz + #include #include #include -- cgit v1.2.3 From 1e0edd4deadbbacd3b35179c233efa26624ab2af Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:58:38 +0000 Subject: ASoC: rsnd: convert to SPDX identifiers Tidyup incoherence between MODULE_LICENSE and header license, too Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/Makefile | 1 + sound/soc/sh/rcar/adg.c | 15 ++++++--------- sound/soc/sh/rcar/cmd.c | 17 +++++++---------- sound/soc/sh/rcar/core.c | 24 ++++++++++-------------- sound/soc/sh/rcar/ctu.c | 15 ++++++--------- sound/soc/sh/rcar/dma.c | 17 +++++++---------- sound/soc/sh/rcar/dvc.c | 16 ++++++---------- sound/soc/sh/rcar/gen.c | 16 ++++++---------- sound/soc/sh/rcar/mix.c | 14 +++++--------- sound/soc/sh/rcar/rsnd.h | 17 +++++++---------- sound/soc/sh/rcar/src.c | 16 ++++++---------- sound/soc/sh/rcar/ssi.c | 22 +++++++++------------- sound/soc/sh/rcar/ssiu.c | 15 ++++++--------- 13 files changed, 82 insertions(+), 123 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile index 9c3d5aed99d1..5d1ff8ef26f9 100644 --- a/sound/soc/sh/rcar/Makefile +++ b/sound/soc/sh/rcar/Makefile @@ -1,2 +1,3 @@ +# SPDX-License-Identifier: GPL-2.0 snd-soc-rcar-objs := core.o gen.o dma.o adg.o ssi.o ssiu.o src.o ctu.o mix.o dvc.o cmd.o obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 4672688cac32..3a3064dda57f 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -1,12 +1,9 @@ -/* - * Helper routines for R-Car sound ADG. - * - * Copyright (C) 2013 Kuninori Morimoto - * - * This file is subject to the terms and conditions of the GNU General Public - * License. See the file "COPYING" in the main directory of this archive - * for more details. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Helper routines for R-Car sound ADG. +// +// Copyright (C) 2013 Kuninori Morimoto + #include #include "rsnd.h" diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index 5900fb535a2b..d8043ad33540 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -1,13 +1,10 @@ -/* - * Renesas R-Car CMD support - * - * Copyright (C) 2015 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car CMD support +// +// Copyright (C) 2015 Renesas Solutions Corp. +// Kuninori Morimoto + #include "rsnd.h" struct rsnd_cmd { diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f237002180c0..eac22fef4543 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1,16 +1,12 @@ -/* - * Renesas R-Car SRU/SCU/SSIU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * Based on fsi.c - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SRU/SCU/SSIU/SSI support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto +// +// Based on fsi.c +// Kuninori Morimoto /* * Renesas R-Car sound device structure @@ -1606,7 +1602,7 @@ static struct platform_driver rsnd_driver = { }; module_platform_driver(rsnd_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Renesas R-Car audio driver"); MODULE_AUTHOR("Kuninori Morimoto "); MODULE_ALIAS("platform:rcar-pcm-audio"); diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 83be7d3ae0a8..6a55aa753003 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -1,12 +1,9 @@ -/* - * ctu.c - * - * Copyright (c) 2015 Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ctu.c +// +// Copyright (c) 2015 Kuninori Morimoto + #include "rsnd.h" #define CTU_NAME_SIZE 16 diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index ef82b94d038b..fe63ef8600d0 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -1,13 +1,10 @@ -/* - * Renesas R-Car Audio DMAC support - * - * Copyright (C) 2015 Renesas Electronics Corp. - * Copyright (c) 2015 Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car Audio DMAC support +// +// Copyright (C) 2015 Renesas Electronics Corp. +// Copyright (c) 2015 Kuninori Morimoto + #include #include #include "rsnd.h" diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index ca1780e0b830..2b16e0ce6bc5 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -1,13 +1,9 @@ -/* - * Renesas R-Car DVC support - * - * Copyright (C) 2014 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car DVC support +// +// Copyright (C) 2014 Renesas Solutions Corp. +// Kuninori Morimoto /* * Playback Volume diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 25642e92dae0..0230301fe078 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -1,13 +1,9 @@ -/* - * Renesas R-Car Gen1 SRU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car Gen1 SRU/SSI support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto /* * #define DEBUG diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 1881b2de9126..8e3b57eaa708 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -1,12 +1,8 @@ -/* - * mix.c - * - * Copyright (c) 2015 Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// mix.c +// +// Copyright (c) 2015 Kuninori Morimoto /* * CTUn MIXn diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 6d7280d2d9be..96d93330b1e1 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -1,13 +1,10 @@ -/* - * Renesas R-Car - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto + #ifndef RSND_H #define RSND_H diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 6c72d1a81cf5..beccfbac7581 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -1,13 +1,9 @@ -/* - * Renesas R-Car SRC support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SRC support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto /* * you can enable below define if you don't need diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 6e1166ec24a0..3d9ea100a64f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1,16 +1,12 @@ -/* - * Renesas R-Car SSIU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * Based on fsi.c - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SSIU/SSI support +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto +// +// Based on fsi.c +// Kuninori Morimoto /* * you can enable below define if you don't need diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 47bdba9fc582..016fbf5ac242 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -1,12 +1,9 @@ -/* - * Renesas R-Car SSIU support - * - * Copyright (c) 2015 Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Renesas R-Car SSIU support +// +// Copyright (c) 2015 Kuninori Morimoto + #include "rsnd.h" #define SSIU_NAME "ssiu" -- cgit v1.2.3 From 0026c551bacd9654eaa6ee8e4aa19d52730e74f4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:58:54 +0000 Subject: ASoC: migor: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/migor.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index ecb057ff9fbb..8739c9f60672 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -1,12 +1,8 @@ -/* - * ALSA SoC driver for Migo-R - * - * Copyright (C) 2009-2010 Guennadi Liakhovetski - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC driver for Migo-R +// +// Copyright (C) 2009-2010 Guennadi Liakhovetski #include #include -- cgit v1.2.3 From ddfe227c0cbfb97b37afe123e014b9bc4df217b5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:59:11 +0000 Subject: ASoC: dma-sh7760: convert to SPDX identifiers Tidyup incoherence between MODULE_LICENSE and header license, too Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/dma-sh7760.c | 26 ++++++++++++-------------- 1 file changed, 12 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 2dc3b762fdd9..922fb6aa3ed1 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -1,16 +1,14 @@ -/* - * SH7760 ("camelot") DMABRG audio DMA unit support - * - * Copyright (C) 2007 Manuel Lauss - * licensed under the terms outlined in the file COPYING at the root - * of the linux kernel sources. - * - * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which - * trigger an interrupt when one half of the programmed transfer size - * has been xmitted. - * - * FIXME: little-endian only for now - */ +// SPDX-License-Identifier: GPL-2.0 +// +// SH7760 ("camelot") DMABRG audio DMA unit support +// +// Copyright (C) 2007 Manuel Lauss +// +// The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which +// trigger an interrupt when one half of the programmed transfer size +// has been xmitted. +// +// FIXME: little-endian only for now #include #include @@ -341,6 +339,6 @@ static struct platform_driver sh7760_pcm_driver = { module_platform_driver(sh7760_pcm_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver"); MODULE_AUTHOR("Manuel Lauss "); -- cgit v1.2.3 From 04433977b164c8ed0b65cd023c87c74da0fff2bf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:59:27 +0000 Subject: ASoC: sh7760-ac97: convert to SPDX identifiers Tidyup incoherence between MODULE_LICENSE and header license, too Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/sh7760-ac97.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 4a3568a9bf59..4bb4c13cf860 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -1,10 +1,8 @@ -/* - * Generic AC97 sound support for SH7760 - * - * (c) 2007 Manuel Lauss - * - * Licensed under the GPLv2. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Generic AC97 sound support for SH7760 +// +// (c) 2007 Manuel Lauss #include #include @@ -68,6 +66,6 @@ static void __exit sh7760_ac97_exit(void) module_init(sh7760_ac97_init); module_exit(sh7760_ac97_exit); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Generic SH7760 AC97 sound machine"); MODULE_AUTHOR("Manuel Lauss "); -- cgit v1.2.3 From 7cc90a5cadb1733d95d3c2bc147cbcf7843aa585 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:52:00 +0000 Subject: ASoC: rsnd: has .symmetric_rates if SSIs are sharing WS pin If SSIs are sharing WS pin, it should has .symmetric_rates. This patch sets it. Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 6 ++++++ sound/soc/sh/rcar/ssi.c | 5 +++-- 2 files changed, 9 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index af04d41a4274..6bbdddef426e 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1085,6 +1085,12 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, of_node_put(capture); } + if (rsnd_ssi_is_pin_sharing(io_capture) || + rsnd_ssi_is_pin_sharing(io_playback)) { + /* should have symmetric_rates if pin sharing */ + drv->symmetric_rates = 1; + } + dev_dbg(dev, "%s (%s/%s)\n", rdai->name, rsnd_io_to_mod_ssi(io_playback) ? "play" : " -- ", rsnd_io_to_mod_ssi(io_capture) ? "capture" : " -- "); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 9538f76f8e20..4e605648918b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1055,9 +1055,10 @@ struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id) int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod) { - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + if (!mod) + return 0; - return !!(rsnd_flags_has(ssi, RSND_SSI_CLK_PIN_SHARE)); + return !!(rsnd_flags_has(rsnd_mod_to_ssi(mod), RSND_SSI_CLK_PIN_SHARE)); } static u32 *rsnd_ssi_get_status(struct rsnd_dai_stream *io, -- cgit v1.2.3 From 203cdf51f28820bee7893b4be392847418e6f4ec Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:52:17 +0000 Subject: ASoC: rsnd: SSI parent cares SWSP bit SSICR has SWSP bit (= Serial WS Polarity) which decides WS pin 1st channel polarity (low or hi). This bit shouldn't exchange after running. Current SSI "parent" doesn't care SSICR, just controls clock only. Because of this behavior, if platform uses SSI0 as playback, SSI1 as capture, and if user starts capture -> playback order, SSI0 SSICR::SWSP bit exchanged 0 -> 1 during captureing, and it makes capture noise. This patch cares SSICR on SSI parent, too. Special thanks to Yokoyama-san Reported-by: Hiroyuki Yokoyama Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 32 +++++++++++++++++++++----------- 1 file changed, 21 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4e605648918b..98dd120d830a 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -37,6 +37,7 @@ #define CHNL_4 (1 << 22) /* Channels */ #define CHNL_6 (2 << 22) /* Channels */ #define CHNL_8 (3 << 22) /* Channels */ +#define DWL_MASK (7 << 19) /* Data Word Length mask */ #define DWL_8 (0 << 19) /* Data Word Length */ #define DWL_16 (1 << 19) /* Data Word Length */ #define DWL_18 (2 << 19) /* Data Word Length */ @@ -353,21 +354,18 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, struct rsnd_dai *rdai = rsnd_io_to_rdai(io); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - u32 cr_own; - u32 cr_mode; - u32 wsr; + u32 cr_own = ssi->cr_own; + u32 cr_mode = ssi->cr_mode; + u32 wsr = ssi->wsr; int is_tdm; - if (rsnd_ssi_is_parent(mod, io)) - return; - is_tdm = rsnd_runtime_is_ssi_tdm(io); /* * always use 32bit system word. * see also rsnd_ssi_master_clk_enable() */ - cr_own = FORCE | SWL_32; + cr_own |= FORCE | SWL_32; if (rdai->bit_clk_inv) cr_own |= SCKP; @@ -377,9 +375,18 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_own |= SDTA; if (rdai->sys_delay) cr_own |= DEL; + + /* + * We shouldn't exchange SWSP after running. + * This means, parent needs to care it. + */ + if (rsnd_ssi_is_parent(mod, io)) + goto init_end; + if (rsnd_io_is_play(io)) cr_own |= TRMD; + cr_own &= ~DWL_MASK; switch (snd_pcm_format_width(runtime->format)) { case 16: cr_own |= DWL_16; @@ -406,7 +413,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, wsr |= WS_MODE; cr_own |= CHNL_8; } - +init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; ssi->wsr = wsr; @@ -470,15 +477,18 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, return -EIO; } - if (!rsnd_ssi_is_parent(mod, io)) - ssi->cr_own = 0; - rsnd_ssi_master_clk_stop(mod, io); rsnd_mod_power_off(mod); ssi->usrcnt--; + if (!ssi->usrcnt) { + ssi->cr_own = 0; + ssi->cr_mode = 0; + ssi->wsr = 0; + } + return 0; } -- cgit v1.2.3 From 6e56e5d04191aa20e08430dcb203c081fa247e93 Mon Sep 17 00:00:00 2001 From: Agrawal, Akshu Date: Thu, 7 Jun 2018 14:48:43 +0800 Subject: ASoC: AMD: Add NULL pointer check Fix crash in those platforms whose machine driver does not expose platform_info. For those platforms we rely on default value and select I2SSP channel. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 77203841c535..1458b5048498 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -773,7 +773,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, if (WARN_ON(!rtd)) return -EINVAL; - rtd->i2s_instance = pinfo->i2s_instance; + if (pinfo) + rtd->i2s_instance = pinfo->i2s_instance; if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); -- cgit v1.2.3 From 187e01d0d56d1fd682dfaafb0b45d332abec6387 Mon Sep 17 00:00:00 2001 From: olivier moysan Date: Mon, 11 Jun 2018 17:13:59 +0200 Subject: ASoC: stm32: sai: add iec958 controls support Add support of iec958 controls for STM32 SAI. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/Kconfig | 1 + sound/soc/stm/stm32_sai_sub.c | 139 ++++++++++++++++++++++++++++++++++++++---- 2 files changed, 128 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig index 48f9ddd94016..9b2681397dba 100644 --- a/sound/soc/stm/Kconfig +++ b/sound/soc/stm/Kconfig @@ -6,6 +6,7 @@ config SND_SOC_STM32_SAI depends on SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO + select SND_PCM_IEC958 help Say Y if you want to enable SAI for STM32 diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index cfeb219e1d78..c4f15ea14197 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -96,7 +96,8 @@ * @slot_mask: rx or tx active slots mask. set at init or at runtime * @data_size: PCM data width. corresponds to PCM substream width. * @spdif_frm_cnt: S/PDIF playback frame counter - * @spdif_status_bits: S/PDIF status bits + * @snd_aes_iec958: iec958 data + * @ctrl_lock: control lock */ struct stm32_sai_sub_data { struct platform_device *pdev; @@ -125,7 +126,8 @@ struct stm32_sai_sub_data { int slot_mask; int data_size; unsigned int spdif_frm_cnt; - unsigned char spdif_status_bits[SAI_IEC60958_STATUS_BYTES]; + struct snd_aes_iec958 iec958; + struct mutex ctrl_lock; /* protect resources accessed by controls */ }; enum stm32_sai_fifo_th { @@ -184,10 +186,6 @@ static bool stm32_sai_sub_writeable_reg(struct device *dev, unsigned int reg) } } -static const unsigned char default_status_bits[SAI_IEC60958_STATUS_BYTES] = { - 0, 0, 0, IEC958_AES3_CON_FS_48000, -}; - static const struct regmap_config stm32_sai_sub_regmap_config_f4 = { .reg_bits = 32, .reg_stride = 4, @@ -210,6 +208,49 @@ static const struct regmap_config stm32_sai_sub_regmap_config_h7 = { .fast_io = true, }; +static int snd_pcm_iec958_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + + return 0; +} + +static int snd_pcm_iec958_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol); + + mutex_lock(&sai->ctrl_lock); + memcpy(uctl->value.iec958.status, sai->iec958.status, 4); + mutex_unlock(&sai->ctrl_lock); + + return 0; +} + +static int snd_pcm_iec958_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct stm32_sai_sub_data *sai = snd_kcontrol_chip(kcontrol); + + mutex_lock(&sai->ctrl_lock); + memcpy(sai->iec958.status, uctl->value.iec958.status, 4); + mutex_unlock(&sai->ctrl_lock); + + return 0; +} + +static const struct snd_kcontrol_new iec958_ctls = { + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE), + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = snd_pcm_iec958_info, + .get = snd_pcm_iec958_get, + .put = snd_pcm_iec958_put, +}; + static irqreturn_t stm32_sai_isr(int irq, void *devid) { struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid; @@ -619,6 +660,59 @@ static void stm32_sai_set_frame(struct snd_soc_dai *cpu_dai) } } +static void stm32_sai_init_iec958_status(struct stm32_sai_sub_data *sai) +{ + unsigned char *cs = sai->iec958.status; + + cs[0] = IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_NONE; + cs[1] = IEC958_AES1_CON_GENERAL; + cs[2] = IEC958_AES2_CON_SOURCE_UNSPEC | IEC958_AES2_CON_CHANNEL_UNSPEC; + cs[3] = IEC958_AES3_CON_CLOCK_1000PPM | IEC958_AES3_CON_FS_NOTID; +} + +static void stm32_sai_set_iec958_status(struct stm32_sai_sub_data *sai, + struct snd_pcm_runtime *runtime) +{ + if (!runtime) + return; + + /* Force the sample rate according to runtime rate */ + mutex_lock(&sai->ctrl_lock); + switch (runtime->rate) { + case 22050: + sai->iec958.status[3] = IEC958_AES3_CON_FS_22050; + break; + case 44100: + sai->iec958.status[3] = IEC958_AES3_CON_FS_44100; + break; + case 88200: + sai->iec958.status[3] = IEC958_AES3_CON_FS_88200; + break; + case 176400: + sai->iec958.status[3] = IEC958_AES3_CON_FS_176400; + break; + case 24000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_24000; + break; + case 48000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_48000; + break; + case 96000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_96000; + break; + case 192000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_192000; + break; + case 32000: + sai->iec958.status[3] = IEC958_AES3_CON_FS_32000; + break; + default: + sai->iec958.status[3] = IEC958_AES3_CON_FS_NOTID; + break; + } + mutex_unlock(&sai->ctrl_lock); +} + static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, struct snd_pcm_hw_params *params) { @@ -709,7 +803,11 @@ static int stm32_sai_hw_params(struct snd_pcm_substream *substream, sai->data_size = params_width(params); - if (!STM_SAI_PROTOCOL_IS_SPDIF(sai)) { + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { + /* Rate not already set in runtime structure */ + substream->runtime->rate = params_rate(params); + stm32_sai_set_iec958_status(sai, substream->runtime); + } else { ret = stm32_sai_set_slots(cpu_dai); if (ret < 0) return ret; @@ -789,6 +887,20 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, sai->substream = NULL; } +static int stm32_sai_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *cpu_dai) +{ + struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); + + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { + dev_dbg(&sai->pdev->dev, "%s: register iec controls", __func__); + return snd_ctl_add(rtd->pcm->card, + snd_ctl_new1(&iec958_ctls, sai)); + } + + return 0; +} + static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); @@ -809,6 +921,10 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) else snd_soc_dai_init_dma_data(cpu_dai, NULL, &sai->dma_params); + /* Next settings are not relevant for spdif mode */ + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) + return 0; + cr1_mask = SAI_XCR1_RX_TX; if (STM_SAI_IS_CAPTURE(sai)) cr1 |= SAI_XCR1_RX_TX; @@ -820,10 +936,6 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) sai->synco, sai->synci); } - if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) - memcpy(sai->spdif_status_bits, default_status_bits, - sizeof(default_status_bits)); - cr1_mask |= SAI_XCR1_SYNCEN_MASK; cr1 |= SAI_XCR1_SYNCEN_SET(sai->sync); @@ -861,7 +973,7 @@ static int stm32_sai_pcm_process_spdif(struct snd_pcm_substream *substream, /* Set channel status bit */ byte = frm_cnt >> 3; mask = 1 << (frm_cnt - (byte << 3)); - if (sai->spdif_status_bits[byte] & mask) + if (sai->iec958.status[byte] & mask) *ptr |= 0x04000000; ptr++; @@ -888,6 +1000,7 @@ static const struct snd_pcm_hardware stm32_sai_pcm_hw = { static struct snd_soc_dai_driver stm32_sai_playback_dai[] = { { .probe = stm32_sai_dai_probe, + .pcm_new = stm32_sai_pcm_new, .id = 1, /* avoid call to fmt_single_name() */ .playback = { .channels_min = 1, @@ -998,6 +1111,7 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, dev_err(&pdev->dev, "S/PDIF IEC60958 not supported\n"); return -EINVAL; } + stm32_sai_init_iec958_status(sai); sai->spdif = true; sai->master = true; } @@ -1114,6 +1228,7 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) sai->id = (uintptr_t)of_id->data; sai->pdev = pdev; + mutex_init(&sai->ctrl_lock); platform_set_drvdata(pdev, sai); sai->pdata = dev_get_drvdata(pdev->dev.parent); -- cgit v1.2.3 From a56df73ba5960848f60f609c68770d2638bf1dd5 Mon Sep 17 00:00:00 2001 From: Alexey Khoroshilov Date: Sun, 10 Jun 2018 01:20:54 +0300 Subject: ASoC: rockchip: put device_node on remove snd_rk_mc_probe() gets a couple of device nodes with of_parse_phandle(), but there is no release of them. The patch adds remove handler and proper error handling in the probe. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_rt5645.c | 27 ++++++++++++++++++++++++--- 1 file changed, 24 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 4db4fd56db35..881c32498808 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -181,7 +181,8 @@ static int snd_rk_mc_probe(struct platform_device *pdev) if (!rk_dailink.cpu_of_node) { dev_err(&pdev->dev, "Property 'rockchip,i2s-controller' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_codec_of_node; } rk_dailink.platform_of_node = rk_dailink.cpu_of_node; @@ -190,17 +191,36 @@ static int snd_rk_mc_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "Soc parse card name failed %d\n", ret); - return ret; + goto put_cpu_of_node; } ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "Soc register card failed %d\n", ret); - return ret; + goto put_cpu_of_node; } return ret; + +put_cpu_of_node: + of_node_put(rk_dailink.cpu_of_node); + rk_dailink.cpu_of_node = NULL; +put_codec_of_node: + of_node_put(rk_dailink.codec_of_node); + rk_dailink.codec_of_node = NULL; + + return ret; +} + +static int snd_rk_mc_remove(struct platform_device *pdev) +{ + of_node_put(rk_dailink.cpu_of_node); + rk_dailink.cpu_of_node = NULL; + of_node_put(rk_dailink.codec_of_node); + rk_dailink.codec_of_node = NULL; + + return 0; } static const struct of_device_id rockchip_rt5645_of_match[] = { @@ -212,6 +232,7 @@ MODULE_DEVICE_TABLE(of, rockchip_rt5645_of_match); static struct platform_driver snd_rk_mc_driver = { .probe = snd_rk_mc_probe, + .remove = snd_rk_mc_remove, .driver = { .name = DRV_NAME, .pm = &snd_soc_pm_ops, -- cgit v1.2.3 From 75b31192fe6ad20b42276b20ee3bdf1493216d63 Mon Sep 17 00:00:00 2001 From: Jianqun Xu Date: Fri, 8 Jun 2018 16:31:09 +0800 Subject: ASoC: rockchip: add config for rockchip dmaengine pcm register This patch makes the rockchip i2s pcm configurable by adding rockchip pcm config for devm_snd_dmaengine_pcm_register. Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown --- sound/soc/rockchip/Makefile | 3 ++- sound/soc/rockchip/rockchip_i2s.c | 3 ++- sound/soc/rockchip/rockchip_pcm.c | 45 +++++++++++++++++++++++++++++++++++++++ sound/soc/rockchip/rockchip_pcm.h | 14 ++++++++++++ 4 files changed, 63 insertions(+), 2 deletions(-) create mode 100644 sound/soc/rockchip/rockchip_pcm.c create mode 100644 sound/soc/rockchip/rockchip_pcm.h (limited to 'sound/soc') diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 05b078e7b87f..65e814d46006 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,10 +1,11 @@ # SPDX-License-Identifier: GPL-2.0 # ROCKCHIP Platform Support snd-soc-rockchip-i2s-objs := rockchip_i2s.o +snd-soc-rockchip-pcm-objs := rockchip_pcm.o snd-soc-rockchip-pdm-objs := rockchip_pdm.o snd-soc-rockchip-spdif-objs := rockchip_spdif.o -obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-pcm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 950823d69e9c..60d43d53a8f5 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -22,6 +22,7 @@ #include #include "rockchip_i2s.h" +#include "rockchip_pcm.h" #define DRV_NAME "rockchip-i2s" @@ -674,7 +675,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = rockchip_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); return ret; diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c new file mode 100644 index 000000000000..f77538319221 --- /dev/null +++ b/sound/soc/rockchip/rockchip_pcm.c @@ -0,0 +1,45 @@ +/* + * Copyright (c) 2018 Rockchip Electronics Co. Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include + +#include +#include +#include +#include + +#include "rockchip_pcm.h" + +static const struct snd_pcm_hardware snd_rockchip_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 52, + .buffer_bytes_max = 64 * 1024, + .fifo_size = 32, +}; + +static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = { + .pcm_hardware = &snd_rockchip_hardware, + .prealloc_buffer_size = 32 * 1024, +}; + +int rockchip_pcm_platform_register(struct device *dev) +{ + return devm_snd_dmaengine_pcm_register(dev, &rk_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_COMPAT); +} +EXPORT_SYMBOL_GPL(rockchip_pcm_platform_register); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_pcm.h b/sound/soc/rockchip/rockchip_pcm.h new file mode 100644 index 000000000000..d6c36115c60a --- /dev/null +++ b/sound/soc/rockchip/rockchip_pcm.h @@ -0,0 +1,14 @@ +/* + * Copyright (c) 2018 Rockchip Electronics Co. Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ROCKCHIP_PCM_H +#define _ROCKCHIP_PCM_H + +int rockchip_pcm_platform_register(struct device *dev); + +#endif -- cgit v1.2.3 From 4f29b663c08d369fe320a148179996c94cf7d01b Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Thu, 7 Jun 2018 15:50:48 +0200 Subject: ASoC: rt1305: Use ULL suffixes for 64-bit constants With gcc 4.1.2: sound/soc/codecs/rt1305.c: In function ‘rt1305_calibrate’: sound/soc/codecs/rt1305.c:1069: warning: integer constant is too large for ‘long’ type sound/soc/codecs/rt1305.c:1086: warning: integer constant is too large for ‘long’ type Add the missing "ULL" suffixes to fix this. Fixes: 29bc643ddd7efb74 ("ASoC: rt1305: Add RT1305/RT1306 amplifier driver") Signed-off-by: Geert Uytterhoeven Signed-off-by: Mark Brown --- sound/soc/codecs/rt1305.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index f4c8c45f4010..421b8fb2fa04 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -1066,7 +1066,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305) pr_debug("Left_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl); pr_info("Left channel %d.%dohm\n", (r0ohm/10), (r0ohm%10)); - r0l = 562949953421312; + r0l = 562949953421312ULL; if (rhl != 0) do_div(r0l, rhl); pr_debug("Left_r0 = 0x%llx\n", r0l); @@ -1083,7 +1083,7 @@ static void rt1305_calibrate(struct rt1305_priv *rt1305) pr_debug("Right_rhl = 0x%x rh=0x%x rl=0x%x\n", rhl, rh, rl); pr_info("Right channel %d.%dohm\n", (r0ohm/10), (r0ohm%10)); - r0r = 562949953421312; + r0r = 562949953421312ULL; if (rhl != 0) do_div(r0r, rhl); pr_debug("Right_r0 = 0x%llx\n", r0r); -- cgit v1.2.3 From d5a1826c32fa2ec2b161a89df904c6977f7ec44c Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Fri, 8 Jun 2018 10:19:25 +0200 Subject: ASoC: Intel: bytcr_rt5640: Add quirk for the Chuwi Vi10 tablet Add a quirk for the Chuwi Vi10 tablet, this tablet uses IN1 for the internal mic rather then the default IN3 and it uses JD2 rather then JD1 for its jack-detect switch. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 33065ba294a9..5c4f9ea40f57 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -463,6 +463,22 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* Chuwi Vi10 (CWI505) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"), + DMI_MATCH(DMI_BOARD_NAME, "BYT-PF02"), + DMI_MATCH(DMI_SYS_VENDOR, "ilife"), + DMI_MATCH(DMI_PRODUCT_NAME, "S165"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), -- cgit v1.2.3 From 62c2c9fcac4341d306dda4cf400b77e7e124480a Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Mon, 11 Jun 2018 17:32:12 +0900 Subject: ASoC: simple-card-utils: move hp and mic detect gpios from simple-card This patch moves headphone and microphone jack detection gpios from simple-card driver. It is preparing for using this feature from other drivers. Signed-off-by: Katsuhiro Suzuki Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 15 ++++++++ sound/soc/generic/simple-card-utils.c | 59 ++++++++++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 64 ----------------------------------- 3 files changed, 74 insertions(+), 64 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 7e25afce6566..f82acef3b992 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -12,6 +12,11 @@ #include +#define asoc_simple_card_init_hp(card, sjack, prefix) \ + asoc_simple_card_init_jack(card, sjack, 1, prefix) +#define asoc_simple_card_init_mic(card, sjack, prefix) \ + asoc_simple_card_init_jack(card, sjack, 0, prefix) + struct asoc_simple_dai { const char *name; unsigned int sysclk; @@ -28,6 +33,12 @@ struct asoc_simple_card_data { u32 convert_channels; }; +struct asoc_simple_jack { + struct snd_soc_jack jack; + struct snd_soc_jack_pin pin; + struct snd_soc_jack_gpio gpio; +}; + int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *node, struct device_node *codec, @@ -107,4 +118,8 @@ int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, char *prefix); +int asoc_simple_card_init_jack(struct snd_soc_card *card, + struct asoc_simple_jack *sjack, + int is_hp, char *prefix); + #endif /* __SIMPLE_CARD_UTILS_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 3751a07de6aa..4398c9580929 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -8,9 +8,13 @@ * published by the Free Software Foundation. */ #include +#include +#include #include #include +#include #include +#include #include void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, @@ -419,6 +423,61 @@ int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_widgets); +int asoc_simple_card_init_jack(struct snd_soc_card *card, + struct asoc_simple_jack *sjack, + int is_hp, char *prefix) +{ + struct device *dev = card->dev; + enum of_gpio_flags flags; + char prop[128]; + char *pin_name; + char *gpio_name; + int mask; + int det; + + if (!prefix) + prefix = ""; + + sjack->gpio.gpio = -ENOENT; + + if (is_hp) { + snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); + pin_name = "Headphones"; + gpio_name = "Headphone detection"; + mask = SND_JACK_HEADPHONE; + } else { + snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix); + pin_name = "Mic Jack"; + gpio_name = "Mic detection"; + mask = SND_JACK_MICROPHONE; + } + + det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags); + if (det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + if (gpio_is_valid(det)) { + sjack->pin.pin = pin_name; + sjack->pin.mask = mask; + + sjack->gpio.name = gpio_name; + sjack->gpio.report = mask; + sjack->gpio.gpio = det; + sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); + sjack->gpio.debounce_time = 150; + + snd_soc_card_jack_new(card, pin_name, mask, + &sjack->jack, + &sjack->pin, 1); + + snd_soc_jack_add_gpios(&sjack->jack, 1, + &sjack->gpio); + } + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_init_jack); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto "); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 8b374af86a6e..a6477a022156 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -10,23 +10,14 @@ */ #include #include -#include #include #include -#include #include #include -#include #include #include #include -struct asoc_simple_jack { - struct snd_soc_jack jack; - struct snd_soc_jack_pin pin; - struct snd_soc_jack_gpio gpio; -}; - struct simple_card_data { struct snd_soc_card snd_card; struct simple_dai_props { @@ -49,61 +40,6 @@ struct simple_card_data { #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," -#define asoc_simple_card_init_hp(card, sjack, prefix)\ - asoc_simple_card_init_jack(card, sjack, 1, prefix) -#define asoc_simple_card_init_mic(card, sjack, prefix)\ - asoc_simple_card_init_jack(card, sjack, 0, prefix) -static int asoc_simple_card_init_jack(struct snd_soc_card *card, - struct asoc_simple_jack *sjack, - int is_hp, char *prefix) -{ - struct device *dev = card->dev; - enum of_gpio_flags flags; - char prop[128]; - char *pin_name; - char *gpio_name; - int mask; - int det; - - sjack->gpio.gpio = -ENOENT; - - if (is_hp) { - snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); - pin_name = "Headphones"; - gpio_name = "Headphone detection"; - mask = SND_JACK_HEADPHONE; - } else { - snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix); - pin_name = "Mic Jack"; - gpio_name = "Mic detection"; - mask = SND_JACK_MICROPHONE; - } - - det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags); - if (det == -EPROBE_DEFER) - return -EPROBE_DEFER; - - if (gpio_is_valid(det)) { - sjack->pin.pin = pin_name; - sjack->pin.mask = mask; - - sjack->gpio.name = gpio_name; - sjack->gpio.report = mask; - sjack->gpio.gpio = det; - sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); - sjack->gpio.debounce_time = 150; - - snd_soc_card_jack_new(card, pin_name, mask, - &sjack->jack, - &sjack->pin, 1); - - snd_soc_jack_add_gpios(&sjack->jack, 1, - &sjack->gpio); - } - - return 0; -} - static int asoc_simple_card_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; -- cgit v1.2.3 From 8d1bd113a194407f9ad083403ea1cf92108edf5c Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Mon, 11 Jun 2018 17:32:13 +0900 Subject: ASoC: simple-card: move hp and mic detection to soc_card probe This patch moves headphone and microphone detection to probe() of snd_soc_card from init() of snd_soc_dai_link. This is because init() is called (and an input device /dev/input/eventX is created too) twice or above if simple card has two or more DAI links. Signed-off-by: Katsuhiro Suzuki Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index a6477a022156..c5b6e04cd926 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -149,14 +149,6 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - ret = asoc_simple_card_init_hp(rtd->card, &priv->hp_jack, PREFIX); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_mic(rtd->card, &priv->mic_jack, PREFIX); - if (ret < 0) - return ret; - return 0; } @@ -350,6 +342,22 @@ card_parse_end: return ret; } +static int asoc_simple_soc_card_probe(struct snd_soc_card *card) +{ + struct simple_card_data *priv = snd_soc_card_get_drvdata(card); + int ret; + + ret = asoc_simple_card_init_hp(card, &priv->hp_jack, PREFIX); + if (ret < 0) + return ret; + + ret = asoc_simple_card_init_mic(card, &priv->mic_jack, PREFIX); + if (ret < 0) + return ret; + + return 0; +} + static int asoc_simple_card_probe(struct platform_device *pdev) { struct simple_card_data *priv; @@ -385,6 +393,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) card->dev = dev; card->dai_link = priv->dai_link; card->num_links = num; + card->probe = asoc_simple_soc_card_probe; if (np && of_device_is_available(np)) { -- cgit v1.2.3 From f6de35cc145fb55d842db94e74841ecd8382e012 Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Mon, 11 Jun 2018 17:32:14 +0900 Subject: ASoC: audio-graph-card: add hp and mic detect gpios same as simple-card This patch adds headphone and microphone jack detection gpios as same as simple-card driver. Signed-off-by: Katsuhiro Suzuki Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index d93bacacbd5b..a2a3e630f11c 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -21,7 +21,6 @@ #include #include #include -#include #include struct graph_card_data { @@ -32,6 +31,8 @@ struct graph_card_data { unsigned int mclk_fs; } *dai_props; unsigned int mclk_fs; + struct asoc_simple_jack hp_jack; + struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; struct gpio_desc *pa_gpio; }; @@ -278,6 +279,22 @@ static int asoc_graph_get_dais_count(struct device *dev) return count; } +static int asoc_graph_soc_card_probe(struct snd_soc_card *card) +{ + struct graph_card_data *priv = snd_soc_card_get_drvdata(card); + int ret; + + ret = asoc_simple_card_init_hp(card, &priv->hp_jack, NULL); + if (ret < 0) + return ret; + + ret = asoc_simple_card_init_mic(card, &priv->mic_jack, NULL); + if (ret < 0) + return ret; + + return 0; +} + static int asoc_graph_card_probe(struct platform_device *pdev) { struct graph_card_data *priv; @@ -319,6 +336,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) card->num_links = num; card->dapm_widgets = asoc_graph_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(asoc_graph_card_dapm_widgets); + card->probe = asoc_graph_soc_card_probe; ret = asoc_graph_card_parse_of(priv); if (ret < 0) { -- cgit v1.2.3 From a0d847c380ba47886fcca4168698eef51c69f109 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 Jun 2018 05:51:46 +0000 Subject: ASoC: rsnd: add rsnd_daidrv_get() rsnd priv has many parameters. On __rsnd_dai_probe() it uses rsnd_rdai_get() to get rdai pointer, but is using priv->daidrv directly to get daidrvhv, but it is confusable for reader. This patch adds rsnd_daidrv_get() to get daidrv from priv. Now reader can understand that rdai and daidrv are related. Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index eac22fef4543..6091e0916085 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -548,6 +548,15 @@ struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id) return priv->rdai + id; } +static struct snd_soc_dai_driver +*rsnd_daidrv_get(struct rsnd_priv *priv, int id) +{ + if ((id < 0) || (id >= rsnd_rdai_nr(priv))) + return NULL; + + return priv->daidrv + id; +} + #define rsnd_dai_to_priv(dai) snd_soc_dai_get_drvdata(dai) static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai) { @@ -1033,7 +1042,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, int io_i; rdai = rsnd_rdai_get(priv, dai_i); - drv = priv->daidrv + dai_i; + drv = rsnd_daidrv_get(priv, dai_i); io_playback = &rdai->playback; io_capture = &rdai->capture; -- cgit v1.2.3 From d5c4e972d512ae0b59108ca92b9b35bc5cf5c14e Mon Sep 17 00:00:00 2001 From: Rohit Kumar Date: Wed, 6 Jun 2018 14:25:24 +0530 Subject: ASoC: qcom: apq8096: set card as device drvdata snd_soc_card is retrieved as device drvdata during unbind(). Set it as drvdata during bind() to avoid memory corruption during unbind(). Signed-off-by: Rohit kumar Acked-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/apq8096.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 561cd429e6f2..239b8cb77bdb 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -140,6 +140,7 @@ static int apq8096_bind(struct device *dev) component_bind_all(dev, card); card->dev = dev; + dev_set_drvdata(dev, card); ret = apq8096_sbc_parse_of(card); if (ret) { dev_err(dev, "Error parsing OF data\n"); -- cgit v1.2.3 From 510e419cb85798915c6426c496a164ec6328d1a7 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 6 Jun 2018 10:35:04 +0100 Subject: ASoC: twl6040: make pointer dmic_codec_dev static The pointer dmic_codec_dev is local to the source and does not need to be in global scope, so make it static. Cleans up sparse warning: warning: symbol 'dmic_codec_dev' was not declared. Should it be static? Signed-off-by: Colin Ian King Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-abe-twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 15ccbf479c96..d5ae9eb8c756 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -40,7 +40,7 @@ struct abe_twl6040 { int mclk_freq; /* MCLK frequency speed for twl6040 */ }; -struct platform_device *dmic_codec_dev; +static struct platform_device *dmic_codec_dev; static int omap_abe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) -- cgit v1.2.3 From e380be7c557c4ddb1cb71404c10e0bfb3daf4644 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 6 Jun 2018 10:57:19 +0100 Subject: ASoC: ak5558: make two structures static The structure ak5558_pm and soc_codec_dev_ak5558 are local to the source and do not need to be in global scope, so make them static. Also make soc_codec_dev_ak5558 static. Cleans up sparse warnings: warning: symbol 'ak5558_pm' was not declared. Should it be static? warning: symbol 'soc_codec_dev_ak5558' was not declared. Should it be static? Signed-off-by: Colin Ian King Reviewed-by: Daniel Baluta Signed-off-by: Mark Brown --- sound/soc/codecs/ak5558.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index f4ed5cc40661..448bb90c9c8e 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -322,13 +322,13 @@ static int __maybe_unused ak5558_runtime_resume(struct device *dev) return regcache_sync(ak5558->regmap); } -const struct dev_pm_ops ak5558_pm = { +static const struct dev_pm_ops ak5558_pm = { SET_RUNTIME_PM_OPS(ak5558_runtime_suspend, ak5558_runtime_resume, NULL) SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) }; -struct snd_soc_component_driver soc_codec_dev_ak5558 = { +static const struct snd_soc_component_driver soc_codec_dev_ak5558 = { .probe = ak5558_probe, .remove = ak5558_remove, .controls = ak5558_snd_controls, -- cgit v1.2.3 From 62624f72592b1d8e756e699cd6ade2be379f95a9 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 6 Jun 2018 11:31:23 +0100 Subject: ASoC: ak4458: make structure soc_codec_dev_ak4458 static const The structure soc_codec_dev_ak4458 is local to the source and do not need to be in global scope and can be const, make it static const. Cleans up sparse warnings: warning: symbol 'soc_codec_dev_ak4458' was not declared. Should it be static? Signed-off-by: Colin Ian King Signed-off-by: Mark Brown --- sound/soc/codecs/ak4458.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 31ec0ba2e639..299ada4dfaa0 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -558,7 +558,7 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev) } #endif /* CONFIG_PM */ -struct snd_soc_component_driver soc_codec_dev_ak4458 = { +static const struct snd_soc_component_driver soc_codec_dev_ak4458 = { .probe = ak4458_probe, .remove = ak4458_remove, .controls = ak4458_snd_controls, -- cgit v1.2.3 From 58f7d470c85801f55bbae2d3c93fe1a3d34aa143 Mon Sep 17 00:00:00 2001 From: Steven Eckhoff Date: Mon, 4 Jun 2018 15:45:40 -0500 Subject: ASoC: TSCS42xx: Add mic bias boost control Add mic bias boost control Signed-off-by: Steven Eckhoff Signed-off-by: Mark Brown --- sound/soc/codecs/tscs42xx.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index d18ff17719cc..743194052fc4 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -644,6 +644,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { /* Input Channel Map */ SOC_ENUM("Input Channel Map", ch_map_select_enum), + /* Mic Bias */ + SOC_SINGLE("Mic Bias Boost Switch", 0x71, 0x07, 1, 0), + /* Coefficient Ram */ COEFF_RAM_CTL("Cascade1L BiQuad1", BIQUAD_SIZE, 0x00), COEFF_RAM_CTL("Cascade1L BiQuad2", BIQUAD_SIZE, 0x05), -- cgit v1.2.3 From 19d996cc3ad698379bcd8482dabe78abe3dc8d96 Mon Sep 17 00:00:00 2001 From: Steven Eckhoff Date: Mon, 4 Jun 2018 15:46:28 -0500 Subject: ASoC: TSCS42xx: Remove Playback/Capture in names These aren't needed and some userspace apps don't work consistently with them. Remove Playback/Capture from control names Signed-off-by: Steven Eckhoff Signed-off-by: Mark Brown --- sound/soc/codecs/tscs42xx.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index 743194052fc4..5d596e4dab0c 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -625,19 +625,19 @@ static int bytes_info_ext(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { /* Volumes */ - SOC_DOUBLE_R_TLV("Headphone Playback Volume", R_HPVOLL, R_HPVOLR, + SOC_DOUBLE_R_TLV("Headphone Volume", R_HPVOLL, R_HPVOLR, FB_HPVOLL, 0x7F, 0, hpvol_scale), - SOC_DOUBLE_R_TLV("Speaker Playback Volume", R_SPKVOLL, R_SPKVOLR, + SOC_DOUBLE_R_TLV("Speaker Volume", R_SPKVOLL, R_SPKVOLR, FB_SPKVOLL, 0x7F, 0, spkvol_scale), - SOC_DOUBLE_R_TLV("Master Playback Volume", R_DACVOLL, R_DACVOLR, + SOC_DOUBLE_R_TLV("Master Volume", R_DACVOLL, R_DACVOLR, FB_DACVOLL, 0xFF, 0, dacvol_scale), - SOC_DOUBLE_R_TLV("PCM Capture Volume", R_ADCVOLL, R_ADCVOLR, + SOC_DOUBLE_R_TLV("PCM Volume", R_ADCVOLL, R_ADCVOLR, FB_ADCVOLL, 0xFF, 0, adcvol_scale), - SOC_DOUBLE_R_TLV("Master Capture Volume", R_INVOLL, R_INVOLR, + SOC_DOUBLE_R_TLV("Input Volume", R_INVOLL, R_INVOLR, FB_INVOLL, 0x3F, 0, invol_scale), /* INSEL */ - SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", R_INSELL, R_INSELR, + SOC_DOUBLE_R_TLV("Mic Boost Volume", R_INSELL, R_INSELR, FB_INSELL_MICBSTL, FV_INSELL_MICBSTL_30DB, 0, mic_boost_scale), @@ -736,9 +736,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { R_CLECTL, FB_CLECTL_LIMIT_EN, 1, 0), SOC_SINGLE("Comp Switch", R_CLECTL, FB_CLECTL_COMP_EN, 1, 0), - SOC_SINGLE_TLV("CLE Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("CLE Make-Up Gain Volume", R_MUGAIN, FB_MUGAIN_CLEMUG, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("Comp Thresh Playback Volume", + SOC_SINGLE_TLV("Comp Thresh Volume", R_COMPTH, FB_COMPTH, 0xff, 0, compth_scale), SOC_ENUM("Comp Ratio", compressor_ratio_enum), SND_SOC_BYTES("Comp Atk Time", R_CATKTCL, 2), @@ -769,9 +769,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC1 Phase Invert Switch", R_DACMBCMUG1, FB_DACMBCMUG1_PHASE, 1, 0), - SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Volume", R_DACMBCMUG1, FB_DACMBCMUG1_MUGAIN, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Playback Volume", + SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Volume", R_DACMBCTHR1, FB_DACMBCTHR1_THRESH, 0xff, 0, compth_scale), SOC_ENUM("DAC MBC1 Comp Ratio", dac_mbc1_compressor_ratio_enum), @@ -781,9 +781,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC2 Phase Invert Switch", R_DACMBCMUG2, FB_DACMBCMUG2_PHASE, 1, 0), - SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Volume", R_DACMBCMUG2, FB_DACMBCMUG2_MUGAIN, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Playback Volume", + SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Volume", R_DACMBCTHR2, FB_DACMBCTHR2_THRESH, 0xff, 0, compth_scale), SOC_ENUM("DAC MBC2 Comp Ratio", dac_mbc2_compressor_ratio_enum), @@ -793,9 +793,9 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC3 Phase Invert Switch", R_DACMBCMUG3, FB_DACMBCMUG3_PHASE, 1, 0), - SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Playback Volume", + SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Volume", R_DACMBCMUG3, FB_DACMBCMUG3_MUGAIN, 0x1f, 0, mugain_scale), - SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Playback Volume", + SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Volume", R_DACMBCTHR3, FB_DACMBCTHR3_THRESH, 0xff, 0, compth_scale), SOC_ENUM("DAC MBC3 Comp Ratio", dac_mbc3_compressor_ratio_enum), -- cgit v1.2.3 From 53af408cd9f2d1c66bc081c1d82797bfe44af3e5 Mon Sep 17 00:00:00 2001 From: Steven Eckhoff Date: Mon, 4 Jun 2018 15:47:02 -0500 Subject: ASoC: TSCS42xx: Add headphone auto switching Add headphone auto switching controls Signed-off-by: Steven Eckhoff Signed-off-by: Mark Brown --- sound/soc/codecs/tscs42xx.c | 6 ++++++ sound/soc/codecs/tscs42xx.h | 8 ++++++++ 2 files changed, 14 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index 5d596e4dab0c..7396a6e5277e 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -647,6 +647,12 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { /* Mic Bias */ SOC_SINGLE("Mic Bias Boost Switch", 0x71, 0x07, 1, 0), + /* Headphone Auto Switching */ + SOC_SINGLE("Headphone Auto Switching Switch", + R_CTL, FB_CTL_HPSWEN, 1, 0), + SOC_SINGLE("Headphone Detect Polarity Toggle Switch", + R_CTL, FB_CTL_HPSWPOL, 1, 0), + /* Coefficient Ram */ COEFF_RAM_CTL("Cascade1L BiQuad1", BIQUAD_SIZE, 0x00), COEFF_RAM_CTL("Cascade1L BiQuad2", BIQUAD_SIZE, 0x05), diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h index 814c8f3c4a68..6b3a21081635 100644 --- a/sound/soc/codecs/tscs42xx.h +++ b/sound/soc/codecs/tscs42xx.h @@ -34,6 +34,7 @@ enum { #define R_DACSR 0x19 #define R_PWRM1 0x1A #define R_PWRM2 0x1B +#define R_CTL 0x1C #define R_CONFIG0 0x1F #define R_CONFIG1 0x20 #define R_DMICCTL 0x24 @@ -1110,6 +1111,13 @@ enum { #define RV_PWRM2_VREF_DISABLE \ RV(FV_PWRM2_VREF_DISABLE, FB_PWRM2_VREF) +/****************************** + * R_CTL (0x1C) * + ******************************/ + +/* Fiel Offsets */ +#define FB_CTL_HPSWEN 7 +#define FB_CTL_HPSWPOL 6 /****************************** * R_CONFIG0 (0x1F) * -- cgit v1.2.3 From 0ddce71c21f03fd19867c4939d3ca710f37cdf1a Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 7 Jun 2018 16:37:38 +0800 Subject: ASoC: rt5682: add rt5682 codec driver This is the initial codec driver for rt5682. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rt5682.txt | 50 + include/sound/rt5682.h | 40 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5682.c | 2682 ++++++++++++++++++++ sound/soc/codecs/rt5682.h | 1324 ++++++++++ 6 files changed, 4104 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5682.txt create mode 100644 include/sound/rt5682.h create mode 100644 sound/soc/codecs/rt5682.c create mode 100644 sound/soc/codecs/rt5682.h (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt new file mode 100644 index 000000000000..312e9a129530 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -0,0 +1,50 @@ +RT5682 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5682" or "realtek,rt5682i" + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using GPIO5 pin as dmic1 data pin + +- realtek,dmic1-clk-pin + 0: using GPIO1 pin as dmic1 clock pin + 1: using GPIO3 pin as dmic1 clock pin + +- realtek,jd-src + 0: No JD is used + 1: using JD1 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes) for RT5682: + + * DMIC L1 + * DMIC R1 + * IN1P + * HPOL + * HPOR + +Example: + +rt5682 { + compatible = "realtek,rt5682i"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = ; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; +}; diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h new file mode 100644 index 000000000000..0251797ab438 --- /dev/null +++ b/include/sound/rt5682.h @@ -0,0 +1,40 @@ +/* + * linux/sound/rt5682.h -- Platform data for RT5682 + * + * Copyright 2018 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5682_H +#define __LINUX_SND_RT5682_H + +enum rt5682_dmic1_data_pin { + RT5682_DMIC1_NULL, + RT5682_DMIC1_DATA_GPIO2, + RT5682_DMIC1_DATA_GPIO5, +}; + +enum rt5682_dmic1_clk_pin { + RT5682_DMIC1_CLK_GPIO1, + RT5682_DMIC1_CLK_GPIO3, +}; + +enum rt5682_jd_src { + RT5682_JD_NULL, + RT5682_JD1, +}; + +struct rt5682_platform_data { + + int ldo1_en; /* GPIO for LDO1_EN */ + + enum rt5682_dmic1_data_pin dmic1_data_pin; + enum rt5682_dmic1_clk_pin dmic1_clk_pin; + enum rt5682_jd_src jd_src; +}; + +#endif + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 63cf62e9c9aa..f6b8d4bf8796 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -141,6 +141,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5668 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER + select SND_SOC_RT5682 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIRF_AUDIO_CODEC @@ -778,6 +779,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5668=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y + default y if SND_SOC_RT5682=y default y if SND_SOC_RT1305=y default m if SND_SOC_RT5514=m default m if SND_SOC_RT5616=m @@ -791,6 +793,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5668=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m + default m if SND_SOC_RT5682=m default m if SND_SOC_RT1305=m config SND_SOC_RL6347A @@ -871,6 +874,9 @@ config SND_SOC_RT5677_SPI tristate default SND_SOC_RT5677 && SPI +config SND_SOC_RT5682 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e023fdf85221..e43d99a039d3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -146,6 +146,7 @@ snd-soc-rt5668-objs := rt5668.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o +snd-soc-rt5682-objs := rt5682.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -405,6 +406,7 @@ obj-$(CONFIG_SND_SOC_RT5668) += snd-soc-rt5668.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o +obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c new file mode 100644 index 000000000000..61a97301bcfa --- /dev/null +++ b/sound/soc/codecs/rt5682.c @@ -0,0 +1,2682 @@ +/* + * rt5682.c -- RT5682 ALSA SoC audio component driver + * + * Copyright 2018 Realtek Semiconductor Corp. + * Author: Bard Liao + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "rl6231.h" +#include "rt5682.h" + +#define RT5682_NUM_SUPPLIES 3 + +static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = { + "AVDD", + "MICVDD", + "VBAT", +}; + +struct rt5682_priv { + struct snd_soc_component *component; + struct rt5682_platform_data pdata; + struct regmap *regmap; + struct snd_soc_jack *hs_jack; + struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES]; + struct delayed_work jack_detect_work; + struct delayed_work jd_check_work; + struct mutex calibrate_mutex; + + int sysclk; + int sysclk_src; + int lrck[RT5682_AIFS]; + int bclk[RT5682_AIFS]; + int master[RT5682_AIFS]; + + int pll_src; + int pll_in; + int pll_out; + + int jack_type; +}; + +static const struct reg_sequence patch_list[] = { + {0x01c1, 0x1000}, +}; + +static const struct reg_default rt5682_reg[] = { + {0x0002, 0x8080}, + {0x0003, 0x8000}, + {0x0005, 0x0000}, + {0x0006, 0x0000}, + {0x0008, 0x800f}, + {0x000b, 0x0000}, + {0x0010, 0x4040}, + {0x0011, 0x0000}, + {0x0012, 0x1404}, + {0x0013, 0x1000}, + {0x0014, 0xa00a}, + {0x0015, 0x0404}, + {0x0016, 0x0404}, + {0x0019, 0xafaf}, + {0x001c, 0x2f2f}, + {0x001f, 0x0000}, + {0x0022, 0x5757}, + {0x0023, 0x0039}, + {0x0024, 0x000b}, + {0x0026, 0xc0c4}, + {0x0029, 0x8080}, + {0x002a, 0xa0a0}, + {0x002b, 0x0300}, + {0x0030, 0x0000}, + {0x003c, 0x0080}, + {0x0044, 0x0c0c}, + {0x0049, 0x0000}, + {0x0061, 0x0000}, + {0x0062, 0x0000}, + {0x0063, 0x003f}, + {0x0064, 0x0000}, + {0x0065, 0x0000}, + {0x0066, 0x0030}, + {0x0067, 0x0000}, + {0x006b, 0x0000}, + {0x006c, 0x0000}, + {0x006d, 0x2200}, + {0x006e, 0x0a10}, + {0x0070, 0x8000}, + {0x0071, 0x8000}, + {0x0073, 0x0000}, + {0x0074, 0x0000}, + {0x0075, 0x0002}, + {0x0076, 0x0001}, + {0x0079, 0x0000}, + {0x007a, 0x0000}, + {0x007b, 0x0000}, + {0x007c, 0x0100}, + {0x007e, 0x0000}, + {0x0080, 0x0000}, + {0x0081, 0x0000}, + {0x0082, 0x0000}, + {0x0083, 0x0000}, + {0x0084, 0x0000}, + {0x0085, 0x0000}, + {0x0086, 0x0005}, + {0x0087, 0x0000}, + {0x0088, 0x0000}, + {0x008c, 0x0003}, + {0x008d, 0x0000}, + {0x008e, 0x0060}, + {0x008f, 0x1000}, + {0x0091, 0x0c26}, + {0x0092, 0x0073}, + {0x0093, 0x0000}, + {0x0094, 0x0080}, + {0x0098, 0x0000}, + {0x009a, 0x0000}, + {0x009b, 0x0000}, + {0x009c, 0x0000}, + {0x009d, 0x0000}, + {0x009e, 0x100c}, + {0x009f, 0x0000}, + {0x00a0, 0x0000}, + {0x00a3, 0x0002}, + {0x00a4, 0x0001}, + {0x00ae, 0x2040}, + {0x00af, 0x0000}, + {0x00b6, 0x0000}, + {0x00b7, 0x0000}, + {0x00b8, 0x0000}, + {0x00b9, 0x0002}, + {0x00be, 0x0000}, + {0x00c0, 0x0160}, + {0x00c1, 0x82a0}, + {0x00c2, 0x0000}, + {0x00d0, 0x0000}, + {0x00d1, 0x2244}, + {0x00d2, 0x3300}, + {0x00d3, 0x2200}, + {0x00d4, 0x0000}, + {0x00d9, 0x0009}, + {0x00da, 0x0000}, + {0x00db, 0x0000}, + {0x00dc, 0x00c0}, + {0x00dd, 0x2220}, + {0x00de, 0x3131}, + {0x00df, 0x3131}, + {0x00e0, 0x3131}, + {0x00e2, 0x0000}, + {0x00e3, 0x4000}, + {0x00e4, 0x0aa0}, + {0x00e5, 0x3131}, + {0x00e6, 0x3131}, + {0x00e7, 0x3131}, + {0x00e8, 0x3131}, + {0x00ea, 0xb320}, + {0x00eb, 0x0000}, + {0x00f0, 0x0000}, + {0x00f1, 0x00d0}, + {0x00f2, 0x00d0}, + {0x00f6, 0x0000}, + {0x00fa, 0x0000}, + {0x00fb, 0x0000}, + {0x00fc, 0x0000}, + {0x00fd, 0x0000}, + {0x00fe, 0x10ec}, + {0x00ff, 0x6530}, + {0x0100, 0xa0a0}, + {0x010b, 0x0000}, + {0x010c, 0xae00}, + {0x010d, 0xaaa0}, + {0x010e, 0x8aa2}, + {0x010f, 0x02a2}, + {0x0110, 0xc000}, + {0x0111, 0x04a2}, + {0x0112, 0x2800}, + {0x0113, 0x0000}, + {0x0117, 0x0100}, + {0x0125, 0x0410}, + {0x0132, 0x6026}, + {0x0136, 0x5555}, + {0x0138, 0x3700}, + {0x013a, 0x2000}, + {0x013b, 0x2000}, + {0x013c, 0x2005}, + {0x013f, 0x0000}, + {0x0142, 0x0000}, + {0x0145, 0x0002}, + {0x0146, 0x0000}, + {0x0147, 0x0000}, + {0x0148, 0x0000}, + {0x0149, 0x0000}, + {0x0150, 0x79a1}, + {0x0151, 0x0000}, + {0x0160, 0x4ec0}, + {0x0161, 0x0080}, + {0x0162, 0x0200}, + {0x0163, 0x0800}, + {0x0164, 0x0000}, + {0x0165, 0x0000}, + {0x0166, 0x0000}, + {0x0167, 0x000f}, + {0x0168, 0x000f}, + {0x0169, 0x0021}, + {0x0190, 0x413d}, + {0x0194, 0x0000}, + {0x0195, 0x0000}, + {0x0197, 0x0022}, + {0x0198, 0x0000}, + {0x0199, 0x0000}, + {0x01af, 0x0000}, + {0x01b0, 0x0400}, + {0x01b1, 0x0000}, + {0x01b2, 0x0000}, + {0x01b3, 0x0000}, + {0x01b4, 0x0000}, + {0x01b5, 0x0000}, + {0x01b6, 0x01c3}, + {0x01b7, 0x02a0}, + {0x01b8, 0x03e9}, + {0x01b9, 0x1389}, + {0x01ba, 0xc351}, + {0x01bb, 0x0009}, + {0x01bc, 0x0018}, + {0x01bd, 0x002a}, + {0x01be, 0x004c}, + {0x01bf, 0x0097}, + {0x01c0, 0x433d}, + {0x01c2, 0x0000}, + {0x01c3, 0x0000}, + {0x01c4, 0x0000}, + {0x01c5, 0x0000}, + {0x01c6, 0x0000}, + {0x01c7, 0x0000}, + {0x01c8, 0x40af}, + {0x01c9, 0x0702}, + {0x01ca, 0x0000}, + {0x01cb, 0x0000}, + {0x01cc, 0x5757}, + {0x01cd, 0x5757}, + {0x01ce, 0x5757}, + {0x01cf, 0x5757}, + {0x01d0, 0x5757}, + {0x01d1, 0x5757}, + {0x01d2, 0x5757}, + {0x01d3, 0x5757}, + {0x01d4, 0x5757}, + {0x01d5, 0x5757}, + {0x01d6, 0x0000}, + {0x01d7, 0x0008}, + {0x01d8, 0x0029}, + {0x01d9, 0x3333}, + {0x01da, 0x0000}, + {0x01db, 0x0004}, + {0x01dc, 0x0000}, + {0x01de, 0x7c00}, + {0x01df, 0x0320}, + {0x01e0, 0x06a1}, + {0x01e1, 0x0000}, + {0x01e2, 0x0000}, + {0x01e3, 0x0000}, + {0x01e4, 0x0000}, + {0x01e6, 0x0001}, + {0x01e7, 0x0000}, + {0x01e8, 0x0000}, + {0x01ea, 0x0000}, + {0x01eb, 0x0000}, + {0x01ec, 0x0000}, + {0x01ed, 0x0000}, + {0x01ee, 0x0000}, + {0x01ef, 0x0000}, + {0x01f0, 0x0000}, + {0x01f1, 0x0000}, + {0x01f2, 0x0000}, + {0x01f3, 0x0000}, + {0x01f4, 0x0000}, + {0x0210, 0x6297}, + {0x0211, 0xa005}, + {0x0212, 0x824c}, + {0x0213, 0xf7ff}, + {0x0214, 0xf24c}, + {0x0215, 0x0102}, + {0x0216, 0x00a3}, + {0x0217, 0x0048}, + {0x0218, 0xa2c0}, + {0x0219, 0x0400}, + {0x021a, 0x00c8}, + {0x021b, 0x00c0}, + {0x021c, 0x0000}, + {0x0250, 0x4500}, + {0x0251, 0x40b3}, + {0x0252, 0x0000}, + {0x0253, 0x0000}, + {0x0254, 0x0000}, + {0x0255, 0x0000}, + {0x0256, 0x0000}, + {0x0257, 0x0000}, + {0x0258, 0x0000}, + {0x0259, 0x0000}, + {0x025a, 0x0005}, + {0x0270, 0x0000}, + {0x02ff, 0x0110}, + {0x0300, 0x001f}, + {0x0301, 0x032c}, + {0x0302, 0x5f21}, + {0x0303, 0x4000}, + {0x0304, 0x4000}, + {0x0305, 0x06d5}, + {0x0306, 0x8000}, + {0x0307, 0x0700}, + {0x0310, 0x4560}, + {0x0311, 0xa4a8}, + {0x0312, 0x7418}, + {0x0313, 0x0000}, + {0x0314, 0x0006}, + {0x0315, 0xffff}, + {0x0316, 0xc400}, + {0x0317, 0x0000}, + {0x03c0, 0x7e00}, + {0x03c1, 0x8000}, + {0x03c2, 0x8000}, + {0x03c3, 0x8000}, + {0x03c4, 0x8000}, + {0x03c5, 0x8000}, + {0x03c6, 0x8000}, + {0x03c7, 0x8000}, + {0x03c8, 0x8000}, + {0x03c9, 0x8000}, + {0x03ca, 0x8000}, + {0x03cb, 0x8000}, + {0x03cc, 0x8000}, + {0x03d0, 0x0000}, + {0x03d1, 0x0000}, + {0x03d2, 0x0000}, + {0x03d3, 0x0000}, + {0x03d4, 0x2000}, + {0x03d5, 0x2000}, + {0x03d6, 0x0000}, + {0x03d7, 0x0000}, + {0x03d8, 0x2000}, + {0x03d9, 0x2000}, + {0x03da, 0x2000}, + {0x03db, 0x2000}, + {0x03dc, 0x0000}, + {0x03dd, 0x0000}, + {0x03de, 0x0000}, + {0x03df, 0x2000}, + {0x03e0, 0x0000}, + {0x03e1, 0x0000}, + {0x03e2, 0x0000}, + {0x03e3, 0x0000}, + {0x03e4, 0x0000}, + {0x03e5, 0x0000}, + {0x03e6, 0x0000}, + {0x03e7, 0x0000}, + {0x03e8, 0x0000}, + {0x03e9, 0x0000}, + {0x03ea, 0x0000}, + {0x03eb, 0x0000}, + {0x03ec, 0x0000}, + {0x03ed, 0x0000}, + {0x03ee, 0x0000}, + {0x03ef, 0x0000}, + {0x03f0, 0x0800}, + {0x03f1, 0x0800}, + {0x03f2, 0x0800}, + {0x03f3, 0x0800}, +}; + +static bool rt5682_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682_RESET: + case RT5682_CBJ_CTRL_2: + case RT5682_INT_ST_1: + case RT5682_4BTN_IL_CMD_1: + case RT5682_AJD1_CTRL: + case RT5682_HP_CALIB_CTRL_1: + case RT5682_DEVICE_ID: + case RT5682_I2C_MODE: + case RT5682_HP_CALIB_CTRL_10: + case RT5682_EFUSE_CTRL_2: + case RT5682_JD_TOP_VC_VTRL: + case RT5682_HP_IMP_SENS_CTRL_19: + case RT5682_IL_CMD_1: + case RT5682_SAR_IL_CMD_2: + case RT5682_SAR_IL_CMD_4: + case RT5682_SAR_IL_CMD_10: + case RT5682_SAR_IL_CMD_11: + case RT5682_EFUSE_CTRL_6...RT5682_EFUSE_CTRL_11: + case RT5682_HP_CALIB_STA_1...RT5682_HP_CALIB_STA_11: + return true; + default: + return false; + } +} + +static bool rt5682_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5682_RESET: + case RT5682_VERSION_ID: + case RT5682_VENDOR_ID: + case RT5682_DEVICE_ID: + case RT5682_HP_CTRL_1: + case RT5682_HP_CTRL_2: + case RT5682_HPL_GAIN: + case RT5682_HPR_GAIN: + case RT5682_I2C_CTRL: + case RT5682_CBJ_BST_CTRL: + case RT5682_CBJ_CTRL_1: + case RT5682_CBJ_CTRL_2: + case RT5682_CBJ_CTRL_3: + case RT5682_CBJ_CTRL_4: + case RT5682_CBJ_CTRL_5: + case RT5682_CBJ_CTRL_6: + case RT5682_CBJ_CTRL_7: + case RT5682_DAC1_DIG_VOL: + case RT5682_STO1_ADC_DIG_VOL: + case RT5682_STO1_ADC_BOOST: + case RT5682_HP_IMP_GAIN_1: + case RT5682_HP_IMP_GAIN_2: + case RT5682_SIDETONE_CTRL: + case RT5682_STO1_ADC_MIXER: + case RT5682_AD_DA_MIXER: + case RT5682_STO1_DAC_MIXER: + case RT5682_A_DAC1_MUX: + case RT5682_DIG_INF2_DATA: + case RT5682_REC_MIXER: + case RT5682_CAL_REC: + case RT5682_ALC_BACK_GAIN: + case RT5682_PWR_DIG_1: + case RT5682_PWR_DIG_2: + case RT5682_PWR_ANLG_1: + case RT5682_PWR_ANLG_2: + case RT5682_PWR_ANLG_3: + case RT5682_PWR_MIXER: + case RT5682_PWR_VOL: + case RT5682_CLK_DET: + case RT5682_RESET_LPF_CTRL: + case RT5682_RESET_HPF_CTRL: + case RT5682_DMIC_CTRL_1: + case RT5682_I2S1_SDP: + case RT5682_I2S2_SDP: + case RT5682_ADDA_CLK_1: + case RT5682_ADDA_CLK_2: + case RT5682_I2S1_F_DIV_CTRL_1: + case RT5682_I2S1_F_DIV_CTRL_2: + case RT5682_TDM_CTRL: + case RT5682_TDM_ADDA_CTRL_1: + case RT5682_TDM_ADDA_CTRL_2: + case RT5682_DATA_SEL_CTRL_1: + case RT5682_TDM_TCON_CTRL: + case RT5682_GLB_CLK: + case RT5682_PLL_CTRL_1: + case RT5682_PLL_CTRL_2: + case RT5682_PLL_TRACK_1: + case RT5682_PLL_TRACK_2: + case RT5682_PLL_TRACK_3: + case RT5682_PLL_TRACK_4: + case RT5682_PLL_TRACK_5: + case RT5682_PLL_TRACK_6: + case RT5682_PLL_TRACK_11: + case RT5682_SDW_REF_CLK: + case RT5682_DEPOP_1: + case RT5682_DEPOP_2: + case RT5682_HP_CHARGE_PUMP_1: + case RT5682_HP_CHARGE_PUMP_2: + case RT5682_MICBIAS_1: + case RT5682_MICBIAS_2: + case RT5682_PLL_TRACK_12: + case RT5682_PLL_TRACK_14: + case RT5682_PLL2_CTRL_1: + case RT5682_PLL2_CTRL_2: + case RT5682_PLL2_CTRL_3: + case RT5682_PLL2_CTRL_4: + case RT5682_RC_CLK_CTRL: + case RT5682_I2S_M_CLK_CTRL_1: + case RT5682_I2S2_F_DIV_CTRL_1: + case RT5682_I2S2_F_DIV_CTRL_2: + case RT5682_EQ_CTRL_1: + case RT5682_EQ_CTRL_2: + case RT5682_IRQ_CTRL_1: + case RT5682_IRQ_CTRL_2: + case RT5682_IRQ_CTRL_3: + case RT5682_IRQ_CTRL_4: + case RT5682_INT_ST_1: + case RT5682_GPIO_CTRL_1: + case RT5682_GPIO_CTRL_2: + case RT5682_GPIO_CTRL_3: + case RT5682_HP_AMP_DET_CTRL_1: + case RT5682_HP_AMP_DET_CTRL_2: + case RT5682_MID_HP_AMP_DET: + case RT5682_LOW_HP_AMP_DET: + case RT5682_DELAY_BUF_CTRL: + case RT5682_SV_ZCD_1: + case RT5682_SV_ZCD_2: + case RT5682_IL_CMD_1: + case RT5682_IL_CMD_2: + case RT5682_IL_CMD_3: + case RT5682_IL_CMD_4: + case RT5682_IL_CMD_5: + case RT5682_IL_CMD_6: + case RT5682_4BTN_IL_CMD_1: + case RT5682_4BTN_IL_CMD_2: + case RT5682_4BTN_IL_CMD_3: + case RT5682_4BTN_IL_CMD_4: + case RT5682_4BTN_IL_CMD_5: + case RT5682_4BTN_IL_CMD_6: + case RT5682_4BTN_IL_CMD_7: + case RT5682_ADC_STO1_HP_CTRL_1: + case RT5682_ADC_STO1_HP_CTRL_2: + case RT5682_AJD1_CTRL: + case RT5682_JD1_THD: + case RT5682_JD2_THD: + case RT5682_JD_CTRL_1: + case RT5682_DUMMY_1: + case RT5682_DUMMY_2: + case RT5682_DUMMY_3: + case RT5682_DAC_ADC_DIG_VOL1: + case RT5682_BIAS_CUR_CTRL_2: + case RT5682_BIAS_CUR_CTRL_3: + case RT5682_BIAS_CUR_CTRL_4: + case RT5682_BIAS_CUR_CTRL_5: + case RT5682_BIAS_CUR_CTRL_6: + case RT5682_BIAS_CUR_CTRL_7: + case RT5682_BIAS_CUR_CTRL_8: + case RT5682_BIAS_CUR_CTRL_9: + case RT5682_BIAS_CUR_CTRL_10: + case RT5682_VREF_REC_OP_FB_CAP_CTRL: + case RT5682_CHARGE_PUMP_1: + case RT5682_DIG_IN_CTRL_1: + case RT5682_PAD_DRIVING_CTRL: + case RT5682_SOFT_RAMP_DEPOP: + case RT5682_CHOP_DAC: + case RT5682_CHOP_ADC: + case RT5682_CALIB_ADC_CTRL: + case RT5682_VOL_TEST: + case RT5682_SPKVDD_DET_STA: + case RT5682_TEST_MODE_CTRL_1: + case RT5682_TEST_MODE_CTRL_2: + case RT5682_TEST_MODE_CTRL_3: + case RT5682_TEST_MODE_CTRL_4: + case RT5682_TEST_MODE_CTRL_5: + case RT5682_PLL1_INTERNAL: + case RT5682_PLL2_INTERNAL: + case RT5682_STO_NG2_CTRL_1: + case RT5682_STO_NG2_CTRL_2: + case RT5682_STO_NG2_CTRL_3: + case RT5682_STO_NG2_CTRL_4: + case RT5682_STO_NG2_CTRL_5: + case RT5682_STO_NG2_CTRL_6: + case RT5682_STO_NG2_CTRL_7: + case RT5682_STO_NG2_CTRL_8: + case RT5682_STO_NG2_CTRL_9: + case RT5682_STO_NG2_CTRL_10: + case RT5682_STO1_DAC_SIL_DET: + case RT5682_SIL_PSV_CTRL1: + case RT5682_SIL_PSV_CTRL2: + case RT5682_SIL_PSV_CTRL3: + case RT5682_SIL_PSV_CTRL4: + case RT5682_SIL_PSV_CTRL5: + case RT5682_HP_IMP_SENS_CTRL_01: + case RT5682_HP_IMP_SENS_CTRL_02: + case RT5682_HP_IMP_SENS_CTRL_03: + case RT5682_HP_IMP_SENS_CTRL_04: + case RT5682_HP_IMP_SENS_CTRL_05: + case RT5682_HP_IMP_SENS_CTRL_06: + case RT5682_HP_IMP_SENS_CTRL_07: + case RT5682_HP_IMP_SENS_CTRL_08: + case RT5682_HP_IMP_SENS_CTRL_09: + case RT5682_HP_IMP_SENS_CTRL_10: + case RT5682_HP_IMP_SENS_CTRL_11: + case RT5682_HP_IMP_SENS_CTRL_12: + case RT5682_HP_IMP_SENS_CTRL_13: + case RT5682_HP_IMP_SENS_CTRL_14: + case RT5682_HP_IMP_SENS_CTRL_15: + case RT5682_HP_IMP_SENS_CTRL_16: + case RT5682_HP_IMP_SENS_CTRL_17: + case RT5682_HP_IMP_SENS_CTRL_18: + case RT5682_HP_IMP_SENS_CTRL_19: + case RT5682_HP_IMP_SENS_CTRL_20: + case RT5682_HP_IMP_SENS_CTRL_21: + case RT5682_HP_IMP_SENS_CTRL_22: + case RT5682_HP_IMP_SENS_CTRL_23: + case RT5682_HP_IMP_SENS_CTRL_24: + case RT5682_HP_IMP_SENS_CTRL_25: + case RT5682_HP_IMP_SENS_CTRL_26: + case RT5682_HP_IMP_SENS_CTRL_27: + case RT5682_HP_IMP_SENS_CTRL_28: + case RT5682_HP_IMP_SENS_CTRL_29: + case RT5682_HP_IMP_SENS_CTRL_30: + case RT5682_HP_IMP_SENS_CTRL_31: + case RT5682_HP_IMP_SENS_CTRL_32: + case RT5682_HP_IMP_SENS_CTRL_33: + case RT5682_HP_IMP_SENS_CTRL_34: + case RT5682_HP_IMP_SENS_CTRL_35: + case RT5682_HP_IMP_SENS_CTRL_36: + case RT5682_HP_IMP_SENS_CTRL_37: + case RT5682_HP_IMP_SENS_CTRL_38: + case RT5682_HP_IMP_SENS_CTRL_39: + case RT5682_HP_IMP_SENS_CTRL_40: + case RT5682_HP_IMP_SENS_CTRL_41: + case RT5682_HP_IMP_SENS_CTRL_42: + case RT5682_HP_IMP_SENS_CTRL_43: + case RT5682_HP_LOGIC_CTRL_1: + case RT5682_HP_LOGIC_CTRL_2: + case RT5682_HP_LOGIC_CTRL_3: + case RT5682_HP_CALIB_CTRL_1: + case RT5682_HP_CALIB_CTRL_2: + case RT5682_HP_CALIB_CTRL_3: + case RT5682_HP_CALIB_CTRL_4: + case RT5682_HP_CALIB_CTRL_5: + case RT5682_HP_CALIB_CTRL_6: + case RT5682_HP_CALIB_CTRL_7: + case RT5682_HP_CALIB_CTRL_9: + case RT5682_HP_CALIB_CTRL_10: + case RT5682_HP_CALIB_CTRL_11: + case RT5682_HP_CALIB_STA_1: + case RT5682_HP_CALIB_STA_2: + case RT5682_HP_CALIB_STA_3: + case RT5682_HP_CALIB_STA_4: + case RT5682_HP_CALIB_STA_5: + case RT5682_HP_CALIB_STA_6: + case RT5682_HP_CALIB_STA_7: + case RT5682_HP_CALIB_STA_8: + case RT5682_HP_CALIB_STA_9: + case RT5682_HP_CALIB_STA_10: + case RT5682_HP_CALIB_STA_11: + case RT5682_SAR_IL_CMD_1: + case RT5682_SAR_IL_CMD_2: + case RT5682_SAR_IL_CMD_3: + case RT5682_SAR_IL_CMD_4: + case RT5682_SAR_IL_CMD_5: + case RT5682_SAR_IL_CMD_6: + case RT5682_SAR_IL_CMD_7: + case RT5682_SAR_IL_CMD_8: + case RT5682_SAR_IL_CMD_9: + case RT5682_SAR_IL_CMD_10: + case RT5682_SAR_IL_CMD_11: + case RT5682_SAR_IL_CMD_12: + case RT5682_SAR_IL_CMD_13: + case RT5682_EFUSE_CTRL_1: + case RT5682_EFUSE_CTRL_2: + case RT5682_EFUSE_CTRL_3: + case RT5682_EFUSE_CTRL_4: + case RT5682_EFUSE_CTRL_5: + case RT5682_EFUSE_CTRL_6: + case RT5682_EFUSE_CTRL_7: + case RT5682_EFUSE_CTRL_8: + case RT5682_EFUSE_CTRL_9: + case RT5682_EFUSE_CTRL_10: + case RT5682_EFUSE_CTRL_11: + case RT5682_JD_TOP_VC_VTRL: + case RT5682_DRC1_CTRL_0: + case RT5682_DRC1_CTRL_1: + case RT5682_DRC1_CTRL_2: + case RT5682_DRC1_CTRL_3: + case RT5682_DRC1_CTRL_4: + case RT5682_DRC1_CTRL_5: + case RT5682_DRC1_CTRL_6: + case RT5682_DRC1_HARD_LMT_CTRL_1: + case RT5682_DRC1_HARD_LMT_CTRL_2: + case RT5682_DRC1_PRIV_1: + case RT5682_DRC1_PRIV_2: + case RT5682_DRC1_PRIV_3: + case RT5682_DRC1_PRIV_4: + case RT5682_DRC1_PRIV_5: + case RT5682_DRC1_PRIV_6: + case RT5682_DRC1_PRIV_7: + case RT5682_DRC1_PRIV_8: + case RT5682_EQ_AUTO_RCV_CTRL1: + case RT5682_EQ_AUTO_RCV_CTRL2: + case RT5682_EQ_AUTO_RCV_CTRL3: + case RT5682_EQ_AUTO_RCV_CTRL4: + case RT5682_EQ_AUTO_RCV_CTRL5: + case RT5682_EQ_AUTO_RCV_CTRL6: + case RT5682_EQ_AUTO_RCV_CTRL7: + case RT5682_EQ_AUTO_RCV_CTRL8: + case RT5682_EQ_AUTO_RCV_CTRL9: + case RT5682_EQ_AUTO_RCV_CTRL10: + case RT5682_EQ_AUTO_RCV_CTRL11: + case RT5682_EQ_AUTO_RCV_CTRL12: + case RT5682_EQ_AUTO_RCV_CTRL13: + case RT5682_ADC_L_EQ_LPF1_A1: + case RT5682_R_EQ_LPF1_A1: + case RT5682_L_EQ_LPF1_H0: + case RT5682_R_EQ_LPF1_H0: + case RT5682_L_EQ_BPF1_A1: + case RT5682_R_EQ_BPF1_A1: + case RT5682_L_EQ_BPF1_A2: + case RT5682_R_EQ_BPF1_A2: + case RT5682_L_EQ_BPF1_H0: + case RT5682_R_EQ_BPF1_H0: + case RT5682_L_EQ_BPF2_A1: + case RT5682_R_EQ_BPF2_A1: + case RT5682_L_EQ_BPF2_A2: + case RT5682_R_EQ_BPF2_A2: + case RT5682_L_EQ_BPF2_H0: + case RT5682_R_EQ_BPF2_H0: + case RT5682_L_EQ_BPF3_A1: + case RT5682_R_EQ_BPF3_A1: + case RT5682_L_EQ_BPF3_A2: + case RT5682_R_EQ_BPF3_A2: + case RT5682_L_EQ_BPF3_H0: + case RT5682_R_EQ_BPF3_H0: + case RT5682_L_EQ_BPF4_A1: + case RT5682_R_EQ_BPF4_A1: + case RT5682_L_EQ_BPF4_A2: + case RT5682_R_EQ_BPF4_A2: + case RT5682_L_EQ_BPF4_H0: + case RT5682_R_EQ_BPF4_H0: + case RT5682_L_EQ_HPF1_A1: + case RT5682_R_EQ_HPF1_A1: + case RT5682_L_EQ_HPF1_H0: + case RT5682_R_EQ_HPF1_H0: + case RT5682_L_EQ_PRE_VOL: + case RT5682_R_EQ_PRE_VOL: + case RT5682_L_EQ_POST_VOL: + case RT5682_R_EQ_POST_VOL: + case RT5682_I2C_MODE: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static const DECLARE_TLV_DB_RANGE(bst_tlv, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0) +); + +/* Interface data select */ +static const char * const rt5682_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682_if2_adc_enum, + RT5682_DIG_INF2_DATA, RT5682_IF2_ADC_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_01_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC1_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_23_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC2_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_45_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC3_SEL_SFT, rt5682_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5682_if1_67_adc_enum, + RT5682_TDM_ADDA_CTRL_1, RT5682_IF1_ADC4_SEL_SFT, rt5682_data_select); + +static const struct snd_kcontrol_new rt5682_if2_adc_swap_mux = + SOC_DAPM_ENUM("IF2 ADC Swap Mux", rt5682_if2_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_01_adc_swap_mux = + SOC_DAPM_ENUM("IF1 01 ADC Swap Mux", rt5682_if1_01_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_23_adc_swap_mux = + SOC_DAPM_ENUM("IF1 23 ADC Swap Mux", rt5682_if1_23_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_45_adc_swap_mux = + SOC_DAPM_ENUM("IF1 45 ADC Swap Mux", rt5682_if1_45_adc_enum); + +static const struct snd_kcontrol_new rt5682_if1_67_adc_swap_mux = + SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682_if1_67_adc_enum); + +static void rt5682_reset(struct regmap *regmap) +{ + regmap_write(regmap, RT5682_RESET, 0); + regmap_write(regmap, RT5682_I2C_MODE, 1); +} +/** + * rt5682_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @component: SoC audio component device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5682 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the component driver will turn on + * ASRC for these filters if ASRC is selected as their clock source. + */ +int rt5682_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src) +{ + + switch (clk_src) { + case RT5682_CLK_SEL_SYS: + case RT5682_CLK_SEL_I2S1_ASRC: + case RT5682_CLK_SEL_I2S2_ASRC: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5682_DA_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682_PLL_TRACK_2, + RT5682_FILTER_CLK_SEL_MASK, + clk_src << RT5682_FILTER_CLK_SEL_SFT); + } + + if (filter_mask & RT5682_AD_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5682_PLL_TRACK_3, + RT5682_FILTER_CLK_SEL_MASK, + clk_src << RT5682_FILTER_CLK_SEL_SFT); + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5682_sel_asrc_clk_src); + +static int rt5682_button_detect(struct snd_soc_component *component) +{ + int btn_type, val; + + val = snd_soc_component_read32(component, RT5682_4BTN_IL_CMD_1); + btn_type = val & 0xfff0; + snd_soc_component_write(component, RT5682_4BTN_IL_CMD_1, val); + pr_debug("%s btn_type=%x\n", __func__, btn_type); + + return btn_type; +} + +static void rt5682_enable_push_button_irq(struct snd_soc_component *component, + bool enable) +{ + if (enable) { + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_EN); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13, + RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_BTN); + snd_soc_component_write(component, RT5682_IL_CMD_1, 0x0040); + snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, + RT5682_4BTN_IL_MASK | RT5682_4BTN_IL_RST_MASK, + RT5682_4BTN_IL_EN | RT5682_4BTN_IL_NOR); + snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3, + RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_EN); + } else { + snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3, + RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_DIS); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_DIS); + snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, + RT5682_4BTN_IL_MASK, RT5682_4BTN_IL_DIS); + snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, + RT5682_4BTN_IL_RST_MASK, RT5682_4BTN_IL_RST); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_13, + RT5682_SAR_SOUR_MASK, RT5682_SAR_SOUR_TYPE); + } +} + +/** + * rt5682_headset_detect - Detect headset. + * @component: SoC audio component device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +static int rt5682_headset_detect(struct snd_soc_component *component, + int jack_insert) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + unsigned int val, count; + + if (jack_insert) { + snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH); + + count = 0; + val = snd_soc_component_read32(component, RT5682_CBJ_CTRL_2) + & RT5682_JACK_TYPE_MASK; + while (val == 0 && count < 50) { + usleep_range(10000, 15000); + val = snd_soc_component_read32(component, + RT5682_CBJ_CTRL_2) & RT5682_JACK_TYPE_MASK; + count++; + } + + switch (val) { + case 0x1: + case 0x2: + rt5682->jack_type = SND_JACK_HEADSET; + rt5682_enable_push_button_irq(component, true); + break; + default: + rt5682->jack_type = SND_JACK_HEADPHONE; + } + + } else { + rt5682_enable_push_button_irq(component, false); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + + rt5682->jack_type = 0; + } + + dev_dbg(component->dev, "jack_type = %d\n", rt5682->jack_type); + return rt5682->jack_type; +} + +static irqreturn_t rt5682_irq(int irq, void *data) +{ + struct rt5682_priv *rt5682 = data; + + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + + return IRQ_HANDLED; +} + +static void rt5682_jd_check_handler(struct work_struct *work) +{ + struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv, + jd_check_work.work); + + if (snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL) + & RT5682_JDH_RS_MASK) { + /* jack out */ + rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0); + + snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + } else { + schedule_delayed_work(&rt5682->jd_check_work, 500); + } +} + +static int rt5682_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + switch (rt5682->pdata.jd_src) { + case RT5682_JD1: + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2, + RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); + snd_soc_component_write(component, RT5682_CBJ_CTRL_1, 0xd042); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_3, + RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_IRQ | RT5682_POW_JDH | + RT5682_POW_ANA, RT5682_POW_IRQ | + RT5682_POW_JDH | RT5682_POW_ANA); + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2, + RT5682_PWR_JDH | RT5682_PWR_JDL, + RT5682_PWR_JDH | RT5682_PWR_JDL); + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK, + RT5682_JD1_EN | RT5682_JD1_POL_NOR); + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + break; + + case RT5682_JD_NULL: + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + break; + + default: + dev_warn(component->dev, "Wrong JD source\n"); + break; + } + + rt5682->hs_jack = hs_jack; + + return 0; +} + +static void rt5682_jack_detect_handler(struct work_struct *work) +{ + struct rt5682_priv *rt5682 = + container_of(work, struct rt5682_priv, jack_detect_work.work); + int val, btn_type; + + while (!rt5682->component) + usleep_range(10000, 15000); + + while (!rt5682->component->card->instantiated) + usleep_range(10000, 15000); + + mutex_lock(&rt5682->calibrate_mutex); + + val = snd_soc_component_read32(rt5682->component, RT5682_AJD1_CTRL) + & RT5682_JDH_RS_MASK; + if (!val) { + /* jack in */ + if (rt5682->jack_type == 0) { + /* jack was out, report jack type */ + rt5682->jack_type = + rt5682_headset_detect(rt5682->component, 1); + } else { + /* jack is already in, report button event */ + rt5682->jack_type = SND_JACK_HEADSET; + btn_type = rt5682_button_detect(rt5682->component); + /** + * rt5682 can report three kinds of button behavior, + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + rt5682->jack_type |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + rt5682->jack_type |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + rt5682->jack_type |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + rt5682->jack_type |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + btn_type = 0; + dev_err(rt5682->component->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + } else { + /* jack out */ + rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0); + } + + snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3)) + schedule_delayed_work(&rt5682->jd_check_work, 0); + else + cancel_delayed_work_sync(&rt5682->jd_check_work); + + mutex_unlock(&rt5682->calibrate_mutex); +} + +static const struct snd_kcontrol_new rt5682_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN, + RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, + RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 175, 0, dac_vol_tlv), + + /* IN Boost Volume */ + SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL, + RT5682_BST_CBJ_SFT, 8, 0, bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("STO1 ADC Capture Switch", RT5682_STO1_ADC_DIG_VOL, + RT5682_L_MUTE_SFT, RT5682_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5682_STO1_ADC_DIG_VOL, + RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682_STO1_ADC_BOOST, + RT5682_STO1_ADC_L_BST_SFT, RT5682_STO1_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), +}; + + +static int rt5682_div_sel(struct rt5682_priv *rt5682, + int target, const int div[], int size) +{ + int i; + + if (rt5682->sysclk < target) { + pr_err("sysclk rate %d is too low\n", + rt5682->sysclk); + return 0; + } + + for (i = 0; i < size - 1; i++) { + pr_info("div[%d]=%d\n", i, div[i]); + if (target * div[i] == rt5682->sysclk) + return i; + if (target * div[i + 1] > rt5682->sysclk) { + pr_err("can't find div for sysclk %d\n", + rt5682->sysclk); + return i; + } + } + + if (target * div[i] < rt5682->sysclk) + pr_err("sysclk rate %d is too high\n", + rt5682->sysclk); + + return size - 1; + +} + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + int idx = -EINVAL; + static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128}; + + idx = rt5682_div_sel(rt5682, 1500000, div, ARRAY_SIZE(div)); + + snd_soc_component_update_bits(component, RT5682_DMIC_CTRL_1, + RT5682_DMIC_CLK_MASK, idx << RT5682_DMIC_CLK_SFT); + + return 0; +} + +static int set_filter_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + int ref, val, reg, sft, mask, idx = -EINVAL; + static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; + static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48}; + + val = snd_soc_component_read32(component, RT5682_GPIO_CTRL_1) & + RT5682_GP4_PIN_MASK; + if (w->shift == RT5682_PWR_ADC_S1F_BIT && + val == RT5682_GP4_PIN_ADCDAT2) + ref = 256 * rt5682->lrck[RT5682_AIF2]; + else + ref = 256 * rt5682->lrck[RT5682_AIF1]; + + idx = rt5682_div_sel(rt5682, ref, div_f, ARRAY_SIZE(div_f)); + + if (w->shift == RT5682_PWR_ADC_S1F_BIT) { + reg = RT5682_PLL_TRACK_3; + sft = RT5682_ADC_OSR_SFT; + mask = RT5682_ADC_OSR_MASK; + } else { + reg = RT5682_PLL_TRACK_2; + sft = RT5682_DAC_OSR_SFT; + mask = RT5682_DAC_OSR_MASK; + } + + snd_soc_component_update_bits(component, reg, + RT5682_FILTER_CLK_DIV_MASK, idx << RT5682_FILTER_CLK_DIV_SFT); + + /* select over sample rate */ + for (idx = 0; idx < ARRAY_SIZE(div_o); idx++) { + if (rt5682->sysclk <= 12288000 * div_o[idx]) + break; + } + + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1, + mask, idx << sft); + + return 0; +} + +static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + val = snd_soc_component_read32(component, RT5682_GLB_CLK); + val &= RT5682_SCLK_SRC_MASK; + if (val == RT5682_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int is_using_asrc(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (w->shift) { + case RT5682_ADC_STO1_ASRC_SFT: + reg = RT5682_PLL_TRACK_3; + shift = RT5682_FILTER_CLK_SEL_SFT; + break; + case RT5682_DAC_STO1_ASRC_SFT: + reg = RT5682_PLL_TRACK_2; + shift = RT5682_FILTER_CLK_SEL_SFT; + break; + default: + return 0; + } + + val = (snd_soc_component_read32(component, reg) >> shift) & 0xf; + switch (val) { + case RT5682_CLK_SEL_I2S1_ASRC: + case RT5682_CLK_SEL_I2S2_ASRC: + return 1; + default: + return 0; + } + +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5682_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5682_STO1_ADC_MIXER, + RT5682_M_STO1_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER, + RT5682_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER, + RT5682_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5682_AD_DA_MIXER, + RT5682_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5682_AD_DA_MIXER, + RT5682_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_L1_STO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5682_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_L1_STO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5682_STO1_DAC_MIXER, + RT5682_M_DAC_R1_STO_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5682_rec1_l_mix[] = { + SOC_DAPM_SINGLE("CBJ Switch", RT5682_REC_MIXER, + RT5682_M_CBJ_RM1_L_SFT, 1, 1), +}; + +/* STO1 ADC1 Source */ +/* MX-26 [13] [5] */ +static const char * const rt5682_sto1_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc1l_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC1L_SRC_SFT, rt5682_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc1l_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc1r_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC1R_SRC_SFT, rt5682_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc1r_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5682_sto1_adc1r_enum); + +/* STO1 ADC Source */ +/* MX-26 [11:10] [3:2] */ +static const char * const rt5682_sto1_adc_src[] = { + "ADC1 L", "ADC1 R" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adcl_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADCL_SRC_SFT, rt5682_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682_sto1_adcl_mux = + SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5682_sto1_adcl_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adcr_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADCR_SRC_SFT, rt5682_sto1_adc_src); + +static const struct snd_kcontrol_new rt5682_sto1_adcr_mux = + SOC_DAPM_ENUM("Stereo1 ADCR Source", rt5682_sto1_adcr_enum); + +/* STO1 ADC2 Source */ +/* MX-26 [12] [4] */ +static const char * const rt5682_sto1_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc2l_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC2L_SRC_SFT, rt5682_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc2l_mux = + SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5682_sto1_adc2l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_sto1_adc2r_enum, RT5682_STO1_ADC_MIXER, + RT5682_STO1_ADC2R_SRC_SFT, rt5682_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5682_sto1_adc2r_mux = + SOC_DAPM_ENUM("Stereo1 ADC2R Source", rt5682_sto1_adc2r_enum); + +/* MX-79 [6:4] I2S1 ADC data location */ +static const unsigned int rt5682_if1_adc_slot_values[] = { + 0, + 2, + 4, + 6, +}; + +static const char * const rt5682_if1_adc_slot_src[] = { + "Slot 0", "Slot 2", "Slot 4", "Slot 6" +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_if1_adc_slot_enum, + RT5682_TDM_CTRL, RT5682_TDM_ADC_LCA_SFT, RT5682_TDM_ADC_LCA_MASK, + rt5682_if1_adc_slot_src, rt5682_if1_adc_slot_values); + +static const struct snd_kcontrol_new rt5682_if1_adc_slot_mux = + SOC_DAPM_ENUM("IF1 ADC Slot location", rt5682_if1_adc_slot_enum); + +/* Analog DAC L1 Source, Analog DAC R1 Source*/ +/* MX-2B [4], MX-2B [0]*/ +static const char * const rt5682_alg_dac1_src[] = { + "Stereo1 DAC Mixer", "DAC1" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5682_alg_dac_l1_enum, RT5682_A_DAC1_MUX, + RT5682_A_DACL1_SFT, rt5682_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682_alg_dac_l1_mux = + SOC_DAPM_ENUM("Analog DAC L1 Source", rt5682_alg_dac_l1_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5682_alg_dac_r1_enum, RT5682_A_DAC1_MUX, + RT5682_A_DACR1_SFT, rt5682_alg_dac1_src); + +static const struct snd_kcontrol_new rt5682_alg_dac_r1_mux = + SOC_DAPM_ENUM("Analog DAC R1 Source", rt5682_alg_dac_r1_enum); + +/* Out Switch */ +static const struct snd_kcontrol_new hpol_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, + RT5682_L_MUTE_SFT, 1, 1); +static const struct snd_kcontrol_new hpor_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, + RT5682_R_MUTE_SFT, 1, 1); + +static int rt5682_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write(component, + RT5682_HP_LOGIC_CTRL_2, 0x0012); + snd_soc_component_write(component, + RT5682_HP_CTRL_2, 0x6000); + snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1, + RT5682_NG2_EN_MASK, RT5682_NG2_EN); + snd_soc_component_update_bits(component, + RT5682_DEPOP_1, 0x60, 0x60); + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, + RT5682_DEPOP_1, 0x60, 0x0); + snd_soc_component_write(component, + RT5682_HP_CTRL_2, 0x0000); + break; + + default: + return 0; + } + + return 0; + +} + +static int set_dmic_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /*Add delay to avoid pop noise*/ + msleep(150); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5655_set_verf(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (w->shift) { + case RT5682_PWR_VREF1_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV1, 0); + break; + + case RT5682_PWR_VREF2_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); + break; + + default: + break; + } + break; + + case SND_SOC_DAPM_POST_PMU: + usleep_range(15000, 20000); + switch (w->shift) { + case RT5682_PWR_VREF1_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV1, + RT5682_PWR_FV1); + break; + + case RT5682_PWR_VREF2_BIT: + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, + RT5682_PWR_FV2); + break; + + default: + break; + } + break; + + default: + return 0; + } + + return 0; +} + +static const unsigned int rt5682_adcdat_pin_values[] = { + 1, + 3, +}; + +static const char * const rt5682_adcdat_pin_select[] = { + "ADCDAT1", + "ADCDAT2", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_adcdat_pin_enum, + RT5682_GPIO_CTRL_1, RT5682_GP4_PIN_SFT, RT5682_GP4_PIN_MASK, + rt5682_adcdat_pin_select, rt5682_adcdat_pin_values); + +static const struct snd_kcontrol_new rt5682_adcdat_pin_ctrl = + SOC_DAPM_ENUM("ADCDAT", rt5682_adcdat_pin_enum); + +static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5682_PWR_ANLG_3, RT5682_PWR_LDO2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT5682_PWR_ANLG_3, RT5682_PWR_PLL_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, + rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, + rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_DAC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AD ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_AD_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DA ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_DA_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC ASRC", 1, RT5682_PLL_TRACK_1, + RT5682_DMIC_ASRC_SFT, 0, NULL, 0), + + /* Input Side */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5682_PWR_ANLG_2, RT5682_PWR_MB1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5682_PWR_ANLG_2, RT5682_PWR_MB2_BIT, + 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + + SND_SOC_DAPM_INPUT("IN1P"), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5682_DMIC_CTRL_1, + RT5682_DMIC_1_EN_SFT, 0, set_dmic_power, SND_SOC_DAPM_POST_PMU), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, + 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("CBJ Power", RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ_BIT, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682_rec1_l_mix, + ARRAY_SIZE(rt5682_rec1_l_mix)), + SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5682_PWR_ANLG_2, + RT5682_PWR_RM1_L_BIT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1 L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC1 R", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SUPPLY("ADC1 L Power", RT5682_PWR_DIG_1, + RT5682_PWR_ADC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 R Power", RT5682_PWR_DIG_1, + RT5682_PWR_ADC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 clock", RT5682_CHOP_ADC, + RT5682_CKGEN_ADC1_SFT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc1l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc1r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc2l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adc2r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adcl_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R Mux", SND_SOC_NOPM, 0, 0, + &rt5682_sto1_adcr_mux), + SND_SOC_DAPM_MUX("IF1_ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_adc_slot_mux), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5682_PWR_DIG_2, + RT5682_PWR_ADC_S1F_BIT, 0, set_filter_clk, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5682_STO1_ADC_DIG_VOL, + RT5682_L_MUTE_SFT, 1, rt5682_sto1_adc_l_mix, + ARRAY_SIZE(rt5682_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682_STO1_ADC_DIG_VOL, + RT5682_R_MUTE_SFT, 1, rt5682_sto1_adc_r_mix, + ARRAY_SIZE(rt5682_sto1_adc_r_mix)), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5682_PWR_DIG_1, RT5682_PWR_I2S1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5682_PWR_DIG_1, RT5682_PWR_I2S2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_01_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 23 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_23_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 45 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_45_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 67 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if1_67_adc_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5682_if2_adc_swap_mux), + + SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0, + &rt5682_adcdat_pin_ctrl), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, + RT5682_I2S1_SDP, RT5682_SEL_ADCDAT_SFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, + RT5682_I2S2_SDP, RT5682_I2S2_PIN_CFG_SFT, 1), + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5682_dac_l_mix, ARRAY_SIZE(rt5682_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5682_dac_r_mix, ARRAY_SIZE(rt5682_dac_r_mix)), + + /* DAC channel Mux */ + SND_SOC_DAPM_MUX("DAC L1 Source", SND_SOC_NOPM, 0, 0, + &rt5682_alg_dac_l1_mux), + SND_SOC_DAPM_MUX("DAC R1 Source", SND_SOC_NOPM, 0, 0, + &rt5682_alg_dac_r1_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5682_PWR_DIG_2, + RT5682_PWR_DAC_S1F_BIT, 0, set_filter_clk, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5682_sto1_dac_l_mix, ARRAY_SIZE(rt5682_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5682_sto1_dac_r_mix, ARRAY_SIZE(rt5682_sto1_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5682_PWR_DIG_1, + RT5682_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5682_PWR_DIG_1, + RT5682_PWR_DAC_R1_BIT, 0), + SND_SOC_DAPM_SUPPLY_S("DAC 1 Clock", 3, RT5682_CHOP_DAC, + RT5682_CKGEN_DAC1_SFT, 0, NULL, 0), + + /* HPO */ + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5682_hp_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_SUPPLY("HP Amp L", RT5682_PWR_ANLG_1, + RT5682_PWR_HA_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1, + RT5682_PWR_HA_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1, + RT5682_PUMP_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1, + RT5682_CAPLESS_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("HPOL Playback", SND_SOC_NOPM, 0, 0, + &hpol_switch), + SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0, + &hpor_switch), + + /* CLK DET */ + SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682_CLK_DET, + RT5682_SYS_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL1", RT5682_CLK_DET, + RT5682_PLL1_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL2", RT5682_CLK_DET, + RT5682_PLL2_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET", RT5682_CLK_DET, + RT5682_POW_CLK_DET_SFT, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + +}; + +static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { + /*PLL*/ + {"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + + /*ASRC*/ + {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, + {"ADC STO1 ASRC", NULL, "AD ASRC"}, + {"ADC STO1 ASRC", NULL, "CLKDET"}, + {"DAC STO1 ASRC", NULL, "DA ASRC"}, + {"DAC STO1 ASRC", NULL, "CLKDET"}, + + /*Vref*/ + {"MICBIAS1", NULL, "Vref1"}, + {"MICBIAS1", NULL, "Vref2"}, + {"MICBIAS2", NULL, "Vref1"}, + {"MICBIAS2", NULL, "Vref2"}, + + {"CLKDET SYS", NULL, "CLKDET"}, + + {"IN1P", NULL, "LDO2"}, + + {"BST1 CBJ", NULL, "IN1P"}, + {"BST1 CBJ", NULL, "CBJ Power"}, + {"CBJ Power", NULL, "Vref2"}, + + {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, + {"RECMIX1L", NULL, "RECMIX1L Power"}, + + {"ADC1 L", NULL, "RECMIX1L"}, + {"ADC1 L", NULL, "ADC1 L Power"}, + {"ADC1 L", NULL, "ADC1 clock"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC CLK", NULL, "DMIC ASRC"}, + + {"Stereo1 ADC L Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC L Mux", "ADC1 R", "ADC1 R"}, + {"Stereo1 ADC R Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC R Mux", "ADC1 R", "ADC1 R"}, + + {"Stereo1 ADC L1 Mux", "ADC", "Stereo1 ADC L Mux"}, + {"Stereo1 ADC L1 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + {"Stereo1 ADC L2 Mux", "DMIC", "DMIC L1"}, + {"Stereo1 ADC L2 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + + {"Stereo1 ADC R1 Mux", "ADC", "Stereo1 ADC R Mux"}, + {"Stereo1 ADC R1 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + {"Stereo1 ADC R2 Mux", "DMIC", "DMIC R1"}, + {"Stereo1 ADC R2 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + + {"Stereo1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux"}, + {"Stereo1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux"}, + {"Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux"}, + {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, + {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, + + {"IF1 01 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + + {"IF1_ADC Mux", "Slot 0", "IF1 01 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"}, + {"IF1_ADC Mux", NULL, "I2S1"}, + {"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"}, + {"AIF1TX", NULL, "ADCDAT Mux"}, + {"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"}, + {"AIF2TX", NULL, "ADCDAT Mux"}, + + {"IF1 DAC1 L", NULL, "AIF1RX"}, + {"IF1 DAC1 L", NULL, "I2S1"}, + {"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"}, + {"IF1 DAC1 R", NULL, "AIF1RX"}, + {"IF1 DAC1 R", NULL, "I2S1"}, + {"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"}, + + {"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, + {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"}, + {"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, + {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"}, + + {"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"}, + {"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"}, + + {"Stereo1 DAC MIXR", "DAC R1 Switch", "DAC1 MIXR"}, + {"Stereo1 DAC MIXR", "DAC L1 Switch", "DAC1 MIXL"}, + + {"DAC L1 Source", "DAC1", "DAC1 MIXL"}, + {"DAC L1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXL"}, + {"DAC R1 Source", "DAC1", "DAC1 MIXR"}, + {"DAC R1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXR"}, + + {"DAC L1", NULL, "DAC L1 Source"}, + {"DAC R1", NULL, "DAC R1 Source"}, + + {"DAC L1", NULL, "DAC 1 Clock"}, + {"DAC R1", NULL, "DAC 1 Clock"}, + + {"HP Amp", NULL, "DAC L1"}, + {"HP Amp", NULL, "DAC R1"}, + {"HP Amp", NULL, "HP Amp L"}, + {"HP Amp", NULL, "HP Amp R"}, + {"HP Amp", NULL, "Capless"}, + {"HP Amp", NULL, "Charge Pump"}, + {"HP Amp", NULL, "CLKDET SYS"}, + {"HP Amp", NULL, "CBJ Power"}, + {"HP Amp", NULL, "Vref2"}, + {"HPOL Playback", "Switch", "HP Amp"}, + {"HPOR Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HPOL Playback"}, + {"HPOR", NULL, "HPOR Playback"}, +}; + +static int rt5682_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int cl, val = 0; + + if (tx_mask || rx_mask) + snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2, + RT5682_TDM_EN, RT5682_TDM_EN); + else + snd_soc_component_update_bits(component, RT5682_TDM_ADDA_CTRL_2, + RT5682_TDM_EN, 0); + + switch (slots) { + case 4: + val |= RT5682_TDM_TX_CH_4; + val |= RT5682_TDM_RX_CH_4; + break; + case 6: + val |= RT5682_TDM_TX_CH_6; + val |= RT5682_TDM_RX_CH_6; + break; + case 8: + val |= RT5682_TDM_TX_CH_8; + val |= RT5682_TDM_RX_CH_8; + break; + case 2: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682_TDM_CTRL, + RT5682_TDM_TX_CH_MASK | RT5682_TDM_RX_CH_MASK, val); + + switch (slot_width) { + case 8: + if (tx_mask || rx_mask) + return -EINVAL; + cl = RT5682_I2S1_TX_CHL_8 | RT5682_I2S1_RX_CHL_8; + break; + case 16: + val = RT5682_TDM_CL_16; + cl = RT5682_I2S1_TX_CHL_16 | RT5682_I2S1_RX_CHL_16; + break; + case 20: + val = RT5682_TDM_CL_20; + cl = RT5682_I2S1_TX_CHL_20 | RT5682_I2S1_RX_CHL_20; + break; + case 24: + val = RT5682_TDM_CL_24; + cl = RT5682_I2S1_TX_CHL_24 | RT5682_I2S1_RX_CHL_24; + break; + case 32: + val = RT5682_TDM_CL_32; + cl = RT5682_I2S1_TX_CHL_32 | RT5682_I2S1_RX_CHL_32; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_CL_MASK, val); + snd_soc_component_update_bits(component, RT5682_I2S1_SDP, + RT5682_I2S1_TX_CHL_MASK | RT5682_I2S1_RX_CHL_MASK, cl); + + return 0; +} + + +static int rt5682_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int len_1 = 0, len_2 = 0; + int pre_div, frame_size; + + rt5682->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5682->sysclk, rt5682->lrck[dai->id]); + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", + frame_size); + return -EINVAL; + } + + dev_dbg(dai->dev, "lrck is %dHz and pre_div is %d for iis %d\n", + rt5682->lrck[dai->id], pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + len_1 |= RT5682_I2S1_DL_20; + len_2 |= RT5682_I2S2_DL_20; + break; + case 24: + len_1 |= RT5682_I2S1_DL_24; + len_2 |= RT5682_I2S2_DL_24; + break; + case 32: + len_1 |= RT5682_I2S1_DL_32; + len_2 |= RT5682_I2S2_DL_24; + break; + case 8: + len_1 |= RT5682_I2S2_DL_8; + len_2 |= RT5682_I2S2_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682_AIF1: + snd_soc_component_update_bits(component, RT5682_I2S1_SDP, + RT5682_I2S1_DL_MASK, len_1); + if (rt5682->master[RT5682_AIF1]) { + snd_soc_component_update_bits(component, + RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK, + pre_div << RT5682_I2S_M_DIV_SFT); + } + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, + RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK, + RT5682_I2S1_MONO_EN); + else + snd_soc_component_update_bits(component, + RT5682_I2S1_SDP, RT5682_I2S1_MONO_MASK, + RT5682_I2S1_MONO_DIS); + break; + case RT5682_AIF2: + snd_soc_component_update_bits(component, RT5682_I2S2_SDP, + RT5682_I2S2_DL_MASK, len_2); + if (rt5682->master[RT5682_AIF2]) { + snd_soc_component_update_bits(component, + RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_M_PD_MASK, + pre_div << RT5682_I2S2_M_PD_SFT); + } + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, + RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK, + RT5682_I2S2_MONO_EN); + else + snd_soc_component_update_bits(component, + RT5682_I2S2_SDP, RT5682_I2S2_MONO_MASK, + RT5682_I2S2_MONO_DIS); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5682_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, tdm_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5682->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt5682->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5682_I2S_BP_INV; + tdm_ctrl |= RT5682_TDM_S_BP_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + if (dai->id == RT5682_AIF1) + tdm_ctrl |= RT5682_TDM_S_LP_INV | RT5682_TDM_M_BP_INV; + else + return -EINVAL; + break; + case SND_SOC_DAIFMT_IB_IF: + if (dai->id == RT5682_AIF1) + tdm_ctrl |= RT5682_TDM_S_BP_INV | RT5682_TDM_S_LP_INV | + RT5682_TDM_M_BP_INV | RT5682_TDM_M_LP_INV; + else + return -EINVAL; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5682_I2S_DF_LEFT; + tdm_ctrl |= RT5682_TDM_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5682_I2S_DF_PCM_A; + tdm_ctrl |= RT5682_TDM_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5682_I2S_DF_PCM_B; + tdm_ctrl |= RT5682_TDM_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5682_AIF1: + snd_soc_component_update_bits(component, RT5682_I2S1_SDP, + RT5682_I2S_DF_MASK, reg_val); + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_MS_MASK | RT5682_TDM_S_BP_MASK | + RT5682_TDM_DF_MASK | RT5682_TDM_M_BP_MASK | + RT5682_TDM_M_LP_MASK | RT5682_TDM_S_LP_MASK, + tdm_ctrl | rt5682->master[dai->id]); + break; + case RT5682_AIF2: + if (rt5682->master[dai->id] == 0) + reg_val |= RT5682_I2S2_MS_S; + snd_soc_component_update_bits(component, RT5682_I2S2_SDP, + RT5682_I2S2_MS_MASK | RT5682_I2S_BP_MASK | + RT5682_I2S_DF_MASK, reg_val); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5682_set_component_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, src = 0; + + if (freq == rt5682->sysclk && clk_id == rt5682->sysclk_src) + return 0; + + switch (clk_id) { + case RT5682_SCLK_S_MCLK: + reg_val |= RT5682_SCLK_SRC_MCLK; + src = RT5682_CLK_SRC_MCLK; + break; + case RT5682_SCLK_S_PLL1: + reg_val |= RT5682_SCLK_SRC_PLL1; + src = RT5682_CLK_SRC_PLL1; + break; + case RT5682_SCLK_S_PLL2: + reg_val |= RT5682_SCLK_SRC_PLL2; + src = RT5682_CLK_SRC_PLL2; + break; + case RT5682_SCLK_S_RCCLK: + reg_val |= RT5682_SCLK_SRC_RCCLK; + src = RT5682_CLK_SRC_RCCLK; + break; + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_SCLK_SRC_MASK, reg_val); + + if (rt5682->master[RT5682_AIF2]) { + snd_soc_component_update_bits(component, + RT5682_I2S_M_CLK_CTRL_1, RT5682_I2S2_SRC_MASK, + src << RT5682_I2S2_SRC_SFT); + } + + rt5682->sysclk = freq; + rt5682->sysclk_src = clk_id; + + dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static int rt5682_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5682->pll_src && freq_in == rt5682->pll_in && + freq_out == rt5682->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + + rt5682->pll_in = 0; + rt5682->pll_out = 0; + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_SCLK_SRC_MASK, RT5682_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5682_PLL1_S_MCLK: + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_MCLK); + break; + case RT5682_PLL1_S_BCLK1: + snd_soc_component_update_bits(component, RT5682_GLB_CLK, + RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_BCLK1); + break; + default: + dev_err(component->dev, "Unknown PLL Source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_write(component, RT5682_PLL_CTRL_1, + pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code); + snd_soc_component_write(component, RT5682_PLL_CTRL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT | + pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST); + + rt5682->pll_in = freq_in; + rt5682->pll_out = freq_out; + rt5682->pll_src = source; + + return 0; +} + +static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + rt5682->bclk[dai->id] = ratio; + + switch (ratio) { + case 64: + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2, + RT5682_I2S2_BCLK_MS2_MASK, + RT5682_I2S2_BCLK_MS2_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_2, + RT5682_I2S2_BCLK_MS2_MASK, + RT5682_I2S2_BCLK_MS2_32); + break; + default: + dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio); + return -EINVAL; + } + + return 0; +} + +static int rt5682_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_PWR_MB | RT5682_PWR_BG, + RT5682_PWR_MB | RT5682_PWR_BG); + regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, + RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO, + RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_PWR_MB, RT5682_PWR_MB); + regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, + RT5682_DIG_GATE_CTRL, RT5682_DIG_GATE_CTRL); + break; + case SND_SOC_BIAS_OFF: + regmap_update_bits(rt5682->regmap, RT5682_PWR_DIG_1, + RT5682_DIG_GATE_CTRL | RT5682_PWR_LDO, 0); + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_PWR_MB | RT5682_PWR_BG, 0); + break; + + default: + break; + } + + return 0; +} + +static int rt5682_probe(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + rt5682->component = component; + + return 0; +} + +static void rt5682_remove(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + rt5682_reset(rt5682->regmap); +} + +#ifdef CONFIG_PM +static int rt5682_suspend(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5682->regmap, true); + regcache_mark_dirty(rt5682->regmap); + return 0; +} + +static int rt5682_resume(struct snd_soc_component *component) +{ + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5682->regmap, false); + regcache_sync(rt5682->regmap); + + return 0; +} +#else +#define rt5682_suspend NULL +#define rt5682_resume NULL +#endif + +#define RT5682_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5682_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt5682_aif1_dai_ops = { + .hw_params = rt5682_hw_params, + .set_fmt = rt5682_set_dai_fmt, + .set_tdm_slot = rt5682_set_tdm_slot, +}; + +static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = { + .hw_params = rt5682_hw_params, + .set_fmt = rt5682_set_dai_fmt, + .set_bclk_ratio = rt5682_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt5682_dai[] = { + { + .name = "rt5682-aif1", + .id = RT5682_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .ops = &rt5682_aif1_dai_ops, + }, + { + .name = "rt5682-aif2", + .id = RT5682_AIF2, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .ops = &rt5682_aif2_dai_ops, + }, +}; + +static const struct snd_soc_component_driver soc_component_dev_rt5682 = { + .probe = rt5682_probe, + .remove = rt5682_remove, + .suspend = rt5682_suspend, + .resume = rt5682_resume, + .set_bias_level = rt5682_set_bias_level, + .controls = rt5682_snd_controls, + .num_controls = ARRAY_SIZE(rt5682_snd_controls), + .dapm_widgets = rt5682_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5682_dapm_widgets), + .dapm_routes = rt5682_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5682_dapm_routes), + .set_sysclk = rt5682_set_component_sysclk, + .set_pll = rt5682_set_component_pll, + .set_jack = rt5682_set_jack_detect, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config rt5682_regmap = { + .reg_bits = 16, + .val_bits = 16, + .max_register = RT5682_I2C_MODE, + .volatile_reg = rt5682_volatile_register, + .readable_reg = rt5682_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5682_reg, + .num_reg_defaults = ARRAY_SIZE(rt5682_reg), + .use_single_rw = true, +}; + +static const struct i2c_device_id rt5682_i2c_id[] = { + {"rt5682", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5682_i2c_id); + +static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) +{ + + device_property_read_u32(dev, "realtek,dmic1-data-pin", + &rt5682->pdata.dmic1_data_pin); + device_property_read_u32(dev, "realtek,dmic1-clk-pin", + &rt5682->pdata.dmic1_clk_pin); + device_property_read_u32(dev, "realtek,jd-src", + &rt5682->pdata.jd_src); + + rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, + "realtek,ldo1-en-gpios", 0); + + return 0; +} + +static void rt5682_calibrate(struct rt5682_priv *rt5682) +{ + int value, count; + + mutex_lock(&rt5682->calibrate_mutex); + + rt5682_reset(rt5682->regmap); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf); + usleep_range(15000, 20000); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8001); + regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + regmap_write(rt5682->regmap, RT5682_STO1_DAC_MIXER, 0x2080); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x4040); + regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0069); + regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); + regmap_write(rt5682->regmap, RT5682_HP_CTRL_2, 0x6000); + regmap_write(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, 0x0f26); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7f05); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); + regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_9, 0x000f); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8d01); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); + regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1); + regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311); + regmap_write(rt5682->regmap, RT5682_RESET_HPF_CTRL, 0x0000); + regmap_write(rt5682->regmap, RT5682_ADC_STO1_HP_CTRL_1, 0x3320); + + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00); + + for (count = 0; count < 60; count++) { + regmap_read(rt5682->regmap, RT5682_HP_CALIB_STA_1, &value); + if (!(value & 0x8000)) + break; + + usleep_range(10000, 10005); + } + + if (count >= 60) + pr_err("HP Calibration Failure\n"); + + /* restore settings */ + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); + + mutex_unlock(&rt5682->calibrate_mutex); + +} + +static int rt5682_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5682_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5682_priv *rt5682; + int i, ret; + unsigned int val; + + rt5682 = devm_kzalloc(&i2c->dev, sizeof(struct rt5682_priv), + GFP_KERNEL); + + if (rt5682 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5682); + + if (pdata) + rt5682->pdata = *pdata; + else + rt5682_parse_dt(rt5682, &i2c->dev); + + rt5682->regmap = devm_regmap_init_i2c(i2c, &rt5682_regmap); + if (IS_ERR(rt5682->regmap)) { + ret = PTR_ERR(rt5682->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(rt5682->supplies); i++) + rt5682->supplies[i].supply = rt5682_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(rt5682->supplies), + rt5682->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(rt5682->supplies), + rt5682->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + if (gpio_is_valid(rt5682->pdata.ldo1_en)) { + if (devm_gpio_request_one(&i2c->dev, rt5682->pdata.ldo1_en, + GPIOF_OUT_INIT_HIGH, "rt5682")) + dev_err(&i2c->dev, "Fail gpio_request gpio_ldo\n"); + } + + /* Sleep for 300 ms miniumum */ + usleep_range(300000, 350000); + + regmap_write(rt5682->regmap, RT5682_I2C_MODE, 0x1); + usleep_range(10000, 15000); + + regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val); + if (val != DEVICE_ID) { + pr_err("Device with ID register %x is not rt5682\n", val); + return -ENODEV; + } + + rt5682_reset(rt5682->regmap); + + rt5682_calibrate(rt5682); + + ret = regmap_register_patch(rt5682->regmap, patch_list, + ARRAY_SIZE(patch_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0000); + + /* DMIC pin*/ + if (rt5682->pdata.dmic1_data_pin != RT5682_DMIC1_NULL) { + switch (rt5682->pdata.dmic1_data_pin) { + case RT5682_DMIC1_DATA_GPIO2: /* share with LRCK2 */ + regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1, + RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO2); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP2_PIN_MASK, RT5682_GP2_PIN_DMIC_SDA); + break; + + case RT5682_DMIC1_DATA_GPIO5: /* share with DACDAT1 */ + regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1, + RT5682_DMIC_1_DP_MASK, RT5682_DMIC_1_DP_GPIO5); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP5_PIN_MASK, RT5682_GP5_PIN_DMIC_SDA); + break; + + default: + dev_warn(&i2c->dev, "invalid DMIC_DAT pin\n"); + break; + } + + switch (rt5682->pdata.dmic1_clk_pin) { + case RT5682_DMIC1_CLK_GPIO1: /* share with IRQ */ + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_DMIC_CLK); + break; + + case RT5682_DMIC1_CLK_GPIO3: /* share with BCLK2 */ + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP3_PIN_MASK, RT5682_GP3_PIN_DMIC_CLK); + break; + + default: + dev_warn(&i2c->dev, "invalid DMIC_CLK pin\n"); + break; + } + } + + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK, + RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK, + RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1); + regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + + INIT_DELAYED_WORK(&rt5682->jack_detect_work, + rt5682_jack_detect_handler); + INIT_DELAYED_WORK(&rt5682->jd_check_work, + rt5682_jd_check_handler); + + mutex_init(&rt5682->calibrate_mutex); + + if (i2c->irq) { + ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, + rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5682", rt5682); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + + } + + return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682, + rt5682_dai, ARRAY_SIZE(rt5682_dai)); +} + +static int rt5682_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + + return 0; +} + +static void rt5682_i2c_shutdown(struct i2c_client *client) +{ + struct rt5682_priv *rt5682 = i2c_get_clientdata(client); + + rt5682_reset(rt5682->regmap); +} + +#ifdef CONFIG_OF +static const struct of_device_id rt5682_of_match[] = { + {.compatible = "realtek,rt5682i"}, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5682_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5682_acpi_match[] = { + {"10EC5682", 0,}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5682_acpi_match); +#endif + +static struct i2c_driver rt5682_i2c_driver = { + .driver = { + .name = "rt5682", + .of_match_table = of_match_ptr(rt5682_of_match), + .acpi_match_table = ACPI_PTR(rt5682_acpi_match), + }, + .probe = rt5682_i2c_probe, + .remove = rt5682_i2c_remove, + .shutdown = rt5682_i2c_shutdown, + .id_table = rt5682_i2c_id, +}; +module_i2c_driver(rt5682_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5682 driver"); +MODULE_AUTHOR("Bard Liao "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h new file mode 100644 index 000000000000..8068140ebe3f --- /dev/null +++ b/sound/soc/codecs/rt5682.h @@ -0,0 +1,1324 @@ +/* + * rt5682.h -- RT5682/RT5658 ALSA SoC audio driver + * + * Copyright 2018 Realtek Microelectronics + * Author: Bard Liao + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5682_H__ +#define __RT5682_H__ + +#include + +#define DEVICE_ID 0x6530 + +/* Info */ +#define RT5682_RESET 0x0000 +#define RT5682_VERSION_ID 0x00fd +#define RT5682_VENDOR_ID 0x00fe +#define RT5682_DEVICE_ID 0x00ff +/* I/O - Output */ +#define RT5682_HP_CTRL_1 0x0002 +#define RT5682_HP_CTRL_2 0x0003 +#define RT5682_HPL_GAIN 0x0005 +#define RT5682_HPR_GAIN 0x0006 + +#define RT5682_I2C_CTRL 0x0008 + +/* I/O - Input */ +#define RT5682_CBJ_BST_CTRL 0x000b +#define RT5682_CBJ_CTRL_1 0x0010 +#define RT5682_CBJ_CTRL_2 0x0011 +#define RT5682_CBJ_CTRL_3 0x0012 +#define RT5682_CBJ_CTRL_4 0x0013 +#define RT5682_CBJ_CTRL_5 0x0014 +#define RT5682_CBJ_CTRL_6 0x0015 +#define RT5682_CBJ_CTRL_7 0x0016 +/* I/O - ADC/DAC/DMIC */ +#define RT5682_DAC1_DIG_VOL 0x0019 +#define RT5682_STO1_ADC_DIG_VOL 0x001c +#define RT5682_STO1_ADC_BOOST 0x001f +#define RT5682_HP_IMP_GAIN_1 0x0022 +#define RT5682_HP_IMP_GAIN_2 0x0023 +/* Mixer - D-D */ +#define RT5682_SIDETONE_CTRL 0x0024 +#define RT5682_STO1_ADC_MIXER 0x0026 +#define RT5682_AD_DA_MIXER 0x0029 +#define RT5682_STO1_DAC_MIXER 0x002a +#define RT5682_A_DAC1_MUX 0x002b +#define RT5682_DIG_INF2_DATA 0x0030 +/* Mixer - ADC */ +#define RT5682_REC_MIXER 0x003c +#define RT5682_CAL_REC 0x0044 +#define RT5682_ALC_BACK_GAIN 0x0049 +/* Power */ +#define RT5682_PWR_DIG_1 0x0061 +#define RT5682_PWR_DIG_2 0x0062 +#define RT5682_PWR_ANLG_1 0x0063 +#define RT5682_PWR_ANLG_2 0x0064 +#define RT5682_PWR_ANLG_3 0x0065 +#define RT5682_PWR_MIXER 0x0066 +#define RT5682_PWR_VOL 0x0067 +/* Clock Detect */ +#define RT5682_CLK_DET 0x006b +/* Filter Auto Reset */ +#define RT5682_RESET_LPF_CTRL 0x006c +#define RT5682_RESET_HPF_CTRL 0x006d +/* DMIC */ +#define RT5682_DMIC_CTRL_1 0x006e +/* Format - ADC/DAC */ +#define RT5682_I2S1_SDP 0x0070 +#define RT5682_I2S2_SDP 0x0071 +#define RT5682_ADDA_CLK_1 0x0073 +#define RT5682_ADDA_CLK_2 0x0074 +#define RT5682_I2S1_F_DIV_CTRL_1 0x0075 +#define RT5682_I2S1_F_DIV_CTRL_2 0x0076 +/* Format - TDM Control */ +#define RT5682_TDM_CTRL 0x0079 +#define RT5682_TDM_ADDA_CTRL_1 0x007a +#define RT5682_TDM_ADDA_CTRL_2 0x007b +#define RT5682_DATA_SEL_CTRL_1 0x007c +#define RT5682_TDM_TCON_CTRL 0x007e +/* Function - Analog */ +#define RT5682_GLB_CLK 0x0080 +#define RT5682_PLL_CTRL_1 0x0081 +#define RT5682_PLL_CTRL_2 0x0082 +#define RT5682_PLL_TRACK_1 0x0083 +#define RT5682_PLL_TRACK_2 0x0084 +#define RT5682_PLL_TRACK_3 0x0085 +#define RT5682_PLL_TRACK_4 0x0086 +#define RT5682_PLL_TRACK_5 0x0087 +#define RT5682_PLL_TRACK_6 0x0088 +#define RT5682_PLL_TRACK_11 0x008c +#define RT5682_SDW_REF_CLK 0x008d +#define RT5682_DEPOP_1 0x008e +#define RT5682_DEPOP_2 0x008f +#define RT5682_HP_CHARGE_PUMP_1 0x0091 +#define RT5682_HP_CHARGE_PUMP_2 0x0092 +#define RT5682_MICBIAS_1 0x0093 +#define RT5682_MICBIAS_2 0x0094 +#define RT5682_PLL_TRACK_12 0x0098 +#define RT5682_PLL_TRACK_14 0x009a +#define RT5682_PLL2_CTRL_1 0x009b +#define RT5682_PLL2_CTRL_2 0x009c +#define RT5682_PLL2_CTRL_3 0x009d +#define RT5682_PLL2_CTRL_4 0x009e +#define RT5682_RC_CLK_CTRL 0x009f +#define RT5682_I2S_M_CLK_CTRL_1 0x00a0 +#define RT5682_I2S2_F_DIV_CTRL_1 0x00a3 +#define RT5682_I2S2_F_DIV_CTRL_2 0x00a4 +/* Function - Digital */ +#define RT5682_EQ_CTRL_1 0x00ae +#define RT5682_EQ_CTRL_2 0x00af +#define RT5682_IRQ_CTRL_1 0x00b6 +#define RT5682_IRQ_CTRL_2 0x00b7 +#define RT5682_IRQ_CTRL_3 0x00b8 +#define RT5682_IRQ_CTRL_4 0x00b9 +#define RT5682_INT_ST_1 0x00be +#define RT5682_GPIO_CTRL_1 0x00c0 +#define RT5682_GPIO_CTRL_2 0x00c1 +#define RT5682_GPIO_CTRL_3 0x00c2 +#define RT5682_HP_AMP_DET_CTRL_1 0x00d0 +#define RT5682_HP_AMP_DET_CTRL_2 0x00d1 +#define RT5682_MID_HP_AMP_DET 0x00d2 +#define RT5682_LOW_HP_AMP_DET 0x00d3 +#define RT5682_DELAY_BUF_CTRL 0x00d4 +#define RT5682_SV_ZCD_1 0x00d9 +#define RT5682_SV_ZCD_2 0x00da +#define RT5682_IL_CMD_1 0x00db +#define RT5682_IL_CMD_2 0x00dc +#define RT5682_IL_CMD_3 0x00dd +#define RT5682_IL_CMD_4 0x00de +#define RT5682_IL_CMD_5 0x00df +#define RT5682_IL_CMD_6 0x00e0 +#define RT5682_4BTN_IL_CMD_1 0x00e2 +#define RT5682_4BTN_IL_CMD_2 0x00e3 +#define RT5682_4BTN_IL_CMD_3 0x00e4 +#define RT5682_4BTN_IL_CMD_4 0x00e5 +#define RT5682_4BTN_IL_CMD_5 0x00e6 +#define RT5682_4BTN_IL_CMD_6 0x00e7 +#define RT5682_4BTN_IL_CMD_7 0x00e8 + +#define RT5682_ADC_STO1_HP_CTRL_1 0x00ea +#define RT5682_ADC_STO1_HP_CTRL_2 0x00eb +#define RT5682_AJD1_CTRL 0x00f0 +#define RT5682_JD1_THD 0x00f1 +#define RT5682_JD2_THD 0x00f2 +#define RT5682_JD_CTRL_1 0x00f6 +/* General Control */ +#define RT5682_DUMMY_1 0x00fa +#define RT5682_DUMMY_2 0x00fb +#define RT5682_DUMMY_3 0x00fc + +#define RT5682_DAC_ADC_DIG_VOL1 0x0100 +#define RT5682_BIAS_CUR_CTRL_2 0x010b +#define RT5682_BIAS_CUR_CTRL_3 0x010c +#define RT5682_BIAS_CUR_CTRL_4 0x010d +#define RT5682_BIAS_CUR_CTRL_5 0x010e +#define RT5682_BIAS_CUR_CTRL_6 0x010f +#define RT5682_BIAS_CUR_CTRL_7 0x0110 +#define RT5682_BIAS_CUR_CTRL_8 0x0111 +#define RT5682_BIAS_CUR_CTRL_9 0x0112 +#define RT5682_BIAS_CUR_CTRL_10 0x0113 +#define RT5682_VREF_REC_OP_FB_CAP_CTRL 0x0117 +#define RT5682_CHARGE_PUMP_1 0x0125 +#define RT5682_DIG_IN_CTRL_1 0x0132 +#define RT5682_PAD_DRIVING_CTRL 0x0136 +#define RT5682_SOFT_RAMP_DEPOP 0x0138 +#define RT5682_CHOP_DAC 0x013a +#define RT5682_CHOP_ADC 0x013b +#define RT5682_CALIB_ADC_CTRL 0x013c +#define RT5682_VOL_TEST 0x013f +#define RT5682_SPKVDD_DET_STA 0x0142 +#define RT5682_TEST_MODE_CTRL_1 0x0145 +#define RT5682_TEST_MODE_CTRL_2 0x0146 +#define RT5682_TEST_MODE_CTRL_3 0x0147 +#define RT5682_TEST_MODE_CTRL_4 0x0148 +#define RT5682_TEST_MODE_CTRL_5 0x0149 +#define RT5682_PLL1_INTERNAL 0x0150 +#define RT5682_PLL2_INTERNAL 0x0151 +#define RT5682_STO_NG2_CTRL_1 0x0160 +#define RT5682_STO_NG2_CTRL_2 0x0161 +#define RT5682_STO_NG2_CTRL_3 0x0162 +#define RT5682_STO_NG2_CTRL_4 0x0163 +#define RT5682_STO_NG2_CTRL_5 0x0164 +#define RT5682_STO_NG2_CTRL_6 0x0165 +#define RT5682_STO_NG2_CTRL_7 0x0166 +#define RT5682_STO_NG2_CTRL_8 0x0167 +#define RT5682_STO_NG2_CTRL_9 0x0168 +#define RT5682_STO_NG2_CTRL_10 0x0169 +#define RT5682_STO1_DAC_SIL_DET 0x0190 +#define RT5682_SIL_PSV_CTRL1 0x0194 +#define RT5682_SIL_PSV_CTRL2 0x0195 +#define RT5682_SIL_PSV_CTRL3 0x0197 +#define RT5682_SIL_PSV_CTRL4 0x0198 +#define RT5682_SIL_PSV_CTRL5 0x0199 +#define RT5682_HP_IMP_SENS_CTRL_01 0x01af +#define RT5682_HP_IMP_SENS_CTRL_02 0x01b0 +#define RT5682_HP_IMP_SENS_CTRL_03 0x01b1 +#define RT5682_HP_IMP_SENS_CTRL_04 0x01b2 +#define RT5682_HP_IMP_SENS_CTRL_05 0x01b3 +#define RT5682_HP_IMP_SENS_CTRL_06 0x01b4 +#define RT5682_HP_IMP_SENS_CTRL_07 0x01b5 +#define RT5682_HP_IMP_SENS_CTRL_08 0x01b6 +#define RT5682_HP_IMP_SENS_CTRL_09 0x01b7 +#define RT5682_HP_IMP_SENS_CTRL_10 0x01b8 +#define RT5682_HP_IMP_SENS_CTRL_11 0x01b9 +#define RT5682_HP_IMP_SENS_CTRL_12 0x01ba +#define RT5682_HP_IMP_SENS_CTRL_13 0x01bb +#define RT5682_HP_IMP_SENS_CTRL_14 0x01bc +#define RT5682_HP_IMP_SENS_CTRL_15 0x01bd +#define RT5682_HP_IMP_SENS_CTRL_16 0x01be +#define RT5682_HP_IMP_SENS_CTRL_17 0x01bf +#define RT5682_HP_IMP_SENS_CTRL_18 0x01c0 +#define RT5682_HP_IMP_SENS_CTRL_19 0x01c1 +#define RT5682_HP_IMP_SENS_CTRL_20 0x01c2 +#define RT5682_HP_IMP_SENS_CTRL_21 0x01c3 +#define RT5682_HP_IMP_SENS_CTRL_22 0x01c4 +#define RT5682_HP_IMP_SENS_CTRL_23 0x01c5 +#define RT5682_HP_IMP_SENS_CTRL_24 0x01c6 +#define RT5682_HP_IMP_SENS_CTRL_25 0x01c7 +#define RT5682_HP_IMP_SENS_CTRL_26 0x01c8 +#define RT5682_HP_IMP_SENS_CTRL_27 0x01c9 +#define RT5682_HP_IMP_SENS_CTRL_28 0x01ca +#define RT5682_HP_IMP_SENS_CTRL_29 0x01cb +#define RT5682_HP_IMP_SENS_CTRL_30 0x01cc +#define RT5682_HP_IMP_SENS_CTRL_31 0x01cd +#define RT5682_HP_IMP_SENS_CTRL_32 0x01ce +#define RT5682_HP_IMP_SENS_CTRL_33 0x01cf +#define RT5682_HP_IMP_SENS_CTRL_34 0x01d0 +#define RT5682_HP_IMP_SENS_CTRL_35 0x01d1 +#define RT5682_HP_IMP_SENS_CTRL_36 0x01d2 +#define RT5682_HP_IMP_SENS_CTRL_37 0x01d3 +#define RT5682_HP_IMP_SENS_CTRL_38 0x01d4 +#define RT5682_HP_IMP_SENS_CTRL_39 0x01d5 +#define RT5682_HP_IMP_SENS_CTRL_40 0x01d6 +#define RT5682_HP_IMP_SENS_CTRL_41 0x01d7 +#define RT5682_HP_IMP_SENS_CTRL_42 0x01d8 +#define RT5682_HP_IMP_SENS_CTRL_43 0x01d9 +#define RT5682_HP_LOGIC_CTRL_1 0x01da +#define RT5682_HP_LOGIC_CTRL_2 0x01db +#define RT5682_HP_LOGIC_CTRL_3 0x01dc +#define RT5682_HP_CALIB_CTRL_1 0x01de +#define RT5682_HP_CALIB_CTRL_2 0x01df +#define RT5682_HP_CALIB_CTRL_3 0x01e0 +#define RT5682_HP_CALIB_CTRL_4 0x01e1 +#define RT5682_HP_CALIB_CTRL_5 0x01e2 +#define RT5682_HP_CALIB_CTRL_6 0x01e3 +#define RT5682_HP_CALIB_CTRL_7 0x01e4 +#define RT5682_HP_CALIB_CTRL_9 0x01e6 +#define RT5682_HP_CALIB_CTRL_10 0x01e7 +#define RT5682_HP_CALIB_CTRL_11 0x01e8 +#define RT5682_HP_CALIB_STA_1 0x01ea +#define RT5682_HP_CALIB_STA_2 0x01eb +#define RT5682_HP_CALIB_STA_3 0x01ec +#define RT5682_HP_CALIB_STA_4 0x01ed +#define RT5682_HP_CALIB_STA_5 0x01ee +#define RT5682_HP_CALIB_STA_6 0x01ef +#define RT5682_HP_CALIB_STA_7 0x01f0 +#define RT5682_HP_CALIB_STA_8 0x01f1 +#define RT5682_HP_CALIB_STA_9 0x01f2 +#define RT5682_HP_CALIB_STA_10 0x01f3 +#define RT5682_HP_CALIB_STA_11 0x01f4 +#define RT5682_SAR_IL_CMD_1 0x0210 +#define RT5682_SAR_IL_CMD_2 0x0211 +#define RT5682_SAR_IL_CMD_3 0x0212 +#define RT5682_SAR_IL_CMD_4 0x0213 +#define RT5682_SAR_IL_CMD_5 0x0214 +#define RT5682_SAR_IL_CMD_6 0x0215 +#define RT5682_SAR_IL_CMD_7 0x0216 +#define RT5682_SAR_IL_CMD_8 0x0217 +#define RT5682_SAR_IL_CMD_9 0x0218 +#define RT5682_SAR_IL_CMD_10 0x0219 +#define RT5682_SAR_IL_CMD_11 0x021a +#define RT5682_SAR_IL_CMD_12 0x021b +#define RT5682_SAR_IL_CMD_13 0x021c +#define RT5682_EFUSE_CTRL_1 0x0250 +#define RT5682_EFUSE_CTRL_2 0x0251 +#define RT5682_EFUSE_CTRL_3 0x0252 +#define RT5682_EFUSE_CTRL_4 0x0253 +#define RT5682_EFUSE_CTRL_5 0x0254 +#define RT5682_EFUSE_CTRL_6 0x0255 +#define RT5682_EFUSE_CTRL_7 0x0256 +#define RT5682_EFUSE_CTRL_8 0x0257 +#define RT5682_EFUSE_CTRL_9 0x0258 +#define RT5682_EFUSE_CTRL_10 0x0259 +#define RT5682_EFUSE_CTRL_11 0x025a +#define RT5682_JD_TOP_VC_VTRL 0x0270 +#define RT5682_DRC1_CTRL_0 0x02ff +#define RT5682_DRC1_CTRL_1 0x0300 +#define RT5682_DRC1_CTRL_2 0x0301 +#define RT5682_DRC1_CTRL_3 0x0302 +#define RT5682_DRC1_CTRL_4 0x0303 +#define RT5682_DRC1_CTRL_5 0x0304 +#define RT5682_DRC1_CTRL_6 0x0305 +#define RT5682_DRC1_HARD_LMT_CTRL_1 0x0306 +#define RT5682_DRC1_HARD_LMT_CTRL_2 0x0307 +#define RT5682_DRC1_PRIV_1 0x0310 +#define RT5682_DRC1_PRIV_2 0x0311 +#define RT5682_DRC1_PRIV_3 0x0312 +#define RT5682_DRC1_PRIV_4 0x0313 +#define RT5682_DRC1_PRIV_5 0x0314 +#define RT5682_DRC1_PRIV_6 0x0315 +#define RT5682_DRC1_PRIV_7 0x0316 +#define RT5682_DRC1_PRIV_8 0x0317 +#define RT5682_EQ_AUTO_RCV_CTRL1 0x03c0 +#define RT5682_EQ_AUTO_RCV_CTRL2 0x03c1 +#define RT5682_EQ_AUTO_RCV_CTRL3 0x03c2 +#define RT5682_EQ_AUTO_RCV_CTRL4 0x03c3 +#define RT5682_EQ_AUTO_RCV_CTRL5 0x03c4 +#define RT5682_EQ_AUTO_RCV_CTRL6 0x03c5 +#define RT5682_EQ_AUTO_RCV_CTRL7 0x03c6 +#define RT5682_EQ_AUTO_RCV_CTRL8 0x03c7 +#define RT5682_EQ_AUTO_RCV_CTRL9 0x03c8 +#define RT5682_EQ_AUTO_RCV_CTRL10 0x03c9 +#define RT5682_EQ_AUTO_RCV_CTRL11 0x03ca +#define RT5682_EQ_AUTO_RCV_CTRL12 0x03cb +#define RT5682_EQ_AUTO_RCV_CTRL13 0x03cc +#define RT5682_ADC_L_EQ_LPF1_A1 0x03d0 +#define RT5682_R_EQ_LPF1_A1 0x03d1 +#define RT5682_L_EQ_LPF1_H0 0x03d2 +#define RT5682_R_EQ_LPF1_H0 0x03d3 +#define RT5682_L_EQ_BPF1_A1 0x03d4 +#define RT5682_R_EQ_BPF1_A1 0x03d5 +#define RT5682_L_EQ_BPF1_A2 0x03d6 +#define RT5682_R_EQ_BPF1_A2 0x03d7 +#define RT5682_L_EQ_BPF1_H0 0x03d8 +#define RT5682_R_EQ_BPF1_H0 0x03d9 +#define RT5682_L_EQ_BPF2_A1 0x03da +#define RT5682_R_EQ_BPF2_A1 0x03db +#define RT5682_L_EQ_BPF2_A2 0x03dc +#define RT5682_R_EQ_BPF2_A2 0x03dd +#define RT5682_L_EQ_BPF2_H0 0x03de +#define RT5682_R_EQ_BPF2_H0 0x03df +#define RT5682_L_EQ_BPF3_A1 0x03e0 +#define RT5682_R_EQ_BPF3_A1 0x03e1 +#define RT5682_L_EQ_BPF3_A2 0x03e2 +#define RT5682_R_EQ_BPF3_A2 0x03e3 +#define RT5682_L_EQ_BPF3_H0 0x03e4 +#define RT5682_R_EQ_BPF3_H0 0x03e5 +#define RT5682_L_EQ_BPF4_A1 0x03e6 +#define RT5682_R_EQ_BPF4_A1 0x03e7 +#define RT5682_L_EQ_BPF4_A2 0x03e8 +#define RT5682_R_EQ_BPF4_A2 0x03e9 +#define RT5682_L_EQ_BPF4_H0 0x03ea +#define RT5682_R_EQ_BPF4_H0 0x03eb +#define RT5682_L_EQ_HPF1_A1 0x03ec +#define RT5682_R_EQ_HPF1_A1 0x03ed +#define RT5682_L_EQ_HPF1_H0 0x03ee +#define RT5682_R_EQ_HPF1_H0 0x03ef +#define RT5682_L_EQ_PRE_VOL 0x03f0 +#define RT5682_R_EQ_PRE_VOL 0x03f1 +#define RT5682_L_EQ_POST_VOL 0x03f2 +#define RT5682_R_EQ_POST_VOL 0x03f3 +#define RT5682_I2C_MODE 0xffff + + +/* global definition */ +#define RT5682_L_MUTE (0x1 << 15) +#define RT5682_L_MUTE_SFT 15 +#define RT5682_VOL_L_MUTE (0x1 << 14) +#define RT5682_VOL_L_SFT 14 +#define RT5682_R_MUTE (0x1 << 7) +#define RT5682_R_MUTE_SFT 7 +#define RT5682_VOL_R_MUTE (0x1 << 6) +#define RT5682_VOL_R_SFT 6 +#define RT5682_L_VOL_MASK (0x3f << 8) +#define RT5682_L_VOL_SFT 8 +#define RT5682_R_VOL_MASK (0x3f) +#define RT5682_R_VOL_SFT 0 + +/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ +#define RT5682_G_HP (0xf << 8) +#define RT5682_G_HP_SFT 8 +#define RT5682_G_STO_DA_DMIX (0xf) +#define RT5682_G_STO_DA_SFT 0 + +/* CBJ Control (0x000b) */ +#define RT5682_BST_CBJ_MASK (0xf << 8) +#define RT5682_BST_CBJ_SFT 8 + +/* Embeeded Jack and Type Detection Control 1 (0x0010) */ +#define RT5682_EMB_JD_EN (0x1 << 15) +#define RT5682_EMB_JD_EN_SFT 15 +#define RT5682_EMB_JD_RST (0x1 << 14) +#define RT5682_JD_MODE (0x1 << 13) +#define RT5682_JD_MODE_SFT 13 +#define RT5682_DET_TYPE (0x1 << 12) +#define RT5682_DET_TYPE_SFT 12 +#define RT5682_POLA_EXT_JD_MASK (0x1 << 11) +#define RT5682_POLA_EXT_JD_LOW (0x1 << 11) +#define RT5682_POLA_EXT_JD_HIGH (0x0 << 11) +#define RT5682_EXT_JD_DIG (0x1 << 9) +#define RT5682_POL_FAST_OFF_MASK (0x1 << 8) +#define RT5682_POL_FAST_OFF_HIGH (0x1 << 8) +#define RT5682_POL_FAST_OFF_LOW (0x0 << 8) +#define RT5682_FAST_OFF_MASK (0x1 << 7) +#define RT5682_FAST_OFF_EN (0x1 << 7) +#define RT5682_FAST_OFF_DIS (0x0 << 7) +#define RT5682_VREF_POW_MASK (0x1 << 6) +#define RT5682_VREF_POW_FSM (0x0 << 6) +#define RT5682_VREF_POW_REG (0x1 << 6) +#define RT5682_MB1_PATH_MASK (0x1 << 5) +#define RT5682_CTRL_MB1_REG (0x1 << 5) +#define RT5682_CTRL_MB1_FSM (0x0 << 5) +#define RT5682_MB2_PATH_MASK (0x1 << 4) +#define RT5682_CTRL_MB2_REG (0x1 << 4) +#define RT5682_CTRL_MB2_FSM (0x0 << 4) +#define RT5682_TRIG_JD_MASK (0x1 << 3) +#define RT5682_TRIG_JD_HIGH (0x1 << 3) +#define RT5682_TRIG_JD_LOW (0x0 << 3) +#define RT5682_MIC_CAP_MASK (0x1 << 1) +#define RT5682_MIC_CAP_HS (0x1 << 1) +#define RT5682_MIC_CAP_HP (0x0 << 1) +#define RT5682_MIC_CAP_SRC_MASK (0x1) +#define RT5682_MIC_CAP_SRC_REG (0x1) +#define RT5682_MIC_CAP_SRC_ANA (0x0) + +/* Embeeded Jack and Type Detection Control 2 (0x0011) */ +#define RT5682_EXT_JD_SRC (0x7 << 4) +#define RT5682_EXT_JD_SRC_SFT 4 +#define RT5682_EXT_JD_SRC_GPIO_JD1 (0x0 << 4) +#define RT5682_EXT_JD_SRC_GPIO_JD2 (0x1 << 4) +#define RT5682_EXT_JD_SRC_JDH (0x2 << 4) +#define RT5682_EXT_JD_SRC_JDL (0x3 << 4) +#define RT5682_EXT_JD_SRC_MANUAL (0x4 << 4) +#define RT5682_JACK_TYPE_MASK (0x3) + +/* Combo Jack and Type Detection Control 3 (0x0012) */ +#define RT5682_CBJ_IN_BUF_EN (0x1 << 7) + +/* Combo Jack and Type Detection Control 4 (0x0013) */ +#define RT5682_SEL_SHT_MID_TON_MASK (0x3 << 12) +#define RT5682_SEL_SHT_MID_TON_2 (0x0 << 12) +#define RT5682_SEL_SHT_MID_TON_3 (0x1 << 12) +#define RT5682_CBJ_JD_TEST_MASK (0x1 << 6) +#define RT5682_CBJ_JD_TEST_NORM (0x0 << 6) +#define RT5682_CBJ_JD_TEST_MODE (0x1 << 6) + +/* DAC1 Digital Volume (0x0019) */ +#define RT5682_DAC_L1_VOL_MASK (0xff << 8) +#define RT5682_DAC_L1_VOL_SFT 8 +#define RT5682_DAC_R1_VOL_MASK (0xff) +#define RT5682_DAC_R1_VOL_SFT 0 + +/* ADC Digital Volume Control (0x001c) */ +#define RT5682_ADC_L_VOL_MASK (0x7f << 8) +#define RT5682_ADC_L_VOL_SFT 8 +#define RT5682_ADC_R_VOL_MASK (0x7f) +#define RT5682_ADC_R_VOL_SFT 0 + +/* Stereo1 ADC Boost Gain Control (0x001f) */ +#define RT5682_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5682_STO1_ADC_L_BST_SFT 14 +#define RT5682_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5682_STO1_ADC_R_BST_SFT 12 + +/* Sidetone Control (0x0024) */ +#define RT5682_ST_SRC_SEL (0x1 << 8) +#define RT5682_ST_SRC_SFT 8 +#define RT5682_ST_EN_MASK (0x1 << 6) +#define RT5682_ST_DIS (0x0 << 6) +#define RT5682_ST_EN (0x1 << 6) +#define RT5682_ST_EN_SFT 6 + +/* Stereo1 ADC Mixer Control (0x0026) */ +#define RT5682_M_STO1_ADC_L1 (0x1 << 15) +#define RT5682_M_STO1_ADC_L1_SFT 15 +#define RT5682_M_STO1_ADC_L2 (0x1 << 14) +#define RT5682_M_STO1_ADC_L2_SFT 14 +#define RT5682_STO1_ADC1L_SRC_MASK (0x1 << 13) +#define RT5682_STO1_ADC1L_SRC_SFT 13 +#define RT5682_STO1_ADC1_SRC_ADC (0x1 << 13) +#define RT5682_STO1_ADC1_SRC_DACMIX (0x0 << 13) +#define RT5682_STO1_ADC2L_SRC_MASK (0x1 << 12) +#define RT5682_STO1_ADC2L_SRC_SFT 12 +#define RT5682_STO1_ADCL_SRC_MASK (0x3 << 10) +#define RT5682_STO1_ADCL_SRC_SFT 10 +#define RT5682_STO1_DD_L_SRC_MASK (0x1 << 9) +#define RT5682_STO1_DD_L_SRC_SFT 9 +#define RT5682_STO1_DMIC_SRC_MASK (0x1 << 8) +#define RT5682_STO1_DMIC_SRC_SFT 8 +#define RT5682_STO1_DMIC_SRC_DMIC2 (0x1 << 8) +#define RT5682_STO1_DMIC_SRC_DMIC1 (0x0 << 8) +#define RT5682_M_STO1_ADC_R1 (0x1 << 7) +#define RT5682_M_STO1_ADC_R1_SFT 7 +#define RT5682_M_STO1_ADC_R2 (0x1 << 6) +#define RT5682_M_STO1_ADC_R2_SFT 6 +#define RT5682_STO1_ADC1R_SRC_MASK (0x1 << 5) +#define RT5682_STO1_ADC1R_SRC_SFT 5 +#define RT5682_STO1_ADC2R_SRC_MASK (0x1 << 4) +#define RT5682_STO1_ADC2R_SRC_SFT 4 +#define RT5682_STO1_ADCR_SRC_MASK (0x3 << 2) +#define RT5682_STO1_ADCR_SRC_SFT 2 + +/* ADC Mixer to DAC Mixer Control (0x0029) */ +#define RT5682_M_ADCMIX_L (0x1 << 15) +#define RT5682_M_ADCMIX_L_SFT 15 +#define RT5682_M_DAC1_L (0x1 << 14) +#define RT5682_M_DAC1_L_SFT 14 +#define RT5682_DAC1_R_SEL_MASK (0x1 << 10) +#define RT5682_DAC1_R_SEL_SFT 10 +#define RT5682_DAC1_L_SEL_MASK (0x1 << 8) +#define RT5682_DAC1_L_SEL_SFT 8 +#define RT5682_M_ADCMIX_R (0x1 << 7) +#define RT5682_M_ADCMIX_R_SFT 7 +#define RT5682_M_DAC1_R (0x1 << 6) +#define RT5682_M_DAC1_R_SFT 6 + +/* Stereo1 DAC Mixer Control (0x002a) */ +#define RT5682_M_DAC_L1_STO_L (0x1 << 15) +#define RT5682_M_DAC_L1_STO_L_SFT 15 +#define RT5682_G_DAC_L1_STO_L_MASK (0x1 << 14) +#define RT5682_G_DAC_L1_STO_L_SFT 14 +#define RT5682_M_DAC_R1_STO_L (0x1 << 13) +#define RT5682_M_DAC_R1_STO_L_SFT 13 +#define RT5682_G_DAC_R1_STO_L_MASK (0x1 << 12) +#define RT5682_G_DAC_R1_STO_L_SFT 12 +#define RT5682_M_DAC_L1_STO_R (0x1 << 7) +#define RT5682_M_DAC_L1_STO_R_SFT 7 +#define RT5682_G_DAC_L1_STO_R_MASK (0x1 << 6) +#define RT5682_G_DAC_L1_STO_R_SFT 6 +#define RT5682_M_DAC_R1_STO_R (0x1 << 5) +#define RT5682_M_DAC_R1_STO_R_SFT 5 +#define RT5682_G_DAC_R1_STO_R_MASK (0x1 << 4) +#define RT5682_G_DAC_R1_STO_R_SFT 4 + +/* Analog DAC1 Input Source Control (0x002b) */ +#define RT5682_M_ST_STO_L (0x1 << 9) +#define RT5682_M_ST_STO_L_SFT 9 +#define RT5682_M_ST_STO_R (0x1 << 8) +#define RT5682_M_ST_STO_R_SFT 8 +#define RT5682_DAC_L1_SRC_MASK (0x3 << 4) +#define RT5682_A_DACL1_SFT 4 +#define RT5682_DAC_R1_SRC_MASK (0x3) +#define RT5682_A_DACR1_SFT 0 + +/* Digital Interface Data Control (0x0030) */ +#define RT5682_IF2_ADC_SEL_MASK (0x3 << 0) +#define RT5682_IF2_ADC_SEL_SFT 0 + +/* REC Left Mixer Control 2 (0x003c) */ +#define RT5682_G_CBJ_RM1_L (0x7 << 10) +#define RT5682_G_CBJ_RM1_L_SFT 10 +#define RT5682_M_CBJ_RM1_L (0x1 << 7) +#define RT5682_M_CBJ_RM1_L_SFT 7 + +/* Power Management for Digital 1 (0x0061) */ +#define RT5682_PWR_I2S1 (0x1 << 15) +#define RT5682_PWR_I2S1_BIT 15 +#define RT5682_PWR_I2S2 (0x1 << 14) +#define RT5682_PWR_I2S2_BIT 14 +#define RT5682_PWR_DAC_L1 (0x1 << 11) +#define RT5682_PWR_DAC_L1_BIT 11 +#define RT5682_PWR_DAC_R1 (0x1 << 10) +#define RT5682_PWR_DAC_R1_BIT 10 +#define RT5682_PWR_LDO (0x1 << 8) +#define RT5682_PWR_LDO_BIT 8 +#define RT5682_PWR_ADC_L1 (0x1 << 4) +#define RT5682_PWR_ADC_L1_BIT 4 +#define RT5682_PWR_ADC_R1 (0x1 << 3) +#define RT5682_PWR_ADC_R1_BIT 3 +#define RT5682_DIG_GATE_CTRL (0x1 << 0) +#define RT5682_DIG_GATE_CTRL_SFT 0 + + +/* Power Management for Digital 2 (0x0062) */ +#define RT5682_PWR_ADC_S1F (0x1 << 15) +#define RT5682_PWR_ADC_S1F_BIT 15 +#define RT5682_PWR_DAC_S1F (0x1 << 10) +#define RT5682_PWR_DAC_S1F_BIT 10 + +/* Power Management for Analog 1 (0x0063) */ +#define RT5682_PWR_VREF1 (0x1 << 15) +#define RT5682_PWR_VREF1_BIT 15 +#define RT5682_PWR_FV1 (0x1 << 14) +#define RT5682_PWR_FV1_BIT 14 +#define RT5682_PWR_VREF2 (0x1 << 13) +#define RT5682_PWR_VREF2_BIT 13 +#define RT5682_PWR_FV2 (0x1 << 12) +#define RT5682_PWR_FV2_BIT 12 +#define RT5682_LDO1_DBG_MASK (0x3 << 10) +#define RT5682_PWR_MB (0x1 << 9) +#define RT5682_PWR_MB_BIT 9 +#define RT5682_PWR_BG (0x1 << 7) +#define RT5682_PWR_BG_BIT 7 +#define RT5682_LDO1_BYPASS_MASK (0x1 << 6) +#define RT5682_LDO1_BYPASS (0x1 << 6) +#define RT5682_LDO1_NOT_BYPASS (0x0 << 6) +#define RT5682_PWR_MA_BIT 6 +#define RT5682_LDO1_DVO_MASK (0x3 << 4) +#define RT5682_LDO1_DVO_09 (0x0 << 4) +#define RT5682_LDO1_DVO_10 (0x1 << 4) +#define RT5682_LDO1_DVO_12 (0x2 << 4) +#define RT5682_LDO1_DVO_14 (0x3 << 4) +#define RT5682_HP_DRIVER_MASK (0x3 << 2) +#define RT5682_HP_DRIVER_1X (0x0 << 2) +#define RT5682_HP_DRIVER_3X (0x1 << 2) +#define RT5682_HP_DRIVER_5X (0x3 << 2) +#define RT5682_PWR_HA_L (0x1 << 1) +#define RT5682_PWR_HA_L_BIT 1 +#define RT5682_PWR_HA_R (0x1 << 0) +#define RT5682_PWR_HA_R_BIT 0 + +/* Power Management for Analog 2 (0x0064) */ +#define RT5682_PWR_MB1 (0x1 << 11) +#define RT5682_PWR_MB1_PWR_DOWN (0x0 << 11) +#define RT5682_PWR_MB1_BIT 11 +#define RT5682_PWR_MB2 (0x1 << 10) +#define RT5682_PWR_MB2_PWR_DOWN (0x0 << 10) +#define RT5682_PWR_MB2_BIT 10 +#define RT5682_PWR_JDH (0x1 << 3) +#define RT5682_PWR_JDH_BIT 3 +#define RT5682_PWR_JDL (0x1 << 2) +#define RT5682_PWR_JDL_BIT 2 +#define RT5682_PWR_RM1_L (0x1 << 1) +#define RT5682_PWR_RM1_L_BIT 1 + +/* Power Management for Analog 3 (0x0065) */ +#define RT5682_PWR_CBJ (0x1 << 9) +#define RT5682_PWR_CBJ_BIT 9 +#define RT5682_PWR_PLL (0x1 << 6) +#define RT5682_PWR_PLL_BIT 6 +#define RT5682_PWR_PLL2B (0x1 << 5) +#define RT5682_PWR_PLL2B_BIT 5 +#define RT5682_PWR_PLL2F (0x1 << 4) +#define RT5682_PWR_PLL2F_BIT 4 +#define RT5682_PWR_LDO2 (0x1 << 2) +#define RT5682_PWR_LDO2_BIT 2 +#define RT5682_PWR_DET_SPKVDD (0x1 << 1) +#define RT5682_PWR_DET_SPKVDD_BIT 1 + +/* Power Management for Mixer (0x0066) */ +#define RT5682_PWR_STO1_DAC_L (0x1 << 5) +#define RT5682_PWR_STO1_DAC_L_BIT 5 +#define RT5682_PWR_STO1_DAC_R (0x1 << 4) +#define RT5682_PWR_STO1_DAC_R_BIT 4 + +/* MCLK and System Clock Detection Control (0x006b) */ +#define RT5682_SYS_CLK_DET (0x1 << 15) +#define RT5682_SYS_CLK_DET_SFT 15 +#define RT5682_PLL1_CLK_DET (0x1 << 14) +#define RT5682_PLL1_CLK_DET_SFT 14 +#define RT5682_PLL2_CLK_DET (0x1 << 13) +#define RT5682_PLL2_CLK_DET_SFT 13 +#define RT5682_POW_CLK_DET2_SFT 8 +#define RT5682_POW_CLK_DET_SFT 0 + +/* Digital Microphone Control 1 (0x006e) */ +#define RT5682_DMIC_1_EN_MASK (0x1 << 15) +#define RT5682_DMIC_1_EN_SFT 15 +#define RT5682_DMIC_1_DIS (0x0 << 15) +#define RT5682_DMIC_1_EN (0x1 << 15) +#define RT5682_DMIC_1_DP_MASK (0x3 << 4) +#define RT5682_DMIC_1_DP_SFT 4 +#define RT5682_DMIC_1_DP_GPIO2 (0x0 << 4) +#define RT5682_DMIC_1_DP_GPIO5 (0x1 << 4) +#define RT5682_DMIC_CLK_MASK (0xf << 0) +#define RT5682_DMIC_CLK_SFT 0 + +/* I2S1 Audio Serial Data Port Control (0x0070) */ +#define RT5682_SEL_ADCDAT_MASK (0x1 << 15) +#define RT5682_SEL_ADCDAT_OUT (0x0 << 15) +#define RT5682_SEL_ADCDAT_IN (0x1 << 15) +#define RT5682_SEL_ADCDAT_SFT 15 +#define RT5682_I2S1_TX_CHL_MASK (0x7 << 12) +#define RT5682_I2S1_TX_CHL_SFT 12 +#define RT5682_I2S1_TX_CHL_16 (0x0 << 12) +#define RT5682_I2S1_TX_CHL_20 (0x1 << 12) +#define RT5682_I2S1_TX_CHL_24 (0x2 << 12) +#define RT5682_I2S1_TX_CHL_32 (0x3 << 12) +#define RT5682_I2S1_TX_CHL_8 (0x4 << 12) +#define RT5682_I2S1_RX_CHL_MASK (0x7 << 8) +#define RT5682_I2S1_RX_CHL_SFT 8 +#define RT5682_I2S1_RX_CHL_16 (0x0 << 8) +#define RT5682_I2S1_RX_CHL_20 (0x1 << 8) +#define RT5682_I2S1_RX_CHL_24 (0x2 << 8) +#define RT5682_I2S1_RX_CHL_32 (0x3 << 8) +#define RT5682_I2S1_RX_CHL_8 (0x4 << 8) +#define RT5682_I2S1_MONO_MASK (0x1 << 7) +#define RT5682_I2S1_MONO_EN (0x1 << 7) +#define RT5682_I2S1_MONO_DIS (0x0 << 7) +#define RT5682_I2S2_MONO_MASK (0x1 << 6) +#define RT5682_I2S2_MONO_EN (0x1 << 6) +#define RT5682_I2S2_MONO_DIS (0x0 << 6) +#define RT5682_I2S1_DL_MASK (0x7 << 4) +#define RT5682_I2S1_DL_SFT 4 +#define RT5682_I2S1_DL_16 (0x0 << 4) +#define RT5682_I2S1_DL_20 (0x1 << 4) +#define RT5682_I2S1_DL_24 (0x2 << 4) +#define RT5682_I2S1_DL_32 (0x3 << 4) +#define RT5682_I2S1_DL_8 (0x4 << 4) + +/* I2S1/2 Audio Serial Data Port Control (0x0070)(0x0071) */ +#define RT5682_I2S2_MS_MASK (0x1 << 15) +#define RT5682_I2S2_MS_SFT 15 +#define RT5682_I2S2_MS_M (0x0 << 15) +#define RT5682_I2S2_MS_S (0x1 << 15) +#define RT5682_I2S2_PIN_CFG_MASK (0x1 << 14) +#define RT5682_I2S2_PIN_CFG_SFT 14 +#define RT5682_I2S2_CLK_SEL_MASK (0x1 << 11) +#define RT5682_I2S2_CLK_SEL_SFT 11 +#define RT5682_I2S2_OUT_MASK (0x1 << 9) +#define RT5682_I2S2_OUT_SFT 9 +#define RT5682_I2S2_OUT_UM (0x0 << 9) +#define RT5682_I2S2_OUT_M (0x1 << 9) +#define RT5682_I2S_BP_MASK (0x1 << 8) +#define RT5682_I2S_BP_SFT 8 +#define RT5682_I2S_BP_NOR (0x0 << 8) +#define RT5682_I2S_BP_INV (0x1 << 8) +#define RT5682_I2S2_MONO_EN (0x1 << 6) +#define RT5682_I2S2_MONO_DIS (0x0 << 6) +#define RT5682_I2S2_DL_MASK (0x3 << 4) +#define RT5682_I2S2_DL_SFT 4 +#define RT5682_I2S2_DL_16 (0x0 << 4) +#define RT5682_I2S2_DL_20 (0x1 << 4) +#define RT5682_I2S2_DL_24 (0x2 << 4) +#define RT5682_I2S2_DL_8 (0x3 << 4) +#define RT5682_I2S_DF_MASK (0x7) +#define RT5682_I2S_DF_SFT 0 +#define RT5682_I2S_DF_I2S (0x0) +#define RT5682_I2S_DF_LEFT (0x1) +#define RT5682_I2S_DF_PCM_A (0x2) +#define RT5682_I2S_DF_PCM_B (0x3) +#define RT5682_I2S_DF_PCM_A_N (0x6) +#define RT5682_I2S_DF_PCM_B_N (0x7) + +/* ADC/DAC Clock Control 1 (0x0073) */ +#define RT5682_ADC_OSR_MASK (0xf << 12) +#define RT5682_ADC_OSR_SFT 12 +#define RT5682_ADC_OSR_D_1 (0x0 << 12) +#define RT5682_ADC_OSR_D_2 (0x1 << 12) +#define RT5682_ADC_OSR_D_4 (0x2 << 12) +#define RT5682_ADC_OSR_D_6 (0x3 << 12) +#define RT5682_ADC_OSR_D_8 (0x4 << 12) +#define RT5682_ADC_OSR_D_12 (0x5 << 12) +#define RT5682_ADC_OSR_D_16 (0x6 << 12) +#define RT5682_ADC_OSR_D_24 (0x7 << 12) +#define RT5682_ADC_OSR_D_32 (0x8 << 12) +#define RT5682_ADC_OSR_D_48 (0x9 << 12) +#define RT5682_I2S_M_DIV_MASK (0xf << 12) +#define RT5682_I2S_M_DIV_SFT 8 +#define RT5682_I2S_M_D_1 (0x0 << 8) +#define RT5682_I2S_M_D_2 (0x1 << 8) +#define RT5682_I2S_M_D_3 (0x2 << 8) +#define RT5682_I2S_M_D_4 (0x3 << 8) +#define RT5682_I2S_M_D_6 (0x4 << 8) +#define RT5682_I2S_M_D_8 (0x5 << 8) +#define RT5682_I2S_M_D_12 (0x6 << 8) +#define RT5682_I2S_M_D_16 (0x7 << 8) +#define RT5682_I2S_M_D_24 (0x8 << 8) +#define RT5682_I2S_M_D_32 (0x9 << 8) +#define RT5682_I2S_M_D_48 (0x10 << 8) +#define RT5682_I2S_CLK_SRC_MASK (0x7 << 4) +#define RT5682_I2S_CLK_SRC_SFT 4 +#define RT5682_I2S_CLK_SRC_MCLK (0x0 << 4) +#define RT5682_I2S_CLK_SRC_PLL1 (0x1 << 4) +#define RT5682_I2S_CLK_SRC_PLL2 (0x2 << 4) +#define RT5682_I2S_CLK_SRC_SDW (0x3 << 4) +#define RT5682_I2S_CLK_SRC_RCCLK (0x4 << 4) /* 25M */ +#define RT5682_DAC_OSR_MASK (0xf << 0) +#define RT5682_DAC_OSR_SFT 0 +#define RT5682_DAC_OSR_D_1 (0x0 << 0) +#define RT5682_DAC_OSR_D_2 (0x1 << 0) +#define RT5682_DAC_OSR_D_4 (0x2 << 0) +#define RT5682_DAC_OSR_D_6 (0x3 << 0) +#define RT5682_DAC_OSR_D_8 (0x4 << 0) +#define RT5682_DAC_OSR_D_12 (0x5 << 0) +#define RT5682_DAC_OSR_D_16 (0x6 << 0) +#define RT5682_DAC_OSR_D_24 (0x7 << 0) +#define RT5682_DAC_OSR_D_32 (0x8 << 0) +#define RT5682_DAC_OSR_D_48 (0x9 << 0) + +/* ADC/DAC Clock Control 2 (0x0074) */ +#define RT5682_I2S2_BCLK_MS2_MASK (0x1 << 11) +#define RT5682_I2S2_BCLK_MS2_SFT 11 +#define RT5682_I2S2_BCLK_MS2_32 (0x0 << 11) +#define RT5682_I2S2_BCLK_MS2_64 (0x1 << 11) + + +/* TDM control 1 (0x0079) */ +#define RT5682_TDM_TX_CH_MASK (0x3 << 12) +#define RT5682_TDM_TX_CH_2 (0x0 << 12) +#define RT5682_TDM_TX_CH_4 (0x1 << 12) +#define RT5682_TDM_TX_CH_6 (0x2 << 12) +#define RT5682_TDM_TX_CH_8 (0x3 << 12) +#define RT5682_TDM_RX_CH_MASK (0x3 << 8) +#define RT5682_TDM_RX_CH_2 (0x0 << 8) +#define RT5682_TDM_RX_CH_4 (0x1 << 8) +#define RT5682_TDM_RX_CH_6 (0x2 << 8) +#define RT5682_TDM_RX_CH_8 (0x3 << 8) +#define RT5682_TDM_ADC_LCA_MASK (0xf << 4) +#define RT5682_TDM_ADC_LCA_SFT 4 +#define RT5682_TDM_ADC_DL_SFT 0 + +/* TDM control 2 (0x007a) */ +#define RT5682_IF1_ADC1_SEL_SFT 14 +#define RT5682_IF1_ADC2_SEL_SFT 12 +#define RT5682_IF1_ADC3_SEL_SFT 10 +#define RT5682_IF1_ADC4_SEL_SFT 8 +#define RT5682_TDM_ADC_SEL_SFT 4 + +/* TDM control 3 (0x007b) */ +#define RT5682_TDM_EN (0x1 << 7) + +/* TDM/I2S control (0x007e) */ +#define RT5682_TDM_S_BP_MASK (0x1 << 15) +#define RT5682_TDM_S_BP_SFT 15 +#define RT5682_TDM_S_BP_NOR (0x0 << 15) +#define RT5682_TDM_S_BP_INV (0x1 << 15) +#define RT5682_TDM_S_LP_MASK (0x1 << 14) +#define RT5682_TDM_S_LP_SFT 14 +#define RT5682_TDM_S_LP_NOR (0x0 << 14) +#define RT5682_TDM_S_LP_INV (0x1 << 14) +#define RT5682_TDM_DF_MASK (0x7 << 11) +#define RT5682_TDM_DF_SFT 11 +#define RT5682_TDM_DF_I2S (0x0 << 11) +#define RT5682_TDM_DF_LEFT (0x1 << 11) +#define RT5682_TDM_DF_PCM_A (0x2 << 11) +#define RT5682_TDM_DF_PCM_B (0x3 << 11) +#define RT5682_TDM_DF_PCM_A_N (0x6 << 11) +#define RT5682_TDM_DF_PCM_B_N (0x7 << 11) +#define RT5682_TDM_CL_MASK (0x3 << 4) +#define RT5682_TDM_CL_16 (0x0 << 4) +#define RT5682_TDM_CL_20 (0x1 << 4) +#define RT5682_TDM_CL_24 (0x2 << 4) +#define RT5682_TDM_CL_32 (0x3 << 4) +#define RT5682_TDM_M_BP_MASK (0x1 << 2) +#define RT5682_TDM_M_BP_SFT 2 +#define RT5682_TDM_M_BP_NOR (0x0 << 2) +#define RT5682_TDM_M_BP_INV (0x1 << 2) +#define RT5682_TDM_M_LP_MASK (0x1 << 1) +#define RT5682_TDM_M_LP_SFT 1 +#define RT5682_TDM_M_LP_NOR (0x0 << 1) +#define RT5682_TDM_M_LP_INV (0x1 << 1) +#define RT5682_TDM_MS_MASK (0x1 << 0) +#define RT5682_TDM_MS_SFT 0 +#define RT5682_TDM_MS_M (0x0 << 0) +#define RT5682_TDM_MS_S (0x1 << 0) + +/* Global Clock Control (0x0080) */ +#define RT5682_SCLK_SRC_MASK (0x7 << 13) +#define RT5682_SCLK_SRC_SFT 13 +#define RT5682_SCLK_SRC_MCLK (0x0 << 13) +#define RT5682_SCLK_SRC_PLL1 (0x1 << 13) +#define RT5682_SCLK_SRC_PLL2 (0x2 << 13) +#define RT5682_SCLK_SRC_SDW (0x3 << 13) +#define RT5682_SCLK_SRC_RCCLK (0x4 << 13) +#define RT5682_PLL1_SRC_MASK (0x3 << 10) +#define RT5682_PLL1_SRC_SFT 10 +#define RT5682_PLL1_SRC_MCLK (0x0 << 10) +#define RT5682_PLL1_SRC_BCLK1 (0x1 << 10) +#define RT5682_PLL1_SRC_SDW (0x2 << 10) +#define RT5682_PLL1_SRC_RC (0x3 << 10) +#define RT5682_PLL2_SRC_MASK (0x3 << 8) +#define RT5682_PLL2_SRC_SFT 8 +#define RT5682_PLL2_SRC_MCLK (0x0 << 8) +#define RT5682_PLL2_SRC_BCLK1 (0x1 << 8) +#define RT5682_PLL2_SRC_SDW (0x2 << 8) +#define RT5682_PLL2_SRC_RC (0x3 << 8) + + + +#define RT5682_PLL_INP_MAX 40000000 +#define RT5682_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x0081) */ +#define RT5682_PLL_N_MAX 0x001ff +#define RT5682_PLL_N_MASK (RT5682_PLL_N_MAX << 7) +#define RT5682_PLL_N_SFT 7 +#define RT5682_PLL_K_MAX 0x001f +#define RT5682_PLL_K_MASK (RT5682_PLL_K_MAX) +#define RT5682_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x0082) */ +#define RT5682_PLL_M_MAX 0x00f +#define RT5682_PLL_M_MASK (RT5682_PLL_M_MAX << 12) +#define RT5682_PLL_M_SFT 12 +#define RT5682_PLL_M_BP (0x1 << 11) +#define RT5682_PLL_M_BP_SFT 11 +#define RT5682_PLL_K_BP (0x1 << 10) +#define RT5682_PLL_K_BP_SFT 10 +#define RT5682_PLL_RST (0x1 << 1) + +/* PLL tracking mode 1 (0x0083) */ +#define RT5682_DA_ASRC_MASK (0x1 << 13) +#define RT5682_DA_ASRC_SFT 13 +#define RT5682_DAC_STO1_ASRC_MASK (0x1 << 12) +#define RT5682_DAC_STO1_ASRC_SFT 12 +#define RT5682_AD_ASRC_MASK (0x1 << 8) +#define RT5682_AD_ASRC_SFT 8 +#define RT5682_AD_ASRC_SEL_MASK (0x1 << 4) +#define RT5682_AD_ASRC_SEL_SFT 4 +#define RT5682_DMIC_ASRC_MASK (0x1 << 3) +#define RT5682_DMIC_ASRC_SFT 3 +#define RT5682_ADC_STO1_ASRC_MASK (0x1 << 2) +#define RT5682_ADC_STO1_ASRC_SFT 2 +#define RT5682_DA_ASRC_SEL_MASK (0x1 << 0) +#define RT5682_DA_ASRC_SEL_SFT 0 + +/* PLL tracking mode 2 3 (0x0084)(0x0085)*/ +#define RT5682_FILTER_CLK_SEL_MASK (0x7 << 12) +#define RT5682_FILTER_CLK_SEL_SFT 12 +#define RT5682_FILTER_CLK_DIV_MASK (0xf << 8) +#define RT5682_FILTER_CLK_DIV_SFT 8 + +/* ASRC Control 4 (0x0086) */ +#define RT5682_ASRCIN_FTK_N1_MASK (0x3 << 14) +#define RT5682_ASRCIN_FTK_N1_SFT 14 +#define RT5682_ASRCIN_FTK_N2_MASK (0x3 << 12) +#define RT5682_ASRCIN_FTK_N2_SFT 12 +#define RT5682_ASRCIN_FTK_M1_MASK (0x7 << 8) +#define RT5682_ASRCIN_FTK_M1_SFT 8 +#define RT5682_ASRCIN_FTK_M2_MASK (0x7 << 4) +#define RT5682_ASRCIN_FTK_M2_SFT 4 + +/* SoundWire reference clk (0x008d) */ +#define RT5682_PLL2_OUT_MASK (0x1 << 8) +#define RT5682_PLL2_OUT_98M (0x0 << 8) +#define RT5682_PLL2_OUT_49M (0x1 << 8) +#define RT5682_SDW_REF_2_MASK (0xf << 4) +#define RT5682_SDW_REF_2_SFT 4 +#define RT5682_SDW_REF_2_48K (0x0 << 4) +#define RT5682_SDW_REF_2_96K (0x1 << 4) +#define RT5682_SDW_REF_2_192K (0x2 << 4) +#define RT5682_SDW_REF_2_32K (0x3 << 4) +#define RT5682_SDW_REF_2_24K (0x4 << 4) +#define RT5682_SDW_REF_2_16K (0x5 << 4) +#define RT5682_SDW_REF_2_12K (0x6 << 4) +#define RT5682_SDW_REF_2_8K (0x7 << 4) +#define RT5682_SDW_REF_2_44K (0x8 << 4) +#define RT5682_SDW_REF_2_88K (0x9 << 4) +#define RT5682_SDW_REF_2_176K (0xa << 4) +#define RT5682_SDW_REF_2_353K (0xb << 4) +#define RT5682_SDW_REF_2_22K (0xc << 4) +#define RT5682_SDW_REF_2_384K (0xd << 4) +#define RT5682_SDW_REF_2_11K (0xe << 4) +#define RT5682_SDW_REF_1_MASK (0xf << 0) +#define RT5682_SDW_REF_1_SFT 0 +#define RT5682_SDW_REF_1_48K (0x0 << 0) +#define RT5682_SDW_REF_1_96K (0x1 << 0) +#define RT5682_SDW_REF_1_192K (0x2 << 0) +#define RT5682_SDW_REF_1_32K (0x3 << 0) +#define RT5682_SDW_REF_1_24K (0x4 << 0) +#define RT5682_SDW_REF_1_16K (0x5 << 0) +#define RT5682_SDW_REF_1_12K (0x6 << 0) +#define RT5682_SDW_REF_1_8K (0x7 << 0) +#define RT5682_SDW_REF_1_44K (0x8 << 0) +#define RT5682_SDW_REF_1_88K (0x9 << 0) +#define RT5682_SDW_REF_1_176K (0xa << 0) +#define RT5682_SDW_REF_1_353K (0xb << 0) +#define RT5682_SDW_REF_1_22K (0xc << 0) +#define RT5682_SDW_REF_1_384K (0xd << 0) +#define RT5682_SDW_REF_1_11K (0xe << 0) + +/* Depop Mode Control 1 (0x008e) */ +#define RT5682_PUMP_EN (0x1 << 3) +#define RT5682_PUMP_EN_SFT 3 +#define RT5682_CAPLESS_EN (0x1 << 0) +#define RT5682_CAPLESS_EN_SFT 0 + +/* Depop Mode Control 2 (0x8f) */ +#define RT5682_RAMP_MASK (0x1 << 12) +#define RT5682_RAMP_SFT 12 +#define RT5682_RAMP_DIS (0x0 << 12) +#define RT5682_RAMP_EN (0x1 << 12) +#define RT5682_BPS_MASK (0x1 << 11) +#define RT5682_BPS_SFT 11 +#define RT5682_BPS_DIS (0x0 << 11) +#define RT5682_BPS_EN (0x1 << 11) +#define RT5682_FAST_UPDN_MASK (0x1 << 10) +#define RT5682_FAST_UPDN_SFT 10 +#define RT5682_FAST_UPDN_DIS (0x0 << 10) +#define RT5682_FAST_UPDN_EN (0x1 << 10) +#define RT5682_VLO_MASK (0x1 << 7) +#define RT5682_VLO_SFT 7 +#define RT5682_VLO_3V (0x0 << 7) +#define RT5682_VLO_33V (0x1 << 7) + +/* HPOUT charge pump 1 (0x0091) */ +#define RT5682_OSW_L_MASK (0x1 << 11) +#define RT5682_OSW_L_SFT 11 +#define RT5682_OSW_L_DIS (0x0 << 11) +#define RT5682_OSW_L_EN (0x1 << 11) +#define RT5682_OSW_R_MASK (0x1 << 10) +#define RT5682_OSW_R_SFT 10 +#define RT5682_OSW_R_DIS (0x0 << 10) +#define RT5682_OSW_R_EN (0x1 << 10) +#define RT5682_PM_HP_MASK (0x3 << 8) +#define RT5682_PM_HP_SFT 8 +#define RT5682_PM_HP_LV (0x0 << 8) +#define RT5682_PM_HP_MV (0x1 << 8) +#define RT5682_PM_HP_HV (0x2 << 8) +#define RT5682_IB_HP_MASK (0x3 << 6) +#define RT5682_IB_HP_SFT 6 +#define RT5682_IB_HP_125IL (0x0 << 6) +#define RT5682_IB_HP_25IL (0x1 << 6) +#define RT5682_IB_HP_5IL (0x2 << 6) +#define RT5682_IB_HP_1IL (0x3 << 6) + +/* Micbias Control1 (0x93) */ +#define RT5682_MIC1_OV_MASK (0x3 << 14) +#define RT5682_MIC1_OV_SFT 14 +#define RT5682_MIC1_OV_2V7 (0x0 << 14) +#define RT5682_MIC1_OV_2V4 (0x1 << 14) +#define RT5682_MIC1_OV_2V25 (0x3 << 14) +#define RT5682_MIC1_OV_1V8 (0x4 << 14) +#define RT5682_MIC1_CLK_MASK (0x1 << 13) +#define RT5682_MIC1_CLK_SFT 13 +#define RT5682_MIC1_CLK_DIS (0x0 << 13) +#define RT5682_MIC1_CLK_EN (0x1 << 13) +#define RT5682_MIC1_OVCD_MASK (0x1 << 12) +#define RT5682_MIC1_OVCD_SFT 12 +#define RT5682_MIC1_OVCD_DIS (0x0 << 12) +#define RT5682_MIC1_OVCD_EN (0x1 << 12) +#define RT5682_MIC1_OVTH_MASK (0x3 << 10) +#define RT5682_MIC1_OVTH_SFT 10 +#define RT5682_MIC1_OVTH_768UA (0x0 << 10) +#define RT5682_MIC1_OVTH_960UA (0x1 << 10) +#define RT5682_MIC1_OVTH_1152UA (0x2 << 10) +#define RT5682_MIC1_OVTH_1960UA (0x3 << 10) +#define RT5682_MIC2_OV_MASK (0x3 << 8) +#define RT5682_MIC2_OV_SFT 8 +#define RT5682_MIC2_OV_2V7 (0x0 << 8) +#define RT5682_MIC2_OV_2V4 (0x1 << 8) +#define RT5682_MIC2_OV_2V25 (0x3 << 8) +#define RT5682_MIC2_OV_1V8 (0x4 << 8) +#define RT5682_MIC2_CLK_MASK (0x1 << 7) +#define RT5682_MIC2_CLK_SFT 7 +#define RT5682_MIC2_CLK_DIS (0x0 << 7) +#define RT5682_MIC2_CLK_EN (0x1 << 7) +#define RT5682_MIC2_OVTH_MASK (0x3 << 4) +#define RT5682_MIC2_OVTH_SFT 4 +#define RT5682_MIC2_OVTH_768UA (0x0 << 4) +#define RT5682_MIC2_OVTH_960UA (0x1 << 4) +#define RT5682_MIC2_OVTH_1152UA (0x2 << 4) +#define RT5682_MIC2_OVTH_1960UA (0x3 << 4) +#define RT5682_PWR_MB_MASK (0x1 << 3) +#define RT5682_PWR_MB_SFT 3 +#define RT5682_PWR_MB_PD (0x0 << 3) +#define RT5682_PWR_MB_PU (0x1 << 3) + +/* Micbias Control2 (0x0094) */ +#define RT5682_PWR_CLK25M_MASK (0x1 << 9) +#define RT5682_PWR_CLK25M_SFT 9 +#define RT5682_PWR_CLK25M_PD (0x0 << 9) +#define RT5682_PWR_CLK25M_PU (0x1 << 9) +#define RT5682_PWR_CLK1M_MASK (0x1 << 8) +#define RT5682_PWR_CLK1M_SFT 8 +#define RT5682_PWR_CLK1M_PD (0x0 << 8) +#define RT5682_PWR_CLK1M_PU (0x1 << 8) + +/* RC Clock Control (0x009f) */ +#define RT5682_POW_IRQ (0x1 << 15) +#define RT5682_POW_JDH (0x1 << 14) +#define RT5682_POW_JDL (0x1 << 13) +#define RT5682_POW_ANA (0x1 << 12) + +/* I2S Master Mode Clock Control 1 (0x00a0) */ +#define RT5682_CLK_SRC_MCLK (0x0) +#define RT5682_CLK_SRC_PLL1 (0x1) +#define RT5682_CLK_SRC_PLL2 (0x2) +#define RT5682_CLK_SRC_SDW (0x3) +#define RT5682_CLK_SRC_RCCLK (0x4) +#define RT5682_I2S_PD_1 (0x0) +#define RT5682_I2S_PD_2 (0x1) +#define RT5682_I2S_PD_3 (0x2) +#define RT5682_I2S_PD_4 (0x3) +#define RT5682_I2S_PD_6 (0x4) +#define RT5682_I2S_PD_8 (0x5) +#define RT5682_I2S_PD_12 (0x6) +#define RT5682_I2S_PD_16 (0x7) +#define RT5682_I2S_PD_24 (0x8) +#define RT5682_I2S_PD_32 (0x9) +#define RT5682_I2S_PD_48 (0xa) +#define RT5682_I2S2_SRC_MASK (0x3 << 4) +#define RT5682_I2S2_SRC_SFT 4 +#define RT5682_I2S2_M_PD_MASK (0xf << 0) +#define RT5682_I2S2_M_PD_SFT 0 + +/* IRQ Control 1 (0x00b6) */ +#define RT5682_JD1_PULSE_EN_MASK (0x1 << 10) +#define RT5682_JD1_PULSE_EN_SFT 10 +#define RT5682_JD1_PULSE_DIS (0x0 << 10) +#define RT5682_JD1_PULSE_EN (0x1 << 10) + +/* IRQ Control 2 (0x00b7) */ +#define RT5682_JD1_EN_MASK (0x1 << 15) +#define RT5682_JD1_EN_SFT 15 +#define RT5682_JD1_DIS (0x0 << 15) +#define RT5682_JD1_EN (0x1 << 15) +#define RT5682_JD1_POL_MASK (0x1 << 13) +#define RT5682_JD1_POL_NOR (0x0 << 13) +#define RT5682_JD1_POL_INV (0x1 << 13) + +/* IRQ Control 3 (0x00b8) */ +#define RT5682_IL_IRQ_MASK (0x1 << 7) +#define RT5682_IL_IRQ_DIS (0x0 << 7) +#define RT5682_IL_IRQ_EN (0x1 << 7) + +/* GPIO Control 1 (0x00c0) */ +#define RT5682_GP1_PIN_MASK (0x3 << 14) +#define RT5682_GP1_PIN_SFT 14 +#define RT5682_GP1_PIN_GPIO1 (0x0 << 14) +#define RT5682_GP1_PIN_IRQ (0x1 << 14) +#define RT5682_GP1_PIN_DMIC_CLK (0x2 << 14) +#define RT5682_GP2_PIN_MASK (0x3 << 12) +#define RT5682_GP2_PIN_SFT 12 +#define RT5682_GP2_PIN_GPIO2 (0x0 << 12) +#define RT5682_GP2_PIN_LRCK2 (0x1 << 12) +#define RT5682_GP2_PIN_DMIC_SDA (0x2 << 12) +#define RT5682_GP3_PIN_MASK (0x3 << 10) +#define RT5682_GP3_PIN_SFT 10 +#define RT5682_GP3_PIN_GPIO3 (0x0 << 10) +#define RT5682_GP3_PIN_BCLK2 (0x1 << 10) +#define RT5682_GP3_PIN_DMIC_CLK (0x2 << 10) +#define RT5682_GP4_PIN_MASK (0x3 << 8) +#define RT5682_GP4_PIN_SFT 8 +#define RT5682_GP4_PIN_GPIO4 (0x0 << 8) +#define RT5682_GP4_PIN_ADCDAT1 (0x1 << 8) +#define RT5682_GP4_PIN_DMIC_CLK (0x2 << 8) +#define RT5682_GP4_PIN_ADCDAT2 (0x3 << 8) +#define RT5682_GP5_PIN_MASK (0x3 << 6) +#define RT5682_GP5_PIN_SFT 6 +#define RT5682_GP5_PIN_GPIO5 (0x0 << 6) +#define RT5682_GP5_PIN_DACDAT1 (0x1 << 6) +#define RT5682_GP5_PIN_DMIC_SDA (0x2 << 6) +#define RT5682_GP6_PIN_MASK (0x1 << 5) +#define RT5682_GP6_PIN_SFT 5 +#define RT5682_GP6_PIN_GPIO6 (0x0 << 5) +#define RT5682_GP6_PIN_LRCK1 (0x1 << 5) + +/* GPIO Control 2 (0x00c1)*/ +#define RT5682_GP1_PF_MASK (0x1 << 15) +#define RT5682_GP1_PF_IN (0x0 << 15) +#define RT5682_GP1_PF_OUT (0x1 << 15) +#define RT5682_GP1_OUT_MASK (0x1 << 14) +#define RT5682_GP1_OUT_L (0x0 << 14) +#define RT5682_GP1_OUT_H (0x1 << 14) +#define RT5682_GP2_PF_MASK (0x1 << 13) +#define RT5682_GP2_PF_IN (0x0 << 13) +#define RT5682_GP2_PF_OUT (0x1 << 13) +#define RT5682_GP2_OUT_MASK (0x1 << 12) +#define RT5682_GP2_OUT_L (0x0 << 12) +#define RT5682_GP2_OUT_H (0x1 << 12) +#define RT5682_GP3_PF_MASK (0x1 << 11) +#define RT5682_GP3_PF_IN (0x0 << 11) +#define RT5682_GP3_PF_OUT (0x1 << 11) +#define RT5682_GP3_OUT_MASK (0x1 << 10) +#define RT5682_GP3_OUT_L (0x0 << 10) +#define RT5682_GP3_OUT_H (0x1 << 10) +#define RT5682_GP4_PF_MASK (0x1 << 9) +#define RT5682_GP4_PF_IN (0x0 << 9) +#define RT5682_GP4_PF_OUT (0x1 << 9) +#define RT5682_GP4_OUT_MASK (0x1 << 8) +#define RT5682_GP4_OUT_L (0x0 << 8) +#define RT5682_GP4_OUT_H (0x1 << 8) +#define RT5682_GP5_PF_MASK (0x1 << 7) +#define RT5682_GP5_PF_IN (0x0 << 7) +#define RT5682_GP5_PF_OUT (0x1 << 7) +#define RT5682_GP5_OUT_MASK (0x1 << 6) +#define RT5682_GP5_OUT_L (0x0 << 6) +#define RT5682_GP5_OUT_H (0x1 << 6) +#define RT5682_GP6_PF_MASK (0x1 << 5) +#define RT5682_GP6_PF_IN (0x0 << 5) +#define RT5682_GP6_PF_OUT (0x1 << 5) +#define RT5682_GP6_OUT_MASK (0x1 << 4) +#define RT5682_GP6_OUT_L (0x0 << 4) +#define RT5682_GP6_OUT_H (0x1 << 4) + + +/* GPIO Status (0x00c2) */ +#define RT5682_GP6_STA (0x1 << 6) +#define RT5682_GP5_STA (0x1 << 5) +#define RT5682_GP4_STA (0x1 << 4) +#define RT5682_GP3_STA (0x1 << 3) +#define RT5682_GP2_STA (0x1 << 2) +#define RT5682_GP1_STA (0x1 << 1) + +/* Soft volume and zero cross control 1 (0x00d9) */ +#define RT5682_SV_MASK (0x1 << 15) +#define RT5682_SV_SFT 15 +#define RT5682_SV_DIS (0x0 << 15) +#define RT5682_SV_EN (0x1 << 15) +#define RT5682_ZCD_MASK (0x1 << 10) +#define RT5682_ZCD_SFT 10 +#define RT5682_ZCD_PD (0x0 << 10) +#define RT5682_ZCD_PU (0x1 << 10) +#define RT5682_SV_DLY_MASK (0xf) +#define RT5682_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0x00da) */ +#define RT5682_ZCD_BST1_CBJ_MASK (0x1 << 7) +#define RT5682_ZCD_BST1_CBJ_SFT 7 +#define RT5682_ZCD_BST1_CBJ_DIS (0x0 << 7) +#define RT5682_ZCD_BST1_CBJ_EN (0x1 << 7) +#define RT5682_ZCD_RECMIX_MASK (0x1) +#define RT5682_ZCD_RECMIX_SFT 0 +#define RT5682_ZCD_RECMIX_DIS (0x0) +#define RT5682_ZCD_RECMIX_EN (0x1) + +/* 4 Button Inline Command Control 2 (0x00e3) */ +#define RT5682_4BTN_IL_MASK (0x1 << 15) +#define RT5682_4BTN_IL_EN (0x1 << 15) +#define RT5682_4BTN_IL_DIS (0x0 << 15) +#define RT5682_4BTN_IL_RST_MASK (0x1 << 14) +#define RT5682_4BTN_IL_NOR (0x1 << 14) +#define RT5682_4BTN_IL_RST (0x0 << 14) + +/* Analog JD Control (0x00f0) */ +#define RT5682_JDH_RS_MASK (0x1 << 4) +#define RT5682_JDH_NO_PLUG (0x1 << 4) +#define RT5682_JDH_PLUG (0x0 << 4) + +/* Chopper and Clock control for DAC (0x013a)*/ +#define RT5682_CKXEN_DAC1_MASK (0x1 << 13) +#define RT5682_CKXEN_DAC1_SFT 13 +#define RT5682_CKGEN_DAC1_MASK (0x1 << 12) +#define RT5682_CKGEN_DAC1_SFT 12 + +/* Chopper and Clock control for ADC (0x013b)*/ +#define RT5682_CKXEN_ADC1_MASK (0x1 << 13) +#define RT5682_CKXEN_ADC1_SFT 13 +#define RT5682_CKGEN_ADC1_MASK (0x1 << 12) +#define RT5682_CKGEN_ADC1_SFT 12 + +/* Volume test (0x013f)*/ +#define RT5682_SEL_CLK_VOL_MASK (0x1 << 15) +#define RT5682_SEL_CLK_VOL_EN (0x1 << 15) +#define RT5682_SEL_CLK_VOL_DIS (0x0 << 15) + +/* Test Mode Control 1 (0x0145) */ +#define RT5682_AD2DA_LB_MASK (0x1 << 10) +#define RT5682_AD2DA_LB_SFT 10 + +/* Stereo Noise Gate Control 1 (0x0160) */ +#define RT5682_NG2_EN_MASK (0x1 << 15) +#define RT5682_NG2_EN (0x1 << 15) +#define RT5682_NG2_DIS (0x0 << 15) + +/* Stereo1 DAC Silence Detection Control (0x0190) */ +#define RT5682_DEB_STO_DAC_MASK (0x7 << 4) +#define RT5682_DEB_80_MS (0x0 << 4) + +/* SAR ADC Inline Command Control 1 (0x0210) */ +#define RT5682_SAR_BUTT_DET_MASK (0x1 << 15) +#define RT5682_SAR_BUTT_DET_EN (0x1 << 15) +#define RT5682_SAR_BUTT_DET_DIS (0x0 << 15) +#define RT5682_SAR_BUTDET_MODE_MASK (0x1 << 14) +#define RT5682_SAR_BUTDET_POW_SAV (0x1 << 14) +#define RT5682_SAR_BUTDET_POW_NORM (0x0 << 14) +#define RT5682_SAR_BUTDET_RST_MASK (0x1 << 13) +#define RT5682_SAR_BUTDET_RST_NORMAL (0x1 << 13) +#define RT5682_SAR_BUTDET_RST (0x0 << 13) +#define RT5682_SAR_POW_MASK (0x1 << 12) +#define RT5682_SAR_POW_EN (0x1 << 12) +#define RT5682_SAR_POW_DIS (0x0 << 12) +#define RT5682_SAR_RST_MASK (0x1 << 11) +#define RT5682_SAR_RST_NORMAL (0x1 << 11) +#define RT5682_SAR_RST (0x0 << 11) +#define RT5682_SAR_BYPASS_MASK (0x1 << 10) +#define RT5682_SAR_BYPASS_EN (0x1 << 10) +#define RT5682_SAR_BYPASS_DIS (0x0 << 10) +#define RT5682_SAR_SEL_MB1_MASK (0x1 << 9) +#define RT5682_SAR_SEL_MB1_SEL (0x1 << 9) +#define RT5682_SAR_SEL_MB1_NOSEL (0x0 << 9) +#define RT5682_SAR_SEL_MB2_MASK (0x1 << 8) +#define RT5682_SAR_SEL_MB2_SEL (0x1 << 8) +#define RT5682_SAR_SEL_MB2_NOSEL (0x0 << 8) +#define RT5682_SAR_SEL_MODE_MASK (0x1 << 7) +#define RT5682_SAR_SEL_MODE_CMP (0x1 << 7) +#define RT5682_SAR_SEL_MODE_ADC (0x0 << 7) +#define RT5682_SAR_SEL_MB1_MB2_MASK (0x1 << 5) +#define RT5682_SAR_SEL_MB1_MB2_AUTO (0x1 << 5) +#define RT5682_SAR_SEL_MB1_MB2_MANU (0x0 << 5) +#define RT5682_SAR_SEL_SIGNAL_MASK (0x1 << 4) +#define RT5682_SAR_SEL_SIGNAL_AUTO (0x1 << 4) +#define RT5682_SAR_SEL_SIGNAL_MANU (0x0 << 4) + +/* SAR ADC Inline Command Control 13 (0x021c) */ +#define RT5682_SAR_SOUR_MASK (0x3f) +#define RT5682_SAR_SOUR_BTN (0x3f) +#define RT5682_SAR_SOUR_TYPE (0x0) + + +/* System Clock Source */ +enum { + RT5682_SCLK_S_MCLK, + RT5682_SCLK_S_PLL1, + RT5682_SCLK_S_PLL2, + RT5682_SCLK_S_RCCLK, +}; + +/* PLL Source */ +enum { + RT5682_PLL1_S_MCLK, + RT5682_PLL1_S_BCLK1, + RT5682_PLL1_S_RCCLK, +}; + +enum { + RT5682_AIF1, + RT5682_AIF2, + RT5682_AIFS +}; + +/* filter mask */ +enum { + RT5682_DA_STEREO1_FILTER = 0x1, + RT5682_AD_STEREO1_FILTER = (0x1 << 1), +}; + +enum { + RT5682_CLK_SEL_SYS, + RT5682_CLK_SEL_I2S1_ASRC, + RT5682_CLK_SEL_I2S2_ASRC, +}; + +int rt5682_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src); + +#endif /* __RT5682_H__ */ -- cgit v1.2.3 From a7dc662c6a7b9209df600c64b16d33d72dbf56b1 Mon Sep 17 00:00:00 2001 From: Stefan Agner Date: Sun, 17 Jun 2018 15:45:29 +0200 Subject: ASoC: codecs: PCM1789: unconditionally flush work Work is guaranteed to be initialized on exit. Drop the unnecessary if statement and always call flush_work. This fixes a warning seen with clang: sound/soc/codecs/pcm1789.c:265:13: warning: address of 'priv->work' will always evaluate to 'true' [-Wpointer-bool-conversion] if (&priv->work) ~~ ~~~~~~^~~~ Signed-off-by: Stefan Agner Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1789.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/pcm1789.c b/sound/soc/codecs/pcm1789.c index 21f15219b3ad..8df6447c76a6 100644 --- a/sound/soc/codecs/pcm1789.c +++ b/sound/soc/codecs/pcm1789.c @@ -262,8 +262,7 @@ int pcm1789_common_exit(struct device *dev) { struct pcm1789_private *priv = dev_get_drvdata(dev); - if (&priv->work) - flush_work(&priv->work); + flush_work(&priv->work); return 0; } -- cgit v1.2.3 From a4519526ebbd261e36425fa1c269515ee0648ab2 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 12:50:01 +0200 Subject: ASoC: pxa: add devicetree support Add the devicetree support, so that the driver can be used in a devictree platform. Signed-off-by: Robert Jarzmik Signed-off-by: Mark Brown --- sound/arm/pxa2xx-ac97-lib.c | 12 ++++++++++++ sound/soc/pxa/pxa2xx-ac97.c | 12 ++++++++++++ 2 files changed, 24 insertions(+) (limited to 'sound/soc') diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 5950a9e218d9..8eafd3d3dff6 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -19,6 +19,7 @@ #include #include #include +#include #include @@ -337,6 +338,17 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) dev_err(&dev->dev, "Invalid reset GPIO %d\n", pdata->reset_gpio); } + } else if (!pdata && dev->dev.of_node) { + pdata = devm_kzalloc(&dev->dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + pdata->reset_gpio = of_get_named_gpio(dev->dev.of_node, + "reset-gpios", 0); + if (pdata->reset_gpio == -ENOENT) + pdata->reset_gpio = -1; + else if (pdata->reset_gpio < 0) + return pdata->reset_gpio; + reset_gpio = pdata->reset_gpio; } else { if (cpu_is_pxa27x()) reset_gpio = 113; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 803818aabee9..5738a0abcd6a 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -238,6 +238,17 @@ static const struct snd_soc_component_driver pxa_ac97_component = { .name = "pxa-ac97", }; +#ifdef CONFIG_OF +static const struct of_device_id pxa2xx_ac97_dt_ids[] = { + { .compatible = "marvell,pxa250-ac97", }, + { .compatible = "marvell,pxa270-ac97", }, + { .compatible = "marvell,pxa300-ac97", }, + { } +}; +MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids); + +#endif + static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; @@ -296,6 +307,7 @@ static struct platform_driver pxa2xx_ac97_driver = { #ifdef CONFIG_PM_SLEEP .pm = &pxa2xx_ac97_pm_ops, #endif + .of_match_table = of_match_ptr(pxa2xx_ac97_dt_ids), }, }; -- cgit v1.2.3 From 7c5dfd549617b87db8e891ff4ecaa4a582b6c4cc Mon Sep 17 00:00:00 2001 From: Alexey Khoroshilov Date: Sat, 16 Jun 2018 01:22:58 +0300 Subject: ASoC: tegra: fix device_node refcounting tegra_rt5677_probe() gets a couple of device nodes with of_parse_phandle(), but there is no release of them. The patch adds the release to tegra_rt5677_remove() and to error handling paths in the probe. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov Reviewed-by: Nicholas Mc Guire Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_rt5677.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index 0e4805c7b4ca..7081f15302cc 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -264,13 +264,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_rt5677_dai.platform_of_node = tegra_rt5677_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -283,6 +283,13 @@ static int tegra_rt5677_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_put_cpu_of_node: + of_node_put(tegra_rt5677_dai.cpu_of_node); + tegra_rt5677_dai.cpu_of_node = NULL; + tegra_rt5677_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_rt5677_dai.codec_of_node); + tegra_rt5677_dai.codec_of_node = NULL; err: return ret; } @@ -296,6 +303,12 @@ static int tegra_rt5677_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + tegra_rt5677_dai.platform_of_node = NULL; + of_node_put(tegra_rt5677_dai.codec_of_node); + tegra_rt5677_dai.codec_of_node = NULL; + of_node_put(tegra_rt5677_dai.cpu_of_node); + tegra_rt5677_dai.cpu_of_node = NULL; + return 0; } -- cgit v1.2.3 From 6cea3590820819049df5945136b8a5acd72ed0f8 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 3 Jun 2018 15:42:31 +0200 Subject: ASoC: Intel: bytcr_rt5640: Add quirk for the Nuvison/TMax TM800W560 tablet Add a quirk for the Nuvison/TMax TM800W560 tablet, this tablet uses IN1 for the internal mic rather then the default IN3 and it uses JD2 rather then JD1 for its not-inverted jack-detect switch. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 5c4f9ea40f57..8571f41767ef 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -565,6 +565,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_DIFF_MIC | BYT_RT5640_MCLK_EN), }, + { /* Nuvison/TMax TM800W560 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TMAX"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TM800W560L"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Pipo W4 */ .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), -- cgit v1.2.3 From f12a0a3c4cc6f594d7c2ea361f2396ae5c518d2c Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 3 Jun 2018 15:42:32 +0200 Subject: ASoC: Intel: bytcr_rt5640: Fix Acer Iconia 8 over-current detect threshold Change the over-current detect threshold on the Acer Iconia 8 from 2000ua to 1500uA, this fixes headset button presses not being detected. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 8571f41767ef..7456566c5648 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -404,7 +404,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | BYT_RT5640_JD_SRC_JD1_IN4P | - BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_TH_1500UA | BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), -- cgit v1.2.3 From cd31b80736852d34bc1072f3e579a6fd73a244e7 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 17 Jun 2018 19:02:17 +0200 Subject: ARM: pxa: change SSP DMA channels allocation Now the dma_slave_map is available for PXA architecture, switch the SSP device to it. This specifically means that : - for platform data based machines, the DMA requestor channels are extracted from the slave map, where pxa-ssp-dai. is a 1-1 match to ssp., and the channels are either "rx" or "tx". - for device tree platforms, the dma node should be hooked into the pxa2xx-ac97 or pxa-ssp-dai node. Signed-off-by: Robert Jarzmik Acked-by: Daniel Mack --- arch/arm/plat-pxa/ssp.c | 47 ---------------------------------------------- include/linux/pxa2xx_ssp.h | 2 -- sound/soc/pxa/pxa-ssp.c | 5 ++--- 3 files changed, 2 insertions(+), 52 deletions(-) (limited to 'sound/soc') diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c index ba13f793fbce..ed36dcab80f1 100644 --- a/arch/arm/plat-pxa/ssp.c +++ b/arch/arm/plat-pxa/ssp.c @@ -127,53 +127,6 @@ static int pxa_ssp_probe(struct platform_device *pdev) if (IS_ERR(ssp->clk)) return PTR_ERR(ssp->clk); - if (dev->of_node) { - struct of_phandle_args dma_spec; - struct device_node *np = dev->of_node; - int ret; - - /* - * FIXME: we should allocate the DMA channel from this - * context and pass the channel down to the ssp users. - * For now, we lookup the rx and tx indices manually - */ - - /* rx */ - ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", - 0, &dma_spec); - - if (ret) { - dev_err(dev, "Can't parse dmas property\n"); - return -ENODEV; - } - ssp->drcmr_rx = dma_spec.args[0]; - of_node_put(dma_spec.np); - - /* tx */ - ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", - 1, &dma_spec); - if (ret) { - dev_err(dev, "Can't parse dmas property\n"); - return -ENODEV; - } - ssp->drcmr_tx = dma_spec.args[0]; - of_node_put(dma_spec.np); - } else { - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (res == NULL) { - dev_err(dev, "no SSP RX DRCMR defined\n"); - return -ENODEV; - } - ssp->drcmr_rx = res->start; - - res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (res == NULL) { - dev_err(dev, "no SSP TX DRCMR defined\n"); - return -ENODEV; - } - ssp->drcmr_tx = res->start; - } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (res == NULL) { dev_err(dev, "no memory resource defined\n"); diff --git a/include/linux/pxa2xx_ssp.h b/include/linux/pxa2xx_ssp.h index 8461b18e4608..03a7ca46735b 100644 --- a/include/linux/pxa2xx_ssp.h +++ b/include/linux/pxa2xx_ssp.h @@ -212,8 +212,6 @@ struct ssp_device { int type; int use_count; int irq; - int drcmr_rx; - int drcmr_tx; struct device_node *of_node; }; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 6fc986080130..0b441338bdd4 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -105,9 +105,8 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) return -ENOMEM; - - dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - &ssp->drcmr_tx : &ssp->drcmr_rx; + dma->chan_name = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "tx" : "rx"; snd_soc_dai_set_dma_data(cpu_dai, substream, dma); -- cgit v1.2.3 From 5fd46e649ee63259f2197625662477ac67a69e79 Mon Sep 17 00:00:00 2001 From: Mac Chiang Date: Tue, 19 Jun 2018 14:16:06 +0800 Subject: ASoC: Intel: kbl_da7219_max98357a: add fe_ops for kbl Audio Capture Port platform support fixed constraint hw_prams as Stereo, 48KHz, 16 bits. This fixed the headset mic recorded noise due to mono capturing request from some apps. e.g. online Voice Recorder Signed-off-by: Louis Collard Signed-off-by: Mac Chiang Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 94294c27d1db..7961f1fd18bd 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -380,6 +380,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .trigger = { SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", -- cgit v1.2.3 From 95555f580dca21fac5ea35c10fa92fa034bd403f Mon Sep 17 00:00:00 2001 From: Naveen Manohar Date: Mon, 18 Jun 2018 13:29:35 -0500 Subject: ASoC: Intel: broxton: reduce machine name for bxt_da7219_max98357a Use truncated names in bxt id table and bxt_da7219_max98357a machine as platform device id table expects names to be less then 20chars. Signed-off-by: Naveen Manohar Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 4 ++-- sound/soc/intel/skylake/skl.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 40eb979d5ac1..3aba5bcf806a 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -586,7 +586,7 @@ static int broxton_audio_probe(struct platform_device *pdev) static struct platform_driver broxton_audio = { .probe = broxton_audio_probe, .driver = { - .name = "bxt_da7219_max98357a_i2s", + .name = "bxt_da7219_max98357a", .pm = &snd_soc_pm_ops, }, }; @@ -599,4 +599,4 @@ MODULE_AUTHOR("Rohit Ainapure "); MODULE_AUTHOR("Harsha Priya "); MODULE_AUTHOR("Conrad Cooke "); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:bxt_da7219_max98357a_i2s"); +MODULE_ALIAS("platform:bxt_da7219_max98357a"); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index f0d9793f872a..0a8f0768e987 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -1093,7 +1093,7 @@ static struct snd_soc_acpi_mach sst_bxtp_devdata[] = { }, { .id = "DLGS7219", - .drv_name = "bxt_da7219_max98357a_i2s", + .drv_name = "bxt_da7219_max98357a", .fw_filename = "intel/dsp_fw_bxtn.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &bxt_codecs, -- cgit v1.2.3 From 5f15f267daf81a4c7c2a1cd2a0d6743ec7fc8b59 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 18 Jun 2018 13:29:36 -0500 Subject: ASoC: Intel: Skylake: cleanup before moving ACPI tables There is no need to deal with DMICs if the DSP is not present and there is no ACPI machine ID found. Simplify before moving these ACPI tables to sound/soc/intel/common Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 0a8f0768e987..6dec748e8949 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -500,10 +500,12 @@ static int skl_find_machine(struct skl *skl, void *driver_data) skl->mach = mach; skl->fw_name = mach->fw_filename; - pdata = skl->mach->pdata; + pdata = mach->pdata; - if (mach->pdata) + if (pdata) { skl->use_tplg_pcm = pdata->use_tplg_pcm; + pdata->dmic_num = skl_get_dmic_geo(skl); + } return 0; } @@ -930,8 +932,6 @@ static int skl_probe(struct pci_dev *pci, pci_set_drvdata(skl->pci, ebus); - skl_dmic_data.dmic_num = skl_get_dmic_geo(skl); - /* check if dsp is there */ if (bus->ppcap) { /* create device for dsp clk */ -- cgit v1.2.3 From cbaa7f0bdbee1969bb311c641abbd0d2af6ba861 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 18 Jun 2018 13:29:37 -0500 Subject: ASoC: Intel: move SKL+ codec ACPI tables to common directory No functionality change, just move to common tables to make it easier to deal with SOF and share the same machine drivers - as done previously for BYT/CHT/HSW/BDW. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- include/sound/soc-acpi-intel-match.h | 5 + sound/soc/intel/common/Makefile | 6 +- sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 35 +++++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 29 ++++ sound/soc/intel/common/soc-acpi-intel-glk-match.c | 23 +++ sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 91 ++++++++++++ sound/soc/intel/common/soc-acpi-intel-skl-match.c | 47 +++++++ sound/soc/intel/skylake/skl.c | 162 +--------------------- 8 files changed, 241 insertions(+), 157 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-bxt-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-cnl-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-glk-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-kbl-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-skl-match.c (limited to 'sound/soc') diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index 9da6388c20a1..917ddd0f2762 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -29,5 +29,10 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_legacy_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[]; #endif diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 7379d8830c39..915a34cdc8ac 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -3,7 +3,11 @@ snd-soc-sst-dsp-objs := sst-dsp.o snd-soc-sst-acpi-objs := sst-acpi.o snd-soc-sst-ipc-objs := sst-ipc.o snd-soc-sst-firmware-objs := sst-firmware.o -snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o soc-acpi-intel-hsw-bdw-match.o +snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o \ + soc-acpi-intel-hsw-bdw-match.o \ + soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \ + soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \ + soc-acpi-intel-cnl-match.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c new file mode 100644 index 000000000000..569f1de97e82 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -0,0 +1,35 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-bxt-match.c - tables and support for BXT ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include +#include + +static struct snd_soc_acpi_codecs bxt_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { + { + .id = "INT343A", + .drv_name = "bxt_alc298s_i2s", + .fw_filename = "intel/dsp_fw_bxtn.bin", + }, + { + .id = "DLGS7219", + .drv_name = "bxt_da7219_max98357a", + .fw_filename = "intel/dsp_fw_bxtn.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &bxt_codecs, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_bxt_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c new file mode 100644 index 000000000000..b83ee053d417 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -0,0 +1,29 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-cnl-match.c - tables and support for CNL ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include +#include +#include "../skylake/skl.h" + +static struct skl_machine_pdata cnl_pdata = { + .use_tplg_pcm = true, +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { + { + .id = "INT34C2", + .drv_name = "cnl_rt274", + .fw_filename = "intel/dsp_fw_cnl.bin", + .pdata = &cnl_pdata, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c new file mode 100644 index 000000000000..dee09439e7cb --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -0,0 +1,23 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-glk-match.c - tables and support for GLK ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include +#include + +struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { + { + .id = "INT343A", + .drv_name = "glk_alc298s_i2s", + .fw_filename = "intel/dsp_fw_glk.bin", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_glk_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c new file mode 100644 index 000000000000..0ee173ca437d --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -0,0 +1,91 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-kbl-match.c - tables and support for KBL ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include +#include +#include "../skylake/skl.h" + +static struct skl_machine_pdata skl_dmic_data; + +static struct snd_soc_acpi_codecs kbl_codecs = { + .num_codecs = 1, + .codecs = {"10508825"} +}; + +static struct snd_soc_acpi_codecs kbl_poppy_codecs = { + .num_codecs = 1, + .codecs = {"10EC5663"} +}; + +static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = { + .num_codecs = 2, + .codecs = {"10EC5663", "10EC5514"} +}; + +static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { + { + .id = "INT343A", + .drv_name = "kbl_alc286s_i2s", + .fw_filename = "intel/dsp_fw_kbl.bin", + }, + { + .id = "INT343B", + .drv_name = "kbl_n88l25_s4567", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98357A", + .drv_name = "kbl_n88l25_m98357a", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98927", + .drv_name = "kbl_r5514_5663_max", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_5663_5514_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98927", + .drv_name = "kbl_rt5663_m98927", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_poppy_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "10EC5663", + .drv_name = "kbl_rt5663", + .fw_filename = "intel/dsp_fw_kbl.bin", + }, + { + .id = "DLGS7219", + .drv_name = "kbl_da7219_max98357a", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_7219_98357_codecs, + .pdata = &skl_dmic_data, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-skl-match.c b/sound/soc/intel/common/soc-acpi-intel-skl-match.c new file mode 100644 index 000000000000..0c9c0edd35b3 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-skl-match.c @@ -0,0 +1,47 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-skl-match.c - tables and support for SKL ACPI enumeration. + * + * Copyright (c) 2018, Intel Corporation. + * + */ + +#include +#include +#include "../skylake/skl.h" + +static struct skl_machine_pdata skl_dmic_data; + +static struct snd_soc_acpi_codecs skl_codecs = { + .num_codecs = 1, + .codecs = {"10508825"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[] = { + { + .id = "INT343A", + .drv_name = "skl_alc286s_i2s", + .fw_filename = "intel/dsp_fw_release.bin", + }, + { + .id = "INT343B", + .drv_name = "skl_n88l25_s4567", + .fw_filename = "intel/dsp_fw_release.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &skl_codecs, + .pdata = &skl_dmic_data, + }, + { + .id = "MX98357A", + .drv_name = "skl_n88l25_m98357a", + .fw_filename = "intel/dsp_fw_release.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &skl_codecs, + .pdata = &skl_dmic_data, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_skl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 6dec748e8949..670ff9aaca55 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -29,6 +29,7 @@ #include #include #include +#include #include #include #include @@ -36,8 +37,6 @@ #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" -static struct skl_machine_pdata skl_dmic_data; - /* * initialize the PCI registers */ @@ -1026,172 +1025,23 @@ static void skl_remove(struct pci_dev *pci) dev_set_drvdata(&pci->dev, NULL); } -static struct snd_soc_acpi_codecs skl_codecs = { - .num_codecs = 1, - .codecs = {"10508825"} -}; - -static struct snd_soc_acpi_codecs kbl_codecs = { - .num_codecs = 1, - .codecs = {"10508825"} -}; - -static struct snd_soc_acpi_codecs bxt_codecs = { - .num_codecs = 1, - .codecs = {"MX98357A"} -}; - -static struct snd_soc_acpi_codecs kbl_poppy_codecs = { - .num_codecs = 1, - .codecs = {"10EC5663"} -}; - -static struct snd_soc_acpi_codecs kbl_5663_5514_codecs = { - .num_codecs = 2, - .codecs = {"10EC5663", "10EC5514"} -}; - -static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = { - .num_codecs = 1, - .codecs = {"MX98357A"} -}; - -static struct skl_machine_pdata cnl_pdata = { - .use_tplg_pcm = true, -}; - -static struct snd_soc_acpi_mach sst_skl_devdata[] = { - { - .id = "INT343A", - .drv_name = "skl_alc286s_i2s", - .fw_filename = "intel/dsp_fw_release.bin", - }, - { - .id = "INT343B", - .drv_name = "skl_n88l25_s4567", - .fw_filename = "intel/dsp_fw_release.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &skl_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98357A", - .drv_name = "skl_n88l25_m98357a", - .fw_filename = "intel/dsp_fw_release.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &skl_codecs, - .pdata = &skl_dmic_data - }, - {} -}; - -static struct snd_soc_acpi_mach sst_bxtp_devdata[] = { - { - .id = "INT343A", - .drv_name = "bxt_alc298s_i2s", - .fw_filename = "intel/dsp_fw_bxtn.bin", - }, - { - .id = "DLGS7219", - .drv_name = "bxt_da7219_max98357a", - .fw_filename = "intel/dsp_fw_bxtn.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &bxt_codecs, - }, - {} -}; - -static struct snd_soc_acpi_mach sst_kbl_devdata[] = { - { - .id = "INT343A", - .drv_name = "kbl_alc286s_i2s", - .fw_filename = "intel/dsp_fw_kbl.bin", - }, - { - .id = "INT343B", - .drv_name = "kbl_n88l25_s4567", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98357A", - .drv_name = "kbl_n88l25_m98357a", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98927", - .drv_name = "kbl_r5514_5663_max", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_5663_5514_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "MX98927", - .drv_name = "kbl_rt5663_m98927", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_poppy_codecs, - .pdata = &skl_dmic_data - }, - { - .id = "10EC5663", - .drv_name = "kbl_rt5663", - .fw_filename = "intel/dsp_fw_kbl.bin", - }, - { - .id = "DLGS7219", - .drv_name = "kbl_da7219_max98357a", - .fw_filename = "intel/dsp_fw_kbl.bin", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &kbl_7219_98357_codecs, - .pdata = &skl_dmic_data - }, - - {} -}; - -static struct snd_soc_acpi_mach sst_glk_devdata[] = { - { - .id = "INT343A", - .drv_name = "glk_alc298s_i2s", - .fw_filename = "intel/dsp_fw_glk.bin", - }, - {} -}; - -static const struct snd_soc_acpi_mach sst_cnl_devdata[] = { - { - .id = "INT34C2", - .drv_name = "cnl_rt274", - .fw_filename = "intel/dsp_fw_cnl.bin", - .pdata = &cnl_pdata, - }, - {} -}; - /* PCI IDs */ static const struct pci_device_id skl_ids[] = { /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), - .driver_data = (unsigned long)&sst_skl_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_skl_machines}, /* BXT-P */ { PCI_DEVICE(0x8086, 0x5a98), - .driver_data = (unsigned long)&sst_bxtp_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_bxt_machines}, /* KBL */ { PCI_DEVICE(0x8086, 0x9D71), - .driver_data = (unsigned long)&sst_kbl_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_kbl_machines}, /* GLK */ { PCI_DEVICE(0x8086, 0x3198), - .driver_data = (unsigned long)&sst_glk_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_glk_machines}, /* CNL */ { PCI_DEVICE(0x8086, 0x9dc8), - .driver_data = (unsigned long)&sst_cnl_devdata}, + .driver_data = (unsigned long)&snd_soc_acpi_intel_cnl_machines}, { 0, } }; MODULE_DEVICE_TABLE(pci, skl_ids); -- cgit v1.2.3 From 65a33883c778befcb85ef45285763fd8ac1b2ba3 Mon Sep 17 00:00:00 2001 From: Naveen Manohar Date: Mon, 18 Jun 2018 13:29:38 -0500 Subject: ASoC: Intel: common: Add Geminilake Dialog+Maxim machine driver entry This patch adds da7219_max98357a machine driver entry into machine table Signed-off-by: Naveen Manohar Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-glk-match.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index dee09439e7cb..5902aa1d0ee3 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -9,12 +9,24 @@ #include #include +static struct snd_soc_acpi_codecs glk_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { { .id = "INT343A", .drv_name = "glk_alc298s_i2s", .fw_filename = "intel/dsp_fw_glk.bin", }, + { + .id = "DLGS7219", + .drv_name = "glk_da7219_max98357a", + .fw_filename = "intel/dsp_fw_glk.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &glk_codecs, + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_glk_machines); -- cgit v1.2.3 From e6d298fd4a4454dd121343323e3f00a27f8819a4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 18 Jun 2018 13:29:39 -0500 Subject: ASoC: Intel: common: add firmware/topology information for SOF No functionality change for Skylake driver, add relevant names needed by SOF for BXT/APL, GLK and CNL. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 3 +++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 3 +++ sound/soc/intel/common/soc-acpi-intel-glk-match.c | 6 ++++++ 3 files changed, 12 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c index 569f1de97e82..50869eddbb3b 100644 --- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -26,6 +26,9 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { .fw_filename = "intel/dsp_fw_bxtn.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &bxt_codecs, + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-da7219.tplg", + .asoc_plat_name = "0000:00:0e.0", }, {}, }; diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index b83ee053d417..ec8e28e7b937 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -20,6 +20,9 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { .drv_name = "cnl_rt274", .fw_filename = "intel/dsp_fw_cnl.bin", .pdata = &cnl_pdata, + .sof_fw_filename = "intel/sof-cnl.ri", + .sof_tplg_filename = "intel/sof-cnl-rt274.tplg", + .asoc_plat_name = "0000:00:1f.3", }, {}, }; diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index 5902aa1d0ee3..305875af71ca 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -19,6 +19,9 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { .id = "INT343A", .drv_name = "glk_alc298s_i2s", .fw_filename = "intel/dsp_fw_glk.bin", + .sof_fw_filename = "intel/sof-glk.ri", + .sof_tplg_filename = "intel/sof-glk-alc298.tplg", + .asoc_plat_name = "0000:00:0e.0", }, { .id = "DLGS7219", @@ -26,6 +29,9 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { .fw_filename = "intel/dsp_fw_glk.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &glk_codecs, + .sof_fw_filename = "intel/sof-glk.ri", + .sof_tplg_filename = "intel/sof-glk-da7219.tplg", + .asoc_plat_name = "0000:00:0e.0", }, {}, }; -- cgit v1.2.3 From b45350135b9241b64cc91ccc8dddca2ee4dc25d7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 18 Jun 2018 13:29:41 -0500 Subject: ASoC: Intel: common: add entries for SOF-based machine drivers While we are at it, add entries for machine drivers that are used on SOF-based platforms. The drivers will be submitted upstream after the core SOF patches, but there's no harm in adding these references now. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c index 50869eddbb3b..f39386e540d3 100644 --- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -30,6 +30,27 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { .sof_tplg_filename = "intel/sof-apl-da7219.tplg", .asoc_plat_name = "0000:00:0e.0", }, + { + .id = "104C5122", + .drv_name = "bxt-pcm512x", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-pcm512x.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + { + .id = "1AEC8804", + .drv_name = "bxt-wm8804", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-wm8804.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, + { + .id = "INT34C3", + .drv_name = "bxt_tdf8532", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-tdf8532.tplg", + .asoc_plat_name = "0000:00:0e.0", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_bxt_machines); -- cgit v1.2.3 From f0d9034b290d8bad590e843c2a1081eb47d813ad Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 18 Jun 2018 13:29:42 -0500 Subject: ASoC: Intel: common: fix copy/paste issue with SOF/broadwell topology file There are two commercially-available Broadwell platforms based on I2S (Dell XPS13 and 'Samus' Pixel 2015 Chromebook). Fix a copy/paste issue to allow each platform to enable different features if needed when SOF is enabled Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index e0e8c8c27528..53e7acdfbb81 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -45,7 +45,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { .drv_name = "bdw-rt5677", .fw_filename = "intel/IntcSST2.bin", .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt286.tplg", + .sof_tplg_filename = "intel/reef-bdw-rt5677.tplg", .asoc_plat_name = "haswell-pcm-audio", }, { -- cgit v1.2.3 From 244e293690a6e07cbdfa11af1977488d91931eed Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 19 Jun 2018 16:22:09 +0100 Subject: ASoC: pcm: Tidy up open/hw_params handling Currently, the core will continue processing open/hw_params component callbacks after one has failed even though it will abort immediately afterwards. This is unnecessary and also has the issue that close/hw_free will be called on the component which failed open/hw_params which could result in issues if the driver doesn't expect this behaviour. Update the core to abort processing open/hw_params when an error is hit and only call close/hw_free for those components that were successfully opened. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 116 ++++++++++++++++++++++++++++------------------------ 1 file changed, 62 insertions(+), 54 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5e7ae47a9658..45b52f7b9690 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -448,6 +448,29 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) hw->rate_max = min_not_zero(hw->rate_max, rate_max); } +static int soc_pcm_components_close(struct snd_pcm_substream *substream, + struct snd_soc_component *last) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (component == last) + break; + + if (!component->driver->ops || + !component->driver->ops->close) + continue; + + component->driver->ops->close(substream); + } + + return 0; +} + /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls @@ -462,7 +485,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; const char *codec_dai_name = "multicodec"; - int i, ret = 0, __ret; + int i, ret = 0; pinctrl_pm_select_default_state(cpu_dai->dev); for (i = 0; i < rtd->num_codecs; i++) @@ -486,7 +509,6 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - ret = 0; for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -494,16 +516,15 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) !component->driver->ops->open) continue; - __ret = component->driver->ops->open(substream); - if (__ret < 0) { + ret = component->driver->ops->open(substream); + if (ret < 0) { dev_err(component->dev, "ASoC: can't open component %s: %d\n", - component->name, __ret); - ret = __ret; + component->name, ret); + goto component_err; } } - if (ret < 0) - goto component_err; + component = NULL; for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; @@ -612,15 +633,7 @@ codec_dai_err: } component_err: - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->close) - continue; - - component->driver->ops->close(substream); - } + soc_pcm_components_close(substream, component); if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); @@ -714,15 +727,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->close) - continue; - - component->driver->ops->close(substream); - } + soc_pcm_components_close(substream, NULL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (snd_soc_runtime_ignore_pmdown_time(rtd)) { @@ -874,6 +879,29 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, return 0; } +static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_component *last) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (component == last) + break; + + if (!component->driver->ops || + !component->driver->ops->hw_free) + continue; + + component->driver->ops->hw_free(substream); + } + + return 0; +} + /* * Called by ALSA when the hardware params are set by application. This * function can also be called multiple times and can allocate buffers @@ -886,7 +914,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret = 0, __ret; + int i, ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (rtd->dai_link->ops->hw_params) { @@ -944,7 +972,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (ret < 0) goto interface_err; - ret = 0; for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -952,16 +979,15 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, !component->driver->ops->hw_params) continue; - __ret = component->driver->ops->hw_params(substream, params); - if (__ret < 0) { + ret = component->driver->ops->hw_params(substream, params); + if (ret < 0) { dev_err(component->dev, "ASoC: %s hw params failed: %d\n", - component->name, __ret); - ret = __ret; + component->name, ret); + goto component_err; } } - if (ret < 0) - goto component_err; + component = NULL; /* store the parameters for each DAIs */ cpu_dai->rate = params_rate(params); @@ -977,15 +1003,7 @@ out: return ret; component_err: - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->hw_free) - continue; - - component->driver->ops->hw_free(substream); - } + soc_pcm_components_hw_free(substream, component); if (cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); @@ -1014,8 +1032,6 @@ codec_err: static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component; - struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; @@ -1052,15 +1068,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) rtd->dai_link->ops->hw_free(substream); /* free any component resources */ - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; - - if (!component->driver->ops || - !component->driver->ops->hw_free) - continue; - - component->driver->ops->hw_free(substream); - } + soc_pcm_components_hw_free(substream, NULL); /* now free hw params for the DAIs */ for (i = 0; i < rtd->num_codecs; i++) { -- cgit v1.2.3 From 7f7cca08abf4c6c58ff1037cff5beec5a10a7da3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Jun 2018 11:56:21 +0100 Subject: ASoC: wm_adsp: Simplify handling of alg offset and length The current code that reads the algorithm list from the DSP is somewhat unclear, it converts directly from bytes to registers using a hard coded divide by 2. Most offsets are usually handled in DSP words within the driver and there is a function specifically for converting from words to register addresses. So update the handling to use these. This also removes the assumption that the registers are 16-bit word addressed, which will no longer be true on some of our newer parts. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 29 ++++++++++++++++++++--------- 1 file changed, 20 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 2fcdd84021a5..07c17acc8a4f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1871,9 +1871,11 @@ static void wm_adsp_ctl_fixup_base(struct wm_adsp *dsp, } static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, + const struct wm_adsp_region *mem, unsigned int pos, unsigned int len) { void *alg; + unsigned int reg; int ret; __be32 val; @@ -1888,7 +1890,9 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, } /* Read the terminator first to validate the length */ - ret = regmap_raw_read(dsp->regmap, pos + len, &val, sizeof(val)); + reg = wm_adsp_region_to_reg(mem, pos + len); + + ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val)); if (ret != 0) { adsp_err(dsp, "Failed to read algorithm list end: %d\n", ret); @@ -1897,13 +1901,18 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, if (be32_to_cpu(val) != 0xbedead) adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbedead\n", - pos + len, be32_to_cpu(val)); + reg, be32_to_cpu(val)); + + /* Convert length from DSP words to bytes */ + len *= sizeof(u32); alg = kcalloc(len, 2, GFP_KERNEL | GFP_DMA); if (!alg) return ERR_PTR(-ENOMEM); - ret = regmap_raw_read(dsp->regmap, pos, alg, len * 2); + reg = wm_adsp_region_to_reg(mem, pos); + + ret = regmap_raw_read(dsp->regmap, reg, alg, len); if (ret != 0) { adsp_err(dsp, "Failed to read algorithm list: %d\n", ret); kfree(alg); @@ -2002,10 +2011,11 @@ static int wm_adsp1_setup_algs(struct wm_adsp *dsp) if (IS_ERR(alg_region)) return PTR_ERR(alg_region); - pos = sizeof(adsp1_id) / 2; - len = (sizeof(*adsp1_alg) * n_algs) / 2; + /* Calculate offset and length in DSP words */ + pos = sizeof(adsp1_id) / sizeof(u32); + len = (sizeof(*adsp1_alg) * n_algs) / sizeof(u32); - adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len); + adsp1_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len); if (IS_ERR(adsp1_alg)) return PTR_ERR(adsp1_alg); @@ -2113,10 +2123,11 @@ static int wm_adsp2_setup_algs(struct wm_adsp *dsp) if (IS_ERR(alg_region)) return PTR_ERR(alg_region); - pos = sizeof(adsp2_id) / 2; - len = (sizeof(*adsp2_alg) * n_algs) / 2; + /* Calculate offset and length in DSP words */ + pos = sizeof(adsp2_id) / sizeof(u32); + len = (sizeof(*adsp2_alg) * n_algs) / sizeof(u32); - adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem->base + pos, len); + adsp2_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len); if (IS_ERR(adsp2_alg)) return PTR_ERR(adsp2_alg); -- cgit v1.2.3 From 1b31de922e28de2bf2078b7a1e341ad4aee6aa03 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Jun 2018 11:56:20 +0100 Subject: ASoC: arizona: Set compressed IRQ to a wake source The current code is not setting the compressed IRQ as a wake source. Normally this doesn't cause any issues as the CODEC IRQ is set as a wake source by the jack detection code and the CODEC only produces a single IRQ line. However if the system is not using jack detection the compressed audio IRQ should still function as a wake source, as such directly set the compressed audio IRQ as a wake source. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/cs47l24.c | 8 ++++++++ sound/soc/codecs/wm5102.c | 8 ++++++++ sound/soc/codecs/wm5110.c | 8 ++++++++ 3 files changed, 24 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 196e9c343aeb..0da52ead91e0 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1283,6 +1283,12 @@ static int cs47l24_probe(struct platform_device *pdev) return ret; } + ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1); + if (ret != 0) + dev_warn(&pdev->dev, + "Failed to set compressed IRQ as a wake source: %d\n", + ret); + arizona_init_common(arizona); ret = arizona_init_vol_limit(arizona); @@ -1306,6 +1312,7 @@ static int cs47l24_probe(struct platform_device *pdev) err_spk_irqs: arizona_free_spk_irqs(arizona); err_dsp_irq: + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, cs47l24); return ret; @@ -1323,6 +1330,7 @@ static int cs47l24_remove(struct platform_device *pdev) arizona_free_spk_irqs(arizona); + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, cs47l24); return 0; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 1ac83388d1b8..a01a0c0e01eb 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2094,6 +2094,12 @@ static int wm5102_probe(struct platform_device *pdev) return ret; } + ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1); + if (ret != 0) + dev_warn(&pdev->dev, + "Failed to set compressed IRQ as a wake source: %d\n", + ret); + arizona_init_common(arizona); ret = arizona_init_vol_limit(arizona); @@ -2117,6 +2123,7 @@ static int wm5102_probe(struct platform_device *pdev) err_spk_irqs: arizona_free_spk_irqs(arizona); err_dsp_irq: + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5102); return ret; @@ -2133,6 +2140,7 @@ static int wm5102_remove(struct platform_device *pdev) arizona_free_spk_irqs(arizona); + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5102); return 0; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index fb9835dcd836..00c735c585d9 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2455,6 +2455,12 @@ static int wm5110_probe(struct platform_device *pdev) return ret; } + ret = arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 1); + if (ret != 0) + dev_warn(&pdev->dev, + "Failed to set compressed IRQ as a wake source: %d\n", + ret); + arizona_init_common(arizona); ret = arizona_init_vol_limit(arizona); @@ -2478,6 +2484,7 @@ static int wm5110_probe(struct platform_device *pdev) err_spk_irqs: arizona_free_spk_irqs(arizona); err_dsp_irq: + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5110); return ret; @@ -2496,6 +2503,7 @@ static int wm5110_remove(struct platform_device *pdev) arizona_free_spk_irqs(arizona); + arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5110); return 0; -- cgit v1.2.3 From 4f722a6a736765310bc95dd3879f1545e5f64303 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 19 Jun 2018 14:00:37 -0500 Subject: ASoC: Intel: common: rename 'reef' to 'sof' in ACPI matching table Align with firmware tools, no functionality change Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 40 ++++++++--------- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 52 +++++++++++----------- .../intel/common/soc-acpi-intel-hsw-bdw-match.c | 16 +++---- 3 files changed, 54 insertions(+), 54 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index bfe1ca68a542..4daa8a4f0c0c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -59,8 +59,8 @@ static struct snd_soc_acpi_mach byt_thinkpad_10 = { .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5670.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5670.tplg", .asoc_plat_name = "sst-mfld-platform", }; @@ -98,8 +98,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5640", .machine_quirk = byt_quirk, - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5640.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -107,8 +107,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcr_rt5640", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5640", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5640.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -116,8 +116,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcr_rt5640", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5640", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5640.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -125,8 +125,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcr_rt5651", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcr_rt5651", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5651.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5651.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -134,8 +134,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-da7213.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -143,8 +143,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_0f28.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-da7213.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* some Baytrail platforms rely on RT5645, use CHT machine driver */ @@ -153,8 +153,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5645.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -162,8 +162,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-rt5645.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* use CHT driver to Baytrail Chromebooks */ @@ -172,8 +172,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .drv_name = "cht-bsw-max98090", .fw_filename = "intel/fw_sst_0f28.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-byt.ri", - .sof_tplg_filename = "intel/reef-byt-max98090.tplg", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-max98090.tplg", .asoc_plat_name = "sst-mfld-platform", }, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index ad1eb2d644be..3c3f2f8585d7 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -44,8 +44,8 @@ static struct snd_soc_acpi_mach cht_surface_mach = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }; @@ -68,8 +68,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5670.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5670.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -77,8 +77,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5672", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5670.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5670.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -86,8 +86,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -95,8 +95,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -104,8 +104,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-rt5645", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5645.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5645.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -113,8 +113,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-max98090", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-max98090.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-max98090.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -131,8 +131,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-da7213.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -140,8 +140,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_da7213", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_da7213", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-da7213.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-da7213.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -149,8 +149,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcht_es8316", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcht_es8316", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-es8316.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-es8316.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* some CHT-T platforms rely on RT5640, use Baytrail machine driver */ @@ -160,8 +160,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcr_rt5640", .machine_quirk = cht_quirk, - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5640.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, { @@ -169,8 +169,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcr_rt5640", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcr_rt5640", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5640.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5640.tplg", .asoc_plat_name = "sst-mfld-platform", }, /* some CHT-T platforms rely on RT5651, use Baytrail machine driver */ @@ -179,8 +179,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "bytcr_rt5651", .fw_filename = "intel/fw_sst_22a8.bin", .board = "bytcr_rt5651", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-rt5651.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-rt5651.tplg", .asoc_plat_name = "sst-mfld-platform", }, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index 53e7acdfbb81..494a0ea9b029 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -23,8 +23,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_haswell_machines[] = { .id = "INT33CA", .drv_name = "haswell-audio", .fw_filename = "intel/IntcSST1.bin", - .sof_fw_filename = "intel/reef-hsw.ri", - .sof_tplg_filename = "intel/reef-hsw.tplg", + .sof_fw_filename = "intel/sof-hsw.ri", + .sof_tplg_filename = "intel/sof-hsw.tplg", .asoc_plat_name = "haswell-pcm-audio", }, {} @@ -36,24 +36,24 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { .id = "INT343A", .drv_name = "broadwell-audio", .fw_filename = "intel/IntcSST2.bin", - .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt286.tplg", + .sof_fw_filename = "intel/sof-bdw.ri", + .sof_tplg_filename = "intel/sof-bdw-rt286.tplg", .asoc_plat_name = "haswell-pcm-audio", }, { .id = "RT5677CE", .drv_name = "bdw-rt5677", .fw_filename = "intel/IntcSST2.bin", - .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt5677.tplg", + .sof_fw_filename = "intel/sof-bdw.ri", + .sof_tplg_filename = "intel/sof-bdw-rt5677.tplg", .asoc_plat_name = "haswell-pcm-audio", }, { .id = "INT33CA", .drv_name = "haswell-audio", .fw_filename = "intel/IntcSST2.bin", - .sof_fw_filename = "intel/reef-bdw.ri", - .sof_tplg_filename = "intel/reef-bdw-rt5640.tplg", + .sof_fw_filename = "intel/sof-bdw.ri", + .sof_tplg_filename = "intel/sof-bdw-rt5640.tplg", .asoc_plat_name = "haswell-pcm-audio", }, {} -- cgit v1.2.3 From bb450fa59c0772310ebcc704722dd8a0313bf8ed Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 21 Jun 2018 15:47:35 -0500 Subject: ASoC: Intel: common: fix missing rename from 'reef' to 'sof' Somehow I missed the Nau8824 support which was added in 4.17. Oops Fixes: 4f722a6a736 ("ASoC: Intel: common: rename 'reef' to 'sof' in ACPI matching table") Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 3c3f2f8585d7..91bb99b69601 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -122,8 +122,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .drv_name = "cht-bsw-nau8824", .fw_filename = "intel/fw_sst_22a8.bin", .board = "cht-bsw", - .sof_fw_filename = "intel/reef-cht.ri", - .sof_tplg_filename = "intel/reef-cht-nau8824.tplg", + .sof_fw_filename = "intel/sof-cht.ri", + .sof_tplg_filename = "intel/sof-cht-nau8824.tplg", .asoc_plat_name = "sst-mfld-platform", }, { -- cgit v1.2.3 From a98ec93d7e378db21a6cdbc6d55feaba9fb4b999 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Fri, 22 Jun 2018 02:23:24 +0000 Subject: ASoC: rt5682: use devm_snd_soc_register_component() Using devm_snd_soc_register_component() and drop all of the code related to .remove hook. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 61a97301bcfa..baad177908ab 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2630,17 +2630,11 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, } - return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682, + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_rt5682, rt5682_dai, ARRAY_SIZE(rt5682_dai)); } -static int rt5682_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_component(&i2c->dev); - - return 0; -} - static void rt5682_i2c_shutdown(struct i2c_client *client) { struct rt5682_priv *rt5682 = i2c_get_clientdata(client); @@ -2671,7 +2665,6 @@ static struct i2c_driver rt5682_i2c_driver = { .acpi_match_table = ACPI_PTR(rt5682_acpi_match), }, .probe = rt5682_i2c_probe, - .remove = rt5682_i2c_remove, .shutdown = rt5682_i2c_shutdown, .id_table = rt5682_i2c_id, }; -- cgit v1.2.3 From 2854a214f398f2c3315204a05efff11739fec062 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Fri, 22 Jun 2018 02:23:34 +0000 Subject: ASoC: rt1305: use devm_snd_soc_register_component() Using devm_snd_soc_register_component() and drop all of the code related to .remove hook. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/codecs/rt1305.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index 421b8fb2fa04..c4452efc7970 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -1150,17 +1150,11 @@ static int rt1305_i2c_probe(struct i2c_client *i2c, rt1305_reset(rt1305->regmap); rt1305_calibrate(rt1305); - return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt1305, + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_rt1305, rt1305_dai, ARRAY_SIZE(rt1305_dai)); } -static int rt1305_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_component(&i2c->dev); - - return 0; -} - static void rt1305_i2c_shutdown(struct i2c_client *client) { struct rt1305_priv *rt1305 = i2c_get_clientdata(client); @@ -1180,7 +1174,6 @@ static struct i2c_driver rt1305_i2c_driver = { #endif }, .probe = rt1305_i2c_probe, - .remove = rt1305_i2c_remove, .shutdown = rt1305_i2c_shutdown, .id_table = rt1305_i2c_id, }; -- cgit v1.2.3 From 366f074d047b2538cc5d7e4a820bb6c117a3ccec Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Thu, 21 Jun 2018 11:56:28 +0900 Subject: ASoC: uniphier: remove redundant check of PLL ID This patch removes redudant check of PLL ID. struct uniphier_aio_pll enable member has already been checked at is_valid_pll(). Signed-off-by: Katsuhiro Suzuki Signed-off-by: Mark Brown --- sound/soc/uniphier/aio-cpu.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index 2d9b7dde2ffa..ee90e6c3937c 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -219,15 +219,10 @@ static int uniphier_aio_set_pll(struct snd_soc_dai *dai, int pll_id, unsigned int freq_out) { struct uniphier_aio *aio = uniphier_priv(dai); - struct device *dev = &aio->chip->pdev->dev; int ret; if (!is_valid_pll(aio->chip, pll_id)) return -EINVAL; - if (!aio->chip->plls[pll_id].enable) { - dev_err(dev, "PLL(%d) is not implemented\n", pll_id); - return -ENOTSUPP; - } ret = aio_chip_set_pll(aio->chip, pll_id, freq_out); if (ret < 0) -- cgit v1.2.3 From 3bec6fa3cdd338860d4b7d4110a49292646d801e Mon Sep 17 00:00:00 2001 From: Agrawal, Akshu Date: Thu, 21 Jun 2018 12:58:16 +0800 Subject: ASoC: AMD: Change codec to channel link as per hardware redesign This is a correction to match acutal hardware configuration. The hardware configuration looks like: I2S_BT -> SPK(Max) + DMIC(Adau) I2S_SP -> DA7219 Headset No actual products have been shipped with previous configuration. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 27 +++++++++++++++++++-------- 1 file changed, 19 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index ccddc6650b9c..566bd268be3a 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -148,7 +148,7 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->i2s_instance = I2S_SP_INSTANCE; return da7219_clk_enable(substream); } @@ -163,7 +163,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -178,7 +178,7 @@ static int cz_dmic_startup(struct snd_pcm_substream *substream) struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -204,16 +204,27 @@ static const struct snd_soc_ops cz_dmic_cap_ops = { static struct snd_soc_dai_link cz_dai_7219_98357[] = { { - .name = "amd-da7219-play-cap", - .stream_name = "Playback and Capture", + .name = "amd-da7219-play", + .stream_name = "Playback", .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.3.auto", + .cpu_dai_name = "designware-i2s.1.auto", .codec_dai_name = "da7219-hifi", .codec_name = "i2c-DLGS7219:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, + .ops = &cz_da7219_cap_ops, + }, + { + .name = "amd-da7219-cap", + .stream_name = "Capture", + .platform_name = "acp_audio_dma.0.auto", + .cpu_dai_name = "designware-i2s.2.auto", + .codec_dai_name = "da7219-hifi", + .codec_name = "i2c-DLGS7219:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, .ops = &cz_da7219_cap_ops, }, @@ -221,7 +232,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "amd-max98357-play", .stream_name = "HiFi Playback", .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.1.auto", + .cpu_dai_name = "designware-i2s.3.auto", .codec_dai_name = "HiFi", .codec_name = "MX98357A:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF @@ -233,7 +244,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .name = "dmic", .stream_name = "DMIC Capture", .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.2.auto", + .cpu_dai_name = "designware-i2s.3.auto", .codec_dai_name = "adau7002-hifi", .codec_name = "ADAU7002:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF -- cgit v1.2.3 From 2718c89a233bf8549fdba0925947b2c3cb887a95 Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Thu, 21 Jun 2018 12:58:17 +0800 Subject: ASoC: AMD: Configure channel 1 or channel 0 for capture ST/CZ SoC have 2 channels for capture in the I2SSP path. The DMA though these channels is done using the same dma descriptors. We configure the channel and enable it on the basis of channel selected by machine driver. Machine driver knows which codec sits on which channel and thus sends the information to dma driver. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 43 +++++++++++++++++++--- sound/soc/amd/acp-pcm-dma.c | 71 +++++++++++++++++++++++++++++++++++- sound/soc/amd/acp.h | 4 ++ 3 files changed, 111 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 566bd268be3a..f42606e5879e 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -149,6 +149,7 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); } @@ -172,7 +173,7 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } -static int cz_dmic_startup(struct snd_pcm_substream *substream) +static int cz_dmic0_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; @@ -182,6 +183,17 @@ static int cz_dmic_startup(struct snd_pcm_substream *substream) return da7219_clk_enable(substream); } +static int cz_dmic1_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL0; + return da7219_clk_enable(substream); +} + static void cz_dmic_shutdown(struct snd_pcm_substream *substream) { da7219_clk_disable(); @@ -197,8 +209,13 @@ static const struct snd_soc_ops cz_max_play_ops = { .shutdown = cz_max_shutdown, }; -static const struct snd_soc_ops cz_dmic_cap_ops = { - .startup = cz_dmic_startup, +static const struct snd_soc_ops cz_dmic0_cap_ops = { + .startup = cz_dmic0_startup, + .shutdown = cz_dmic_shutdown, +}; + +static const struct snd_soc_ops cz_dmic1_cap_ops = { + .startup = cz_dmic1_startup, .shutdown = cz_dmic_shutdown, }; @@ -241,8 +258,9 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .ops = &cz_max_play_ops, }, { - .name = "dmic", - .stream_name = "DMIC Capture", + /* C panel DMIC */ + .name = "dmic0", + .stream_name = "DMIC0 Capture", .platform_name = "acp_audio_dma.0.auto", .cpu_dai_name = "designware-i2s.3.auto", .codec_dai_name = "adau7002-hifi", @@ -250,7 +268,20 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, - .ops = &cz_dmic_cap_ops, + .ops = &cz_dmic0_cap_ops, + }, + { + /* A/B panel DMIC */ + .name = "dmic1", + .stream_name = "DMIC1 Capture", + .platform_name = "acp_audio_dma.0.auto", + .cpu_dai_name = "designware-i2s.2.auto", + .codec_dai_name = "adau7002-hifi", + .codec_name = "ADAU7002:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_dmic1_cap_ops, }, }; diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 1458b5048498..3c3d398d0d0b 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -336,6 +336,61 @@ static void config_acp_dma(void __iomem *acp_mmio, rtd->dma_dscr_idx_2, asic_type); } +static void acp_dma_cap_channel_enable(void __iomem *acp_mmio, + u16 cap_channel) +{ + u32 val, ch_reg, imr_reg, res_reg; + + switch (cap_channel) { + case CAP_CHANNEL1: + ch_reg = mmACP_I2SMICSP_RER1; + res_reg = mmACP_I2SMICSP_RCR1; + imr_reg = mmACP_I2SMICSP_IMR1; + break; + case CAP_CHANNEL0: + default: + ch_reg = mmACP_I2SMICSP_RER0; + res_reg = mmACP_I2SMICSP_RCR0; + imr_reg = mmACP_I2SMICSP_IMR0; + break; + } + val = acp_reg_read(acp_mmio, + mmACP_I2S_16BIT_RESOLUTION_EN); + if (val & ACP_I2S_MIC_16BIT_RESOLUTION_EN) { + acp_reg_write(0x0, acp_mmio, ch_reg); + /* Set 16bit resolution on capture */ + acp_reg_write(0x2, acp_mmio, res_reg); + } + val = acp_reg_read(acp_mmio, imr_reg); + val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK; + val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; + acp_reg_write(val, acp_mmio, imr_reg); + acp_reg_write(0x1, acp_mmio, ch_reg); +} + +static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, + u16 cap_channel) +{ + u32 val, ch_reg, imr_reg; + + switch (cap_channel) { + case CAP_CHANNEL1: + imr_reg = mmACP_I2SMICSP_IMR1; + ch_reg = mmACP_I2SMICSP_RER1; + break; + case CAP_CHANNEL0: + default: + imr_reg = mmACP_I2SMICSP_IMR0; + ch_reg = mmACP_I2SMICSP_RER0; + break; + } + val = acp_reg_read(acp_mmio, imr_reg); + val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK; + val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; + acp_reg_write(val, acp_mmio, imr_reg); + acp_reg_write(0x0, acp_mmio, ch_reg); +} + /* Start a given DMA channel transfer */ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) { @@ -773,8 +828,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, if (WARN_ON(!rtd)) return -EINVAL; - if (pinfo) + if (pinfo) { rtd->i2s_instance = pinfo->i2s_instance; + rtd->capture_channel = pinfo->capture_channel; + } if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); @@ -990,6 +1047,18 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_start(rtd->acp_mmio, rtd->ch1); acp_dma_start(rtd->acp_mmio, rtd->ch2); } else { + if (rtd->capture_channel == CAP_CHANNEL0) { + acp_dma_cap_channel_disable(rtd->acp_mmio, + CAP_CHANNEL1); + acp_dma_cap_channel_enable(rtd->acp_mmio, + CAP_CHANNEL0); + } + if (rtd->capture_channel == CAP_CHANNEL1) { + acp_dma_cap_channel_disable(rtd->acp_mmio, + CAP_CHANNEL0); + acp_dma_cap_channel_enable(rtd->acp_mmio, + CAP_CHANNEL1); + } acp_dma_start(rtd->acp_mmio, rtd->ch2); acp_dma_start(rtd->acp_mmio, rtd->ch1); } diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 9cd3e96c84d4..3190fdce6307 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -55,6 +55,8 @@ #define I2S_SP_INSTANCE 0x01 #define I2S_BT_INSTANCE 0x02 +#define CAP_CHANNEL0 0x00 +#define CAP_CHANNEL1 0x01 #define ACP_TILE_ON_MASK 0x03 #define ACP_TILE_OFF_MASK 0x02 @@ -125,6 +127,7 @@ struct audio_substream_data { unsigned int order; u16 num_of_pages; u16 i2s_instance; + u16 capture_channel; u16 direction; u16 ch1; u16 ch2; @@ -155,6 +158,7 @@ struct audio_drv_data { */ struct acp_platform_info { u16 i2s_instance; + u16 capture_channel; }; union acp_dma_count { -- cgit v1.2.3 From f4c277b817cc9489fffabffb4e15d2f3b686056c Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Wed, 20 Jun 2018 18:25:20 +0900 Subject: ASoC: soc-pcm: DPCM cares BE channel constraint Current DPCM is caring only FE channel configuration. Sometimes it will be trouble if user selects channel which isn't supported by BE. This patch adds new .dpcm_merged_chan on struct snd_soc_dai_link. DPCM will use FE / BE merged channel if struct snd_soc_dai_link has it. Signed-off-by: Jiada Wang Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ sound/soc/soc-pcm.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 48 insertions(+) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index 1378dcd2128a..f7579f82cc00 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -957,6 +957,8 @@ struct snd_soc_dai_link { /* DPCM used FE & BE merged format */ unsigned int dpcm_merged_format:1; + /* DPCM used FE & BE merged channel */ + unsigned int dpcm_merged_chan:1; /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 45b52f7b9690..19ebfc958b9d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1715,6 +1715,46 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) return formats; } +static void dpcm_runtime_base_chan(struct snd_pcm_substream *substream, + unsigned int *channels_min, + unsigned int *channels_max) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = substream->stream; + + if (!fe->dai_link->dpcm_merged_chan) + return; + + *channels_min = 0; + *channels_max = UINT_MAX; + + /* + * It returns merged BE codec channel; + * if FE want to use it (= dpcm_merged_chan) + */ + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_pcm_stream *codec_stream; + int i; + + for (i = 0; i < be->num_codecs; i++) { + codec_dai_drv = be->codec_dais[i]->driver; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + + *channels_min = max(*channels_min, + codec_stream->channels_min); + *channels_max = min(*channels_max, + codec_stream->channels_max); + } + } +} + static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1722,11 +1762,17 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; u64 format = dpcm_runtime_base_format(substream); + unsigned int channels_min = 0, channels_max = UINT_MAX; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback, format); else dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture, format); + + dpcm_runtime_base_chan(substream, &channels_min, &channels_max); + + runtime->hw.channels_min = max(channels_min, runtime->hw.channels_min); + runtime->hw.channels_max = min(channels_max, runtime->hw.channels_max); } static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd); -- cgit v1.2.3 From 50c678772a0b3f9dc6b04262f27e2c373fe03a4d Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Sun, 24 Jun 2018 20:14:41 +0200 Subject: ASoC: cx20442: Don't ignore regulator_get() errors. In its current shape, the driver just ignores errors returned by regulator_get() at component_probe(). This doesn't hurt on Amstrad Delta board as long as it registers the codec device at late_initcall, when the regulator which depends on basic-mmio-gpio device (probed as late as at dev_initcall) is already available. Otherwise the driver may end up trying to control a codec which is not powered up. Remove that dependency on initialization order by handling the error. If the regulator is not yet available and -ENODEV is returned, convert it to -EPROBE_DEFER to get another chance. Signed-off-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/codecs/cx20442.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 07dd33b09596..ab174b5114dc 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -362,8 +362,27 @@ static int cx20442_component_probe(struct snd_soc_component *component) return -ENOMEM; cx20442->por = regulator_get(component->dev, "POR"); - if (IS_ERR(cx20442->por)) - dev_warn(component->dev, "failed to get the regulator"); + if (IS_ERR(cx20442->por)) { + int err = PTR_ERR(cx20442->por); + + dev_warn(component->dev, "failed to get POR supply (%d)", err); + /* + * When running on a non-dt platform and requested regulator + * is not available, regulator_get() never returns + * -EPROBE_DEFER as it is not able to justify if the regulator + * may still appear later. On the other hand, the board can + * still set full constraints flag at late_initcall in order + * to instruct regulator_get() to return a dummy one if + * sufficient. Hence, if we get -ENODEV here, let's convert + * it to -EPROBE_DEFER and wait for the board to decide or + * let Deferred Probe infrastructure handle this error. + */ + if (err == -ENODEV) + err = -EPROBE_DEFER; + kfree(cx20442); + return err; + } + cx20442->tty = NULL; snd_soc_component_set_drvdata(component, cx20442); -- cgit v1.2.3 From b66c9b911fe6ca188002b342b05c43deab4491a3 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 26 Jun 2018 08:58:35 -0300 Subject: ASoC: soc-utils: Fix unregistration order The unregistration should happen in the opposite order of the registration, so change it accordingly. No real issue has been noticed, but it is good practice to keep the correct unregistration order. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 2d9e98bd1530..a863bb3f66c2 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -381,6 +381,6 @@ int __init snd_soc_util_init(void) void __exit snd_soc_util_exit(void) { - platform_device_unregister(soc_dummy_dev); platform_driver_unregister(&soc_dummy_driver); + platform_device_unregister(soc_dummy_dev); } -- cgit v1.2.3 From f614c9b070ed149bbaac4edefb2b5fcb7755c4b0 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:08 +0100 Subject: ASoC: qdsp6: q6adm: use of_platform_populate/depopulate() Now that the child nodes have there own compatible strings, Use of_platform_populate/depopulate() instead of less common of_platform_device_create()/destroy(). Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6adm.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 9983c665a941..932c3ebfd252 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -64,7 +64,6 @@ struct q6adm { struct aprv2_ibasic_rsp_result_t result; struct mutex lock; wait_queue_head_t matrix_map_wait; - struct platform_device *pdev_routing; }; struct q6adm_cmd_device_open_v5 { @@ -588,7 +587,6 @@ EXPORT_SYMBOL_GPL(q6adm_close); static int q6adm_probe(struct apr_device *adev) { struct device *dev = &adev->dev; - struct device_node *dais_np; struct q6adm *adm; adm = devm_kzalloc(&adev->dev, sizeof(*adm), GFP_KERNEL); @@ -605,22 +603,12 @@ static int q6adm_probe(struct apr_device *adev) INIT_LIST_HEAD(&adm->copps_list); spin_lock_init(&adm->copps_list_lock); - dais_np = of_get_child_by_name(dev->of_node, "routing"); - if (dais_np) { - adm->pdev_routing = of_platform_device_create(dais_np, - "q6routing", dev); - of_node_put(dais_np); - } - - return 0; + return of_platform_populate(dev->of_node, NULL, NULL, dev); } static int q6adm_remove(struct apr_device *adev) { - struct q6adm *adm = dev_get_drvdata(&adev->dev); - - if (adm->pdev_routing) - of_platform_device_destroy(&adm->pdev_routing->dev, NULL); + of_platform_depopulate(&adev->dev); return 0; } -- cgit v1.2.3 From 4aac7e2773030d667491fbb6d97c9f467fdcbc05 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:09 +0100 Subject: ASoC: qdsp6: q6asm: use of_platform_populate/depopulate() Now that the child nodes have there own compatible strings, Use of_platform_populate/depopulate() instead of less common of_platform_device_create()/destroy(). Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 530852385cad..c4fd28f168d5 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -177,7 +177,6 @@ struct q6asm { struct platform_device *pcmdev; spinlock_t slock; struct audio_client *session[MAX_SESSIONS + 1]; - struct platform_device *pdev_dais; }; struct audio_client { @@ -1344,7 +1343,6 @@ EXPORT_SYMBOL_GPL(q6asm_cmd_nowait); static int q6asm_probe(struct apr_device *adev) { struct device *dev = &adev->dev; - struct device_node *dais_np; struct q6asm *q6asm; q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL); @@ -1359,22 +1357,12 @@ static int q6asm_probe(struct apr_device *adev) spin_lock_init(&q6asm->slock); dev_set_drvdata(dev, q6asm); - dais_np = of_get_child_by_name(dev->of_node, "dais"); - if (dais_np) { - q6asm->pdev_dais = of_platform_device_create(dais_np, - "q6asm-dai", dev); - of_node_put(dais_np); - } - - return 0; + return of_platform_populate(dev->of_node, NULL, NULL, dev); } static int q6asm_remove(struct apr_device *adev) { - struct q6asm *q6asm = dev_get_drvdata(&adev->dev); - - if (q6asm->pdev_dais) - of_platform_device_destroy(&q6asm->pdev_dais->dev, NULL); + of_platform_depopulate(&adev->dev); return 0; } -- cgit v1.2.3 From 01afbd45f78cb0557db18c3ba768eea3e9576cfd Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:10 +0100 Subject: ASoC: qdsp6: q6afe: use of_platform_populate/depopulate() Now that the child nodes have there own compatible strings, Use of_platform_populate/depopulate() instead of less common of_platform_device_create()/destroy(). Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 01f43218984b..621b67b34db9 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -316,7 +316,6 @@ struct q6afe { struct mutex lock; struct list_head port_list; spinlock_t port_list_lock; - struct platform_device *pdev_dais; }; struct afe_port_cmd_device_start { @@ -1438,7 +1437,6 @@ static int q6afe_probe(struct apr_device *adev) { struct q6afe *afe; struct device *dev = &adev->dev; - struct device_node *dais_np; afe = devm_kzalloc(dev, sizeof(*afe), GFP_KERNEL); if (!afe) @@ -1453,22 +1451,12 @@ static int q6afe_probe(struct apr_device *adev) dev_set_drvdata(dev, afe); - dais_np = of_get_child_by_name(dev->of_node, "dais"); - if (dais_np) { - afe->pdev_dais = of_platform_device_create(dais_np, - "q6afe-dai", dev); - of_node_put(dais_np); - } - - return 0; + return of_platform_populate(dev->of_node, NULL, NULL, dev); } static int q6afe_remove(struct apr_device *adev) { - struct q6afe *afe = dev_get_drvdata(&adev->dev); - - if (afe->pdev_dais) - of_platform_device_destroy(&afe->pdev_dais->dev, NULL); + of_platform_depopulate(&adev->dev); return 0; } -- cgit v1.2.3 From eb7cc9be6e9ce17e69252a6fc00e75a5b08201ab Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:11 +0100 Subject: ASoC: qdsp6: q6afe-dai: support dt based module loading This patch uses new compatible string to make DT based module loading work. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 5002dd05bf27..1d2e5013c121 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1290,9 +1290,16 @@ static int q6afe_dai_dev_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id q6afe_dai_device_id[] = { + { .compatible = "qcom,q6afe-dais" }, + {}, +}; +MODULE_DEVICE_TABLE(of, q6afe_dai_device_id); + static struct platform_driver q6afe_dai_platform_driver = { .driver = { .name = "q6afe-dai", + .of_match_table = of_match_ptr(q6afe_dai_device_id), }, .probe = q6afe_dai_dev_probe, .remove = q6afe_dai_dev_remove, -- cgit v1.2.3 From 1ce09ef36fb190fa207fdb3e31fc1c8caa292125 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:12 +0100 Subject: ASoC: qdsp6: q6asm-dai: support dt based module loading This patch uses new compatible string to make DT based module loading work. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 349c6a883c63..1196dc7483d2 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -611,9 +611,16 @@ static int q6asm_dai_dev_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id q6asm_dai_device_id[] = { + { .compatible = "qcom,q6asm-dais" }, + {}, +}; +MODULE_DEVICE_TABLE(of, q6asm_dai_device_id); + static struct platform_driver q6asm_dai_platform_driver = { .driver = { .name = "q6asm-dai", + .of_match_table = of_match_ptr(q6asm_dai_device_id), }, .probe = q6asm_dai_probe, .remove = q6asm_dai_dev_remove, -- cgit v1.2.3 From f48bde4bfbcf434d6aef604c1c50d68b12a4bc45 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:13 +0100 Subject: ASoC: qdsp6: q6routing: support dt based module loading This patch uses new compatible string to make DT based module loading work. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 593f66b8622f..ab696bf8d1d3 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -993,9 +993,16 @@ static int q6pcm_routing_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id q6pcm_routing_device_id[] = { + { .compatible = "qcom,q6adm-routing" }, + {}, +}; +MODULE_DEVICE_TABLE(of, q6pcm_routing_device_id); + static struct platform_driver q6pcm_routing_platform_driver = { .driver = { .name = "q6routing", + .of_match_table = of_match_ptr(q6pcm_routing_device_id), }, .probe = q6pcm_routing_probe, .remove = q6pcm_routing_remove, -- cgit v1.2.3 From 2d12c20b98ad610892151da37367f2d018181455 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:14 +0100 Subject: ASoC: qcom: apq8096: remove redundant owner assignment module owner is already set in platform_driver_register(), so remove this redundant assignment. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/apq8096.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 239b8cb77bdb..cab8c4ff7c00 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -246,7 +246,6 @@ static struct platform_driver msm_snd_apq8096_driver = { .remove = apq8096_platform_remove, .driver = { .name = "msm-snd-apq8096", - .owner = THIS_MODULE, .of_match_table = msm_snd_apq8096_dt_match, }, }; -- cgit v1.2.3 From 972562f7aaaa261ec6e1ac14fed0c5bdd0dfb1cb Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:15 +0100 Subject: ASoC: qdsp6: q6routing: add proper error check q6adm_open can return error pointer or a null in error cases. Fix the return handling. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index ab696bf8d1d3..c80fdbc2442e 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -310,7 +310,7 @@ int q6routing_stream_open(int fedai_id, int perf_mode, session->channels, topology, perf_mode, session->bits_per_sample, 0, 0); - if (!copp) { + if (IS_ERR_OR_NULL(copp)) { mutex_unlock(&routing_data->lock); return -EINVAL; } -- cgit v1.2.3 From f339155a4063cf3177bb38f6e83760951148e86b Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 26 Jun 2018 10:20:16 +0100 Subject: ASoC: qdsp6: q6asm: remove unused struct q6asm member pcmdev in struct q6asm seems be left over and unused, so just remove it. Signed-off-by: Srinivas Kandagatla Acked-by: Niklas Cassel Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index c4fd28f168d5..2b2c7233bb5f 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -174,7 +174,6 @@ struct q6asm { struct device *dev; struct q6core_svc_api_info ainfo; wait_queue_head_t mem_wait; - struct platform_device *pcmdev; spinlock_t slock; struct audio_client *session[MAX_SESSIONS + 1]; }; -- cgit v1.2.3 From fc7c460fbb4003bf3f5d2b435079c85888644663 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:26 +0200 Subject: ASoC: Intel: bytcr_rt5651: Add BYT_RT5651_DEFAULT_QUIRKS define Almost all boards use the mclk and use the same jack-detect settings, add a BYT_RT5651_DEFAULT_QUIRKS define for this. This shaves of some lines and makes it easier to see which settings are unique to a certain model. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 25 +++++++++---------------- 1 file changed, 9 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 987720e203f9..735b8312e275 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -77,6 +77,11 @@ enum { #define BYT_RT5651_SSP0_AIF1 BIT(20) #define BYT_RT5651_SSP0_AIF2 BIT(21) +#define BYT_RT5651_DEFAULT_QUIRKS (BYT_RT5651_MCLK_EN | \ + BYT_RT5651_JD1_1 | \ + BYT_RT5651_OVCD_TH_2000UA | \ + BYT_RT5651_OVCD_SF_0P75) + /* jack-detect-source + dmic-en + ovcd-th + -sf + terminating empty entry */ #define MAX_NO_PROPS 5 @@ -86,10 +91,7 @@ struct byt_rt5651_private { }; /* Default: jack-detect on JD1_1, internal mic on in2, headsetmic on in3 */ -static unsigned long byt_rt5651_quirk = BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_0P75 | +static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN2_HS_IN3_MAP; static void log_quirks(struct device *dev) @@ -379,10 +381,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "KIANO"), DMI_MATCH(DMI_PRODUCT_NAME, "KIANO SlimNote 14.2"), }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_0P75 | + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN1_IN2_MAP), }, { @@ -392,10 +391,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_0P75 | + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN2_HS_IN3_MAP), }, { @@ -405,10 +401,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "VIOS"), DMI_MATCH(DMI_PRODUCT_NAME, "LTH17"), }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_JD1_1 | - BYT_RT5651_OVCD_TH_2000UA | - BYT_RT5651_OVCD_SF_1P0 | + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN1_IN2_MAP), }, {} -- cgit v1.2.3 From 10876d24eb40c6bfaa0aabd97e3e143258176c53 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:27 +0200 Subject: ASoC: Intel: bytcr_rt5651: Change default input map from in2 to in1 Further testing on all 6 model x86 tablets with a rt5651 which I have access to for testing has shown that their single (mono) microphone is connected to both IN1 *and* IN2. The previous default mapping of IN2 was based on testing on the same 6 tablets, where the internal mic works fine with a mapping of IN2. But it works fine too with a mapping of IN1. This commit changes the default input mapping to to use IN1 instead of IN2, to match the mapping used for the other mono devices in the DMI quirk table. So that we need less different mappings. The same change is made to the Chuwi Vi8 Plus quirks, which is one of the 6 models tested. This is a preparation patch for simplifying the maps in a follow-up commit. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 735b8312e275..910890de38b0 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -90,9 +90,9 @@ struct byt_rt5651_private { struct snd_soc_jack jack; }; -/* Default: jack-detect on JD1_1, internal mic on in2, headsetmic on in3 */ +/* Default: jack-detect on JD1_1, internal mic on in1, headsetmic on in3 */ static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN2_HS_IN3_MAP; + BYT_RT5651_IN1_HS_IN3_MAP; static void log_quirks(struct device *dev) { @@ -392,7 +392,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN2_HS_IN3_MAP), + BYT_RT5651_IN1_HS_IN3_MAP), }, { /* VIOS LTH17 */ -- cgit v1.2.3 From 366780df3e2d40533cc95a4bf25ddd3b934b5fd3 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:28 +0200 Subject: ASoC: Intel: bytcr_rt5651: Fix IN1_IN2_MAP quirk not being logged Fix the quirk logging code not logging the IN1_IN2_MAP quirk. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 910890de38b0..7cc6e36b7c47 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -102,6 +102,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk IN1_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP) dev_info(dev, "quirk IN2_MAP enabled"); + if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_IN2_MAP) + dev_info(dev, "quirk IN1_IN2_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_HS_IN3_MAP) dev_info(dev, "quirk IN1_HS_IN3_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_HS_IN3_MAP) -- cgit v1.2.3 From fcdf1391caa6f7f01de56eea63e070555771fac7 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:29 +0200 Subject: ASoC: Intel: bytcr_rt5651: Remove IN2 input mappings BYT_RT5651_IN2_MAP was introduced in commit 39712db878a4 ("SoC: intel: byt: Introduce new custom IN2 map"), uses in commit 2fe30129b0a6 ("ASoC: intel: byt: Enable IN2 map quirk for a KIANO laptop"), only to be replaced by a new BYT_RT5651_IN1_IN2_MAP quirk in commit ea261bd02a67 ("ASoC: intel: byt: Introduce new map for dual mics") quickly afterwards, because the KIANO laptop has 2 internal mics on IN1 and IN2 and the headset mic is not in IN1 where the BYT_RT5651_IN2_MAP maps it, but on IN3. Now that the KIANO quirk entry uses BYT_RT5651_IN1_IN2_MAP, there are no users of BYT_RT5651_IN2_MAP left. This makes sense since the headset mic seems to always be connected to IN3, so BYT_RT5651_IN2_MAP is not useful. To deal with BYT_RT5651_IN2_MAP wrongly mapping the headset mic to IN1, BYT_RT5651_IN2_HS_IN3_MAP was added in commit f026e0631780 ("ASoC: Intel: bytcr_rt5651: Add new IN2_HS_IN3 input map and a quirk using it"). This was based on the assumption then some devices have the internal mic connected to IN2 only. Further testing has shown that this is wrong and the internal mic is always connected to IN1 and sometimes to both IN1 and IN2. TL;DR: Both BYT_RT5651_IN2_MAP and BYT_RT5651_IN2_HS_IN3_MAP are based on on wrong assumptions from the past and are no longer useful now, so they can both be removed. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 30 ++---------------------------- 1 file changed, 2 insertions(+), 28 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 7cc6e36b7c47..b8ca7137d8b2 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -40,10 +40,8 @@ enum { BYT_RT5651_DMIC_MAP, BYT_RT5651_IN1_MAP, - BYT_RT5651_IN2_MAP, BYT_RT5651_IN1_IN2_MAP, BYT_RT5651_IN1_HS_IN3_MAP, - BYT_RT5651_IN2_HS_IN3_MAP, }; enum { @@ -100,14 +98,10 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk DMIC_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_MAP) dev_info(dev, "quirk IN1_MAP enabled"); - if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP) - dev_info(dev, "quirk IN2_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_IN2_MAP) dev_info(dev, "quirk IN1_IN2_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_HS_IN3_MAP) dev_info(dev, "quirk IN1_HS_IN3_MAP enabled"); - if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_HS_IN3_MAP) - dev_info(dev, "quirk IN2_HS_IN3_MAP enabled"); if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) { dev_info(dev, "quirk realtek,jack-detect-source %ld\n", BYT_RT5651_JDSRC(byt_rt5651_quirk)); @@ -256,12 +250,6 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = { {"IN2P", NULL, "Headset Mic"}, }; -static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = { - {"Internal Mic", NULL, "micbias1"}, - {"IN1P", NULL, "Headset Mic"}, - {"IN2P", NULL, "Internal Mic"}, -}; - static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { {"Internal Mic", NULL, "micbias1"}, {"IN1P", NULL, "Internal Mic"}, @@ -275,12 +263,6 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_hs_in3_map[] = { {"IN3P", NULL, "Headset Mic"}, }; -static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_hs_in3_map[] = { - {"Internal Mic", NULL, "micbias1"}, - {"IN2P", NULL, "Internal Mic"}, - {"IN3P", NULL, "Headset Mic"}, -}; - static const struct snd_soc_dapm_route byt_rt5651_ssp0_aif1_map[] = { {"ssp0 Tx", NULL, "modem_out"}, {"modem_in", NULL, "ssp0 Rx"}, @@ -462,10 +444,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in1_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_map); break; - case BYT_RT5651_IN2_MAP: - custom_map = byt_rt5651_intmic_in2_map; - num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_map); - break; case BYT_RT5651_IN1_IN2_MAP: custom_map = byt_rt5651_intmic_in1_in2_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_in2_map); @@ -474,10 +452,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in1_hs_in3_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_hs_in3_map); break; - case BYT_RT5651_IN2_HS_IN3_MAP: - custom_map = byt_rt5651_intmic_in2_hs_in3_map; - num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_hs_in3_map); - break; default: custom_map = byt_rt5651_intmic_dmic_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map); @@ -723,9 +697,9 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { const char * const intmic_name[] = - { "dmic", "in1", "in2", "in12", "in1", "in2" }; + { "dmic", "in1", "in12", "in1" }; const char * const hsmic_name[] = - { "in2", "in2", "in1", "in3", "in3", "in3" }; + { "in2", "in2", "in3", "in3" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; -- cgit v1.2.3 From de23147983013591bc4d6812ce441f351dec6b9d Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:30 +0200 Subject: ASoC: Intel: bytcr_rt5651: Fix IN1 map headsetmic mapping The initial bytcr_rt5651 machine driver commit mapped IN2 as the headset mic. In retrospect this is not correct as all known boards have the headset mic on IN3. To workaround this special IN?_HS_IN3 mappings were added. This commit fixes the original IN1 mapping to correctly have the headset mic on IN3, moves all users of the IN1_HS_IN3 mapping over to the fixed IN1_MAP and drops the now no longer needed IN1_HS_IN3 mapping. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 29 +++++++---------------------- 1 file changed, 7 insertions(+), 22 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index b8ca7137d8b2..bf2adb36f455 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -41,7 +41,6 @@ enum { BYT_RT5651_DMIC_MAP, BYT_RT5651_IN1_MAP, BYT_RT5651_IN1_IN2_MAP, - BYT_RT5651_IN1_HS_IN3_MAP, }; enum { @@ -90,7 +89,7 @@ struct byt_rt5651_private { /* Default: jack-detect on JD1_1, internal mic on in1, headsetmic on in3 */ static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_HS_IN3_MAP; + BYT_RT5651_IN1_MAP; static void log_quirks(struct device *dev) { @@ -100,8 +99,6 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk IN1_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_IN2_MAP) dev_info(dev, "quirk IN1_IN2_MAP enabled"); - if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_HS_IN3_MAP) - dev_info(dev, "quirk IN1_HS_IN3_MAP enabled"); if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) { dev_info(dev, "quirk realtek,jack-detect-source %ld\n", BYT_RT5651_JDSRC(byt_rt5651_quirk)); @@ -247,7 +244,7 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = { static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = { {"Internal Mic", NULL, "micbias1"}, {"IN1P", NULL, "Internal Mic"}, - {"IN2P", NULL, "Headset Mic"}, + {"IN3P", NULL, "Headset Mic"}, }; static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { @@ -257,12 +254,6 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { {"IN3P", NULL, "Headset Mic"}, }; -static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_hs_in3_map[] = { - {"Internal Mic", NULL, "micbias1"}, - {"IN1P", NULL, "Internal Mic"}, - {"IN3P", NULL, "Headset Mic"}, -}; - static const struct snd_soc_dapm_route byt_rt5651_ssp0_aif1_map[] = { {"ssp0 Tx", NULL, "modem_out"}, {"modem_in", NULL, "ssp0 Rx"}, @@ -348,7 +339,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), }, - .driver_data = (void *)(BYT_RT5651_IN1_HS_IN3_MAP), + .driver_data = (void *)(BYT_RT5651_IN1_MAP), }, { .callback = byt_rt5651_quirk_cb, @@ -357,7 +348,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"), }, .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_IN1_HS_IN3_MAP), + BYT_RT5651_IN1_MAP), }, { .callback = byt_rt5651_quirk_cb, @@ -376,7 +367,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_HS_IN3_MAP), + BYT_RT5651_IN1_MAP), }, { /* VIOS LTH17 */ @@ -448,10 +439,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in1_in2_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_in2_map); break; - case BYT_RT5651_IN1_HS_IN3_MAP: - custom_map = byt_rt5651_intmic_in1_hs_in3_map; - num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_hs_in3_map); - break; default: custom_map = byt_rt5651_intmic_dmic_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map); @@ -696,10 +683,8 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { - const char * const intmic_name[] = - { "dmic", "in1", "in12", "in1" }; - const char * const hsmic_name[] = - { "in2", "in2", "in3", "in3" }; + const char * const intmic_name[] = { "dmic", "in1", "in12" }; + const char * const hsmic_name[] = { "in2", "in3", "in3" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; -- cgit v1.2.3 From 37c7401e8c1f583d197c096152fc87a58f460277 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:31 +0200 Subject: ASoC: Intel: bytcr_rt5651: Fix DMIC map headsetmic mapping The initial bytcr_rt5651 machine driver commit mapped IN2 as the headset mic. In retrospect this is not correct as all known boards have the headset mic on IN3. This commit fixes the original DMIC mapping to correctly have the headset mic on IN3. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index bf2adb36f455..042334d9be32 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -236,9 +236,9 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = { - {"IN2P", NULL, "Headset Mic"}, {"DMIC L1", NULL, "Internal Mic"}, {"DMIC R1", NULL, "Internal Mic"}, + {"IN3P", NULL, "Headset Mic"}, }; static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = { @@ -684,7 +684,7 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { const char * const intmic_name[] = { "dmic", "in1", "in12" }; - const char * const hsmic_name[] = { "in2", "in3", "in3" }; + const char * const hsmic_name[] = { "in3", "in3", "in3" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; -- cgit v1.2.3 From 8e69cd640097fa7af53fb476dbd3597608f32b10 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:32 +0200 Subject: ASoC: Intel: bytcr_rt5651: Simplify card long-name Now that the headset-mic is always IN3 there is no reason to have the headset-mic mapping in the long-name. This commit simplifies the long name to "bytcr-rt5651--mic". We can safely do this without causing regressions (UCM profile not found due to the longname change) as the UCM profiles are not in upstream alsa-lib yet. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 042334d9be32..e778142b8a6e 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -662,7 +662,7 @@ static struct snd_soc_card byt_rt5651_card = { static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ -static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-spk-*-mic" */ +static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic" */ static bool is_valleyview(void) { @@ -683,8 +683,7 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { - const char * const intmic_name[] = { "dmic", "in1", "in12" }; - const char * const hsmic_name[] = { "in3", "in3", "in3" }; + const char * const mic_name[] = { "dmic", "in1", "in12" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; @@ -831,9 +830,8 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) } snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name), - "bytcr-rt5651-%s-intmic-%s-hsmic", - intmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], - hsmic_name[BYT_RT5651_MAP(byt_rt5651_quirk)]); + "bytcr-rt5651-%s-mic", + mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)]); byt_rt5651_card.long_name = byt_rt5651_long_name; ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_rt5651_card); -- cgit v1.2.3 From 8f250e7009d71e6f3f3aeb95a540c36fc9c03398 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:33 +0200 Subject: ASoC: Intel: bytcr_rt5651: Add BYT_RT5651_HP_LR_SWAPPED quirk One some models (Chuwi Vi8 Plus, Chuwi Hi8 Pro) the headphone output has left and right swapped. This can be fixed in with special mixer settings in the UCM profile, bit this requires these devices loading a different UCM profile. This commit adds a BYT_RT5651_HP_LR_SWAPPED quirk for this and postfixes the longname with "-hp-swapped" if set, so that a different UCM profile will be loaded. We can safely do this without causing regressions (UCM profile not found due to the longname change) as the UCM profiles are not in upstream alsa-lib yet. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 27 +++++++++++++++++++++++---- 1 file changed, 23 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index e778142b8a6e..ffd62eb5c266 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -73,6 +73,7 @@ enum { #define BYT_RT5651_SSP2_AIF2 BIT(19) /* default is using AIF1 */ #define BYT_RT5651_SSP0_AIF1 BIT(20) #define BYT_RT5651_SSP0_AIF2 BIT(21) +#define BYT_RT5651_HP_LR_SWAPPED BIT(22) #define BYT_RT5651_DEFAULT_QUIRKS (BYT_RT5651_MCLK_EN | \ BYT_RT5651_JD1_1 | \ @@ -359,6 +360,17 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN1_IN2_MAP), }, + { + /* Chuwi Hi8 Pro (CWI513) */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), + DMI_MATCH(DMI_PRODUCT_NAME, "X1D3_C806N"), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN1_MAP | + BYT_RT5651_HP_LR_SWAPPED), + }, { /* Chuwi Vi8 Plus (CWI519) */ .callback = byt_rt5651_quirk_cb, @@ -367,7 +379,8 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_MAP), + BYT_RT5651_IN1_MAP | + BYT_RT5651_HP_LR_SWAPPED), }, { /* VIOS LTH17 */ @@ -662,7 +675,7 @@ static struct snd_soc_card byt_rt5651_card = { static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ -static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic" */ +static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic[-swapped-hp]" */ static bool is_valleyview(void) { @@ -687,6 +700,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; const char *i2c_name = NULL; + const char *hp_swapped; bool is_bytcr = false; int ret_val = 0; int dai_index = 0; @@ -829,9 +843,14 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) } } + if (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) + hp_swapped = "-hp-swapped"; + else + hp_swapped = ""; + snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name), - "bytcr-rt5651-%s-mic", - mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)]); + "bytcr-rt5651-%s-mic%s", + mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], hp_swapped); byt_rt5651_card.long_name = byt_rt5651_long_name; ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_rt5651_card); -- cgit v1.2.3 From 55d69c0309acea65fb3dd99a05a665b51630362d Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 24 Jun 2018 16:06:34 +0200 Subject: ASoC: Intel: bytcr_rt5651: Sort DMI table entries alphabetically As we get more entries in the DMI quirk table it is nice to have some sort of ordering in the table, sort it alphabetically. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 41 +++++++++++++++++++---------------- 1 file changed, 22 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index ffd62eb5c266..ba2753e0e12a 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -335,23 +335,29 @@ static int byt_rt5651_quirk_cb(const struct dmi_system_id *id) static const struct dmi_system_id byt_rt5651_quirk_table[] = { { + /* Chuwi Hi8 Pro (CWI513) */ .callback = byt_rt5651_quirk_cb, .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), - DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), + DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), + DMI_MATCH(DMI_PRODUCT_NAME, "X1D3_C806N"), }, - .driver_data = (void *)(BYT_RT5651_IN1_MAP), + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN1_MAP | + BYT_RT5651_HP_LR_SWAPPED), }, { + /* Chuwi Vi8 Plus (CWI519) */ .callback = byt_rt5651_quirk_cb, .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "ADI"), - DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"), + DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), + DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), }, - .driver_data = (void *)(BYT_RT5651_MCLK_EN | - BYT_RT5651_IN1_MAP), + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN1_MAP | + BYT_RT5651_HP_LR_SWAPPED), }, { + /* KIANO SlimNote 14.2 */ .callback = byt_rt5651_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "KIANO"), @@ -361,26 +367,23 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_IN1_IN2_MAP), }, { - /* Chuwi Hi8 Pro (CWI513) */ + /* Minnowboard Max B3 */ .callback = byt_rt5651_quirk_cb, .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), - DMI_MATCH(DMI_PRODUCT_NAME, "X1D3_C806N"), + DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), + DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), }, - .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_MAP | - BYT_RT5651_HP_LR_SWAPPED), + .driver_data = (void *)(BYT_RT5651_IN1_MAP), }, { - /* Chuwi Vi8 Plus (CWI519) */ + /* Minnowboard Turbot */ .callback = byt_rt5651_quirk_cb, .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Hampoo"), - DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), + DMI_MATCH(DMI_SYS_VENDOR, "ADI"), + DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"), }, - .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_MAP | - BYT_RT5651_HP_LR_SWAPPED), + .driver_data = (void *)(BYT_RT5651_MCLK_EN | + BYT_RT5651_IN1_MAP), }, { /* VIOS LTH17 */ -- cgit v1.2.3 From 8d881bb6216d28896112bdc0b42f1f21eb6cd4ee Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 26 Jun 2018 14:11:27 +0200 Subject: ASoC: simple-amplifier: rename dio2125 to simple-amplifer The dio2125 is simple enough that we can make it a generic component. Just rename and sed the dio2125 amplifier driver to simple_amplifier. Suggested-by: Nicolò Veronese Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 10 +-- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/dio2125.c | 120 ----------------------------------- sound/soc/codecs/simple-amplifier.c | 121 ++++++++++++++++++++++++++++++++++++ 4 files changed, 128 insertions(+), 127 deletions(-) delete mode 100644 sound/soc/codecs/dio2125.c create mode 100644 sound/soc/codecs/simple-amplifier.c (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f6b8d4bf8796..53c2d726bedf 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -74,7 +74,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7219 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DIO2125 select SND_SOC_DMIC if GPIOLIB select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER @@ -144,6 +143,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5682 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE + select SND_SOC_SIMPLE_AMPLIFIER select SND_SOC_SIRF_AUDIO_CODEC select SND_SOC_SPDIF select SND_SOC_SSM2305 @@ -573,10 +573,6 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DIO2125 - tristate "Dioo DIO2125 Amplifier" - select GPIOLIB - config SND_SOC_DMIC tristate @@ -897,6 +893,10 @@ config SND_SOC_SIGMADSP_REGMAP tristate select SND_SOC_SIGMADSP +config SND_SOC_SIMPLE_AMPLIFIER + tristate "Simple Audio Amplifier" + select GPIOLIB + config SND_SOC_SIRF_AUDIO_CODEC tristate "SiRF SoC internal audio codec" select REGMAP_MMIO diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e43d99a039d3..f26ded89a1e5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -250,9 +250,9 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o snd-soc-zx-aud96p22-objs := zx_aud96p22.o # Amp -snd-soc-dio2125-objs := dio2125.o snd-soc-max9877-objs := max9877.o snd-soc-max98504-objs := max98504.o +snd-soc-simple-amplifier-objs := simple-amplifier.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-tas2552-objs := tas2552.o @@ -509,7 +509,7 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o # Amp -obj-$(CONFIG_SND_SOC_DIO2125) += snd-soc-dio2125.o obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_MAX98504) += snd-soc-max98504.o +obj-$(CONFIG_SND_SOC_SIMPLE_AMPLIFIER) += snd-soc-simple-amplifier.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o diff --git a/sound/soc/codecs/dio2125.c b/sound/soc/codecs/dio2125.c deleted file mode 100644 index 09451cd44f9b..000000000000 --- a/sound/soc/codecs/dio2125.c +++ /dev/null @@ -1,120 +0,0 @@ -/* - * Copyright (c) 2017 BayLibre, SAS. - * Author: Jerome Brunet - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of version 2 of the GNU General Public License as - * published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see . - * The full GNU General Public License is included in this distribution - * in the file called COPYING. - */ - -#include -#include -#include - -#define DRV_NAME "dio2125" - -struct dio2125 { - struct gpio_desc *gpiod_enable; -}; - -static int drv_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *control, int event) -{ - struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); - struct dio2125 *priv = snd_soc_component_get_drvdata(c); - int val; - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - val = 1; - break; - case SND_SOC_DAPM_PRE_PMD: - val = 0; - break; - default: - WARN(1, "Unexpected event"); - return -EINVAL; - } - - gpiod_set_value_cansleep(priv->gpiod_enable, val); - - return 0; -} - -static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("INL"), - SND_SOC_DAPM_INPUT("INR"), - SND_SOC_DAPM_OUT_DRV_E("DRV", SND_SOC_NOPM, 0, 0, NULL, 0, drv_event, - (SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD)), - SND_SOC_DAPM_OUTPUT("OUTL"), - SND_SOC_DAPM_OUTPUT("OUTR"), -}; - -static const struct snd_soc_dapm_route dio2125_dapm_routes[] = { - { "DRV", NULL, "INL" }, - { "DRV", NULL, "INR" }, - { "OUTL", NULL, "DRV" }, - { "OUTR", NULL, "DRV" }, -}; - -static const struct snd_soc_component_driver dio2125_component_driver = { - .dapm_widgets = dio2125_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(dio2125_dapm_widgets), - .dapm_routes = dio2125_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(dio2125_dapm_routes), -}; - -static int dio2125_probe(struct platform_device *pdev) -{ - struct device *dev = &pdev->dev; - struct dio2125 *priv; - int err; - - priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); - if (priv == NULL) - return -ENOMEM; - platform_set_drvdata(pdev, priv); - - priv->gpiod_enable = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); - if (IS_ERR(priv->gpiod_enable)) { - err = PTR_ERR(priv->gpiod_enable); - if (err != -EPROBE_DEFER) - dev_err(dev, "Failed to get 'enable' gpio: %d", err); - return err; - } - - return devm_snd_soc_register_component(dev, &dio2125_component_driver, - NULL, 0); -} - -#ifdef CONFIG_OF -static const struct of_device_id dio2125_ids[] = { - { .compatible = "dioo,dio2125", }, - { } -}; -MODULE_DEVICE_TABLE(of, dio2125_ids); -#endif - -static struct platform_driver dio2125_driver = { - .driver = { - .name = DRV_NAME, - .of_match_table = of_match_ptr(dio2125_ids), - }, - .probe = dio2125_probe, -}; - -module_platform_driver(dio2125_driver); - -MODULE_DESCRIPTION("ASoC DIO2125 output driver"); -MODULE_AUTHOR("Jerome Brunet "); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/simple-amplifier.c b/sound/soc/codecs/simple-amplifier.c new file mode 100644 index 000000000000..6c27d4afaf3a --- /dev/null +++ b/sound/soc/codecs/simple-amplifier.c @@ -0,0 +1,121 @@ +/* + * Copyright (c) 2017 BayLibre, SAS. + * Author: Jerome Brunet + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of version 2 of the GNU General Public License as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see . + * The full GNU General Public License is included in this distribution + * in the file called COPYING. + */ + +#include +#include +#include + +#define DRV_NAME "simple-amplifier" + +struct simple_amp { + struct gpio_desc *gpiod_enable; +}; + +static int drv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct simple_amp *priv = snd_soc_component_get_drvdata(c); + int val; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + val = 1; + break; + case SND_SOC_DAPM_PRE_PMD: + val = 0; + break; + default: + WARN(1, "Unexpected event"); + return -EINVAL; + } + + gpiod_set_value_cansleep(priv->gpiod_enable, val); + + return 0; +} + +static const struct snd_soc_dapm_widget simple_amp_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("INL"), + SND_SOC_DAPM_INPUT("INR"), + SND_SOC_DAPM_OUT_DRV_E("DRV", SND_SOC_NOPM, 0, 0, NULL, 0, drv_event, + (SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD)), + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route simple_amp_dapm_routes[] = { + { "DRV", NULL, "INL" }, + { "DRV", NULL, "INR" }, + { "OUTL", NULL, "DRV" }, + { "OUTR", NULL, "DRV" }, +}; + +static const struct snd_soc_component_driver simple_amp_component_driver = { + .dapm_widgets = simple_amp_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(simple_amp_dapm_widgets), + .dapm_routes = simple_amp_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(simple_amp_dapm_routes), +}; + +static int simple_amp_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct simple_amp *priv; + int err; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->gpiod_enable = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); + if (IS_ERR(priv->gpiod_enable)) { + err = PTR_ERR(priv->gpiod_enable); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'enable' gpio: %d", err); + return err; + } + + return devm_snd_soc_register_component(dev, + &simple_amp_component_driver, + NULL, 0); +} + +#ifdef CONFIG_OF +static const struct of_device_id simple_amp_ids[] = { + { .compatible = "dioo,dio2125", }, + { } +}; +MODULE_DEVICE_TABLE(of, simple_amp_ids); +#endif + +static struct platform_driver simple_amp_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = of_match_ptr(simple_amp_ids), + }, + .probe = simple_amp_probe, +}; + +module_platform_driver(simple_amp_driver); + +MODULE_DESCRIPTION("ASoC Simple Audio Amplifier driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 8ed237e83ce9ff4b3964a6b096beb1cbd3397d5a Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 26 Jun 2018 14:11:28 +0200 Subject: ASoC: simple-amplifer: add simple-amplifier compatible Add simple-audio-amplifier to the list of available compatible Suggested-by: Nicolò Veronese Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/simple-amplifier.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/simple-amplifier.c b/sound/soc/codecs/simple-amplifier.c index 6c27d4afaf3a..85524acf3e9c 100644 --- a/sound/soc/codecs/simple-amplifier.c +++ b/sound/soc/codecs/simple-amplifier.c @@ -101,6 +101,7 @@ static int simple_amp_probe(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id simple_amp_ids[] = { { .compatible = "dioo,dio2125", }, + { .compatible = "simple-audio-amplifier", }, { } }; MODULE_DEVICE_TABLE(of, simple_amp_ids); -- cgit v1.2.3 From f516d32262a4c0fef3fccdf2a82671f54f5c1e33 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 27 Jun 2018 09:39:37 +0200 Subject: ASoC: tas517x: add tas5707 support Add support for the tas5707 audio power amplifier. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 ++- sound/soc/codecs/tas571x.c | 110 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas571x.h | 16 +++++++ 3 files changed, 130 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 53c2d726bedf..6d1674699385 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -959,8 +959,11 @@ config SND_SOC_TAS5086 depends on I2C config SND_SOC_TAS571X - tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers" + tristate "Texas Instruments TAS571x power amplifiers" depends on I2C + help + Enable support for Texas Instruments TAS5707, TAS5711, TAS5717, + TAS5719 and TAS5721 power amplifiers config SND_SOC_TAS5720 tristate "Texas Instruments TAS5720 Mono Audio amplifier" diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 52f34c94ec25..ca2dfe12344e 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -7,6 +7,9 @@ * TAS5721 support: * Copyright (C) 2016 Petr Kulhavy, Barix AG * + * TAS5707 support: + * Copyright (C) 2018 Jerome Brunet, Baylibre SAS + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -444,6 +447,111 @@ static const struct tas571x_chip tas5711_chip = { .vol_reg_size = 1, }; +static const struct regmap_range tas5707_volatile_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG), + regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG), + regmap_reg_range(TAS5707_CH1_BQ0_REG, TAS5707_CH2_BQ6_REG), +}; + +static const struct regmap_access_table tas5707_volatile_regs = { + .yes_ranges = tas5707_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(tas5707_volatile_regs_range), + +}; + +static const DECLARE_TLV_DB_SCALE(tas5707_volume_tlv, -7900, 50, 1); + +static const char * const tas5707_volume_slew_step_txt[] = { + "256", "512", "1024", "2048", +}; + +static const unsigned int tas5707_volume_slew_step_values[] = { + 3, 0, 1, 2, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(tas5707_volume_slew_step_enum, + TAS571X_VOL_CFG_REG, 0, 0x3, + tas5707_volume_slew_step_txt, + tas5707_volume_slew_step_values); + +static const struct snd_kcontrol_new tas5707_controls[] = { + SOC_SINGLE_TLV("Master Volume", + TAS571X_MVOL_REG, + 0, 0xff, 1, tas5707_volume_tlv), + SOC_DOUBLE_R_TLV("Speaker Volume", + TAS571X_CH1_VOL_REG, + TAS571X_CH2_VOL_REG, + 0, 0xff, 1, tas5707_volume_tlv), + SOC_DOUBLE("Speaker Switch", + TAS571X_SOFT_MUTE_REG, + TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT, + 1, 1), + + SOC_ENUM("Slew Rate Steps", tas5707_volume_slew_step_enum), + + BIQUAD_COEFS("CH1 - Biquad 0", TAS5707_CH1_BQ0_REG), + BIQUAD_COEFS("CH1 - Biquad 1", TAS5707_CH1_BQ1_REG), + BIQUAD_COEFS("CH1 - Biquad 2", TAS5707_CH1_BQ2_REG), + BIQUAD_COEFS("CH1 - Biquad 3", TAS5707_CH1_BQ3_REG), + BIQUAD_COEFS("CH1 - Biquad 4", TAS5707_CH1_BQ4_REG), + BIQUAD_COEFS("CH1 - Biquad 5", TAS5707_CH1_BQ5_REG), + BIQUAD_COEFS("CH1 - Biquad 6", TAS5707_CH1_BQ6_REG), + + BIQUAD_COEFS("CH2 - Biquad 0", TAS5707_CH2_BQ0_REG), + BIQUAD_COEFS("CH2 - Biquad 1", TAS5707_CH2_BQ1_REG), + BIQUAD_COEFS("CH2 - Biquad 2", TAS5707_CH2_BQ2_REG), + BIQUAD_COEFS("CH2 - Biquad 3", TAS5707_CH2_BQ3_REG), + BIQUAD_COEFS("CH2 - Biquad 4", TAS5707_CH2_BQ4_REG), + BIQUAD_COEFS("CH2 - Biquad 5", TAS5707_CH2_BQ5_REG), + BIQUAD_COEFS("CH2 - Biquad 6", TAS5707_CH2_BQ6_REG), +}; + +static const struct reg_default tas5707_reg_defaults[] = { + {TAS571X_CLK_CTRL_REG, 0x6c}, + {TAS571X_DEV_ID_REG, 0x70}, + {TAS571X_ERR_STATUS_REG, 0x00}, + {TAS571X_SYS_CTRL_1_REG, 0xa0}, + {TAS571X_SDI_REG, 0x05}, + {TAS571X_SYS_CTRL_2_REG, 0x40}, + {TAS571X_SOFT_MUTE_REG, 0x00}, + {TAS571X_MVOL_REG, 0xff}, + {TAS571X_CH1_VOL_REG, 0x30}, + {TAS571X_CH2_VOL_REG, 0x30}, + {TAS571X_VOL_CFG_REG, 0x91}, + {TAS571X_MODULATION_LIMIT_REG, 0x02}, + {TAS571X_IC_DELAY_CH1_REG, 0xac}, + {TAS571X_IC_DELAY_CH2_REG, 0x54}, + {TAS571X_IC_DELAY_CH3_REG, 0xac}, + {TAS571X_IC_DELAY_CH4_REG, 0x54}, + {TAS571X_START_STOP_PERIOD_REG, 0x0f}, + {TAS571X_OSC_TRIM_REG, 0x82}, + {TAS571X_BKND_ERR_REG, 0x02}, + {TAS571X_INPUT_MUX_REG, 0x17772}, + {TAS571X_PWM_MUX_REG, 0x1021345}, +}; + +static const struct regmap_config tas5707_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .max_register = 0xff, + .reg_read = tas571x_reg_read, + .reg_write = tas571x_reg_write, + .reg_defaults = tas5707_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5707_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas5707_volatile_regs, +}; + +static const struct tas571x_chip tas5707_chip = { + .supply_names = tas5711_supply_names, + .num_supply_names = ARRAY_SIZE(tas5711_supply_names), + .controls = tas5707_controls, + .num_controls = ARRAY_SIZE(tas5707_controls), + .regmap_config = &tas5707_regmap_config, + .vol_reg_size = 1, +}; + static const char *const tas5717_supply_names[] = { "AVDD", "DVDD", @@ -775,6 +883,7 @@ static int tas571x_i2c_remove(struct i2c_client *client) } static const struct of_device_id tas571x_of_match[] = { + { .compatible = "ti,tas5707", .data = &tas5707_chip, }, { .compatible = "ti,tas5711", .data = &tas5711_chip, }, { .compatible = "ti,tas5717", .data = &tas5717_chip, }, { .compatible = "ti,tas5719", .data = &tas5717_chip, }, @@ -784,6 +893,7 @@ static const struct of_device_id tas571x_of_match[] = { MODULE_DEVICE_TABLE(of, tas571x_of_match); static const struct i2c_device_id tas571x_i2c_id[] = { + { "tas5707", (kernel_ulong_t) &tas5707_chip }, { "tas5711", (kernel_ulong_t) &tas5711_chip }, { "tas5717", (kernel_ulong_t) &tas5717_chip }, { "tas5719", (kernel_ulong_t) &tas5717_chip }, diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index c45677bc26ad..bd23e89cfe79 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -53,6 +53,22 @@ #define TAS571X_PWM_MUX_REG 0x25 /* 20-byte biquad registers */ +#define TAS5707_CH1_BQ0_REG 0x29 +#define TAS5707_CH1_BQ1_REG 0x2a +#define TAS5707_CH1_BQ2_REG 0x2b +#define TAS5707_CH1_BQ3_REG 0x2c +#define TAS5707_CH1_BQ4_REG 0x2d +#define TAS5707_CH1_BQ5_REG 0x2e +#define TAS5707_CH1_BQ6_REG 0x2f + +#define TAS5707_CH2_BQ0_REG 0x30 +#define TAS5707_CH2_BQ1_REG 0x31 +#define TAS5707_CH2_BQ2_REG 0x32 +#define TAS5707_CH2_BQ3_REG 0x33 +#define TAS5707_CH2_BQ4_REG 0x34 +#define TAS5707_CH2_BQ5_REG 0x35 +#define TAS5707_CH2_BQ6_REG 0x36 + #define TAS5717_CH1_BQ0_REG 0x26 #define TAS5717_CH1_BQ1_REG 0x27 #define TAS5717_CH1_BQ2_REG 0x28 -- cgit v1.2.3 From de15d7ff5bef98746fcb76a0db7ac46de48d3560 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 26 Jun 2018 12:07:25 +0200 Subject: ASoC: dpcm: improve runtime update predictability As it is, dpcm_runtime_update() performs the old path and new path update of a frontend before going on to the next frontend DAI. Depending the order of the FEs within the rtd list, the result of the update might be different. For example: * Frontend A connected to backend C, with a 48kHz playback * Frontend B connected to backend D, with a 44.1kHz playback * FE A appears before FE B in the rtd list of the card. If we reparent BE C to FE B (disconnecting BE D): * old path update of FE A will run first, and BE C will get hw_free() and shutdown() * new path update of FE B will run after and BE C, which is stopped, so it will be configured at 44.1kHz, as expected If we reparent BE D to FE A (disconnecting BE C): * new path update of FE A will run first but since BE D is still running at 44.1kHz, it won't be reconfigured (no call to startup() or hw_params()) * old path update of FE B runs after, nothing happens * In this case, we end up with a BE playing at 44.1kHz a stream which is supposed to be played at 48Khz (too slow) To improve this situation, this patch performs all the FE old paths update before going on to update the new paths. With this, the result should no longer depend on the order of the FE within the card rtd list. Please note that there might be a small performance penalty since dpcm_process_paths() is called twice per stream direction. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 165 +++++++++++++++++++++++++++------------------------- 1 file changed, 86 insertions(+), 79 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 19ebfc958b9d..63f96cde046a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2597,106 +2597,113 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) return ret; } -/* Called by DAPM mixer/mux changes to update audio routing between PCMs and - * any DAI links. - */ -int soc_dpcm_runtime_update(struct snd_soc_card *card) +static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) { - struct snd_soc_pcm_runtime *fe; - int old, new, paths; + struct snd_soc_dapm_widget_list *list; + int count, paths; - mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - list_for_each_entry(fe, &card->rtd_list, list) { - struct snd_soc_dapm_widget_list *list; + if (!fe->dai_link->dynamic) + return 0; - /* make sure link is FE */ - if (!fe->dai_link->dynamic) - continue; + /* only check active links */ + if (!fe->cpu_dai->active) + return 0; - /* only check active links */ - if (!fe->cpu_dai->active) - continue; + /* DAPM sync will call this to update DSP paths */ + dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n", + new ? "new" : "old", fe->dai_link->name); - /* DAPM sync will call this to update DSP paths */ - dev_dbg(fe->dev, "ASoC: DPCM runtime update for FE %s\n", - fe->dai_link->name); + /* skip if FE doesn't have playback capability */ + if (!fe->cpu_dai->driver->playback.channels_min || + !fe->codec_dai->driver->playback.channels_min) + goto capture; - /* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min - || !fe->codec_dai->driver->playback.channels_min) - goto capture; - - /* skip if FE isn't currently playing */ - if (!fe->cpu_dai->playback_active - || !fe->codec_dai->playback_active) - goto capture; - - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "playback"); - mutex_unlock(&card->mutex); - return paths; - } + /* skip if FE isn't currently playing */ + if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active) + goto capture; - /* update any new playback paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1); - if (new) { - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "playback"); + return paths; + } - /* update any old playback paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0); - if (old) { + /* update any playback paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new); + if (count) { + if (new) + dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); + } + + dpcm_path_put(&list); + capture: - /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min - || !fe->codec_dai->driver->capture.channels_min) - continue; + /* skip if FE doesn't have capture capability */ + if (!fe->cpu_dai->driver->capture.channels_min || + !fe->codec_dai->driver->capture.channels_min) + return 0; - /* skip if FE isn't currently capturing */ - if (!fe->cpu_dai->capture_active - || !fe->codec_dai->capture_active) - continue; + /* skip if FE isn't currently capturing */ + if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active) + return 0; - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "capture"); - mutex_unlock(&card->mutex); - return paths; - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "capture"); + return paths; + } - /* update any new capture paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1); - if (new) { + /* update any old capture paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new); + if (count) { + if (new) dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - - /* update any old capture paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0); - if (old) { + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); } - mutex_unlock(&card->mutex); + dpcm_path_put(&list); + return 0; } + +/* Called by DAPM mixer/mux changes to update audio routing between PCMs and + * any DAI links. + */ +int soc_dpcm_runtime_update(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *fe; + int ret = 0; + + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + /* shutdown all old paths first */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 0); + if (ret) + goto out; + } + + /* bring new paths up */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 1); + if (ret) + goto out; + } + +out: + mutex_unlock(&card->mutex); + return ret; +} int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) { struct snd_soc_dpcm *dpcm; -- cgit v1.2.3 From c54c1c5ee8e73b7cb752834e52e2129b1dab00bd Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 27 Jun 2018 11:56:53 +0300 Subject: ASoC: qdsp6: qdafe: fix some off by one bugs The > should be >= or we could read one element beyond the end of the port_maps[] array. Fixes: 7fa2d70f9766 ("ASoC: qdsp6: q6afe: Add q6afe driver") Signed-off-by: Dan Carpenter Acked-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 621b67b34db9..671743453fbb 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -776,7 +776,7 @@ static int q6afe_callback(struct apr_device *adev, struct apr_resp_pkt *data) */ int q6afe_get_port_id(int index) { - if (index < 0 || index > AFE_PORT_MAX) + if (index < 0 || index >= AFE_PORT_MAX) return -EINVAL; return port_maps[index].port_id; @@ -1013,7 +1013,7 @@ int q6afe_port_stop(struct q6afe_port *port) port_id = port->id; index = port->token; - if (index < 0 || index > AFE_PORT_MAX) { + if (index < 0 || index >= AFE_PORT_MAX) { dev_err(afe->dev, "AFE port index[%d] invalid!\n", index); return -EINVAL; } @@ -1354,7 +1354,7 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) unsigned long flags; int cfg_type; - if (id < 0 || id > AFE_PORT_MAX) { + if (id < 0 || id >= AFE_PORT_MAX) { dev_err(dev, "AFE port token[%d] invalid!\n", id); return ERR_PTR(-EINVAL); } -- cgit v1.2.3 From 4f2bd18b191a10660782f2f1ccc989b000b2be63 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 27 Jun 2018 11:48:18 +0200 Subject: ASoC: dpcm: extend channel merging to the backend cpu dai Extend dpcm_merge_chan to also check backend cpu dai channels capabilities. Apply the same policy as soc_pcm_init_runtime_hw() for multicodec links and only check cpu dai in this case. Cc: Jiada Wang Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 63f96cde046a..6ee4131941df 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1736,12 +1736,26 @@ static void dpcm_runtime_base_chan(struct snd_pcm_substream *substream, list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; - int i; + struct snd_soc_pcm_stream *cpu_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; + else + cpu_stream = &cpu_dai_drv->capture; + + *channels_min = max(*channels_min, cpu_stream->channels_min); + *channels_max = min(*channels_max, cpu_stream->channels_max); + + /* + * chan min/max cannot be enforced if there are multiple CODEC + * DAIs connected to a single CPU DAI, use CPU DAI's directly + */ + if (be->num_codecs == 1) { + codec_dai_drv = be->codec_dais[0]->driver; - for (i = 0; i < be->num_codecs; i++) { - codec_dai_drv = be->codec_dais[i]->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; else -- cgit v1.2.3 From 4febced15ac8ddb9cf3e603edb111842e4863d9a Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 27 Jun 2018 17:36:38 +0200 Subject: ASoC: dpcm: don't merge format from invalid codec dai When merging codec formats, dpcm_runtime_base_format() should skip the codecs which are not supporting the current stream direction. At the moment, if a BE link has more than one codec, and only one of these codecs has no capture DAI, it becomes impossible to start a capture stream because the merged format would be 0. Skipping invalid codec DAI solves the problem. Fixes: b073ed4e2126 ("ASoC: soc-pcm: DPCM cares BE format") Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-pcm.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5e7ae47a9658..5feae9666822 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1694,6 +1694,14 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) int i; for (i = 0; i < be->num_codecs; i++) { + /* + * Skip CODECs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(be->codec_dais[i], + stream)) + continue; + codec_dai_drv = be->codec_dais[i]->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; -- cgit v1.2.3 From 8f54061d001ad2da24dba89fc48adbbf4c85222b Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Thu, 28 Jun 2018 22:08:37 +0200 Subject: ASoC: pxa: remove the dmaengine compat need As the pxa architecture switched towards the dmaengine slave map, the old compatibility mechanism to acquire the dma requestor line number and priority are not needed anymore. This patch simplifies the dma resource acquisition, using the more generic function dma_request_slave_channel(). Signed-off-by: Robert Jarzmik Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/arm/pxa2xx-ac97.c | 14 ++------------ sound/arm/pxa2xx-pcm-lib.c | 6 +++--- sound/soc/pxa/pxa2xx-ac97.c | 32 +++++--------------------------- sound/soc/pxa/pxa2xx-i2s.c | 6 ++---- 4 files changed, 12 insertions(+), 46 deletions(-) (limited to 'sound/soc') diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 4bc244c40f80..236a63cdaf9f 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -63,28 +63,18 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_legacy_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_out_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_in_req, }; static struct snd_pcm *pxa2xx_ac97_pcm; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index e8da3b8ee721..dcbe7ecc1835 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -125,9 +125,9 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) return ret; - return snd_dmaengine_pcm_open_request_chan(substream, - pxad_filter_fn, - dma_params->filter_data); + return snd_dmaengine_pcm_open( + substream, dma_request_slave_channel(rtd->cpu_dai->dev, + dma_params->chan_name)); } EXPORT_SYMBOL(__pxa2xx_pcm_open); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 5738a0abcd6a..c52b33802bf2 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -68,61 +68,39 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 10, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_out", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 9, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_in", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 8, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .addr = __PREG(MCDR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mic_mono", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 3fb60baf6eab..e7184de0de04 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -82,20 +82,18 @@ static struct pxa_i2s_port pxa_i2s; static struct clk *clk_i2s; static int clk_ena = 0; -static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "tx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_out_req, }; -static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "rx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_in_req, }; static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, -- cgit v1.2.3 From 95acb005fef2aeaeb63c20de98aca0ed5bd0efa2 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 27 Jun 2018 21:33:53 +0200 Subject: ASoC: fold pxa2xx-pcm into its only user, pxa2xx-ac97 Now that the PXA SSP bits are ported over to generic DMA, the pxa2xx-pcm code only has a single user left. This patch folds the remaining bits into its only user and removes the unnecessary glue layer along with its header file. The include dependency to linux/dma/pxa-dma.h is also gone now. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/arm/Kconfig | 5 -- sound/arm/Makefile | 3 -- sound/arm/pxa2xx-ac97.c | 114 ++++++++++++++++++++++++++------------- sound/arm/pxa2xx-pcm-lib.c | 2 - sound/arm/pxa2xx-pcm.c | 129 --------------------------------------------- sound/arm/pxa2xx-pcm.h | 27 ---------- sound/soc/pxa/pxa-ssp.c | 1 - sound/soc/pxa/pxa2xx-pcm.c | 2 - 8 files changed, 76 insertions(+), 207 deletions(-) delete mode 100644 sound/arm/pxa2xx-pcm.c delete mode 100644 sound/arm/pxa2xx-pcm.h (limited to 'sound/soc') diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 65171f6657a2..5fbd47a9177e 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -17,14 +17,9 @@ config SND_ARMAACI select SND_PCM select SND_AC97_CODEC -config SND_PXA2XX_PCM - tristate - select SND_PCM - config SND_PXA2XX_AC97 tristate "AC97 driver for the Intel PXA2xx chip" depends on ARCH_PXA - select SND_PXA2XX_PCM select SND_AC97_CODEC select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 diff --git a/sound/arm/Makefile b/sound/arm/Makefile index e10d5b169565..34c769489877 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -6,9 +6,6 @@ obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o snd-aaci-objs := aaci.o -obj-$(CONFIG_SND_PXA2XX_PCM) += snd-pxa2xx-pcm.o -snd-pxa2xx-pcm-objs := pxa2xx-pcm.o - obj-$(CONFIG_SND_PXA2XX_LIB) += snd-pxa2xx-lib.o snd-pxa2xx-lib-y := pxa2xx-pcm-lib.o snd-pxa2xx-lib-$(CONFIG_SND_PXA2XX_LIB_AC97) += pxa2xx-ac97-lib.o diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 236a63cdaf9f..7d8d7b7199dc 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -15,7 +15,7 @@ #include #include #include -#include +#include #include #include @@ -27,8 +27,6 @@ #include #include -#include "pxa2xx-pcm.h" - static void pxa2xx_ac97_legacy_reset(struct snd_ac97 *ac97) { if (!pxa2xx_ac97_try_cold_reset()) @@ -63,51 +61,46 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_legacy_reset, }; -static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { - .addr = __PREG(PCDR), - .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, - .chan_name = "pcm_pcm_stereo_out", - .maxburst = 32, -}; - -static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { - .addr = __PREG(PCDR), - .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, - .chan_name = "pcm_pcm_stereo_in", - .maxburst = 32, -}; - static struct snd_pcm *pxa2xx_ac97_pcm; static struct snd_ac97 *pxa2xx_ac97_ac97; -static int pxa2xx_ac97_pcm_startup(struct snd_pcm_substream *substream) +static int pxa2xx_ac97_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; pxa2xx_audio_ops_t *platform_ops; - int r; + int ret, i; + + ret = __pxa2xx_pcm_open(substream); + if (ret) + return ret; runtime->hw.channels_min = 2; runtime->hw.channels_max = 2; - r = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - AC97_RATES_FRONT_DAC : AC97_RATES_ADC; - runtime->hw.rates = pxa2xx_ac97_ac97->rates[r]; + i = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + AC97_RATES_FRONT_DAC : AC97_RATES_ADC; + runtime->hw.rates = pxa2xx_ac97_ac97->rates[i]; snd_pcm_limit_hw_rates(runtime); - platform_ops = substream->pcm->card->dev->platform_data; - if (platform_ops && platform_ops->startup) - return platform_ops->startup(substream, platform_ops->priv); - else - return 0; + platform_ops = substream->pcm->card->dev->platform_data; + if (platform_ops && platform_ops->startup) { + ret = platform_ops->startup(substream, platform_ops->priv); + if (ret < 0) + __pxa2xx_pcm_close(substream); + } + + return ret; } -static void pxa2xx_ac97_pcm_shutdown(struct snd_pcm_substream *substream) +static int pxa2xx_ac97_pcm_close(struct snd_pcm_substream *substream) { pxa2xx_audio_ops_t *platform_ops; - platform_ops = substream->pcm->card->dev->platform_data; + platform_ops = substream->pcm->card->dev->platform_data; if (platform_ops && platform_ops->shutdown) platform_ops->shutdown(substream, platform_ops->priv); + + return 0; } static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream) @@ -115,17 +108,15 @@ static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; + int ret; + + ret = __pxa2xx_pcm_prepare(substream); + if (ret < 0) + return ret; + return snd_ac97_set_rate(pxa2xx_ac97_ac97, reg, runtime->rate); } -static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = { - .playback_params = &pxa2xx_ac97_pcm_out, - .capture_params = &pxa2xx_ac97_pcm_in, - .startup = pxa2xx_ac97_pcm_startup, - .shutdown = pxa2xx_ac97_pcm_shutdown, - .prepare = pxa2xx_ac97_pcm_prepare, -}; - #ifdef CONFIG_PM_SLEEP static int pxa2xx_ac97_do_suspend(struct snd_card *card) @@ -183,6 +174,53 @@ static int pxa2xx_ac97_resume(struct device *dev) static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume); #endif +static const struct snd_pcm_ops pxa2xx_pcm_ops = { + .open = pxa2xx_ac97_pcm_open, + .close = pxa2xx_ac97_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = __pxa2xx_pcm_hw_params, + .hw_free = __pxa2xx_pcm_hw_free, + .prepare = pxa2xx_ac97_pcm_prepare, + .trigger = pxa2xx_pcm_trigger, + .pointer = pxa2xx_pcm_pointer, + .mmap = pxa2xx_pcm_mmap, +}; + + +static int pxa2xx_ac97_pcm_new(struct snd_card *card) +{ + struct snd_pcm *pcm; + int stream, ret; + + ret = snd_pcm_new(card, "PXA2xx-PCM", 0, 1, 1, &pcm); + if (ret) + goto out; + + pcm->private_free = pxa2xx_pcm_free_dma_buffers; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + goto out; + + stream = SNDRV_PCM_STREAM_PLAYBACK; + snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); + if (ret) + goto out; + + stream = SNDRV_PCM_STREAM_CAPTURE; + snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); + if (ret) + goto out; + + pxa2xx_ac97_pcm = pcm; + ret = 0; + + out: + return ret; +} + static int pxa2xx_ac97_probe(struct platform_device *dev) { struct snd_card *card; @@ -204,7 +242,7 @@ static int pxa2xx_ac97_probe(struct platform_device *dev) strlcpy(card->driver, dev->dev.driver->name, sizeof(card->driver)); - ret = pxa2xx_pcm_new(card, &pxa2xx_ac97_pcm_client, &pxa2xx_ac97_pcm); + ret = pxa2xx_ac97_pcm_new(card); if (ret) goto err; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index dcbe7ecc1835..b927fa5ddbc0 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -16,8 +16,6 @@ #include #include -#include "pxa2xx-pcm.h" - static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c deleted file mode 100644 index 1c6f4b436de3..000000000000 --- a/sound/arm/pxa2xx-pcm.c +++ /dev/null @@ -1,129 +0,0 @@ -/* - * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include - -#include - -#include -#include -#include - -#include "pxa2xx-pcm.h" - -static int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct pxa2xx_pcm_client *client = substream->private_data; - - __pxa2xx_pcm_prepare(substream); - - return client->prepare(substream); -} - -static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) -{ - struct pxa2xx_pcm_client *client = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd; - int ret; - - ret = __pxa2xx_pcm_open(substream); - if (ret) - goto out; - - rtd = runtime->private_data; - - rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - client->playback_params : client->capture_params; - - ret = client->startup(substream); - if (!ret) - goto err2; - - return 0; - - err2: - __pxa2xx_pcm_close(substream); - out: - return ret; -} - -static int pxa2xx_pcm_close(struct snd_pcm_substream *substream) -{ - struct pxa2xx_pcm_client *client = substream->private_data; - - client->shutdown(substream); - - return __pxa2xx_pcm_close(substream); -} - -static const struct snd_pcm_ops pxa2xx_pcm_ops = { - .open = pxa2xx_pcm_open, - .close = pxa2xx_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = __pxa2xx_pcm_hw_params, - .hw_free = __pxa2xx_pcm_hw_free, - .prepare = pxa2xx_pcm_prepare, - .trigger = pxa2xx_pcm_trigger, - .pointer = pxa2xx_pcm_pointer, - .mmap = pxa2xx_pcm_mmap, -}; - -int pxa2xx_pcm_new(struct snd_card *card, struct pxa2xx_pcm_client *client, - struct snd_pcm **rpcm) -{ - struct snd_pcm *pcm; - int play = client->playback_params ? 1 : 0; - int capt = client->capture_params ? 1 : 0; - int ret; - - ret = snd_pcm_new(card, "PXA2xx-PCM", 0, play, capt, &pcm); - if (ret) - goto out; - - pcm->private_data = client; - pcm->private_free = pxa2xx_pcm_free_dma_buffers; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - goto out; - - if (play) { - int stream = SNDRV_PCM_STREAM_PLAYBACK; - snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); - if (ret) - goto out; - } - if (capt) { - int stream = SNDRV_PCM_STREAM_CAPTURE; - snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); - if (ret) - goto out; - } - - if (rpcm) - *rpcm = pcm; - ret = 0; - - out: - return ret; -} - -EXPORT_SYMBOL(pxa2xx_pcm_new); - -MODULE_AUTHOR("Nicolas Pitre"); -MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h deleted file mode 100644 index 8fa2b7c9e6b8..000000000000 --- a/sound/arm/pxa2xx-pcm.h +++ /dev/null @@ -1,27 +0,0 @@ -/* - * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -struct pxa2xx_runtime_data { - int dma_ch; - struct snd_dmaengine_dai_dma_data *params; -}; - -struct pxa2xx_pcm_client { - struct snd_dmaengine_dai_dma_data *playback_params; - struct snd_dmaengine_dai_dma_data *capture_params; - int (*startup)(struct snd_pcm_substream *); - void (*shutdown)(struct snd_pcm_substream *); - int (*prepare)(struct snd_pcm_substream *); -}; - -extern int pxa2xx_pcm_new(struct snd_card *, struct pxa2xx_pcm_client *, struct snd_pcm **); - diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 0b441338bdd4..c1f4af869289 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -34,7 +34,6 @@ #include #include -#include "../../arm/pxa2xx-pcm.h" #include "pxa-ssp.h" /* diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 8b6a70e94c01..445e691126e5 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -20,8 +20,6 @@ #include #include -#include "../../arm/pxa2xx-pcm.h" - static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { -- cgit v1.2.3 From a7160670b5e2d6b59e0f7a5b7e5bcef3b532c24c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 27 Jun 2018 21:33:54 +0200 Subject: ASoC: pxa: clean up function names in pxa2xx-lib Clean up the namespace a bit and drop the __ prefix of all functions exported by pxa2xx-lib. This improves the readability of the code. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- include/sound/pxa2xx-lib.h | 10 +++++----- sound/arm/pxa2xx-ac97.c | 10 +++++----- sound/arm/pxa2xx-pcm-lib.c | 22 +++++++++++----------- sound/soc/pxa/pxa2xx-pcm.c | 21 +++++++-------------- 4 files changed, 28 insertions(+), 35 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index 63f75450d3db..b43de38de8b2 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -10,14 +10,14 @@ struct snd_pcm_substream; struct snd_pcm_hw_params; struct snd_pcm; -extern int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, +extern int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); -extern int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream); +extern int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd); extern snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream); -extern int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream); -extern int __pxa2xx_pcm_open(struct snd_pcm_substream *substream); -extern int __pxa2xx_pcm_close(struct snd_pcm_substream *substream); +extern int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream); +extern int pxa2xx_pcm_open(struct snd_pcm_substream *substream); +extern int pxa2xx_pcm_close(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); extern int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream); diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 7d8d7b7199dc..0d624337857b 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -70,7 +70,7 @@ static int pxa2xx_ac97_pcm_open(struct snd_pcm_substream *substream) pxa2xx_audio_ops_t *platform_ops; int ret, i; - ret = __pxa2xx_pcm_open(substream); + ret = pxa2xx_pcm_open(substream); if (ret) return ret; @@ -86,7 +86,7 @@ static int pxa2xx_ac97_pcm_open(struct snd_pcm_substream *substream) if (platform_ops && platform_ops->startup) { ret = platform_ops->startup(substream, platform_ops->priv); if (ret < 0) - __pxa2xx_pcm_close(substream); + pxa2xx_pcm_close(substream); } return ret; @@ -110,7 +110,7 @@ static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream) AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; int ret; - ret = __pxa2xx_pcm_prepare(substream); + ret = pxa2xx_pcm_prepare(substream); if (ret < 0) return ret; @@ -178,8 +178,8 @@ static const struct snd_pcm_ops pxa2xx_pcm_ops = { .open = pxa2xx_ac97_pcm_open, .close = pxa2xx_ac97_pcm_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = __pxa2xx_pcm_hw_params, - .hw_free = __pxa2xx_pcm_hw_free, + .hw_params = pxa2xx_pcm_hw_params, + .hw_free = pxa2xx_pcm_hw_free, .prepare = pxa2xx_ac97_pcm_prepare, .trigger = pxa2xx_pcm_trigger, .pointer = pxa2xx_pcm_pointer, diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index b927fa5ddbc0..dc56dbebf441 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -33,8 +33,8 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { .fifo_size = 32, }; -int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -62,14 +62,14 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -EXPORT_SYMBOL(__pxa2xx_pcm_hw_params); +EXPORT_SYMBOL(pxa2xx_pcm_hw_params); -int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) +int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { snd_pcm_set_runtime_buffer(substream, NULL); return 0; } -EXPORT_SYMBOL(__pxa2xx_pcm_hw_free); +EXPORT_SYMBOL(pxa2xx_pcm_hw_free); int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -84,13 +84,13 @@ pxa2xx_pcm_pointer(struct snd_pcm_substream *substream) } EXPORT_SYMBOL(pxa2xx_pcm_pointer); -int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) +int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) { return 0; } -EXPORT_SYMBOL(__pxa2xx_pcm_prepare); +EXPORT_SYMBOL(pxa2xx_pcm_prepare); -int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) +int pxa2xx_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; @@ -127,13 +127,13 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) substream, dma_request_slave_channel(rtd->cpu_dai->dev, dma_params->chan_name)); } -EXPORT_SYMBOL(__pxa2xx_pcm_open); +EXPORT_SYMBOL(pxa2xx_pcm_open); -int __pxa2xx_pcm_close(struct snd_pcm_substream *substream) +int pxa2xx_pcm_close(struct snd_pcm_substream *substream) { return snd_dmaengine_pcm_close_release_chan(substream); } -EXPORT_SYMBOL(__pxa2xx_pcm_close); +EXPORT_SYMBOL(pxa2xx_pcm_close); int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 445e691126e5..da252d1f732e 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -20,8 +20,8 @@ #include #include -static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_dmaengine_dai_dma_data *dma; @@ -33,23 +33,16 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, if (!dma) return 0; - return __pxa2xx_pcm_hw_params(substream, params); -} - -static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) -{ - __pxa2xx_pcm_hw_free(substream); - - return 0; + return pxa2xx_pcm_hw_params(substream, params); } static const struct snd_pcm_ops pxa2xx_pcm_ops = { - .open = __pxa2xx_pcm_open, - .close = __pxa2xx_pcm_close, + .open = pxa2xx_pcm_open, + .close = pxa2xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = pxa2xx_pcm_hw_params, + .hw_params = __pxa2xx_pcm_hw_params, .hw_free = pxa2xx_pcm_hw_free, - .prepare = __pxa2xx_pcm_prepare, + .prepare = pxa2xx_pcm_prepare, .trigger = pxa2xx_pcm_trigger, .pointer = pxa2xx_pcm_pointer, .mmap = pxa2xx_pcm_mmap, -- cgit v1.2.3 From 7afd1b0b2ef9a8120951188a955010ef92bdf885 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 27 Jun 2018 21:33:55 +0200 Subject: ASoC: pxa: move some functions to pxa2xx-lib To get rid of some intermediate platform layers, move pxa2xx_soc_pcm_new() and pxa2xx_pcm_ops in pxa2xx-lib. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- include/sound/pxa2xx-lib.h | 3 +++ sound/arm/pxa2xx-ac97.c | 6 ++--- sound/arm/pxa2xx-pcm-lib.c | 41 ++++++++++++++++++++++++++++++++++ sound/soc/pxa/pxa2xx-pcm.c | 55 ---------------------------------------------- 4 files changed, 47 insertions(+), 58 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index b43de38de8b2..6758fc12fa84 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -8,6 +8,7 @@ /* PCM */ struct snd_pcm_substream; struct snd_pcm_hw_params; +struct snd_soc_pcm_runtime; struct snd_pcm; extern int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, @@ -22,6 +23,8 @@ extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); extern int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream); extern void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm); +extern int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd); +extern const struct snd_pcm_ops pxa2xx_pcm_ops; /* AC97 */ diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 0d624337857b..1f72672262d0 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -174,7 +174,7 @@ static int pxa2xx_ac97_resume(struct device *dev) static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume); #endif -static const struct snd_pcm_ops pxa2xx_pcm_ops = { +static const struct snd_pcm_ops pxa2xx_ac97_pcm_ops = { .open = pxa2xx_ac97_pcm_open, .close = pxa2xx_ac97_pcm_close, .ioctl = snd_pcm_lib_ioctl, @@ -203,13 +203,13 @@ static int pxa2xx_ac97_pcm_new(struct snd_card *card) goto out; stream = SNDRV_PCM_STREAM_PLAYBACK; - snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); + snd_pcm_set_ops(pcm, stream, &pxa2xx_ac97_pcm_ops); ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); if (ret) goto out; stream = SNDRV_PCM_STREAM_CAPTURE; - snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); + snd_pcm_set_ops(pcm, stream, &pxa2xx_ac97_pcm_ops); ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); if (ret) goto out; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index dc56dbebf441..add23d9b4ef6 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -179,6 +179,47 @@ void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm) } EXPORT_SYMBOL(pxa2xx_pcm_free_dma_buffers); +int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} +EXPORT_SYMBOL(pxa2xx_soc_pcm_new); + +const struct snd_pcm_ops pxa2xx_pcm_ops = { + .open = pxa2xx_pcm_open, + .close = pxa2xx_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = pxa2xx_pcm_hw_params, + .hw_free = pxa2xx_pcm_hw_free, + .prepare = pxa2xx_pcm_prepare, + .trigger = pxa2xx_pcm_trigger, + .pointer = pxa2xx_pcm_pointer, + .mmap = pxa2xx_pcm_mmap, +}; +EXPORT_SYMBOL(pxa2xx_pcm_ops); + MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("Intel PXA2xx sound library"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index da252d1f732e..a1df4ec76cbd 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -20,61 +20,6 @@ #include #include -static int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_dmaengine_dai_dma_data *dma; - - dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - return pxa2xx_pcm_hw_params(substream, params); -} - -static const struct snd_pcm_ops pxa2xx_pcm_ops = { - .open = pxa2xx_pcm_open, - .close = pxa2xx_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = __pxa2xx_pcm_hw_params, - .hw_free = pxa2xx_pcm_hw_free, - .prepare = pxa2xx_pcm_prepare, - .trigger = pxa2xx_pcm_trigger, - .pointer = pxa2xx_pcm_pointer, - .mmap = pxa2xx_pcm_mmap, -}; - -static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - static const struct snd_soc_component_driver pxa2xx_soc_platform = { .ops = &pxa2xx_pcm_ops, .pcm_new = pxa2xx_soc_pcm_new, -- cgit v1.2.3 From d767d3ce5c48b3378e20e8cfd5d5379c4ca6001b Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 27 Jun 2018 21:33:57 +0200 Subject: ASoC: pxa: provide PCM ops for ssp, i2s and ac97 components Now that the functions are now available through pxa2xx-lib, hook them up to pxa-sspi, pxa-ac97 and pxa-i2s. This allows DT platforms to use the DAIs without a platform driver. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 3 +++ sound/soc/pxa/pxa2xx-ac97.c | 3 +++ sound/soc/pxa/pxa2xx-i2s.c | 3 +++ 3 files changed, 9 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index c1f4af869289..01d54697ede4 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -841,6 +841,9 @@ static struct snd_soc_dai_driver pxa_ssp_dai = { static const struct snd_soc_component_driver pxa_ssp_component = { .name = "pxa-ssp", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, }; #ifdef CONFIG_OF diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index c52b33802bf2..9f779657bc86 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -214,6 +214,9 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { static const struct snd_soc_component_driver pxa_ac97_component = { .name = "pxa-ac97", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, }; #ifdef CONFIG_OF diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index e7184de0de04..42820121e5b9 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -364,6 +364,9 @@ static struct snd_soc_dai_driver pxa_i2s_dai = { static const struct snd_soc_component_driver pxa_i2s_component = { .name = "pxa-i2s", + .ops = &pxa2xx_pcm_ops, + .pcm_new = pxa2xx_soc_pcm_new, + .pcm_free = pxa2xx_pcm_free_dma_buffers, }; static int pxa2xx_i2s_drv_probe(struct platform_device *pdev) -- cgit v1.2.3 From c7b4f15ddb4f28451679019374027f5223c616ce Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 27 Jun 2018 21:33:58 +0200 Subject: ASoC: pxa: remove bindings from pxa2xx-pcm This platform is no longer needed on DT boards, so let's remove them to avoid confusion. DT bindings should use the CPU DAIs (I2S/SSP/AC97) directly. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt | 15 --------------- sound/soc/pxa/pxa2xx-pcm.c | 9 --------- 2 files changed, 24 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt deleted file mode 100644 index 551fbb8348c2..000000000000 --- a/Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt +++ /dev/null @@ -1,15 +0,0 @@ -DT bindings for ARM PXA2xx PCM platform driver - -This is just a dummy driver that registers the PXA ASoC platform driver. -It does not have any resources assigned. - -Required properties: - - - compatible 'mrvl,pxa-pcm-audio' - -Example: - - pxa_pcm_audio: snd_soc_pxa_audio { - compatible = "mrvl,pxa-pcm-audio"; - }; - diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index a1df4ec76cbd..72eaaef1b426 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -32,18 +32,9 @@ static int pxa2xx_soc_platform_probe(struct platform_device *pdev) NULL, 0); } -#ifdef CONFIG_OF -static const struct of_device_id snd_soc_pxa_audio_match[] = { - { .compatible = "mrvl,pxa-pcm-audio" }, - { } -}; -MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match); -#endif - static struct platform_driver pxa_pcm_driver = { .driver = { .name = "pxa-pcm-audio", - .of_match_table = of_match_ptr(snd_soc_pxa_audio_match), }, .probe = pxa2xx_soc_platform_probe, -- cgit v1.2.3 From 0a94cf3457408058f894cc4d95e58d8e18eb7f75 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 27 Jun 2018 21:33:59 +0200 Subject: ASoC: pxa: make SND_PXA2XX_SOC_I2S selectable Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 960744e46edc..95dcf97a6271 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -31,7 +31,7 @@ config SND_PXA2XX_SOC_I2S tristate config SND_PXA_SOC_SSP - tristate + tristate "Soc Audio via PXA2xx/PXA3xx SSP ports" select PXA_SSP config SND_MMP_SOC_SSPA -- cgit v1.2.3 From f11c5db770ab675d270cb4d5a2bb90923066ef49 Mon Sep 17 00:00:00 2001 From: KaiChieh Chuang Date: Fri, 29 Jun 2018 20:29:44 +0800 Subject: ASoC: mediatek: sub dai use list_head use list_head for sub_dais, since original sub_dais array is sparsely occupied Signed-off-by: KaiChieh Chuang Signed-off-by: Mark Brown --- .../soc/mediatek/common/mtk-afe-platform-driver.c | 64 ++++++++-------------- sound/soc/mediatek/common/mtk-base-afe.h | 6 +- 2 files changed, 28 insertions(+), 42 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 51ec4ff6ed95..697aa50aff9a 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -15,20 +15,12 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) { - struct snd_soc_dai_driver *sub_dai_drivers; + struct mtk_base_afe_dai *dai; size_t num_dai_drivers = 0, dai_idx = 0; - int i; - - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } /* calcualte total dai driver size */ - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) - num_dai_drivers += afe->sub_dais[i].num_dai_drivers; + list_for_each_entry(dai, &afe->sub_dais, list) { + num_dai_drivers += dai->num_dai_drivers; } dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers); @@ -42,19 +34,14 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) if (!afe->dai_drivers) return -ENOMEM; - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) { - sub_dai_drivers = afe->sub_dais[i].dai_drivers; - /* dai driver */ - memcpy(&afe->dai_drivers[dai_idx], - sub_dai_drivers, - afe->sub_dais[i].num_dai_drivers * - sizeof(struct snd_soc_dai_driver)); - dai_idx += afe->sub_dais[i].num_dai_drivers; - } + list_for_each_entry(dai, &afe->sub_dais, list) { + /* dai driver */ + memcpy(&afe->dai_drivers[dai_idx], + dai->dai_drivers, + dai->num_dai_drivers * + sizeof(struct snd_soc_dai_driver)); + dai_idx += dai->num_dai_drivers; } - return 0; } EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai); @@ -62,28 +49,25 @@ EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai); int mtk_afe_add_sub_dai_control(struct snd_soc_component *component) { struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int i; + struct mtk_base_afe_dai *dai; - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } - - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].controls) + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->controls) snd_soc_add_component_controls(component, - afe->sub_dais[i].controls, - afe->sub_dais[i].num_controls); + dai->controls, + dai->num_controls); - if (afe->sub_dais[i].dapm_widgets) + if (dai->dapm_widgets) snd_soc_dapm_new_controls(&component->dapm, - afe->sub_dais[i].dapm_widgets, - afe->sub_dais[i].num_dapm_widgets); - - if (afe->sub_dais[i].dapm_routes) + dai->dapm_widgets, + dai->num_dapm_widgets); + } + /* add routes after all widgets are added */ + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->dapm_routes) snd_soc_dapm_add_routes(&component->dapm, - afe->sub_dais[i].dapm_routes, - afe->sub_dais[i].num_dapm_routes); + dai->dapm_routes, + dai->num_dapm_routes); } snd_soc_dapm_new_widgets(component->dapm.card); diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h index bcf562f029b6..bd8d5e0c6843 100644 --- a/sound/soc/mediatek/common/mtk-base-afe.h +++ b/sound/soc/mediatek/common/mtk-base-afe.h @@ -46,6 +46,7 @@ struct mtk_base_irq_data { }; struct device; +struct list_head; struct mtk_base_afe_memif; struct mtk_base_afe_irq; struct mtk_base_afe_dai; @@ -72,8 +73,7 @@ struct mtk_base_afe { struct mtk_base_afe_irq *irqs; int irqs_size; - struct mtk_base_afe_dai *sub_dais; - int num_sub_dais; + struct list_head sub_dais; struct snd_soc_dai_driver *dai_drivers; unsigned int num_dai_drivers; @@ -110,6 +110,8 @@ struct mtk_base_afe_dai { unsigned int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; unsigned int num_dapm_routes; + + struct list_head list; }; #endif -- cgit v1.2.3 From c1d9b4196ba6b311bd48a9320cd46aa125e0b034 Mon Sep 17 00:00:00 2001 From: KaiChieh Chuang Date: Fri, 29 Jun 2018 20:29:45 +0800 Subject: ASoC: mt6797: sub dai use list_head Signed-off-by: KaiChieh Chuang Signed-off-by: Mark Brown --- sound/soc/mediatek/mt6797/mt6797-afe-common.h | 1 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 65 +++++++++++++++++-------- sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 20 +++++--- sound/soc/mediatek/mt6797/mt6797-dai-hostless.c | 16 ++++-- sound/soc/mediatek/mt6797/mt6797-dai-pcm.c | 19 +++++--- 5 files changed, 81 insertions(+), 40 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-common.h b/sound/soc/mediatek/mt6797/mt6797-afe-common.h index 22eb7b455cf1..4eac9977b2b0 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-common.h +++ b/sound/soc/mediatek/mt6797/mt6797-afe-common.h @@ -10,6 +10,7 @@ #define _MT_6797_AFE_COMMON_H_ #include +#include #include #include "../common/mtk-base-afe.h" diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 6c5dd9fc9976..192f4d7b37b6 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -733,6 +733,34 @@ static const struct snd_soc_component_driver mt6797_afe_component = { .probe = mt6797_afe_component_probe, }; +static int mt6797_dai_memif_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mt6797_memif_dai_driver; + dai->num_dai_drivers = ARRAY_SIZE(mt6797_memif_dai_driver); + + dai->dapm_widgets = mt6797_memif_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mt6797_memif_widgets); + dai->dapm_routes = mt6797_memif_routes; + dai->num_dapm_routes = ARRAY_SIZE(mt6797_memif_routes); + return 0; +} + +typedef int (*dai_register_cb)(struct mtk_base_afe *); +static const dai_register_cb dai_register_cbs[] = { + mt6797_dai_adda_register, + mt6797_dai_pcm_register, + mt6797_dai_hostless_register, + mt6797_dai_memif_register, +}; + static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) { struct mtk_base_afe *afe; @@ -811,29 +839,24 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) } /* init sub_dais */ - afe->num_sub_dais = MT6797_DAI_NUM; - afe->sub_dais = devm_kcalloc(dev, afe->num_sub_dais, - sizeof(*afe->sub_dais), - GFP_KERNEL); - if (!afe->sub_dais) - return -ENOMEM; - - mt6797_dai_adda_register(afe); - mt6797_dai_pcm_register(afe); - mt6797_dai_hostless_register(afe); - - afe->sub_dais[MT6797_MEMIF_DL1].dai_drivers = mt6797_memif_dai_driver; - afe->sub_dais[MT6797_MEMIF_DL1].num_dai_drivers = - ARRAY_SIZE(mt6797_memif_dai_driver); - afe->sub_dais[MT6797_MEMIF_DL1].dapm_widgets = mt6797_memif_widgets; - afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_widgets = - ARRAY_SIZE(mt6797_memif_widgets); - afe->sub_dais[MT6797_MEMIF_DL1].dapm_routes = mt6797_memif_routes; - afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_routes = - ARRAY_SIZE(mt6797_memif_routes); + INIT_LIST_HEAD(&afe->sub_dais); + + for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) { + ret = dai_register_cbs[i](afe); + if (ret) { + dev_warn(afe->dev, "dai register i %d fail, ret %d\n", + i, ret); + return ret; + } + } /* init dai_driver and component_driver */ - mtk_afe_combine_sub_dai(afe); + ret = mtk_afe_combine_sub_dai(afe); + if (ret) { + dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n", + ret); + return ret; + } afe->mtk_afe_hardware = &mt6797_afe_hardware; afe->memif_fs = mt6797_memif_fs; diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c index ad083265ce94..0ac6409c6d61 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c @@ -383,14 +383,20 @@ static struct snd_soc_dai_driver mtk_dai_adda_driver[] = { int mt6797_dai_adda_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_ADDA; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_adda_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_widgets = mtk_dai_adda_widgets; - afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); - afe->sub_dais[id].dapm_routes = mtk_dai_adda_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_adda_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + + dai->dapm_widgets = mtk_dai_adda_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); + dai->dapm_routes = mtk_dai_adda_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); return 0; } diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c index 4cf985b15a11..ed23e6a53b08 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c @@ -100,13 +100,19 @@ static struct snd_soc_dai_driver mtk_dai_hostless_driver[] = { int mt6797_dai_hostless_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_HOSTLESS_LPBK; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_hostless_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_routes = mtk_dai_hostless_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes); + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_hostless_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver); + + dai->dapm_routes = mtk_dai_hostless_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes); return 0; } diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c index 16d5b5067204..3136f0bc7827 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c @@ -298,15 +298,20 @@ static struct snd_soc_dai_driver mtk_dai_pcm_driver[] = { int mt6797_dai_pcm_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_PCM_1; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_pcm_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_widgets = mtk_dai_pcm_widgets; - afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); - afe->sub_dais[id].dapm_routes = mtk_dai_pcm_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); + list_add(&dai->list, &afe->sub_dais); + dai->dai_drivers = mtk_dai_pcm_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + + dai->dapm_widgets = mtk_dai_pcm_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); + dai->dapm_routes = mtk_dai_pcm_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); return 0; } -- cgit v1.2.3 From d573454d9b4f00061da314460908e19476d2ff6d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:30:28 +0000 Subject: ASoC: simple-card: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card.h | 7 ++----- sound/soc/generic/simple-card.c | 17 +++++++---------- 2 files changed, 9 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index a6a2e1547092..d264e5463f22 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -1,12 +1,9 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * ASoC simple sound card support * * Copyright (C) 2012 Renesas Solutions Corp. * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __SIMPLE_CARD_H diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c5b6e04cd926..64bf3560c1d1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -1,13 +1,10 @@ -/* - * ASoC simple sound card support - * - * Copyright (C) 2012 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC simple sound card support +// +// Copyright (C) 2012 Renesas Solutions Corp. +// Kuninori Morimoto + #include #include #include -- cgit v1.2.3 From d613a7f45ebb2f113444630fcbbb8a074c741998 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:30:44 +0000 Subject: ASoC: simple-card-utils: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 8 +++----- sound/soc/generic/simple-card-utils.c | 15 ++++++--------- 2 files changed, 9 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index f82acef3b992..8bc5e2d8b13c 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -1,12 +1,10 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * simple_card_utils.h * * Copyright (c) 2016 Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ + #ifndef __SIMPLE_CARD_UTILS_H #define __SIMPLE_CARD_UTILS_H diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 4398c9580929..d3f3f0fec74c 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1,12 +1,9 @@ -/* - * simple-card-utils.c - * - * Copyright (c) 2016 Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// simple-card-utils.c +// +// Copyright (c) 2016 Kuninori Morimoto + #include #include #include -- cgit v1.2.3 From 9afe58f1cbd1fb5e9426483656ae6f65de4130e4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:30:58 +0000 Subject: ASoC: simple-scu-card.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-scu-card.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 487716559deb..16a83bc51e0e 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -1,15 +1,12 @@ -/* - * ASoC simple SCU sound card support - * - * Copyright (C) 2015 Renesas Solutions Corp. - * Kuninori Morimoto - * - * based on ${LINUX}/sound/soc/generic/simple-card.c - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC simple SCU sound card support +// +// Copyright (C) 2015 Renesas Solutions Corp. +// Kuninori Morimoto +// +// based on ${LINUX}/sound/soc/generic/simple-card.c + #include #include #include -- cgit v1.2.3 From decd896121f965db2fdee0f0475c3e404746b333 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:31:16 +0000 Subject: ASoC: audio-graph-card.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index a2a3e630f11c..2094d2c8919f 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -1,15 +1,12 @@ -/* - * ASoC audio graph sound card support - * - * Copyright (C) 2016 Renesas Solutions Corp. - * Kuninori Morimoto - * - * based on ${LINUX}/sound/soc/generic/simple-card.c - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC audio graph sound card support +// +// Copyright (C) 2016 Renesas Solutions Corp. +// Kuninori Morimoto +// +// based on ${LINUX}/sound/soc/generic/simple-card.c + #include #include #include -- cgit v1.2.3 From ac204c9b030fe2d27cf3a63386811254d99b3ce4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:31:33 +0000 Subject: ASoC: audio-graph-scu-card.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-scu-card.c | 25 +++++++++++-------------- 1 file changed, 11 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 095ef6426d42..92882e392d6c 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -1,17 +1,14 @@ -/* - * ASoC audio graph SCU sound card support - * - * Copyright (C) 2017 Renesas Solutions Corp. - * Kuninori Morimoto - * - * based on - * ${LINUX}/sound/soc/generic/simple-scu-card.c - * ${LINUX}/sound/soc/generic/audio-graph-card.c - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC audio graph SCU sound card support +// +// Copyright (C) 2017 Renesas Solutions Corp. +// Kuninori Morimoto +// +// based on +// ${LINUX}/sound/soc/generic/simple-scu-card.c +// ${LINUX}/sound/soc/generic/audio-graph-card.c + #include #include #include -- cgit v1.2.3 From d1aaa2e68619ca4c1cc05ce7bc029cda5a8cbe87 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:21:54 +0000 Subject: ASoC: soc-io.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-io.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 026cd5347e53..1ff9175e9d5e 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -1,15 +1,10 @@ -/* - * soc-io.c -- ASoC register I/O helpers - * - * Copyright 2009-2011 Wolfson Microelectronics PLC. - * - * Author: Mark Brown - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-io.c -- ASoC register I/O helpers +// +// Copyright 2009-2011 Wolfson Microelectronics PLC. +// +// Author: Mark Brown #include #include -- cgit v1.2.3 From 4eef5a90ca8b2aab61c98853141f9242af4a8339 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:22:30 +0000 Subject: ASoC: soc-ops.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 29 ++++++++++++----------------- 1 file changed, 12 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 7144a51ddfa9..592efb370c44 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -1,20 +1,15 @@ -/* - * soc-ops.c -- Generic ASoC operations - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood - * with code, comments and ideas from :- - * Richard Purdie - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-ops.c -- Generic ASoC operations +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood +// with code, comments and ideas from :- +// Richard Purdie #include #include -- cgit v1.2.3 From ed51758247c51dfdb526d279164cfd925496f187 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:22:44 +0000 Subject: ASoC: soc-pcm.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 28 +++++++++++----------------- 1 file changed, 11 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6ee4131941df..c2a31b51da4f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1,20 +1,14 @@ -/* - * soc-pcm.c -- ALSA SoC PCM - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Authors: Liam Girdwood - * Mark Brown - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-pcm.c -- ALSA SoC PCM +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Authors: Liam Girdwood +// Mark Brown #include #include -- cgit v1.2.3 From 8ab0215c11817fc0a89f1f82cdf10fd4c0eb9e86 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:23:16 +0000 Subject: ASoC: soc-jack.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index b2b16044ae80..c7b990abdbaa 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -1,15 +1,10 @@ -/* - * soc-jack.c -- ALSA SoC jack handling - * - * Copyright 2008 Wolfson Microelectronics PLC. - * - * Author: Mark Brown - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-jack.c -- ALSA SoC jack handling +// +// Copyright 2008 Wolfson Microelectronics PLC. +// +// Author: Mark Brown #include #include -- cgit v1.2.3 From 632628df453ccda727fbcc1e346600162ac995e9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:23:30 +0000 Subject: ASoC: soc-utils.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 22 ++++++++-------------- 1 file changed, 8 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 2d9e98bd1530..ea024236c643 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -1,17 +1,11 @@ -/* - * soc-util.c -- ALSA SoC Audio Layer utility functions - * - * Copyright 2009 Wolfson Microelectronics PLC. - * - * Author: Mark Brown - * Liam Girdwood - * - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-util.c -- ALSA SoC Audio Layer utility functions +// +// Copyright 2009 Wolfson Microelectronics PLC. +// +// Author: Mark Brown +// Liam Girdwood #include #include -- cgit v1.2.3 From 9e14035c7fac144f31a822f0034fe5ed79c9ef8a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:23:45 +0000 Subject: ASoC: soc-devres.c: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-devres.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 7ac745df1412..a9ea172a66a7 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -1,13 +1,8 @@ -/* - * soc-devres.c -- ALSA SoC Audio Layer devres functions - * - * Copyright (C) 2013 Linaro Ltd - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-devres.c -- ALSA SoC Audio Layer devres functions +// +// Copyright (C) 2013 Linaro Ltd #include #include -- cgit v1.2.3 From 7730bb13c7472620b585783f248b2dccd09d1819 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:24:04 +0000 Subject: ASoC: soc-acpi: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc-acpi-intel-match.h | 14 ++------------ include/sound/soc-acpi.h | 13 ++----------- sound/soc/soc-acpi.c | 20 +++++--------------- 3 files changed, 9 insertions(+), 38 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index 917ddd0f2762..bb1d24b703fb 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -1,16 +1,6 @@ - -/* - * Copyright (C) 2017, Intel Corporation. All rights reserved. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License version - * 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. +/* SPDX-License-Identifier: GPL-2.0 * + * Copyright (C) 2017, Intel Corporation. All rights reserved. */ #ifndef __LINUX_SND_SOC_ACPI_INTEL_MATCH_H diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 082224275f52..e45b2330d16a 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -1,15 +1,6 @@ -/* - * Copyright (C) 2013-15, Intel Corporation. All rights reserved. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License version - * 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. +/* SPDX-License-Identifier: GPL-2.0 * + * Copyright (C) 2013-15, Intel Corporation. All rights reserved. */ #ifndef __LINUX_SND_SOC_ACPI_H diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index 3d7e1ff79139..b8e72b52db30 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -1,18 +1,8 @@ -/* - * soc-apci.c - support for ACPI enumeration. - * - * Copyright (c) 2013-15, Intel Corporation. - * - * - * This program is free software; you can redistribute it and/or modify it - * under the terms and conditions of the GNU General Public License, - * version 2, as published by the Free Software Foundation. - * - * This program is distributed in the hope it will be useful, but WITHOUT - * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or - * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for - * more details. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// soc-apci.c - support for ACPI enumeration. +// +// Copyright (c) 2013-15, Intel Corporation. #include -- cgit v1.2.3 From 873486ed4af3e11bfc20832dff7b124ba652bf77 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:24:18 +0000 Subject: ASoC: soc-core: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 11 ++++------- sound/soc/soc-core.c | 41 ++++++++++++++++++----------------------- 2 files changed, 22 insertions(+), 30 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index f7579f82cc00..16f0bf10cc42 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1,13 +1,10 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * linux/sound/soc.h -- ALSA SoC Layer * - * Author: Liam Girdwood - * Created: Aug 11th 2005 + * Author: Liam Girdwood + * Created: Aug 11th 2005 * Copyright: Wolfson Microelectronics. PLC. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __LINUX_SND_SOC_H diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4663de3cf495..68b08781c832 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1,26 +1,21 @@ -/* - * soc-core.c -- ALSA SoC Audio Layer - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood - * with code, comments and ideas from :- - * Richard Purdie - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * TODO: - * o Add hw rules to enforce rates, etc. - * o More testing with other codecs/machines. - * o Add more codecs and platforms to ensure good API coverage. - * o Support TDM on PCM and I2S - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-core.c -- ALSA SoC Audio Layer +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood +// with code, comments and ideas from :- +// Richard Purdie +// +// TODO: +// o Add hw rules to enforce rates, etc. +// o More testing with other codecs/machines. +// o Add more codecs and platforms to ensure good API coverage. +// o Support TDM on PCM and I2S #include #include -- cgit v1.2.3 From c01f3af4d32071915119ffcb933e75d7c165378e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:24:31 +0000 Subject: ASoC: soc-dapm: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 11 ++++------- sound/soc/soc-dapm.c | 42 ++++++++++++++++++------------------------ 2 files changed, 22 insertions(+), 31 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a6ce2de4e20a..af9ef16cc34d 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -1,13 +1,10 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * linux/sound/soc-dapm.h -- ALSA SoC Dynamic Audio Power Management * - * Author: Liam Girdwood - * Created: Aug 11th 2005 + * Author: Liam Girdwood + * Created: Aug 11th 2005 * Copyright: Wolfson Microelectronics. PLC. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __LINUX_SND_SOC_DAPM_H diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a099c3e45504..0602b2888d52 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1,27 +1,21 @@ -/* - * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Features: - * o Changes power status of internal codec blocks depending on the - * dynamic configuration of codec internal audio paths and active - * DACs/ADCs. - * o Platform power domain - can support external components i.e. amps and - * mic/headphone insertion events. - * o Automatic Mic Bias support - * o Jack insertion power event initiation - e.g. hp insertion will enable - * sinks, dacs, etc - * o Delayed power down of audio subsystem to reduce pops between a quick - * device reopen. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-dapm.c -- ALSA SoC Dynamic Audio Power Management +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Author: Liam Girdwood +// +// Features: +// o Changes power status of internal codec blocks depending on the +// dynamic configuration of codec internal audio paths and active +// DACs/ADCs. +// o Platform power domain - can support external components i.e. amps and +// mic/headphone insertion events. +// o Automatic Mic Bias support +// o Jack insertion power event initiation - e.g. hp insertion will enable +// sinks, dacs, etc +// o Delayed power down of audio subsystem to reduce pops between a quick +// device reopen. #include #include -- cgit v1.2.3 From f2b6a1b25fecc48a46c8a41636101af8a41c88a8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:24:45 +0000 Subject: ASoC: soc-topology: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc-topology.h | 7 ++----- sound/soc/soc-topology.c | 47 ++++++++++++++++++++------------------------ 2 files changed, 23 insertions(+), 31 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 401ef2c45d6c..fa4b8413d2e2 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -1,13 +1,10 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * linux/sound/soc-topology.h -- ALSA SoC Firmware Controls and DAPM * * Copyright (C) 2012 Texas Instruments Inc. * Copyright (C) 2015 Intel Corporation. * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * * Simple file API to load FW that includes mixers, coefficients, DAPM graphs, * algorithms, equalisers, DAIs, widgets, FE caps, BE caps, codec link caps etc. */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 05d177d689e2..66e77e020745 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1,29 +1,24 @@ -/* - * soc-topology.c -- ALSA SoC Topology - * - * Copyright (C) 2012 Texas Instruments Inc. - * Copyright (C) 2015 Intel Corporation. - * - * Authors: Liam Girdwood - * K, Mythri P - * Prusty, Subhransu S - * B, Jayachandran - * Abdullah, Omair M - * Jin, Yao - * Lin, Mengdong - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Add support to read audio firmware topology alongside firmware text. The - * topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links, - * equalizers, firmware, coefficients etc. - * - * This file only manages the core ALSA and ASoC components, all other bespoke - * firmware topology data is passed to component drivers for bespoke handling. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-topology.c -- ALSA SoC Topology +// +// Copyright (C) 2012 Texas Instruments Inc. +// Copyright (C) 2015 Intel Corporation. +// +// Authors: Liam Girdwood +// K, Mythri P +// Prusty, Subhransu S +// B, Jayachandran +// Abdullah, Omair M +// Jin, Yao +// Lin, Mengdong +// +// Add support to read audio firmware topology alongside firmware text. The +// topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links, +// equalizers, firmware, coefficients etc. +// +// This file only manages the core ALSA and ASoC components, all other bespoke +// firmware topology data is passed to component drivers for bespoke handling. #include #include -- cgit v1.2.3 From b3ed4c86a700b494fc5058a52531eeb14d6fe00f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:24:57 +0000 Subject: ASoC: soc-compress: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/compress_driver.h | 21 +++------------------ sound/soc/soc-compress.c | 24 +++++++++--------------- 2 files changed, 12 insertions(+), 33 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 9924bc9cbc7c..ea8c93bbb0e0 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -1,27 +1,12 @@ -/* +/* SPDX-License-Identifier: GPL-2.0 + * * compress_driver.h - compress offload driver definations * * Copyright (C) 2011 Intel Corporation * Authors: Vinod Koul * Pierre-Louis Bossart - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * */ + #ifndef __COMPRESS_DRIVER_H #define __COMPRESS_DRIVER_H diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index e095115fa9f9..b9e1673fea51 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -1,18 +1,12 @@ -/* - * soc-compress.c -- ALSA SoC Compress - * - * Copyright (C) 2012 Intel Corp. - * - * Authors: Namarta Kohli - * Ramesh Babu K V - * Vinod Koul - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-compress.c -- ALSA SoC Compress +// +// Copyright (C) 2012 Intel Corp. +// +// Authors: Namarta Kohli +// Ramesh Babu K V +// Vinod Koul #include #include -- cgit v1.2.3 From 1356a6071cf4d7187652cd2b18dfab4763e0dba6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:25:11 +0000 Subject: ASoC: soc-generic-dmaengine-pcm: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/dmaengine_pcm.h | 14 +++----------- sound/soc/soc-generic-dmaengine-pcm.c | 19 +++++-------------- 2 files changed, 8 insertions(+), 25 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index e3481eebdd98..2c4cfaa135a6 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -1,17 +1,9 @@ -/* +/* SPDX-License-Identifier: GPL-2.0+ + * * Copyright (C) 2012, Analog Devices Inc. * Author: Lars-Peter Clausen - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * */ + #ifndef __SOUND_DMAENGINE_PCM_H__ #define __SOUND_DMAENGINE_PCM_H__ diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 56a541b9ff9e..13bdca6e41c5 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -1,17 +1,8 @@ -/* - * Copyright (C) 2013, Analog Devices Inc. - * Author: Lars-Peter Clausen - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright (C) 2013, Analog Devices Inc. +// Author: Lars-Peter Clausen + #include #include #include -- cgit v1.2.3 From 1a8f0a3c13c136951de7ea24ccb148e745db98a2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 2 Jul 2018 06:26:27 +0000 Subject: ASoC: ac97: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/ac97/codec.h | 8 +++----- include/sound/ac97/compat.h | 9 +++------ include/sound/ac97/controller.h | 8 +++----- include/sound/ac97/regs.h | 20 ++------------------ include/sound/ac97_codec.h | 25 +++++-------------------- sound/soc/soc-ac97.c | 29 ++++++++++++----------------- 6 files changed, 28 insertions(+), 71 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/ac97/codec.h b/include/sound/ac97/codec.h index ec04be9ab119..9792d25fa369 100644 --- a/include/sound/ac97/codec.h +++ b/include/sound/ac97/codec.h @@ -1,10 +1,8 @@ -/* - * Copyright (C) 2016 Robert Jarzmik +/* SPDX-License-Identifier: GPL-2.0 * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. + * Copyright (C) 2016 Robert Jarzmik */ + #ifndef __SOUND_AC97_CODEC2_H #define __SOUND_AC97_CODEC2_H diff --git a/include/sound/ac97/compat.h b/include/sound/ac97/compat.h index 1351cba40048..57e19afa31ab 100644 --- a/include/sound/ac97/compat.h +++ b/include/sound/ac97/compat.h @@ -1,14 +1,11 @@ -/* - * Copyright (C) 2016 Robert Jarzmik +/* SPDX-License-Identifier: GPL-2.0 * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. + * Copyright (C) 2016 Robert Jarzmik * * This file is for backward compatibility with snd_ac97 structure and its * multiple usages, such as the snd_ac97_bus and snd_ac97_build_ops. - * */ + #ifndef AC97_COMPAT_H #define AC97_COMPAT_H diff --git a/include/sound/ac97/controller.h b/include/sound/ac97/controller.h index b36ecdd64f14..06b5afb7fa6b 100644 --- a/include/sound/ac97/controller.h +++ b/include/sound/ac97/controller.h @@ -1,10 +1,8 @@ -/* - * Copyright (C) 2016 Robert Jarzmik +/* SPDX-License-Identifier: GPL-2.0 * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. + * Copyright (C) 2016 Robert Jarzmik */ + #ifndef AC97_CONTROLLER_H #define AC97_CONTROLLER_H diff --git a/include/sound/ac97/regs.h b/include/sound/ac97/regs.h index 9a4fa0c3264a..843f73f3705a 100644 --- a/include/sound/ac97/regs.h +++ b/include/sound/ac97/regs.h @@ -1,27 +1,11 @@ -/* +/* SPDX-License-Identifier: GPL-2.0+ + * * Copyright (c) by Jaroslav Kysela * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.1 * by Intel Corporation (http://developer.intel.com). - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * */ - /* * AC'97 codec registers */ diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 89d311a503d3..cc383991c0fe 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -1,30 +1,15 @@ -#ifndef __SOUND_AC97_CODEC_H -#define __SOUND_AC97_CODEC_H - -/* +/* SPDX-License-Identifier: GPL-2.0+ + * * Copyright (c) by Jaroslav Kysela * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.1 * by Intel Corporation (http://developer.intel.com). - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * */ +#ifndef __SOUND_AC97_CODEC_H +#define __SOUND_AC97_CODEC_H + #include #include #include diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 3f424f214bca..c086786e4471 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -1,20 +1,15 @@ -/* - * soc-ac97.c -- ALSA SoC Audio Layer AC97 support - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood - * with code, comments and ideas from :- - * Richard Purdie - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-ac97.c -- ALSA SoC Audio Layer AC97 support +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood +// with code, comments and ideas from :- +// Richard Purdie #include #include -- cgit v1.2.3 From 1581250119daa9426c359d059e2dc14ec04bcc0c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sat, 30 Jun 2018 22:24:33 +0200 Subject: ASoC: pxa: select SND_PXA2XX_LIB for drivers that depend on it Commit d767d3ce5c48b ("ASoC: pxa: provide PCM ops for ssp, i2s and ac97 components") created a build-time dependency to SND_PXA2XX_LIB but missed to reflect that in Kconfig. Reported-by: kbuild test robot Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 95dcf97a6271..2fc02c227f69 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -24,15 +24,18 @@ config SND_PXA2XX_AC97 config SND_PXA2XX_SOC_AC97 tristate select AC97_BUS + select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 select SND_SOC_AC97_BUS config SND_PXA2XX_SOC_I2S + select SND_PXA2XX_LIB tristate config SND_PXA_SOC_SSP tristate "Soc Audio via PXA2xx/PXA3xx SSP ports" select PXA_SSP + select SND_PXA2XX_LIB config SND_MMP_SOC_SSPA tristate -- cgit v1.2.3 From 05739375f1c0a1048fea8b9c4cb54d9e4a891938 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 29 Jun 2018 14:59:40 +0200 Subject: ASoC: pxa-ssp: remove .set_pll() and .set_clkdiv() callbacks The .set_pll() and .set_clkdiv() callbacks are considered legacy and should not be used anymore. In order to support PXA boards on DT platforms, remove them and let the code figure out the correct dividers and PLL base frequencies itself. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- include/linux/pxa2xx_ssp.h | 8 +++ sound/soc/pxa/pxa-ssp.c | 146 ++++++++++++++++++++++----------------------- 2 files changed, 81 insertions(+), 73 deletions(-) (limited to 'sound/soc') diff --git a/include/linux/pxa2xx_ssp.h b/include/linux/pxa2xx_ssp.h index 03a7ca46735b..13b4244d44c1 100644 --- a/include/linux/pxa2xx_ssp.h +++ b/include/linux/pxa2xx_ssp.h @@ -171,6 +171,14 @@ #define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass */ #define SSACD_ACPS(x) ((x) << 4) /* Audio clock PLL select */ #define SSACD_ACDS(x) ((x) << 0) /* Audio clock divider select */ +#define SSACD_ACDS_1 (0) +#define SSACD_ACDS_2 (1) +#define SSACD_ACDS_4 (2) +#define SSACD_ACDS_8 (3) +#define SSACD_ACDS_16 (4) +#define SSACD_ACDS_32 (5) +#define SSACD_SCDB_4X (0) +#define SSACD_SCDB_1X (1) #define SSACD_SCDX8 (1 << 7) /* SYSCLK division ratio select */ /* LPSS SSP */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 01d54697ede4..f8339bb01251 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -41,6 +41,7 @@ */ struct ssp_priv { struct ssp_device *ssp; + unsigned long ssp_clk; unsigned int sysclk; unsigned int dai_fmt; unsigned int configured_dai_fmt; @@ -192,21 +193,6 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) pxa_ssp_write_reg(ssp, SSCR0, sscr0); } -/** - * pxa_ssp_get_clkdiv - get SSP clock divider - */ -static u32 pxa_ssp_get_scr(struct ssp_device *ssp) -{ - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - u32 div; - - if (ssp->type == PXA25x_SSP) - div = ((sscr0 >> 8) & 0xff) * 2 + 2; - else - div = ((sscr0 >> 8) & 0xfff) + 1; - return div; -} - /* * Set the SSP ports SYSCLK. */ @@ -262,67 +248,18 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return 0; } -/* - * Set the SSP clock dividers. - */ -static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); - struct ssp_device *ssp = priv->ssp; - int val; - - switch (div_id) { - case PXA_SSP_AUDIO_DIV_ACDS: - val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_AUDIO_DIV_SCDB: - val = pxa_ssp_read_reg(ssp, SSACD); - val &= ~SSACD_SCDB; - if (ssp->type == PXA3xx_SSP) - val &= ~SSACD_SCDX8; - switch (div) { - case PXA_SSP_CLK_SCDB_1: - val |= SSACD_SCDB; - break; - case PXA_SSP_CLK_SCDB_4: - break; - case PXA_SSP_CLK_SCDB_8: - if (ssp->type == PXA3xx_SSP) - val |= SSACD_SCDX8; - else - return -EINVAL; - break; - default: - return -EINVAL; - } - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_DIV_SCR: - pxa_ssp_set_scr(ssp, div); - break; - default: - return -ENODEV; - } - - return 0; -} - /* * Configure the PLL frequency pxa27x and (afaik - pxa320 only) */ -static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int pxa_ssp_set_pll(struct ssp_priv *priv, unsigned int freq) { - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; if (ssp->type == PXA3xx_SSP) pxa_ssp_write_reg(ssp, SSACDD, 0); - switch (freq_out) { + switch (freq) { case 5622000: break; case 11345000: @@ -353,7 +290,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, u64 tmp = 19968; tmp *= 1000000; - do_div(tmp, freq_out); + do_div(tmp, freq); val = tmp; val = (val << 16) | 64; @@ -363,7 +300,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, dev_dbg(&ssp->pdev->dev, "Using SSACDD %x to supply %uHz\n", - val, freq_out); + val, freq); break; } @@ -568,6 +505,24 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) return 0; } +struct pxa_ssp_clock_mode { + int rate; + int pll; + u8 acds; + u8 scdb; +}; + +static const struct pxa_ssp_clock_mode pxa_ssp_clock_modes[] = { + { .rate = 8000, .pll = 32842000, .acds = SSACD_ACDS_32, .scdb = SSACD_SCDB_4X }, + { .rate = 11025, .pll = 5622000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_4X }, + { .rate = 16000, .pll = 32842000, .acds = SSACD_ACDS_16, .scdb = SSACD_SCDB_4X }, + { .rate = 22050, .pll = 5622000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 44100, .pll = 11345000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 48000, .pll = 12235000, .acds = SSACD_ACDS_2, .scdb = SSACD_SCDB_4X }, + { .rate = 96000, .pll = 12235000, .acds = SSACD_ACDS_4, .scdb = SSACD_SCDB_1X }, + {} +}; + /* * Set the SSP audio DMA parameters and sample size. * Can be called multiple times by oss emulation. @@ -579,11 +534,12 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; int chn = params_channels(params); - u32 sscr0; - u32 sspsp; + u32 sscr0, sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; struct snd_dmaengine_dai_dma_data *dma_data; + int rate = params_rate(params); + int bclk = rate * chn * (width / 8); int ret; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); @@ -623,11 +579,57 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, } pxa_ssp_write_reg(ssp, SSCR0, sscr0); + if (sscr0 & SSCR0_ACS) { + ret = pxa_ssp_set_pll(priv, bclk); + + /* + * If we were able to generate the bclk directly, + * all is fine. Otherwise, look up the closest rate + * from the table and also set the dividers. + */ + + if (ret < 0) { + const struct pxa_ssp_clock_mode *m; + int ssacd, acds; + + for (m = pxa_ssp_clock_modes; m->rate; m++) { + if (m->rate == rate) + break; + } + + if (!m->rate) + return -EINVAL; + + acds = m->acds; + + /* The values in the table are for 16 bits */ + if (width == 32) + acds--; + + ret = pxa_ssp_set_pll(priv, bclk); + if (ret < 0) + return ret; + + ssacd = pxa_ssp_read_reg(ssp, SSACD); + ssacd &= ~(SSACD_ACDS(7) | SSACD_SCDB_1X); + ssacd |= SSACD_ACDS(m->acds); + ssacd |= m->scdb; + pxa_ssp_write_reg(ssp, SSACD, ssacd); + } + } else if (sscr0 & SSCR0_ECS) { + /* + * For setups with external clocking, the PLL and its diviers + * are not active. Instead, the SCR bits in SSCR0 can be used + * to divide the clock. + */ + pxa_ssp_set_scr(ssp, bclk / rate); + } + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: sspsp = pxa_ssp_read_reg(ssp, SSPSP); - if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { + if (((priv->sysclk / bclk) == 64) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. @@ -812,8 +814,6 @@ static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { .trigger = pxa_ssp_trigger, .hw_params = pxa_ssp_hw_params, .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, .set_fmt = pxa_ssp_set_dai_fmt, .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, .set_tristate = pxa_ssp_set_dai_tristate, -- cgit v1.2.3 From 5650729f9a1bbf65b57139d855dabe0a7e6cb494 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 29 Jun 2018 17:09:20 +0200 Subject: ASoC: es7134: remove 64kHz rate from the supported rates 64Khz is actually not supported by the es7134 according to the datasheet Fixes: 9000b59d7a12 ("ASoC: es7134: add es7134 DAC driver") Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/es7134.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index 58515bb1a303..2fbb49f5b278 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -48,7 +48,11 @@ static struct snd_soc_dai_driver es7134_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = (SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE | -- cgit v1.2.3 From a016b11cc41df06b79c0c226e719d0d88373919c Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 29 Jun 2018 17:09:21 +0200 Subject: ASoC: es7134: check if mclk rate is valid For each supported sample rate, the es7134 can work with several mclk / sample rate ratio. Check if ratio we get is actually OK. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/es7134.c | 119 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 117 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index 2fbb49f5b278..698289dc3e22 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -17,6 +17,7 @@ * in the file called COPYING. */ +#include #include #include @@ -24,6 +25,77 @@ * The everest 7134 is a very simple DA converter with no register */ +struct es7134_clock_mode { + unsigned int rate_min; + unsigned int rate_max; + unsigned int *mclk_fs; + unsigned int mclk_fs_num; +}; + +struct es7134_chip { + const struct es7134_clock_mode *modes; + unsigned int mode_num; +}; + +struct es7134_data { + unsigned int mclk; + const struct es7134_chip *chip; +}; + +static int es7134_check_mclk(struct snd_soc_dai *dai, + struct es7134_data *priv, + unsigned int rate) +{ + unsigned int mfs = priv->mclk / rate; + int i, j; + + for (i = 0; i < priv->chip->mode_num; i++) { + const struct es7134_clock_mode *mode = &priv->chip->modes[i]; + + if (rate < mode->rate_min || rate > mode->rate_max) + continue; + + for (j = 0; j < mode->mclk_fs_num; j++) { + if (mode->mclk_fs[j] == mfs) + return 0; + } + + dev_err(dai->dev, "unsupported mclk_fs %u for rate %u\n", + mfs, rate); + return -EINVAL; + } + + /* should not happen */ + dev_err(dai->dev, "unsupported rate: %u\n", rate); + return -EINVAL; +} + +static int es7134_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct es7134_data *priv = snd_soc_dai_get_drvdata(dai); + + /* mclk has not been provided, assume it is OK */ + if (!priv->mclk) + return 0; + + return es7134_check_mclk(dai, priv, params_rate(params)); +} + +static int es7134_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct es7134_data *priv = snd_soc_dai_get_drvdata(dai); + + if (dir == SND_SOC_CLOCK_IN && clk_id == 0) { + priv->mclk = freq; + return 0; + } + + return -ENOTSUPP; +} + static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { fmt &= (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK | @@ -40,6 +112,8 @@ static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) static const struct snd_soc_dai_ops es7134_dai_ops = { .set_fmt = es7134_set_fmt, + .hw_params = es7134_hw_params, + .set_sysclk = es7134_set_sysclk, }; static struct snd_soc_dai_driver es7134_dai = { @@ -62,6 +136,33 @@ static struct snd_soc_dai_driver es7134_dai = { .ops = &es7134_dai_ops, }; +static const struct es7134_clock_mode es7134_modes[] = { + { + /* Single speed mode */ + .rate_min = 8000, + .rate_max = 50000, + .mclk_fs = (unsigned int[]) { 256, 384, 512, 768, 1024 }, + .mclk_fs_num = 5, + }, { + /* Double speed mode */ + .rate_min = 84000, + .rate_max = 100000, + .mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512 }, + .mclk_fs_num = 5, + }, { + /* Quad speed mode */ + .rate_min = 167000, + .rate_max = 192000, + .mclk_fs = (unsigned int[]) { 128, 192, 256 }, + .mclk_fs_num = 3, + }, +}; + +static const struct es7134_chip es7134_chip = { + .modes = es7134_modes, + .mode_num = ARRAY_SIZE(es7134_modes), +}; + static const struct snd_soc_dapm_widget es7134_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("AOUTL"), SND_SOC_DAPM_OUTPUT("AOUTR"), @@ -86,6 +187,20 @@ static const struct snd_soc_component_driver es7134_component_driver = { static int es7134_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; + struct es7134_data *priv; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->chip = of_device_get_match_data(dev); + if (!priv->chip) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + return devm_snd_soc_register_component(&pdev->dev, &es7134_component_driver, &es7134_dai, 1); @@ -93,8 +208,8 @@ static int es7134_probe(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id es7134_ids[] = { - { .compatible = "everest,es7134", }, - { .compatible = "everest,es7144", }, + { .compatible = "everest,es7134", .data = &es7134_chip }, + { .compatible = "everest,es7144", .data = &es7134_chip }, { } }; MODULE_DEVICE_TABLE(of, es7134_ids); -- cgit v1.2.3 From 424e2b4b3521334812d833eef27df77671428698 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 29 Jun 2018 17:09:23 +0200 Subject: ASoC: es7134: Add VDD and AVDD power supplies Add the VDD and AVDD power supplies to the DAPM graph as some board may need to enable a regulator to turn them on. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/es7134.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index 698289dc3e22..5ad59c38fed1 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -167,11 +167,15 @@ static const struct snd_soc_dapm_widget es7134_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("AOUTL"), SND_SOC_DAPM_OUTPUT("AOUTR"), SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDD", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 0, 0), }; static const struct snd_soc_dapm_route es7134_dapm_routes[] = { { "AOUTL", NULL, "DAC" }, { "AOUTR", NULL, "DAC" }, + { "Playback", NULL, "VDD" }, + { "DAC", NULL, "AVDD" }, }; static const struct snd_soc_component_driver es7134_component_driver = { -- cgit v1.2.3 From 2daf3d9962c5a11fb79fe17bae03125df5d60236 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 3 Jul 2018 13:07:25 +0800 Subject: ASoC: rt5682: add button detection mode control We are currently using power saving mode for button detection. However, it will impact the headset recording performance. This patch will switch button detection to normal mode in capture and switch to power saving mode in the end of capture. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index baad177908ab..640d400ca013 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -857,6 +857,8 @@ static int rt5682_button_detect(struct snd_soc_component *component) btn_type = val & 0xfff0; snd_soc_component_write(component, RT5682_4BTN_IL_CMD_1, val); pr_debug("%s btn_type=%x\n", __func__, btn_type); + snd_soc_component_update_bits(component, + RT5682_SAR_IL_CMD_2, 0x10, 0x10); return btn_type; } @@ -1645,6 +1647,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682_STO1_ADC_DIG_VOL, RT5682_R_MUTE_SFT, 1, rt5682_sto1_adc_r_mix, ARRAY_SIZE(rt5682_sto1_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("BTN Detection Mode", RT5682_SAR_IL_CMD_1, + 14, 1, NULL, 0), /* ADC PGA */ SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1807,6 +1811,8 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, + {"ADC Stereo1 Filter", NULL, "BTN Detection Mode"}, + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, -- cgit v1.2.3 From 5f7bdc466c772b3af3145a71724965ecdc03e6bf Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 3 Jul 2018 15:28:45 +0200 Subject: ASoC: es7241: add es7241 codec support Add support for the everest es7241 which is a simple 2 channels analog to digital converter. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es7241.c | 322 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 328 insertions(+) create mode 100644 sound/soc/codecs/es7241.c (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6d1674699385..efb095dbcd71 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -79,6 +79,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_ES7134 + select SND_SOC_ES7241 select SND_SOC_GTM601 select SND_SOC_HDAC_HDMI select SND_SOC_ICS43432 @@ -585,6 +586,9 @@ config SND_SOC_HDMI_CODEC config SND_SOC_ES7134 tristate "Everest Semi ES7134 CODEC" +config SND_SOC_ES7241 + tristate "Everest Semi ES7241 CODEC" + config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f26ded89a1e5..7ae7c85e8219 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -71,6 +71,7 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o +snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o @@ -330,6 +331,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o +obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o diff --git a/sound/soc/codecs/es7241.c b/sound/soc/codecs/es7241.c new file mode 100644 index 000000000000..87991bd4acef --- /dev/null +++ b/sound/soc/codecs/es7241.c @@ -0,0 +1,322 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include + +struct es7241_clock_mode { + unsigned int rate_min; + unsigned int rate_max; + unsigned int *slv_mfs; + unsigned int slv_mfs_num; + unsigned int mst_mfs; + unsigned int mst_m0:1; + unsigned int mst_m1:1; +}; + +struct es7241_chip { + const struct es7241_clock_mode *modes; + unsigned int mode_num; +}; + +struct es7241_data { + struct gpio_desc *reset; + struct gpio_desc *m0; + struct gpio_desc *m1; + unsigned int fmt; + unsigned int mclk; + bool is_slave; + const struct es7241_chip *chip; +}; + +static void es7241_set_mode(struct es7241_data *priv, int m0, int m1) +{ + /* put the device in reset */ + gpiod_set_value_cansleep(priv->reset, 0); + + /* set the mode */ + gpiod_set_value_cansleep(priv->m0, m0); + gpiod_set_value_cansleep(priv->m1, m1); + + /* take the device out of reset - datasheet does not specify a delay */ + gpiod_set_value_cansleep(priv->reset, 1); +} + +static int es7241_set_slave_mode(struct es7241_data *priv, + const struct es7241_clock_mode *mode, + unsigned int mfs) +{ + int j; + + if (!mfs) + goto out_ok; + + for (j = 0; j < mode->slv_mfs_num; j++) { + if (mode->slv_mfs[j] == mfs) + goto out_ok; + } + + return -EINVAL; + +out_ok: + es7241_set_mode(priv, 1, 1); + return 0; +} + +static int es7241_set_master_mode(struct es7241_data *priv, + const struct es7241_clock_mode *mode, + unsigned int mfs) +{ + /* + * We can't really set clock ratio, if the mclk/lrclk is different + * from what we provide, then error out + */ + if (mfs && mfs != mode->mst_mfs) + return -EINVAL; + + es7241_set_mode(priv, mode->mst_m0, mode->mst_m1); + + return 0; +} + +static int es7241_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct es7241_data *priv = snd_soc_dai_get_drvdata(dai); + unsigned int rate = params_rate(params); + unsigned int mfs = priv->mclk / rate; + int i; + + for (i = 0; i < priv->chip->mode_num; i++) { + const struct es7241_clock_mode *mode = &priv->chip->modes[i]; + + if (rate < mode->rate_min || rate >= mode->rate_max) + continue; + + if (priv->is_slave) + return es7241_set_slave_mode(priv, mode, mfs); + else + return es7241_set_master_mode(priv, mode, mfs); + } + + /* should not happen */ + dev_err(dai->dev, "unsupported rate: %u\n", rate); + return -EINVAL; +} + +static int es7241_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct es7241_data *priv = snd_soc_dai_get_drvdata(dai); + + if (dir == SND_SOC_CLOCK_IN && clk_id == 0) { + priv->mclk = freq; + return 0; + } + + return -ENOTSUPP; +} + +static int es7241_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct es7241_data *priv = snd_soc_dai_get_drvdata(dai); + + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) { + dev_err(dai->dev, "Unsupported dai clock inversion\n"); + return -EINVAL; + } + + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != priv->fmt) { + dev_err(dai->dev, "Invalid dai format\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + priv->is_slave = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + priv->is_slave = false; + break; + + default: + dev_err(dai->dev, "Unsupported clock configuration\n"); + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops es7241_dai_ops = { + .set_fmt = es7241_set_fmt, + .hw_params = es7241_hw_params, + .set_sysclk = es7241_set_sysclk, +}; + +static struct snd_soc_dai_driver es7241_dai = { + .name = "es7241-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &es7241_dai_ops, +}; + +static const struct es7241_clock_mode es7241_modes[] = { + { + /* Single speed mode */ + .rate_min = 8000, + .rate_max = 50000, + .slv_mfs = (unsigned int[]) { 256, 384, 512, 768, 1024 }, + .slv_mfs_num = 5, + .mst_mfs = 256, + .mst_m0 = 0, + .mst_m1 = 0, + }, { + /* Double speed mode */ + .rate_min = 50000, + .rate_max = 100000, + .slv_mfs = (unsigned int[]) { 128, 192 }, + .slv_mfs_num = 2, + .mst_mfs = 128, + .mst_m0 = 1, + .mst_m1 = 0, + }, { + /* Quad speed mode */ + .rate_min = 100000, + .rate_max = 200000, + .slv_mfs = (unsigned int[]) { 64 }, + .slv_mfs_num = 1, + .mst_mfs = 64, + .mst_m0 = 0, + .mst_m1 = 1, + }, +}; + +static const struct es7241_chip es7241_chip = { + .modes = es7241_modes, + .mode_num = ARRAY_SIZE(es7241_modes), +}; + +static const struct snd_soc_dapm_widget es7241_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("AINL"), + SND_SOC_DAPM_INPUT("AINR"), + SND_SOC_DAPM_DAC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDDP", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDDD", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("VDDA", 0, 0), +}; + +static const struct snd_soc_dapm_route es7241_dapm_routes[] = { + { "ADC", NULL, "AINL", }, + { "ADC", NULL, "AINR", }, + { "ADC", NULL, "VDDA", }, + { "Capture", NULL, "VDDP", }, + { "Capture", NULL, "VDDD", }, +}; + +static const struct snd_soc_component_driver es7241_component_driver = { + .dapm_widgets = es7241_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es7241_dapm_widgets), + .dapm_routes = es7241_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es7241_dapm_routes), + .idle_bias_on = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static void es7241_parse_fmt(struct device *dev, struct es7241_data *priv) +{ + bool is_leftj; + + /* + * The format is given by a pull resistor on the SDOUT pin: + * pull-up for i2s, pull-down for left justified. + */ + is_leftj = of_property_read_bool(dev->of_node, + "everest,sdout-pull-down"); + if (is_leftj) + priv->fmt = SND_SOC_DAIFMT_LEFT_J; + else + priv->fmt = SND_SOC_DAIFMT_I2S; +} + +static int es7241_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct es7241_data *priv; + int err; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->chip = of_device_get_match_data(dev); + if (!priv->chip) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + es7241_parse_fmt(dev, priv); + + priv->reset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); + if (IS_ERR(priv->reset)) { + err = PTR_ERR(priv->reset); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'reset' gpio: %d", err); + return err; + } + + priv->m0 = devm_gpiod_get_optional(dev, "m0", GPIOD_OUT_LOW); + if (IS_ERR(priv->m0)) { + err = PTR_ERR(priv->m0); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'm0' gpio: %d", err); + return err; + } + + priv->m1 = devm_gpiod_get_optional(dev, "m1", GPIOD_OUT_LOW); + if (IS_ERR(priv->m1)) { + err = PTR_ERR(priv->m1); + if (err != -EPROBE_DEFER) + dev_err(dev, "Failed to get 'm1' gpio: %d", err); + return err; + } + + return devm_snd_soc_register_component(&pdev->dev, + &es7241_component_driver, + &es7241_dai, 1); +} + +#ifdef CONFIG_OF +static const struct of_device_id es7241_ids[] = { + { .compatible = "everest,es7241", .data = &es7241_chip }, + { } +}; +MODULE_DEVICE_TABLE(of, es7241_ids); +#endif + +static struct platform_driver es7241_driver = { + .driver = { + .name = "es7241", + .of_match_table = of_match_ptr(es7241_ids), + }, + .probe = es7241_probe, +}; + +module_platform_driver(es7241_driver); + +MODULE_DESCRIPTION("ASoC ES7241 audio codec driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From e0431de301cbb8e3915261dfff4d0b072738de69 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Mon, 2 Jul 2018 07:17:07 -0500 Subject: ASoC: pxa-ssp: mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index f8339bb01251..ff1e0bd8d407 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -470,6 +470,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; + /* fall through */ case SND_SOC_DAIFMT_DSP_B: sscr0 |= SSCR0_MOD | SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; -- cgit v1.2.3 From 30896d3619bd80486a3f8a75d62ea3b58fc61ad5 Mon Sep 17 00:00:00 2001 From: Daniel Kurtz Date: Mon, 2 Jul 2018 15:19:50 -0600 Subject: ASoC: AMD: Always stop ch2 first Commit 6b116dfb4633a ("ASoC: AMD: make channel 1 dma as circular") made both channels circular, so this comment and logic no longer applies. Always stop ch2 (the channel closest to the output) before ch1. This ensures that the downstream circular DMA channel does not continue to play/capture repeated samples after the upstream circular DMA channel has already stopped. Signed-off-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 17 ++--------------- 1 file changed, 2 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 3c3d398d0d0b..4665ae12e74e 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1067,21 +1067,8 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - /* For playback, non circular dma should be stopped first - * i.e Sysram to acp dma transfer channel(rtd->ch1) should be - * stopped before stopping cirular dma which is acp sram to i2s - * fifo dma transfer channel(rtd->ch2). Where as in Capture - * scenario, i2s fifo to acp sram dma channel(rtd->ch2) stopped - * first before stopping acp sram to sysram which is circular - * dma(rtd->ch1). - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - acp_dma_stop(rtd->acp_mmio, rtd->ch1); - ret = acp_dma_stop(rtd->acp_mmio, rtd->ch2); - } else { - acp_dma_stop(rtd->acp_mmio, rtd->ch2); - ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); - } + acp_dma_stop(rtd->acp_mmio, rtd->ch2); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); rtd->bytescount = 0; break; default: -- cgit v1.2.3 From 715cdce04487fb23d5c10693b3bc01309fea955a Mon Sep 17 00:00:00 2001 From: Daniel Kurtz Date: Mon, 2 Jul 2018 15:19:52 -0600 Subject: ASoC: AMD: Always subtract bytescount It is always correct to subtract out the starting bytescount value. Even in the case of 2^64 byte rollover (292 Million Years in the future @ 48000 Hz) the math still works out. Signed-off-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 4665ae12e74e..034fac3de037 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -995,8 +995,7 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd); - if (bytescount > rtd->bytescount) - bytescount -= rtd->bytescount; + bytescount -= rtd->bytescount; pos = do_div(bytescount, buffersize); return bytes_to_frames(runtime, pos); } -- cgit v1.2.3 From 55af49ac1b8627dfbfa2689af118d994d7a0ba1b Mon Sep 17 00:00:00 2001 From: Daniel Kurtz Date: Mon, 2 Jul 2018 15:19:53 -0600 Subject: ASoC: AMD: Fix Capture DMA channel names On capture, audio data is first copied from I2S to ACP memory, and then to SYSRAM. For each step the channel number increases, so the names in the driver were wrong. Signed-off-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 24 ++++++++++++------------ sound/soc/amd/acp.h | 8 ++++---- 2 files changed, 16 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 034fac3de037..df53412967e1 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -697,31 +697,31 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { + if ((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) != 0) { valid_irq = true; snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); - acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, + acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) != 0) { + if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; - acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) << 16, + acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { + if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & - BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, + BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) { + if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; acp_reg_write((intr_flag & - BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16, + BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } @@ -899,8 +899,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: rtd->pte_offset = ACP_ST_BT_CAPTURE_PTE_OFFSET; - rtd->ch1 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM; - rtd->ch2 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM; + rtd->ch1 = I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM; + rtd->ch2 = ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM; rtd->sram_bank = ACP_SRAM_BANK_4_ADDRESS; rtd->destination = FROM_BLUETOOTH; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; @@ -914,8 +914,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, case I2S_SP_INSTANCE: default: rtd->pte_offset = ACP_CAPTURE_PTE_OFFSET; - rtd->ch1 = ACP_TO_SYSRAM_CH_NUM; - rtd->ch2 = I2S_TO_ACP_DMA_CH_NUM; + rtd->ch1 = I2S_TO_ACP_DMA_CH_NUM; + rtd->ch2 = ACP_TO_SYSRAM_CH_NUM; switch (adata->asic_type) { case CHIP_STONEY: rtd->pte_offset = ACP_ST_CAPTURE_PTE_OFFSET; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 3190fdce6307..0a2240bff62e 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -74,16 +74,16 @@ #define ACP_TO_I2S_DMA_CH_NUM 13 /* Capture DMA channels */ -#define ACP_TO_SYSRAM_CH_NUM 14 -#define I2S_TO_ACP_DMA_CH_NUM 15 +#define I2S_TO_ACP_DMA_CH_NUM 14 +#define ACP_TO_SYSRAM_CH_NUM 15 /* Playback DMA Channels for I2S BT instance */ #define SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM 8 #define ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM 9 /* Capture DMA Channels for I2S BT Instance */ -#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 10 -#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 11 +#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 10 +#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 11 #define NUM_DSCRS_PER_CHANNEL 2 -- cgit v1.2.3 From 8c6b964eddd2c39a9796899b2be099ece1b6c6ca Mon Sep 17 00:00:00 2001 From: Daniel Kurtz Date: Mon, 2 Jul 2018 15:19:54 -0600 Subject: ASoC: AMD: Do not generate interrups for every captured sample On capture, audio data is first copied from I2S to ACP memory, and then from ACP to SYSRAM. The I2S_TO_ACP_DMA interrupt fires on every sample transferred from I2S to ACP memory. That is it fires ~48000 times per second when capturing @ 48 kHz. Since we don't do anything on this interrupt anyway, disable it to save quite a few unnecessary interrupts. The real "work" (calling snd_pcm_period_elapsed()) is done when transfer from ACP to SYSRAM is complete. Signed-off-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 15 --------------- 1 file changed, 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index df53412967e1..cd4d2520ac14 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -412,10 +412,8 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) switch (ch_num) { case ACP_TO_I2S_DMA_CH_NUM: case ACP_TO_SYSRAM_CH_NUM: - case I2S_TO_ACP_DMA_CH_NUM: case ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM: case ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM: - case I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM: dma_ctrl |= ACP_DMA_CNTL_0__DMAChIOCEn_MASK; break; default: @@ -704,12 +702,6 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { - valid_irq = true; - acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, - acp_mmio, mmACP_EXTERNAL_INTR_STAT); - } - if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); @@ -718,13 +710,6 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { - valid_irq = true; - acp_reg_write((intr_flag & - BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, - acp_mmio, mmACP_EXTERNAL_INTR_STAT); - } - if (valid_irq) return IRQ_HANDLED; else -- cgit v1.2.3 From 1a337a1e7885085d224583c766614e5945bde671 Mon Sep 17 00:00:00 2001 From: Daniel Kurtz Date: Mon, 2 Jul 2018 15:19:51 -0600 Subject: ASoC: AMD: Reset bytescount when starting transaction The pointer() callback gets its value by reading the I2S BYTE_COUNT register. This is a 64-bit runnning transaction counter. If a transaction was aborted in the middle of a sample buffer, the counter will stop counting on a number divisible by the buffer size. Since we actually use it as a pointer into an aligned buffer, however, we do want to ensure that it always starts at a number divisible by the buffer size when starting a transaction, hence we reset it whenever starting a transaction. To accomplish this, it wasn't necessary to zero bytescount at the termination of each transaction, so remove this unnecessary code. Signed-off-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index cd4d2520ac14..ab60129f4f26 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1013,7 +1013,6 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) { int ret; - u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -1024,9 +1023,7 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - bytescount = acp_get_byte_count(rtd); - if (rtd->bytescount == 0) - rtd->bytescount = bytescount; + rtd->bytescount = acp_get_byte_count(rtd); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { acp_dma_start(rtd->acp_mmio, rtd->ch1); acp_dma_start(rtd->acp_mmio, rtd->ch2); @@ -1053,7 +1050,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_SUSPEND: acp_dma_stop(rtd->acp_mmio, rtd->ch2); ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); - rtd->bytescount = 0; break; default: ret = -EINVAL; -- cgit v1.2.3 From df61f9f76609456efbc60d495b3235baf7d07691 Mon Sep 17 00:00:00 2001 From: Daniel Kurtz Date: Mon, 2 Jul 2018 15:19:55 -0600 Subject: ASoC: AMD: Simplify trigger handler Now that the I2S channel names are fixed, and DMA data flow order is consistent (ch1 then ch2), we can simplify channel start order: start the upstream channel and then the downstream channel for both playback and capture cases. Signed-off-by: Daniel Kurtz Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index ab60129f4f26..65c1033bd51c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1024,10 +1024,7 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: rtd->bytescount = acp_get_byte_count(rtd); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - acp_dma_start(rtd->acp_mmio, rtd->ch1); - acp_dma_start(rtd->acp_mmio, rtd->ch2); - } else { + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { if (rtd->capture_channel == CAP_CHANNEL0) { acp_dma_cap_channel_disable(rtd->acp_mmio, CAP_CHANNEL1); @@ -1040,9 +1037,9 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_cap_channel_enable(rtd->acp_mmio, CAP_CHANNEL1); } - acp_dma_start(rtd->acp_mmio, rtd->ch2); - acp_dma_start(rtd->acp_mmio, rtd->ch1); } + acp_dma_start(rtd->acp_mmio, rtd->ch1); + acp_dma_start(rtd->acp_mmio, rtd->ch2); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: -- cgit v1.2.3 From 30ddfffd10b73d5d960650ea70b33a8ee0562679 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 3 Jul 2018 17:05:59 +0200 Subject: ASoC: es7134: correct required power supplies Drop AVDD in favor of PVDD to match the names used in the datasheet and only claim PVDD on the es7154. The es7134 and es7144 don't have a separate supply for the digital I/O. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/es7134.c | 44 +++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 41 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index 5ad59c38fed1..80f2936cefed 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -35,6 +35,10 @@ struct es7134_clock_mode { struct es7134_chip { const struct es7134_clock_mode *modes; unsigned int mode_num; + const struct snd_soc_dapm_widget *extra_widgets; + unsigned int extra_widget_num; + const struct snd_soc_dapm_route *extra_routes; + unsigned int extra_route_num; }; struct es7134_data { @@ -110,6 +114,34 @@ static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } +static int es7134_component_probe(struct snd_soc_component *c) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(c); + struct es7134_data *priv = snd_soc_component_get_drvdata(c); + const struct es7134_chip *chip = priv->chip; + int ret; + + if (chip->extra_widget_num) { + ret = snd_soc_dapm_new_controls(dapm, chip->extra_widgets, + chip->extra_widget_num); + if (ret) { + dev_err(c->dev, "failed to add extra widgets\n"); + return ret; + } + } + + if (chip->extra_route_num) { + ret = snd_soc_dapm_add_routes(dapm, chip->extra_routes, + chip->extra_route_num); + if (ret) { + dev_err(c->dev, "failed to add extra routes\n"); + return ret; + } + } + + return 0; +} + static const struct snd_soc_dai_ops es7134_dai_ops = { .set_fmt = es7134_set_fmt, .hw_params = es7134_hw_params, @@ -158,9 +190,16 @@ static const struct es7134_clock_mode es7134_modes[] = { }, }; +/* Digital I/O are also supplied by VDD on the es7134 */ +static const struct snd_soc_dapm_route es7134_extra_routes[] = { + { "Playback", NULL, "VDD", } +}; + static const struct es7134_chip es7134_chip = { .modes = es7134_modes, .mode_num = ARRAY_SIZE(es7134_modes), + .extra_routes = es7134_extra_routes, + .extra_route_num = ARRAY_SIZE(es7134_extra_routes), }; static const struct snd_soc_dapm_widget es7134_dapm_widgets[] = { @@ -168,17 +207,16 @@ static const struct snd_soc_dapm_widget es7134_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("AOUTR"), SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("VDD", 0, 0), - SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 0, 0), }; static const struct snd_soc_dapm_route es7134_dapm_routes[] = { { "AOUTL", NULL, "DAC" }, { "AOUTR", NULL, "DAC" }, - { "Playback", NULL, "VDD" }, - { "DAC", NULL, "AVDD" }, + { "DAC", NULL, "VDD" }, }; static const struct snd_soc_component_driver es7134_component_driver = { + .probe = es7134_component_probe, .dapm_widgets = es7134_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(es7134_dapm_widgets), .dapm_routes = es7134_dapm_routes, -- cgit v1.2.3 From 563c263248ff37dcd743549a0c0932fe2bf83980 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 3 Jul 2018 17:06:00 +0200 Subject: ASoC: es7134: add support for the es7154 Add support for the es7154 which is basically an es7134 with an embedded power amplifier and lower maximum sample rate Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/es7134.c | 60 ++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 59 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index 80f2936cefed..6d7bca7b78ca 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -33,6 +33,7 @@ struct es7134_clock_mode { }; struct es7134_chip { + struct snd_soc_dai_driver *dai_drv; const struct es7134_clock_mode *modes; unsigned int mode_num; const struct snd_soc_dapm_widget *extra_widgets; @@ -196,6 +197,7 @@ static const struct snd_soc_dapm_route es7134_extra_routes[] = { }; static const struct es7134_chip es7134_chip = { + .dai_drv = &es7134_dai, .modes = es7134_modes, .mode_num = ARRAY_SIZE(es7134_modes), .extra_routes = es7134_extra_routes, @@ -227,6 +229,61 @@ static const struct snd_soc_component_driver es7134_component_driver = { .non_legacy_dai_naming = 1, }; +static struct snd_soc_dai_driver es7154_dai = { + .name = "es7154-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S18_3LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &es7134_dai_ops, +}; + +static const struct es7134_clock_mode es7154_modes[] = { + { + /* Single speed mode */ + .rate_min = 8000, + .rate_max = 50000, + .mclk_fs = (unsigned int[]) { 32, 64, 128, 192, 256, + 384, 512, 768, 1024 }, + .mclk_fs_num = 9, + }, { + /* Double speed mode */ + .rate_min = 84000, + .rate_max = 100000, + .mclk_fs = (unsigned int[]) { 128, 192, 256, 384, 512, + 768, 1024}, + .mclk_fs_num = 7, + } +}; + +/* Es7154 has a separate supply for digital I/O */ +static const struct snd_soc_dapm_widget es7154_extra_widgets[] = { + SND_SOC_DAPM_REGULATOR_SUPPLY("PVDD", 0, 0), +}; + +static const struct snd_soc_dapm_route es7154_extra_routes[] = { + { "Playback", NULL, "PVDD", } +}; + +static const struct es7134_chip es7154_chip = { + .dai_drv = &es7154_dai, + .modes = es7154_modes, + .mode_num = ARRAY_SIZE(es7154_modes), + .extra_routes = es7154_extra_routes, + .extra_route_num = ARRAY_SIZE(es7154_extra_routes), + .extra_widgets = es7154_extra_widgets, + .extra_widget_num = ARRAY_SIZE(es7154_extra_widgets), +}; + static int es7134_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; @@ -245,13 +302,14 @@ static int es7134_probe(struct platform_device *pdev) return devm_snd_soc_register_component(&pdev->dev, &es7134_component_driver, - &es7134_dai, 1); + priv->chip->dai_drv, 1); } #ifdef CONFIG_OF static const struct of_device_id es7134_ids[] = { { .compatible = "everest,es7134", .data = &es7134_chip }, { .compatible = "everest,es7144", .data = &es7134_chip }, + { .compatible = "everest,es7154", .data = &es7154_chip }, { } }; MODULE_DEVICE_TABLE(of, es7134_ids); -- cgit v1.2.3 From 73ad0df572901e03fc703b6f114e4442291f45c2 Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu Date: Tue, 3 Jul 2018 17:56:30 +0300 Subject: ASoC: atmel-i2s: Remove unnecessary audio PLL clock (aclk) The generated clock (gclk) driver is able to set aclk as its parent and change its rate alone, if needed. This means that our driver no longer needs to configure aclk and we can let gclk select and configure its clock source. Signed-off-by: Codrin Ciubotariu Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-i2s.c | 46 +++++++-------------------------------------- 1 file changed, 7 insertions(+), 39 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 5d3b5af9fd92..d88c1d995036 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -206,7 +206,6 @@ struct atmel_i2s_dev { struct regmap *regmap; struct clk *pclk; struct clk *gclk; - struct clk *aclk; struct snd_dmaengine_dai_dma_data playback; struct snd_dmaengine_dai_dma_data capture; unsigned int fmt; @@ -303,7 +302,7 @@ static int atmel_i2s_get_gck_param(struct atmel_i2s_dev *dev, int fs) { int i, best; - if (!dev->gclk || !dev->aclk) { + if (!dev->gclk) { dev_err(dev->dev, "cannot generate the I2S Master Clock\n"); return -EINVAL; } @@ -421,7 +420,7 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev, bool enabled) { unsigned int mr, mr_mask; - unsigned long aclk_rate; + unsigned long gclk_rate; int ret; mr = 0; @@ -445,35 +444,18 @@ static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev, /* Disable/unprepare the PMC generated clock. */ clk_disable_unprepare(dev->gclk); - /* Disable/unprepare the PLL audio clock. */ - clk_disable_unprepare(dev->aclk); return 0; } if (!dev->gck_param) return -EINVAL; - aclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1); + gclk_rate = dev->gck_param->mck * (dev->gck_param->imckdiv + 1); - /* Fist change the PLL audio clock frequency ... */ - ret = clk_set_rate(dev->aclk, aclk_rate); + ret = clk_set_rate(dev->gclk, gclk_rate); if (ret) return ret; - /* - * ... then set the PMC generated clock rate to the very same frequency - * to set the gclk parent to aclk. - */ - ret = clk_set_rate(dev->gclk, aclk_rate); - if (ret) - return ret; - - /* Prepare and enable the PLL audio clock first ... */ - ret = clk_prepare_enable(dev->aclk); - if (ret) - return ret; - - /* ... then prepare and enable the PMC generated clock. */ ret = clk_prepare_enable(dev->gclk); if (ret) return ret; @@ -668,28 +650,14 @@ static int atmel_i2s_probe(struct platform_device *pdev) return err; } - /* Get audio clocks to generate the I2S Master Clock (I2S_MCK) */ - dev->aclk = devm_clk_get(&pdev->dev, "aclk"); + /* Get audio clock to generate the I2S Master Clock (I2S_MCK) */ dev->gclk = devm_clk_get(&pdev->dev, "gclk"); - if (IS_ERR(dev->aclk) && IS_ERR(dev->gclk)) { - if (PTR_ERR(dev->aclk) == -EPROBE_DEFER || - PTR_ERR(dev->gclk) == -EPROBE_DEFER) + if (IS_ERR(dev->gclk)) { + if (PTR_ERR(dev->gclk) == -EPROBE_DEFER) return -EPROBE_DEFER; /* Master Mode not supported */ - dev->aclk = NULL; dev->gclk = NULL; - } else if (IS_ERR(dev->gclk)) { - err = PTR_ERR(dev->gclk); - dev_err(&pdev->dev, - "failed to get the PMC generated clock: %d\n", err); - return err; - } else if (IS_ERR(dev->aclk)) { - err = PTR_ERR(dev->aclk); - dev_err(&pdev->dev, - "failed to get the PLL audio clock: %d\n", err); - return err; } - dev->dev = &pdev->dev; dev->regmap = regmap; platform_set_drvdata(pdev, dev); -- cgit v1.2.3 From a655de808cbde6c58b3298e704d786b53f59fb5d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 2 Jul 2018 16:59:54 +0100 Subject: ASoC: core: Allow topology to override machine driver FE DAI link config. Machine drivers statically define a number of DAI links that currently cannot be changed or removed by topology. This means PCMs and platform components cannot be changed by topology at runtime AND machine drivers are tightly coupled to topology. This patch allows topology to override the machine driver DAI link config in order to reuse machine drivers with different topologies and platform components. The patch supports :- 1) create new FE PCMs with a topology defined PCM ID. 2) destroy existing static FE PCMs 3) change the platform component driver. 4) assign any new HW params fixups. 5) assign a new card name prefix to differentiate this topology to userspace. The patch requires no changes to the machine drivers, but does add some platform component flags that the platform component driver can assign before loading topologies. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc.h | 13 +++++++ sound/soc/soc-core.c | 101 +++++++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/soc-pcm.c | 12 ++++++ 3 files changed, 123 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index 16f0bf10cc42..870ba6b64817 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -803,6 +803,14 @@ struct snd_soc_component_driver { unsigned int use_pmdown_time:1; /* care pmdown_time at stop */ unsigned int endianness:1; unsigned int non_legacy_dai_naming:1; + + /* this component uses topology and ignore machine driver FEs */ + const char *ignore_machine; + const char *topology_name_prefix; + int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); + bool use_dai_pcm_id; /* use the DAI link PCM ID as PCM device number */ + int be_pcm_base; /* base device ID for all BE PCMs */ }; struct snd_soc_component { @@ -960,6 +968,9 @@ struct snd_soc_dai_link { /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; + /* Do not create a PCM for this DAI link (Backend link) */ + unsigned int ignore:1; + struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; @@ -999,6 +1010,7 @@ struct snd_soc_card { const char *long_name; const char *driver_name; char dmi_longname[80]; + char topology_shortname[32]; struct device *dev; struct snd_card *snd_card; @@ -1008,6 +1020,7 @@ struct snd_soc_card { struct mutex dapm_mutex; bool instantiated; + bool topology_shortname_created; int (*probe)(struct snd_soc_card *card); int (*late_probe)(struct snd_soc_card *card); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 68b08781c832..00bd58d167dd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -847,6 +847,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, const char *platform_name; int i; + if (dai_link->ignore) + return 0; + dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); if (soc_is_dai_link_bound(card, dai_link)) { @@ -1456,7 +1459,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + int i, ret, num; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, rtd->num, order); @@ -1502,9 +1507,28 @@ static int soc_probe_link_dais(struct snd_soc_card *card, soc_dpcm_debugfs_add(rtd); #endif + num = rtd->num; + + /* + * most drivers will register their PCMs using DAI link ordering but + * topology based drivers can use the DAI link id field to set PCM + * device number and then use rtd + a base offset of the BEs. + */ + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (!component->driver->use_dai_pcm_id) + continue; + + if (rtd->dai_link->no_pcm) + num += component->driver->be_pcm_base; + else + num = rtd->dai_link->id; + } + if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = cpu_dai->driver->compress_new(rtd, rtd->num); + ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); @@ -1514,7 +1538,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, if (!dai_link->params) { /* create the pcm */ - ret = soc_new_pcm(rtd, rtd->num); + ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); @@ -1841,6 +1865,74 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name); #endif /* CONFIG_DMI */ +static void soc_check_tplg_fes(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + const struct snd_soc_component_driver *comp_drv; + struct snd_soc_dai_link *dai_link; + int i; + + list_for_each_entry(component, &component_list, list) { + + /* does this component override FEs ? */ + if (!component->driver->ignore_machine) + continue; + + /* for this machine ? */ + if (strcmp(component->driver->ignore_machine, + card->dev->driver->name)) + continue; + + /* machine matches, so override the rtd data */ + for (i = 0; i < card->num_links; i++) { + + dai_link = &card->dai_link[i]; + + /* ignore this FE */ + if (dai_link->dynamic) { + dai_link->ignore = true; + continue; + } + + dev_info(card->dev, "info: override FE DAI link %s\n", + card->dai_link[i].name); + + /* override platform component */ + dai_link->platform_name = component->name; + + /* convert non BE into BE */ + dai_link->no_pcm = 1; + + /* override any BE fixups */ + dai_link->be_hw_params_fixup = + component->driver->be_hw_params_fixup; + + /* most BE links don't set stream name, so set it to + * dai link name if it's NULL to help bind widgets. + */ + if (!dai_link->stream_name) + dai_link->stream_name = dai_link->name; + } + + /* Inform userspace we are using alternate topology */ + if (component->driver->topology_name_prefix) { + + /* topology shortname created ? */ + if (!card->topology_shortname_created) { + comp_drv = component->driver; + + snprintf(card->topology_shortname, 32, "%s-%s", + comp_drv->topology_name_prefix, + card->name); + card->topology_shortname_created = true; + } + + /* use topology shortname */ + card->name = card->topology_shortname; + } + } +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -1850,6 +1942,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + /* check whether any platform is ignore machine FE and using topology */ + soc_check_tplg_fes(card); + /* bind DAIs */ for (i = 0; i < card->num_links; i++) { ret = soc_bind_dai_link(card, &card->dai_link[i]); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index c2a31b51da4f..b7e67b871c0c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -859,8 +859,20 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret; + /* perform any topology hw_params fixups before DAI */ + if (rtd->dai_link->be_hw_params_fixup) { + ret = rtd->dai_link->be_hw_params_fixup(rtd, params); + if (ret < 0) { + dev_err(rtd->dev, + "ASoC: hw_params topology fixup failed %d\n", + ret); + return ret; + } + } + if (dai->driver->ops->hw_params) { ret = dai->driver->ops->hw_params(substream, params, dai); if (ret < 0) { -- cgit v1.2.3 From 110743189c863e96dc08a581d56c50b965870a3f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:43 +0100 Subject: ASoC: qdsp6: q6afe-dai: do not close port if its not opened afe ports are open as part of prepare, so for use cases like "aplay sample.wav" were sample.wav is not present. This would call port close eventhough port was never opened. DSP would return errors for such use cases. Avoid doing this by checking the port state. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 5002dd05bf27..a373ca5523ff 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -315,6 +315,9 @@ static void q6afe_dai_shutdown(struct snd_pcm_substream *substream, struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); int rc; + if (!dai_data->is_port_started[dai->id]) + return; + rc = q6afe_port_stop(dai_data->port[dai->id]); if (rc < 0) dev_err(dai->dev, "fail to close AFE port (%d)\n", rc); -- cgit v1.2.3 From 5dffc1752cabde6396fca28ff8343febfa524512 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:44 +0100 Subject: ASoC: qdsp6: q6asm-dai: do not close port if its not opened asm ports are open as part of prepare, so for use cases like "aplay sample.wav" were sample.wav is not present. This would call port close eventhough port was never opened. DSP would return errors for such use cases. Avoid doing this by checking the port state. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 349c6a883c63..360936703b3d 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -390,7 +390,9 @@ static int q6asm_dai_close(struct snd_pcm_substream *substream) struct q6asm_dai_rtd *prtd = runtime->private_data; if (prtd->audio_client) { - q6asm_cmd(prtd->audio_client, CMD_CLOSE); + if (prtd->state) + q6asm_cmd(prtd->audio_client, CMD_CLOSE); + q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); q6asm_audio_client_free(prtd->audio_client); -- cgit v1.2.3 From da13ed1d80fe6a4d95043aaf2e0aff292ade5708 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 4 Jul 2018 09:28:28 -0500 Subject: ASoC: nau8825: use 64-bit arithmetic instead of 32-bit Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: 256 * fs * 2 * mclk_src_scaling[i].param Addresses-Coverity-ID: 1339616 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index dc6ea4987b7d..b9fed99d8b5e 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2016,7 +2016,7 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; -- cgit v1.2.3 From dae35d1f4f7dab9ccef20037df91c43e680bad0f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jul 2018 16:01:43 +0200 Subject: ASoC: davinci: Use snd_pcm_stop_xrun() helper Replace open-codes with the standard snd_pcm_stop_xrun() helper. It simplifies codes a lot. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 47c0c821d325..f70db8412c7c 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -320,12 +320,8 @@ static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data) handled_mask |= XUNDRN; substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]; - if (substream) { - snd_pcm_stream_lock_irq(substream); - if (snd_pcm_running(substream)) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irq(substream); - } + if (substream) + snd_pcm_stop_xrun(substream); } if (!handled_mask) @@ -355,12 +351,8 @@ static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) handled_mask |= ROVRN; substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]; - if (substream) { - snd_pcm_stream_lock_irq(substream); - if (snd_pcm_running(substream)) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irq(substream); - } + if (substream) + snd_pcm_stop_xrun(substream); } if (!handled_mask) -- cgit v1.2.3 From 1a42e7e3aff1aa4789378020318dff7432317d25 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jul 2018 16:01:44 +0200 Subject: ASoC: qcom: Use snd_pcm_stop_xrun() helper The XRUN trigger from the driver should be done via snd_pcm_stop_xrun(). It fixes the missing stream locking as a gratis, too. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 31fe78aa207f..d07271ea4c45 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -458,7 +458,7 @@ static irqreturn_t lpass_dma_interrupt_handler( return IRQ_NONE; } dev_warn(soc_runtime->dev, "xrun warning\n"); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(substream); ret = IRQ_HANDLED; } -- cgit v1.2.3 From dc865fb9e7c2251c9585ff6a7bf185d499db13e4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jul 2018 16:01:45 +0200 Subject: ASoC: sti: Use snd_pcm_stop_xrun() helper The XRUN trigger from the driver should be done via snd_pcm_stop_xrun(). It fixes the missing stream locking as a gratis, too. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sti/uniperif_player.c | 6 +++--- sound/soc/sti/uniperif_reader.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index d8b6936e544e..313dab2857ef 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -91,7 +91,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player); /* Stop the player */ - snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(player->substream); } ret = IRQ_HANDLED; @@ -105,7 +105,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player); /* Stop the player */ - snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(player->substream); ret = IRQ_HANDLED; } @@ -138,7 +138,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) dev_err(player->dev, "Underflow recovery failed\n"); /* Stop the player */ - snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(player->substream); ret = IRQ_HANDLED; } diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index ee0055e60852..7b63d35ef428 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -65,7 +65,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) { dev_err(reader->dev, "FIFO error detected\n"); - snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stop_xrun(reader->substream); ret = IRQ_HANDLED; } -- cgit v1.2.3 From b1625fbb3b87affbedf14545b65d69ff182a0611 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jul 2018 16:01:46 +0200 Subject: ASoC: stm32: Use snd_pcm_stop_xrun() helper The XRUN trigger from the driver should be done via snd_pcm_stop_xrun(). It simplifies the locking as well. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index c4f15ea14197..06fba9650ac4 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -300,11 +300,8 @@ static irqreturn_t stm32_sai_isr(int irq, void *devid) status = SNDRV_PCM_STATE_XRUN; } - if (status != SNDRV_PCM_STATE_RUNNING) { - snd_pcm_stream_lock(sai->substream); - snd_pcm_stop(sai->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(sai->substream); - } + if (status != SNDRV_PCM_STATE_RUNNING) + snd_pcm_stop_xrun(sai->substream); return IRQ_HANDLED; } -- cgit v1.2.3 From 25090bc3f36cc3c171ec020dcc89c71db6bd0a67 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:39 +0100 Subject: ASoC: qdsp6: q6afe: Add missing slimbus capture ports Existing code already has support for SLIMbus TX and RX, only thing that was missing from TX side was mapping between virtual to actual DSP port ids. This patch adds those mappings. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 671743453fbb..000775b4bba8 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -514,6 +514,20 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = { SLIMBUS_5_RX, 1, 1}, [SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX, SLIMBUS_6_RX, 1, 1}, + [SLIMBUS_0_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX, + SLIMBUS_0_TX, 0, 1}, + [SLIMBUS_1_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX, + SLIMBUS_1_TX, 0, 1}, + [SLIMBUS_2_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX, + SLIMBUS_2_TX, 0, 1}, + [SLIMBUS_3_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX, + SLIMBUS_3_TX, 0, 1}, + [SLIMBUS_4_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX, + SLIMBUS_4_TX, 0, 1}, + [SLIMBUS_5_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX, + SLIMBUS_5_TX, 0, 1}, + [SLIMBUS_6_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX, + SLIMBUS_6_TX, 0, 1}, [PRIMARY_MI2S_RX] = { AFE_PORT_ID_PRIMARY_MI2S_RX, PRIMARY_MI2S_RX, 1, 1}, [PRIMARY_MI2S_TX] = { AFE_PORT_ID_PRIMARY_MI2S_TX, @@ -1372,6 +1386,13 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) case AFE_PORT_ID_MULTICHAN_HDMI_RX: cfg_type = AFE_PARAM_ID_HDMI_CONFIG; break; + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX: + case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX: case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX: case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX: case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX: -- cgit v1.2.3 From f03d6b1b4d2460a749fb2826aa71e15a66104a88 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:40 +0100 Subject: ASoC: qdsp6: q6afe-dai: add support to slim tx dais This patch adds support to SLIMbus TX dais in AFE module. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 175 +++++++++++++++++++++++++++++++++++---- 1 file changed, 161 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 1d2e5013c121..8dd3683eb367 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -382,23 +382,31 @@ static int q6slim_set_channel_map(struct snd_soc_dai *dai, struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id]; int i; - if (!rx_slot) { - pr_err("%s: rx slot not found\n", __func__); - return -EINVAL; - } + if (dai->id & 0x1) { + /* TX */ + if (!tx_slot) { + pr_err("%s: tx slot not found\n", __func__); + return -EINVAL; + } - for (i = 0; i < rx_num; i++) { - pcfg->slim.ch_mapping[i] = rx_slot[i]; - pr_debug("%s: find number of channels[%d] ch[%d]\n", - __func__, i, rx_slot[i]); - } + for (i = 0; i < tx_num; i++) + pcfg->slim.ch_mapping[i] = tx_slot[i]; + + pcfg->slim.num_channels = tx_num; + + + } else { + if (!rx_slot) { + pr_err("%s: rx slot not found\n", __func__); + return -EINVAL; + } - pcfg->slim.num_channels = rx_num; + for (i = 0; i < rx_num; i++) + pcfg->slim.ch_mapping[i] = rx_slot[i]; - pr_debug("%s: SLIMBUS_%d_RX cnt[%d] ch[%d %d]\n", __func__, - (dai->id - SLIMBUS_0_RX) / 2, rx_num, - pcfg->slim.ch_mapping[0], - pcfg->slim.ch_mapping[1]); + pcfg->slim.num_channels = rx_num; + + } return 0; } @@ -443,6 +451,14 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"Slimbus5 Playback", NULL, "SLIMBUS_5_RX"}, {"Slimbus6 Playback", NULL, "SLIMBUS_6_RX"}, + {"SLIMBUS_0_TX", NULL, "Slimbus Capture"}, + {"SLIMBUS_1_TX", NULL, "Slimbus1 Capture"}, + {"SLIMBUS_2_TX", NULL, "Slimbus2 Capture"}, + {"SLIMBUS_3_TX", NULL, "Slimbus3 Capture"}, + {"SLIMBUS_4_TX", NULL, "Slimbus4 Capture"}, + {"SLIMBUS_5_TX", NULL, "Slimbus5 Capture"}, + {"SLIMBUS_6_TX", NULL, "Slimbus6 Capture"}, + {"Primary MI2S Playback", NULL, "PRI_MI2S_RX"}, {"Secondary MI2S Playback", NULL, "SEC_MI2S_RX"}, {"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"}, @@ -636,6 +652,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .rate_min = 8000, .rate_max = 192000, }, + }, { + .name = "SLIMBUS_0_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_0_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus1 Playback", @@ -654,6 +688,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_1_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + }, { + .name = "SLIMBUS_1_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_1_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus1 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus2 Playback", @@ -672,6 +724,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_2_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_2_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_2_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus2 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus3 Playback", @@ -690,6 +761,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_3_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_3_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_3_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus3 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus4 Playback", @@ -708,6 +798,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_4_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_4_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_4_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus4 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus5 Playback", @@ -726,6 +835,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_5_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_5_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_5_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus5 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Slimbus6 Playback", @@ -744,6 +872,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .id = SLIMBUS_6_RX, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + + }, { + .name = "SLIMBUS_6_TX", + .ops = &q6slim_ops, + .id = SLIMBUS_6_TX, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + .capture = { + .stream_name = "Slimbus6 Capture", + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, }, { .playback = { .stream_name = "Primary MI2S Playback", -- cgit v1.2.3 From 9191ffe2d212f64aa2ec311f4294ba7066d1f8a1 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:41 +0100 Subject: ASoC: qdsp6: q6routing: add slim rx routings This patch add routings mixer controls for slim rx ports. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index c80fdbc2442e..35269b492761 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -68,6 +68,13 @@ { mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" }, \ { mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" }, \ { mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" }, \ + { mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" }, \ + { mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" }, \ + { mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" }, \ + { mix_name, "SLIMBUS_3_TX", "SLIMBUS_3_TX" }, \ + { mix_name, "SLIMBUS_4_TX", "SLIMBUS_4_TX" }, \ + { mix_name, "SLIMBUS_5_TX", "SLIMBUS_5_TX" }, \ + { mix_name, "SLIMBUS_6_TX", "SLIMBUS_6_TX" }, \ { mix_name, "PRIMARY_TDM_TX_0", "PRIMARY_TDM_TX_0"}, \ { mix_name, "PRIMARY_TDM_TX_1", "PRIMARY_TDM_TX_1"}, \ { mix_name, "PRIMARY_TDM_TX_2", "PRIMARY_TDM_TX_2"}, \ @@ -122,6 +129,27 @@ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_1_TX", SLIMBUS_1_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_2_TX", SLIMBUS_2_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_3_TX", SLIMBUS_3_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_4_TX", SLIMBUS_4_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_5_TX", SLIMBUS_5_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SLIMBUS_6_TX", SLIMBUS_6_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ SOC_SINGLE_EXT("PRIMARY_TDM_TX_0", PRIMARY_TDM_TX_0, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ -- cgit v1.2.3 From f1478a1476d45c5d7b070c726f46a7b581d3103c Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:42 +0100 Subject: ASoC: qdsp6: q6afe-dai: Do not overwrite slim dai num_channels num_channels for slim dais are aready set int set_channel_map, do not overwrite them in hw_params. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 8dd3683eb367..074582afda85 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -81,7 +81,6 @@ static int q6slim_hw_params(struct snd_pcm_substream *substream, struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim; - slim->num_channels = params_channels(params); slim->sample_rate = params_rate(params); switch (params_format(params)) { -- cgit v1.2.3 From b999a7a9e72bd2d37b5d03772cedfc4dd45875bf Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 4 Jul 2018 09:18:33 -0500 Subject: ASoC: fsl_spdif: Use 64-bit arithmetic instead of 32-bit Add suffix ULL to constant 64 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: rate[index] * txclk_df * 64 Addresses-Coverity-ID: 1222129 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 9b59d87b61bf..740b90df44bb 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1118,7 +1118,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) { for (txclk_df = 1; txclk_df <= 128; txclk_df++) { - rate_ideal = rate[index] * txclk_df * 64; + rate_ideal = rate[index] * txclk_df * 64ULL; if (round) rate_actual = clk_round_rate(clk, rate_ideal); else -- cgit v1.2.3 From 74b37e299f038c910dc728d736e3f071ba0ead2a Mon Sep 17 00:00:00 2001 From: Andrew Gabbasov Date: Thu, 5 Jul 2018 11:20:04 +0900 Subject: ASoC: rsnd: cmd: Add missing newline to debug message To comply with the style of all kernel messages, add newline to the end of every message. Fixes: 70fb10529f61 ("ASoC: rsnd: add MIX (Mixer) support") Signed-off-by: Andrew Gabbasov Signed-off-by: Jiada Wang Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/cmd.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index d8043ad33540..cc191cd5fb82 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -86,7 +86,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, cmd_case[rsnd_mod_id(src)] << 16; } - dev_dbg(dev, "ctu/mix path = 0x%08x", data); + dev_dbg(dev, "ctu/mix path = 0x%08x\n", data); rsnd_mod_write(mod, CMD_ROUTE_SLCT, data); rsnd_mod_write(mod, CMD_BUSIF_MODE, rsnd_get_busif_shift(io, mod) | 1); -- cgit v1.2.3 From 90eb6b59d311e6facd040124cb5b659a865125b8 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 2 Jul 2018 17:11:00 +0200 Subject: ASoC: pxa-ssp: add support for an external clock in devicetree Allow setting a clock called 'extclk' in the device of the ssp-dai device. If specified, this clock will be set to the mclk rate from the DAI's .set_sysclk() callback. The DAI will also configure itself to use that external clock. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/mrvl,pxa-ssp.txt | 8 +++++++ sound/soc/pxa/pxa-ssp.c | 25 ++++++++++++++++++++++ 2 files changed, 33 insertions(+) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt index 74c9ba6c2823..93b982e9419f 100644 --- a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt +++ b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt @@ -5,6 +5,14 @@ Required properties: compatible Must be "mrvl,pxa-ssp-dai" port A phandle reference to a PXA ssp upstream device +Optional properties: + + clock-names + clocks Through "clock-names" and "clocks", external clocks + can be configured. If a clock names "extclk" exists, + it will be set to the mclk rate of the audio stream + and be used as clock provider of the DAI. + Example: /* upstream device */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index ff1e0bd8d407..69033e1a84e6 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -41,6 +41,7 @@ */ struct ssp_priv { struct ssp_device *ssp; + struct clk *extclk; unsigned long ssp_clk; unsigned int sysclk; unsigned int dai_fmt; @@ -205,6 +206,21 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + if (priv->extclk) { + int ret; + + /* + * For DT based boards, if an extclk is given, use it + * here and configure PXA_SSP_CLK_EXT. + */ + + ret = clk_set_rate(priv->extclk, freq); + if (ret < 0) + return ret; + + clk_id = PXA_SSP_CLK_EXT; + } + dev_dbg(&ssp->pdev->dev, "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", cpu_dai->id, clk_id, freq); @@ -774,6 +790,15 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) ret = -ENODEV; goto err_priv; } + + priv->extclk = devm_clk_get(dev, "extclk"); + if (IS_ERR(priv->extclk)) { + ret = PTR_ERR(priv->extclk); + if (ret == -EPROBE_DEFER) + return ret; + + priv->extclk = NULL; + } } else { priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); if (priv->ssp == NULL) { -- cgit v1.2.3 From f7ddff54d0a0f068442414b48bec7f22aa777de7 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Thu, 5 Jul 2018 08:06:17 -0500 Subject: ASoC: nau8824: use 64-bit arithmetic instead of 32-bit Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: 256 * fs * 2 * mclk_src_scaling[i].param Addresses-Coverity-ID: 1432039 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/nau8824.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 6bd14453f06e..468d5143e2c4 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1274,7 +1274,7 @@ static int nau8824_calc_fll_param(unsigned int fll_in, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; -- cgit v1.2.3 From 8db339d66774821091f73bd0e57c8a7511c5988e Mon Sep 17 00:00:00 2001 From: benjamin.gaignard@linaro.org Date: Fri, 6 Jul 2018 15:07:03 +0200 Subject: ASoC: stm32: replace "%p" with "%pK" The format specifier "%p" can leak kernel addresses. Use "%pK" instead. Signed-off-by: Benjamin Gaignard Signed-off-by: Mark Brown --- sound/soc/stm/stm32_adfsdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index db73fef3e500..0e9373064032 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -149,7 +149,7 @@ static int stm32_afsdm_pcm_cb(const void *data, size_t size, void *private) unsigned int old_pos = priv->pos; unsigned int cur_size = size; - dev_dbg(rtd->dev, "%s: buff_add :%p, pos = %d, size = %zu\n", + dev_dbg(rtd->dev, "%s: buff_add :%pK, pos = %d, size = %zu\n", __func__, &pcm_buff[priv->pos], priv->pos, size); if ((priv->pos + size) > buff_size) { -- cgit v1.2.3 From 9d1310daedae494874fe36a995172e64b06e29a4 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Fri, 6 Jul 2018 15:30:48 +0200 Subject: ASoC: pxa: make SND_PXA_SOC_SSP depend on PLAT_PXA For the moment, we can't enable CONFIG_SND_PXA_SOC_SSP unless we are building for ARM PXA or MMP: WARNING: unmet direct dependencies detected for PXA_SSP Depends on [n]: PLAT_PXA [=n] Selected by [y]: - SND_PXA_SOC_SSP [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] This adds an explicit dependency for it. Fixes: 0a94cf345740 ("ASoC: pxa: make SND_PXA2XX_SOC_I2S selectable") Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 2fc02c227f69..776e148b0aa2 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -34,6 +34,7 @@ config SND_PXA2XX_SOC_I2S config SND_PXA_SOC_SSP tristate "Soc Audio via PXA2xx/PXA3xx SSP ports" + depends on PLAT_PXA select PXA_SSP select SND_PXA2XX_LIB -- cgit v1.2.3 From 5bea327962fa296efd16f2d3369dd339ddd7ce6f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 6 Jul 2018 16:19:14 +0300 Subject: ASoC: adau171x1: Connect playback DAI to the DSP The playback DAI is connected to the DSP and the DSP might be sourcing signals from the playback stream. Add a DAPM route between the two to make sure that the playback DAI is powered up, when the DSP is active. Signed-off-by: Lars-Peter Clausen Signed-off-by: Alexandru Ardelean Signed-off-by: Mark Brown --- sound/soc/codecs/adau17x1.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index ae41edd1c406..57169b8ff14e 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -299,6 +299,7 @@ static const struct snd_soc_dapm_route adau17x1_dsp_dapm_routes[] = { { "DSP", NULL, "Left Decimator" }, { "DSP", NULL, "Right Decimator" }, + { "DSP", NULL, "Playback" }, }; static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { -- cgit v1.2.3 From 81583afe790c5bd86300537783b23f3b12794f03 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sat, 7 Jul 2018 12:22:10 +0200 Subject: ASoC: Intel: bytcr_rt5640: Add quirk for the Lenovo Miix2 8 tablet Add a quirk for the Lenovo Miix2 8 tablet, this tablet uses a digital mic on DMIC1 and has a mono-speaker. The jack-detect uses the default settings.. Reported-and-tested-by: russianneuromancer@ya.ru Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7456566c5648..657910a08261 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -552,6 +552,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Lenovo Miix 2 8 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "20326"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Hiking"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), -- cgit v1.2.3 From fbea16dbc0a31484811c5f335ae344b2bbc66f40 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 1 Jul 2018 20:36:29 +0200 Subject: ASoC: Intel: bytcr_rt5651: Remove is_valleyview helper Remove is_valleyview helper, this is not necessary, we can simply call x86_match_cpu() directly instead. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index ba2753e0e12a..80f47a45cb10 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include @@ -680,17 +681,10 @@ static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic[-swapped-hp]" */ -static bool is_valleyview(void) -{ - static const struct x86_cpu_id cpu_ids[] = { - { X86_VENDOR_INTEL, 6, 55 }, /* Valleyview, Bay Trail */ - {} - }; - - if (!x86_match_cpu(cpu_ids)) - return false; - return true; -} +static const struct x86_cpu_id baytrail_cpu_ids[] = { + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT1 }, /* Valleyview */ + {} +}; struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ u64 aif_value; /* 1: AIF1, 2: AIF2 */ @@ -741,7 +735,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) * swap SSP0 if bytcr is detected * (will be overridden if DMI quirk is detected) */ - if (is_valleyview()) { + if (x86_match_cpu(baytrail_cpu_ids)) { struct sst_platform_info *p_info = mach->pdata; const struct sst_res_info *res_info = p_info->res_info; -- cgit v1.2.3 From 2c375204bfad2f481feb006a82cdb67cc570b670 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 1 Jul 2018 20:36:30 +0200 Subject: ASoC: Intel: bytcr_rt5651: Move getting of codec_dev into probe() Move the getting of the codec_dev, to add device-props to it, out of byt_rt5651_add_codec_device_props() and into its caller, snd_byt_rt5651_mc_probe(). This is a preparation patch for adding support for an external amplifier enable GPIO, which requires further accesses to the codec_dev. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 80f47a45cb10..d920725ce603 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -403,15 +403,10 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { * Note this MUST be called before snd_soc_register_card(), so that the props * are in place before the codec component driver's probe function parses them. */ -static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name) +static int byt_rt5651_add_codec_device_props(struct device *i2c_dev) { struct property_entry props[MAX_NO_PROPS] = {}; - struct device *i2c_dev; - int ret, cnt = 0; - - i2c_dev = bus_find_device_by_name(&i2c_bus_type, NULL, i2c_dev_name); - if (!i2c_dev) - return -EPROBE_DEFER; + int cnt = 0; props[cnt++] = PROPERTY_ENTRY_U32("realtek,jack-detect-source", BYT_RT5651_JDSRC(byt_rt5651_quirk)); @@ -425,10 +420,7 @@ static int byt_rt5651_add_codec_device_props(const char *i2c_dev_name) if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN) props[cnt++] = PROPERTY_ENTRY_BOOL("realtek,dmic-en"); - ret = device_add_properties(i2c_dev, props); - put_device(i2c_dev); - - return ret; + return device_add_properties(i2c_dev, props); } static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) @@ -696,6 +688,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) const char * const mic_name[] = { "dmic", "in1", "in12" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; + struct device *codec_dev; const char *i2c_name = NULL; const char *hp_swapped; bool is_bytcr = false; @@ -731,6 +724,11 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) "%s%s", "i2c-", i2c_name); byt_rt5651_dais[dai_index].codec_name = byt_rt5651_codec_name; + codec_dev = bus_find_device_by_name(&i2c_bus_type, NULL, + byt_rt5651_codec_name); + if (!codec_dev) + return -EPROBE_DEFER; + /* * swap SSP0 if bytcr is detected * (will be overridden if DMI quirk is detected) @@ -794,7 +792,8 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) dmi_check_system(byt_rt5651_quirk_table); /* Must be called before register_card, also see declaration comment. */ - ret_val = byt_rt5651_add_codec_device_props(byt_rt5651_codec_name); + ret_val = byt_rt5651_add_codec_device_props(codec_dev); + put_device(codec_dev); if (ret_val) return ret_val; -- cgit v1.2.3 From 5f6fb23d2e114506ddb8437a3e65f4a20d081013 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 1 Jul 2018 20:36:31 +0200 Subject: ASoC: Intel: bytcr_rt5651: Add support for externar amplifier enable GPIO The rt5651 does not have a built-in speaker amplifier, so it is often used together with an external amplifier. On Cherry Trail boards this external amplifier's enable pin is driven through a GPIO, which is given as the first GPIO in the ACPI resources of the codec fwnode. This commit adds support to the bytcr_rt5651 for this GPIO, fixing the speaker not working on CHT devices with a rt5651 codec. Cc: Carlo Caione Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 65 +++++++++++++++++++++++++++++++++-- 1 file changed, 62 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index d920725ce603..5301205496be 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -26,6 +26,8 @@ #include #include #include +#include +#include #include #include #include @@ -86,6 +88,7 @@ enum { struct byt_rt5651_private { struct clk *mclk; + struct gpio_desc *ext_amp_gpio; struct snd_soc_jack jack; }; @@ -208,6 +211,20 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return 0; } +static int rt5651_ext_amp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); + + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(priv->ext_amp_gpio, 1); + else + gpiod_set_value_cansleep(priv->ext_amp_gpio, 0); + + return 0; +} + static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -217,7 +234,9 @@ static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = { SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - + SND_SOC_DAPM_SUPPLY("Ext Amp Power", SND_SOC_NOPM, 0, 0, + rt5651_ext_amp_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), }; static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { @@ -225,6 +244,7 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { {"Headset Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "Platform Clock"}, {"Speaker", NULL, "Platform Clock"}, + {"Speaker", NULL, "Ext Amp Power"}, {"Line In", NULL, "Platform Clock"}, {"Headset Mic", NULL, "micbias1"}, /* lowercase for rt5651 */ @@ -678,6 +698,18 @@ static const struct x86_cpu_id baytrail_cpu_ids[] = { {} }; +static const struct x86_cpu_id cherrytrail_cpu_ids[] = { + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_AIRMONT }, /* Braswell */ + {} +}; + +static const struct acpi_gpio_params ext_amp_enable_gpios = { 0, 0, false }; + +static const struct acpi_gpio_mapping byt_rt5651_gpios[] = { + { "ext-amp-enable-gpios", &ext_amp_enable_gpios, 1 }, + { }, +}; + struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ u64 aif_value; /* 1: AIF1, 2: AIF2 */ u64 mclock_value; /* usually 25MHz (0x17d7940), ignored */ @@ -793,9 +825,36 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* Must be called before register_card, also see declaration comment. */ ret_val = byt_rt5651_add_codec_device_props(codec_dev); - put_device(codec_dev); - if (ret_val) + if (ret_val) { + put_device(codec_dev); return ret_val; + } + + /* Cherry Trail devices use an external amplifier enable gpio */ + if (x86_match_cpu(cherrytrail_cpu_ids)) { + devm_acpi_dev_add_driver_gpios(codec_dev, byt_rt5651_gpios); + priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child( + &pdev->dev, "ext-amp-enable", 0, + codec_dev->fwnode, + GPIOD_OUT_LOW, "speaker-amp"); + if (IS_ERR(priv->ext_amp_gpio)) { + ret_val = PTR_ERR(priv->ext_amp_gpio); + switch (ret_val) { + case -ENOENT: + priv->ext_amp_gpio = NULL; + break; + default: + dev_err(&pdev->dev, "Failed to get ext-amp-enable GPIO: %d\n", + ret_val); + /* fall through */ + case -EPROBE_DEFER: + put_device(codec_dev); + return ret_val; + } + } + } + + put_device(codec_dev); log_quirks(&pdev->dev); -- cgit v1.2.3 From 8d2d7bcdc1645dc243f7735278675b083c0e506c Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 5 Jul 2018 00:59:31 +0200 Subject: ASoC: rt5651: Fix workqueue cancel vs irq free race on remove On removal we must free the IRQ *before* cancelling the jack-detect work, so that the jack-detect work cannot be rescheduled by the IRQ. Before this commit we were cancelling the jack-detect work from the driver remove callback, while relying on devm to free the IRQ, which happens after the remove callback. This is the wrong order. This commit uses a devm-action to register a devm callback which cancels the work, before requesting the IRQ (devm tears things down in reverse order). This also allows us to remove the now empty remove driver callback. Cc: Carlo Caione Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/codecs/rt5651.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 6b5669f3e85d..39d2c67cd064 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1696,6 +1696,13 @@ static irqreturn_t rt5651_irq(int irq, void *data) return IRQ_HANDLED; } +static void rt5651_cancel_work(void *data) +{ + struct rt5651_priv *rt5651 = data; + + cancel_work_sync(&rt5651->jack_detect_work); +} + static int rt5651_set_jack(struct snd_soc_component *component, struct snd_soc_jack *hp_jack, void *data) { @@ -2036,6 +2043,11 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, INIT_WORK(&rt5651->jack_detect_work, rt5651_jack_detect_work); + /* Make sure work is stopped on probe-error / remove */ + ret = devm_add_action_or_reset(&i2c->dev, rt5651_cancel_work, rt5651); + if (ret) + return ret; + ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5651, rt5651_dai, ARRAY_SIZE(rt5651_dai)); @@ -2043,15 +2055,6 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, return ret; } -static int rt5651_i2c_remove(struct i2c_client *i2c) -{ - struct rt5651_priv *rt5651 = i2c_get_clientdata(i2c); - - cancel_work_sync(&rt5651->jack_detect_work); - - return 0; -} - static struct i2c_driver rt5651_i2c_driver = { .driver = { .name = "rt5651", @@ -2059,7 +2062,6 @@ static struct i2c_driver rt5651_i2c_driver = { .of_match_table = of_match_ptr(rt5651_of_match), }, .probe = rt5651_i2c_probe, - .remove = rt5651_i2c_remove, .id_table = rt5651_i2c_id, }; module_i2c_driver(rt5651_i2c_driver); -- cgit v1.2.3 From 34c906ddacd237511808fb2bbd941e6b91e9095a Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 5 Jul 2018 00:59:32 +0200 Subject: ASoC: rt5651: Allow disabling jack-detect by calling set_jack(NULL) Allow the machine driver to disable jack-detect over a suspend/resume by calling snd_soc_component_set_jack(NULL). Note this renames rt5651_set_jack, where all the jack-enable work was done to rt5651_enable_jack_detect. This function can now no longer fail as it does not request the IRQ anymore. It can still be passed an invalid jack source, but that should never happen, so this is now logged and treated as no jack source. Cc: Carlo Caione Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/codecs/rt5651.c | 55 +++++++++++++++++++++++++++++++---------------- 1 file changed, 36 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 39d2c67cd064..40bd1e70fee7 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1703,14 +1703,10 @@ static void rt5651_cancel_work(void *data) cancel_work_sync(&rt5651->jack_detect_work); } -static int rt5651_set_jack(struct snd_soc_component *component, - struct snd_soc_jack *hp_jack, void *data) +static void rt5651_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hp_jack) { struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); - int ret; - - if (!rt5651->irq) - return -EINVAL; /* IRQ output on GPIO1 */ snd_soc_component_update_bits(component, RT5651_GPIO_CTRL1, @@ -1737,10 +1733,10 @@ static int rt5651_set_jack(struct snd_soc_component *component, RT5651_JD2_IRQ_EN, RT5651_JD2_IRQ_EN); break; case RT5651_JD_NULL: - return 0; + return; default: dev_err(component->dev, "Currently only JD1_1 / JD1_2 / JD2 are supported\n"); - return -EINVAL; + return; } /* Enable jack detect power */ @@ -1774,19 +1770,28 @@ static int rt5651_set_jack(struct snd_soc_component *component, RT5651_MB1_OC_STKY_MASK, RT5651_MB1_OC_STKY_EN); rt5651->hp_jack = hp_jack; - - ret = devm_request_threaded_irq(component->dev, rt5651->irq, NULL, - rt5651_irq, - IRQF_TRIGGER_RISING | - IRQF_TRIGGER_FALLING | - IRQF_ONESHOT, "rt5651", rt5651); - if (ret) { - dev_err(component->dev, "Failed to reguest IRQ: %d\n", ret); - return ret; - } - + enable_irq(rt5651->irq); /* sync initial jack state */ queue_work(system_power_efficient_wq, &rt5651->jack_detect_work); +} + +static void rt5651_disable_jack_detect(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + disable_irq(rt5651->irq); + rt5651_cancel_work(rt5651); + + rt5651->hp_jack = NULL; +} + +static int rt5651_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + rt5651_enable_jack_detect(component, jack); + else + rt5651_disable_jack_detect(component); return 0; } @@ -2048,6 +2053,18 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, if (ret) return ret; + ret = devm_request_irq(&i2c->dev, rt5651->irq, rt5651_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5651", rt5651); + if (ret == 0) { + /* Gets re-enabled by rt5651_set_jack() */ + disable_irq(rt5651->irq); + } else { + dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n", + rt5651->irq, ret); + rt5651->irq = -ENXIO; + } + ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5651, rt5651_dai, ARRAY_SIZE(rt5651_dai)); -- cgit v1.2.3 From df1569f2006b157caa944367d0d431eb4ea08624 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 5 Jul 2018 00:59:33 +0200 Subject: ASoC: rt5651: Add button press support Enable button press detection for headsets by using the ovcd IRQ to get notified of button presses. This is modelled after (almost exactly copied from) the button press code for the rt5640 which has identical ovcd hardware. Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/codecs/rt5651.c | 158 ++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/rt5651.h | 8 +++ 2 files changed, 159 insertions(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 40bd1e70fee7..0462049e739c 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1581,6 +1581,24 @@ static void rt5651_disable_micbias1_for_ovcd(struct snd_soc_component *component snd_soc_dapm_mutex_unlock(dapm); } +static void rt5651_enable_micbias1_ovcd_irq(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, + RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_NOR); + rt5651->ovcd_irq_enabled = true; +} + +static void rt5651_disable_micbias1_ovcd_irq(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, + RT5651_IRQ_MB1_OC_MASK, RT5651_IRQ_MB1_OC_BP); + rt5651->ovcd_irq_enabled = false; +} + static void rt5651_clear_micbias1_ovcd(struct snd_soc_component *component) { snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, @@ -1622,10 +1640,80 @@ static bool rt5651_jack_inserted(struct snd_soc_component *component) return val == 0; } -/* Jack detect timings */ +/* Jack detect and button-press timings */ #define JACK_SETTLE_TIME 100 /* milli seconds */ #define JACK_DETECT_COUNT 5 #define JACK_DETECT_MAXCOUNT 20 /* Aprox. 2 seconds worth of tries */ +#define JACK_UNPLUG_TIME 80 /* milli seconds */ +#define BP_POLL_TIME 10 /* milli seconds */ +#define BP_POLL_MAXCOUNT 200 /* assume something is wrong after this */ +#define BP_THRESHOLD 3 + +static void rt5651_start_button_press_work(struct snd_soc_component *component) +{ + struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); + + rt5651->poll_count = 0; + rt5651->press_count = 0; + rt5651->release_count = 0; + rt5651->pressed = false; + rt5651->press_reported = false; + rt5651_clear_micbias1_ovcd(component); + schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME)); +} + +static void rt5651_button_press_work(struct work_struct *work) +{ + struct rt5651_priv *rt5651 = + container_of(work, struct rt5651_priv, bp_work.work); + struct snd_soc_component *component = rt5651->component; + + /* Check the jack was not removed underneath us */ + if (!rt5651_jack_inserted(component)) + return; + + if (rt5651_micbias1_ovcd(component)) { + rt5651->release_count = 0; + rt5651->press_count++; + /* Remember till after JACK_UNPLUG_TIME wait */ + if (rt5651->press_count >= BP_THRESHOLD) + rt5651->pressed = true; + rt5651_clear_micbias1_ovcd(component); + } else { + rt5651->press_count = 0; + rt5651->release_count++; + } + + /* + * The pins get temporarily shorted on jack unplug, so we poll for + * at least JACK_UNPLUG_TIME milli-seconds before reporting a press. + */ + rt5651->poll_count++; + if (rt5651->poll_count < (JACK_UNPLUG_TIME / BP_POLL_TIME)) { + schedule_delayed_work(&rt5651->bp_work, + msecs_to_jiffies(BP_POLL_TIME)); + return; + } + + if (rt5651->pressed && !rt5651->press_reported) { + dev_dbg(component->dev, "headset button press\n"); + snd_soc_jack_report(rt5651->hp_jack, SND_JACK_BTN_0, + SND_JACK_BTN_0); + rt5651->press_reported = true; + } + + if (rt5651->release_count >= BP_THRESHOLD) { + if (rt5651->press_reported) { + dev_dbg(component->dev, "headset button release\n"); + snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0); + } + /* Re-enable OVCD IRQ to detect next press */ + rt5651_enable_micbias1_ovcd_irq(component); + return; /* Stop polling */ + } + + schedule_delayed_work(&rt5651->bp_work, msecs_to_jiffies(BP_POLL_TIME)); +} static int rt5651_detect_headset(struct snd_soc_component *component) { @@ -1676,15 +1764,58 @@ static void rt5651_jack_detect_work(struct work_struct *work) { struct rt5651_priv *rt5651 = container_of(work, struct rt5651_priv, jack_detect_work); + struct snd_soc_component *component = rt5651->component; int report = 0; - if (rt5651_jack_inserted(rt5651->component)) { - rt5651_enable_micbias1_for_ovcd(rt5651->component); - report = rt5651_detect_headset(rt5651->component); - rt5651_disable_micbias1_for_ovcd(rt5651->component); + if (!rt5651_jack_inserted(component)) { + /* Jack removed, or spurious IRQ? */ + if (rt5651->hp_jack->status & SND_JACK_HEADPHONE) { + if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + cancel_delayed_work_sync(&rt5651->bp_work); + rt5651_disable_micbias1_ovcd_irq(component); + rt5651_disable_micbias1_for_ovcd(component); + } + snd_soc_jack_report(rt5651->hp_jack, 0, + SND_JACK_HEADSET | SND_JACK_BTN_0); + dev_dbg(component->dev, "jack unplugged\n"); + } + } else if (!(rt5651->hp_jack->status & SND_JACK_HEADPHONE)) { + /* Jack inserted */ + WARN_ON(rt5651->ovcd_irq_enabled); + rt5651_enable_micbias1_for_ovcd(component); + report = rt5651_detect_headset(component); + if (report == SND_JACK_HEADSET) { + /* Enable ovcd IRQ for button press detect. */ + rt5651_enable_micbias1_ovcd_irq(component); + } else { + /* No more need for overcurrent detect. */ + rt5651_disable_micbias1_for_ovcd(component); + } + dev_dbg(component->dev, "detect report %#02x\n", report); + snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); + } else if (rt5651->ovcd_irq_enabled && rt5651_micbias1_ovcd(component)) { + dev_dbg(component->dev, "OVCD IRQ\n"); + + /* + * The ovcd IRQ keeps firing while the button is pressed, so + * we disable it and start polling the button until released. + * + * The disable will make the IRQ pin 0 again and since we get + * IRQs on both edges (so as to detect both jack plugin and + * unplug) this means we will immediately get another IRQ. + * The ovcd_irq_enabled check above makes the 2ND IRQ a NOP. + */ + rt5651_disable_micbias1_ovcd_irq(component); + rt5651_start_button_press_work(component); + + /* + * If the jack-detect IRQ flag goes high (unplug) after our + * above rt5651_jack_inserted() check and before we have + * disabled the OVCD IRQ, the IRQ pin will stay high and as + * we react to edges, we miss the unplug event -> recheck. + */ + queue_work(system_long_wq, &rt5651->jack_detect_work); } - - snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); } static irqreturn_t rt5651_irq(int irq, void *data) @@ -1701,6 +1832,7 @@ static void rt5651_cancel_work(void *data) struct rt5651_priv *rt5651 = data; cancel_work_sync(&rt5651->jack_detect_work); + cancel_delayed_work_sync(&rt5651->bp_work); } static void rt5651_enable_jack_detect(struct snd_soc_component *component, @@ -1770,6 +1902,11 @@ static void rt5651_enable_jack_detect(struct snd_soc_component *component, RT5651_MB1_OC_STKY_MASK, RT5651_MB1_OC_STKY_EN); rt5651->hp_jack = hp_jack; + if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + rt5651_enable_micbias1_for_ovcd(component); + rt5651_enable_micbias1_ovcd_irq(component); + } + enable_irq(rt5651->irq); /* sync initial jack state */ queue_work(system_power_efficient_wq, &rt5651->jack_detect_work); @@ -1782,6 +1919,12 @@ static void rt5651_disable_jack_detect(struct snd_soc_component *component) disable_irq(rt5651->irq); rt5651_cancel_work(rt5651); + if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + rt5651_disable_micbias1_ovcd_irq(component); + rt5651_disable_micbias1_for_ovcd(component); + snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0); + } + rt5651->hp_jack = NULL; } @@ -2046,6 +2189,7 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, rt5651->irq = i2c->irq; rt5651->hp_mute = 1; + INIT_DELAYED_WORK(&rt5651->bp_work, rt5651_button_press_work); INIT_WORK(&rt5651->jack_detect_work, rt5651_jack_detect_work); /* Make sure work is stopped on probe-error / remove */ diff --git a/sound/soc/codecs/rt5651.h b/sound/soc/codecs/rt5651.h index 3a0968c53fde..ac6de6fb5414 100644 --- a/sound/soc/codecs/rt5651.h +++ b/sound/soc/codecs/rt5651.h @@ -2071,8 +2071,16 @@ struct rt5651_pll_code { struct rt5651_priv { struct snd_soc_component *component; struct regmap *regmap; + /* Jack and button detect data */ struct snd_soc_jack *hp_jack; struct work_struct jack_detect_work; + struct delayed_work bp_work; + bool ovcd_irq_enabled; + bool pressed; + bool press_reported; + int press_count; + int release_count; + int poll_count; unsigned int jd_src; unsigned int ovcd_th; unsigned int ovcd_sf; -- cgit v1.2.3 From b91f432cbc3326f715b8c3f02ff4066ab398833f Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 5 Jul 2018 00:59:34 +0200 Subject: ASoC: Intel: bytcr_rt5651: Disable jack-detect over suspend/resume Disable jack-detection and thus the codec IRQ over suspend/resume. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 49 +++++++++++++++++++++++++++++++---- 1 file changed, 44 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 5301205496be..2a8f86dfe4cb 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -676,6 +676,48 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { }; /* SoC card */ +static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; +static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ +static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ +static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic[-swapped-hp]" */ + +static int byt_rt5651_suspend(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + + if (!BYT_RT5651_JDSRC(byt_rt5651_quirk)) + return 0; + + list_for_each_entry(component, &card->component_dev_list, card_list) { + if (!strcmp(component->name, byt_rt5651_codec_name)) { + dev_dbg(component->dev, "disabling jack detect before suspend\n"); + snd_soc_component_set_jack(component, NULL, NULL); + break; + } + } + + return 0; +} + +static int byt_rt5651_resume(struct snd_soc_card *card) +{ + struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component; + + if (!BYT_RT5651_JDSRC(byt_rt5651_quirk)) + return 0; + + list_for_each_entry(component, &card->component_dev_list, card_list) { + if (!strcmp(component->name, byt_rt5651_codec_name)) { + dev_dbg(component->dev, "re-enabling jack detect after resume\n"); + snd_soc_component_set_jack(component, &priv->jack, NULL); + break; + } + } + + return 0; +} + static struct snd_soc_card byt_rt5651_card = { .name = "bytcr-rt5651", .owner = THIS_MODULE, @@ -686,13 +728,10 @@ static struct snd_soc_card byt_rt5651_card = { .dapm_routes = byt_rt5651_audio_map, .num_dapm_routes = ARRAY_SIZE(byt_rt5651_audio_map), .fully_routed = true, + .suspend_pre = byt_rt5651_suspend, + .resume_post = byt_rt5651_resume, }; -static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; -static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ -static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ -static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic[-swapped-hp]" */ - static const struct x86_cpu_id baytrail_cpu_ids[] = { { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT1 }, /* Valleyview */ {} -- cgit v1.2.3 From caed9d636e857997e923dfe473b9310de645d916 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 5 Jul 2018 00:59:35 +0200 Subject: ASoC: Intel: bytcr_rt5651: Reporting button presses Enable reporting of button presses now that the codec driver recently has gotten support for this. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 2a8f86dfe4cb..b687043c8425 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -531,13 +532,17 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) { ret = snd_soc_card_jack_new(runtime->card, "Headset", - SND_JACK_HEADSET, &priv->jack, - bytcr_jack_pins, ARRAY_SIZE(bytcr_jack_pins)); + SND_JACK_HEADSET | SND_JACK_BTN_0, + &priv->jack, bytcr_jack_pins, + ARRAY_SIZE(bytcr_jack_pins)); if (ret) { dev_err(runtime->dev, "jack creation failed %d\n", ret); return ret; } + snd_jack_set_key(priv->jack.jack, SND_JACK_BTN_0, + KEY_PLAYPAUSE); + ret = snd_soc_component_set_jack(codec, &priv->jack, NULL); if (ret) return ret; -- cgit v1.2.3 From 435ffb76f8b35be108a52bf1c43233a57b3c72b4 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Jul 2018 12:13:48 +0200 Subject: ASoC: dpcm: rework runtime stream merge The goal of this patch is to simplify a bit dpcm runtime stream merge by removing several local variables. ATM, merge functions return the BE 'filter' values which should then be filtered against the FE stream values. This create a lot of local variable and unnecessary init of min and max. Instead of this, we can pass the FE stream values directly and let the BE filtering functions perform the merge 'in-place' Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 39 +++++++++++++++------------------------ 1 file changed, 15 insertions(+), 24 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b7e67b871c0c..114e6c060cae 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1672,29 +1672,28 @@ unwind: } static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, - struct snd_soc_pcm_stream *stream, - u64 formats) + struct snd_soc_pcm_stream *stream) { runtime->hw.rate_min = stream->rate_min; runtime->hw.rate_max = stream->rate_max; runtime->hw.channels_min = stream->channels_min; runtime->hw.channels_max = stream->channels_max; if (runtime->hw.formats) - runtime->hw.formats &= formats & stream->formats; + runtime->hw.formats &= stream->formats; else - runtime->hw.formats = formats & stream->formats; + runtime->hw.formats = stream->formats; runtime->hw.rates = stream->rates; } -static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) +static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, + u64 *formats) { struct snd_soc_pcm_runtime *fe = substream->private_data; struct snd_soc_dpcm *dpcm; - u64 formats = ULLONG_MAX; int stream = substream->stream; if (!fe->dai_link->dpcm_merged_format) - return formats; + return; /* * It returns merged BE codec format @@ -1714,16 +1713,14 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) else codec_stream = &codec_dai_drv->capture; - formats &= codec_stream->formats; + *formats &= codec_stream->formats; } } - - return formats; } -static void dpcm_runtime_base_chan(struct snd_pcm_substream *substream, - unsigned int *channels_min, - unsigned int *channels_max) +static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, + unsigned int *channels_min, + unsigned int *channels_max) { struct snd_soc_pcm_runtime *fe = substream->private_data; struct snd_soc_dpcm *dpcm; @@ -1732,9 +1729,6 @@ static void dpcm_runtime_base_chan(struct snd_pcm_substream *substream, if (!fe->dai_link->dpcm_merged_chan) return; - *channels_min = 0; - *channels_max = UINT_MAX; - /* * It returns merged BE codec channel; * if FE want to use it (= dpcm_merged_chan) @@ -1781,18 +1775,15 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - u64 format = dpcm_runtime_base_format(substream); - unsigned int channels_min = 0, channels_max = UINT_MAX; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback, format); + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback); else - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture, format); - - dpcm_runtime_base_chan(substream, &channels_min, &channels_max); + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); - runtime->hw.channels_min = max(channels_min, runtime->hw.channels_min); - runtime->hw.channels_max = min(channels_max, runtime->hw.channels_max); + dpcm_runtime_merge_format(substream, &runtime->hw.formats); + dpcm_runtime_merge_chan(substream, &runtime->hw.channels_min, + &runtime->hw.channels_max); } static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd); -- cgit v1.2.3 From baacd8d100d571aa713c3c60b1471b9962e6ec8a Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Jul 2018 12:13:49 +0200 Subject: ASoC: dpcm: add rate merge to the BE stream merge As done for format and channels, add the possibility to merge the backend rates on the frontend rates. This useful if the backend does not support all rates supported by the frontend, or if several backends (cpu and codecs) with different capabilities are connected to the same frontend. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ sound/soc/soc-pcm.c | 60 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 62 insertions(+) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index 870ba6b64817..a4915148f739 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -964,6 +964,8 @@ struct snd_soc_dai_link { unsigned int dpcm_merged_format:1; /* DPCM used FE & BE merged channel */ unsigned int dpcm_merged_chan:1; + /* DPCM used FE & BE merged rate */ + unsigned int dpcm_merged_rate:1; /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 114e6c060cae..4019bc10897c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1769,6 +1769,64 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, } } +static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, + unsigned int *rates, + unsigned int *rate_min, + unsigned int *rate_max) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = substream->stream; + + if (!fe->dai_link->dpcm_merged_rate) + return; + + /* + * It returns merged BE codec channel; + * if FE want to use it (= dpcm_merged_chan) + */ + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_pcm_stream *codec_stream; + struct snd_soc_pcm_stream *cpu_stream; + int i; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; + else + cpu_stream = &cpu_dai_drv->capture; + + *rate_min = max(*rate_min, cpu_stream->rate_min); + *rate_max = min_not_zero(*rate_max, cpu_stream->rate_max); + *rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates); + + for (i = 0; i < be->num_codecs; i++) { + /* + * Skip CODECs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(be->codec_dais[i], + stream)) + continue; + + codec_dai_drv = be->codec_dais[i]->driver; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + + *rate_min = max(*rate_min, codec_stream->rate_min); + *rate_max = min_not_zero(*rate_max, + codec_stream->rate_max); + *rates = snd_pcm_rate_mask_intersect(*rates, + codec_stream->rates); + } + } +} + static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1784,6 +1842,8 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) dpcm_runtime_merge_format(substream, &runtime->hw.formats); dpcm_runtime_merge_chan(substream, &runtime->hw.channels_min, &runtime->hw.channels_max); + dpcm_runtime_merge_rate(substream, &runtime->hw.rates, + &runtime->hw.rate_min, &runtime->hw.rate_max); } static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd); -- cgit v1.2.3 From aefba45539bc4868c1fae336410aec907ee0882a Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 13 Jul 2018 14:50:43 +0200 Subject: ASoC: allow soc-core to pick up name prefixes from component nodes When the component does not match the configuration table provided by the card, let soc-core check the component node for a name prefix Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 28 +++++++++++++++++++++++----- 1 file changed, 23 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 00bd58d167dd..3be0310d5c81 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1193,15 +1193,27 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_remove_dai_link); +static void soc_set_of_name_prefix(struct snd_soc_component *component) +{ + struct device_node *component_of_node = component->dev->of_node; + const char *str; + int ret; + + if (!component_of_node && component->dev->parent) + component_of_node = component->dev->parent->of_node; + + ret = of_property_read_string(component_of_node, "sound-name-prefix", + &str); + if (!ret) + component->name_prefix = str; +} + static void soc_set_name_prefix(struct snd_soc_card *card, struct snd_soc_component *component) { int i; - if (card->codec_conf == NULL) - return; - - for (i = 0; i < card->num_configs; i++) { + for (i = 0; i < card->num_configs && card->codec_conf; i++) { struct snd_soc_codec_conf *map = &card->codec_conf[i]; struct device_node *component_of_node = component->dev->of_node; @@ -1213,8 +1225,14 @@ static void soc_set_name_prefix(struct snd_soc_card *card, if (map->dev_name && strcmp(component->name, map->dev_name)) continue; component->name_prefix = map->name_prefix; - break; + return; } + + /* + * If there is no configuration table or no match in the table, + * check if a prefix is provided in the node + */ + soc_set_of_name_prefix(component); } static int soc_probe_component(struct snd_soc_card *card, -- cgit v1.2.3 From b8110a87b75f948d978c06e130cc68026645c4a1 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 13 Jul 2018 18:05:57 +0300 Subject: ASoC: qdsp6: q6afe-dai: fix a range check in of_q6afe_parse_dai_data() The main thing is that the data->priv[] array has AFE_PORT_MAX elements so the > condition should be >=. But we may as well check for negative values as well just to be safe. Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe dai driver") Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index a373ca5523ff..9ba95956ada8 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1183,7 +1183,7 @@ static void of_q6afe_parse_dai_data(struct device *dev, int id, i, num_lines; ret = of_property_read_u32(node, "reg", &id); - if (ret || id > AFE_PORT_MAX) { + if (ret || id < 0 || id >= AFE_PORT_MAX) { dev_err(dev, "valid dai id not found:%d\n", ret); continue; } -- cgit v1.2.3 From 090345ce7265dd111299e3a63cdc79c3ef924481 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 13 Jul 2018 18:11:04 +0300 Subject: ASoC: qdsp6: q6routing: off by one in routing_hw_params() The data->port_data[] array has AFE_MAX_PORTS elements so the check should be >= instead of > or we write one element beyond the end of the array. Fixes: e3a33673e845 ("ASoC: qdsp6: q6routing: Add q6routing driver") Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 593f66b8622f..7a19d6278406 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -899,7 +899,7 @@ static int routing_hw_params(struct snd_pcm_substream *substream, else path_type = ADM_PATH_LIVE_REC; - if (be_id > AFE_MAX_PORTS) + if (be_id >= AFE_MAX_PORTS) return -EINVAL; session = &data->port_data[be_id]; -- cgit v1.2.3 From d30e23d69981a4b665f5ce8711335df986576389 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sat, 14 Jul 2018 16:01:06 +0100 Subject: ASoC: hdmi-codec: fix routing Commit 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") broke hdmi-codec's routing between it's output "TX" widget and the S/PDIF or I2S streams by renaming the streams. Whether an error occurs or not is dependent on whether there is another widget called "Playback" registered by some other component - if there is, that widget will be (incorrectly) bound to the HDMI codec's "TX" output widget. If we end up connecting "TX" incorrectly, it can result in components not being started, causing no audio output. Since the I2S and S/PDIF streams now have different names, we can't use a static route at component level to describe the relationship, so arrange to dynamically create the route when the DAI driver is probed. Fixes: 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 6fa11888672d..3e5b12de71bb 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -291,10 +291,6 @@ static const struct snd_soc_dapm_widget hdmi_widgets[] = { SND_SOC_DAPM_OUTPUT("TX"), }; -static const struct snd_soc_dapm_route hdmi_routes[] = { - { "TX", NULL, "Playback" }, -}; - enum { DAI_ID_I2S = 0, DAI_ID_SPDIF, @@ -689,9 +685,23 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, return snd_ctl_add(rtd->card->snd_card, kctl); } +static int hdmi_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_dapm_context *dapm; + struct snd_soc_dapm_route route = { + .sink = "TX", + .source = dai->driver->playback.stream_name, + }; + + dapm = snd_soc_component_get_dapm(dai->component); + + return snd_soc_dapm_add_routes(dapm, &route, 1); +} + static const struct snd_soc_dai_driver hdmi_i2s_dai = { .name = "i2s-hifi", .id = DAI_ID_I2S, + .probe = hdmi_dai_probe, .playback = { .stream_name = "I2S Playback", .channels_min = 2, @@ -707,6 +717,7 @@ static const struct snd_soc_dai_driver hdmi_i2s_dai = { static const struct snd_soc_dai_driver hdmi_spdif_dai = { .name = "spdif-hifi", .id = DAI_ID_SPDIF, + .probe = hdmi_dai_probe, .playback = { .stream_name = "SPDIF Playback", .channels_min = 2, @@ -733,8 +744,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, static const struct snd_soc_component_driver hdmi_driver = { .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), - .dapm_routes = hdmi_routes, - .num_dapm_routes = ARRAY_SIZE(hdmi_routes), .of_xlate_dai_id = hdmi_of_xlate_dai_id, .idle_bias_on = 1, .use_pmdown_time = 1, -- cgit v1.2.3 From 8452112baac67c3235d15de67fb190d29bbba98f Mon Sep 17 00:00:00 2001 From: Naveen Manohar Date: Wed, 11 Jul 2018 16:05:36 +0530 Subject: ASoC: Intel: Boards: Add GLK Realtek Maxim I2S machine driver Patch adds Geminilake I2S machine driver which uses following codecs: RT5682 and MAX98357A. Signed-off-by: Naveen Manohar Signed-off-by: Harsha Priya Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 643 ++++++++++++++++++++++++++ 3 files changed, 659 insertions(+) create mode 100644 sound/soc/intel/boards/glk_rt5682_max98357a.c (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 24797482a3d2..cccda87f4b34 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -281,6 +281,20 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH + tristate "GLK with RT5682 and MAX98357A in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_RT5682 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_HDA_DSP_LOADER + help + This adds support for ASoC machine driver for Geminilake platforms + with RT5682 + MAX98357A I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_INTEL_SKYLAKE endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 92b5507291af..87ef8b4058e5 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o snd-soc-sst-bxt-rt298-objs := bxt_rt298.o +snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o @@ -27,6 +28,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o +obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c new file mode 100644 index 000000000000..c4b94e2617c5 --- /dev/null +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -0,0 +1,643 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018 Intel Corporation. + +/* + * Intel Geminilake I2S Machine Driver with MAX98357A & RT5682 Codecs + * + * Modified from: + * Intel Apollolake I2S Machine driver + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "../skylake/skl.h" +#include "../../codecs/rt5682.h" +#include "../../codecs/hdac_hdmi.h" + +/* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */ +#define GLK_PLAT_CLK_FREQ 19200000 +#define RT5682_PLL_FREQ (48000 * 512) +#define GLK_REALTEK_CODEC_DAI "rt5682-aif1" +#define GLK_MAXIM_CODEC_DAI "HiFi" +#define MAXIM_DEV0_NAME "MX98357A:00" +#define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 +#define NAME_SIZE 32 + +static struct snd_soc_jack geminilake_hdmi[3]; + +struct glk_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct glk_card_private { + struct snd_soc_jack geminilake_headset; + struct list_head hdmi_pcm_list; +}; + +enum { + GLK_DPCM_AUDIO_PB = 0, + GLK_DPCM_AUDIO_CP, + GLK_DPCM_AUDIO_HS_PB, + GLK_DPCM_AUDIO_ECHO_REF_CP, + GLK_DPCM_AUDIO_REF_CP, + GLK_DPCM_AUDIO_DMIC_CP, + GLK_DPCM_AUDIO_HDMI1_PB, + GLK_DPCM_AUDIO_HDMI2_PB, + GLK_DPCM_AUDIO_HDMI3_PB, +}; + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop sysclk: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(card->dev, "can't set codec pll: %d\n", ret); + return ret; + } + } + + if (ret) + dev_err(card->dev, "failed to start internal clk: %d\n", ret); + + return ret; +} + +static const struct snd_kcontrol_new geminilake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget geminilake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("HDMI1", NULL), + SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SPK("HDMI3", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route geminilake_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* speaker */ + { "Spk", NULL, "Speaker" }, + + /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, + { "IN1P", NULL, "Headset Mic" }, + + /* digital mics */ + { "DMic", NULL, "SoC DMIC" }, + + /* CODEC BE connections */ + { "HiFi Playback", NULL, "ssp1 Tx" }, + { "ssp1 Tx", NULL, "codec0_out" }, + + { "AIF1 Playback", NULL, "ssp2 Tx" }, + { "ssp2 Tx", NULL, "codec1_out" }, + + { "codec0_in", NULL, "ssp2 Rx" }, + { "ssp2 Rx", NULL, "AIF1 Capture" }, + + { "HDMI1", NULL, "hif5-0 Output" }, + { "HDMI2", NULL, "hif6-0 Output" }, + { "HDMI2", NULL, "hif7-0 Output" }, + + { "hifi3", NULL, "iDisp3 Tx" }, + { "iDisp3 Tx", NULL, "iDisp3_out" }, + { "hifi2", NULL, "iDisp2 Tx" }, + { "iDisp2 Tx", NULL, "iDisp2_out" }, + { "hifi1", NULL, "iDisp1 Tx" }, + { "iDisp1 Tx", NULL, "iDisp1_out" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "DMIC AIF" }, +}; + +static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = DUAL_CHANNEL; + + /* set SSP to 24 bit */ + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_jack *jack; + int ret; + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->geminilake_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->geminilake_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static int geminilake_rt5682_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Set valid bitmask & configuration for I2S in 24 bit */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, 24); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops geminilake_rt5682_ops = { + .hw_params = geminilake_rt5682_hw_params, +}; + +static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct glk_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = GLK_DPCM_AUDIO_HDMI1_PB + dai->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_dapm_context *dapm; + int ret; + + dapm = snd_soc_component_get_dapm(component); + ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); + if (ret) { + dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret); + return ret; + } + + return ret; +} + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + +static int geminilake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* + * set BE channel constraint as user FE channels + */ + channels->min = channels->max = 4; + + return 0; +} + +static int geminilake_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels_quad); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static const struct snd_soc_ops geminilake_dmic_ops = { + .startup = geminilake_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static int geminilake_refcap_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +}; + +static const struct snd_soc_ops geminilake_refcap_ops = { + .startup = geminilake_refcap_startup, +}; + +/* geminilake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link geminilake_dais[] = { + /* Front End DAI links */ + [GLK_DPCM_AUDIO_PB] = { + .name = "Glk Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = geminilake_rt5682_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + [GLK_DPCM_AUDIO_CP] = { + .name = "Glk Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + [GLK_DPCM_AUDIO_HS_PB] = { + .name = "Glk Audio Headset Playback", + .stream_name = "Headset Audio", + .cpu_dai_name = "System Pin2", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dpcm_playback = 1, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Glk Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [GLK_DPCM_AUDIO_REF_CP] = { + .name = "Glk Audio Reference cap", + .stream_name = "Refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_refcap_ops, + }, + [GLK_DPCM_AUDIO_DMIC_CP] = { + .name = "Glk Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_dmic_ops, + }, + [GLK_DPCM_AUDIO_HDMI1_PB] = { + .name = "Glk HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI2_PB] = { + .name = "Glk HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI3_PB] = { + .name = "Glk HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + /* Back End DAI links */ + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .id = 0, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = MAXIM_DEV0_NAME, + .codec_dai_name = GLK_MAXIM_CODEC_DAI, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .dpcm_playback = 1, + }, + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .id = 1, + .cpu_dai_name = "SSP2 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = "i2c-10EC5682:00", + .codec_dai_name = GLK_REALTEK_CODEC_DAI, + .init = geminilake_rt5682_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .ops = &geminilake_rt5682_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 2, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:0e.0", + .ignore_suspend = 1, + .be_hw_params_fixup = geminilake_dmic_fixup, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +static int glk_card_late_probe(struct snd_soc_card *card) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + char jack_name[NAME_SIZE]; + struct glk_hdmi_pcm *pcm; + int err = 0; + int i = 0; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &geminilake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &geminilake_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +/* geminilake audio machine driver for SPT + RT5682 */ +static struct snd_soc_card glk_audio_card_rt5682_m98357a = { + .name = "glkrt5682max", + .owner = THIS_MODULE, + .dai_link = geminilake_dais, + .num_links = ARRAY_SIZE(geminilake_dais), + .controls = geminilake_controls, + .num_controls = ARRAY_SIZE(geminilake_controls), + .dapm_widgets = geminilake_widgets, + .num_dapm_widgets = ARRAY_SIZE(geminilake_widgets), + .dapm_routes = geminilake_map, + .num_dapm_routes = ARRAY_SIZE(geminilake_map), + .fully_routed = true, + .late_probe = glk_card_late_probe, +}; + +static int geminilake_audio_probe(struct platform_device *pdev) +{ + struct glk_card_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + glk_audio_card_rt5682_m98357a.dev = &pdev->dev; + snd_soc_card_set_drvdata(&glk_audio_card_rt5682_m98357a, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &glk_audio_card_rt5682_m98357a); +} + +static const struct platform_device_id glk_board_ids[] = { + { + .name = "glk_rt5682_max98357a", + .driver_data = + (kernel_ulong_t)&glk_audio_card_rt5682_m98357a, + }, + { } +}; + +static struct platform_driver geminilake_audio = { + .probe = geminilake_audio_probe, + .driver = { + .name = "glk_rt5682_max98357a", + .pm = &snd_soc_pm_ops, + }, + .id_table = glk_board_ids, +}; +module_platform_driver(geminilake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Geminilake Audio Machine driver-RT5682 & MAX98357A in I2S mode"); +MODULE_AUTHOR("Naveen Manohar "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:glk_rt5682_max98357a"); -- cgit v1.2.3 From fa9d2f17c23fb3ea6b659b1bfe4ca10551a19e56 Mon Sep 17 00:00:00 2001 From: Agrawal, Akshu Date: Mon, 16 Jul 2018 15:02:40 +0800 Subject: ASoC: AMD: Send correct channel for configuring DMA descriptors Earlier, ch1 was used to define ACP-SYSMEM transfer and ch2 for ACP-I2S transfer. With recent patches ch1 is used to define channel order number 1 and ch2 as channel order number 2. Thus, Playback: ch1:SYSMEM->ACP ch2:ACP->I2S Capture: ch1:I2S->ACP ch1:ACP->SYSMEM Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 26 ++++++++++++++++++++++---- 1 file changed, 22 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 65c1033bd51c..eeb867767252 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -322,17 +322,27 @@ static void config_acp_dma(void __iomem *acp_mmio, struct audio_substream_data *rtd, u32 asic_type) { + u16 ch_acp_sysmem, ch_acp_i2s; + acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, rtd->pte_offset); + + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + ch_acp_sysmem = rtd->ch1; + ch_acp_i2s = rtd->ch2; + } else { + ch_acp_i2s = rtd->ch1; + ch_acp_sysmem = rtd->ch2; + } /* Configure System memory <-> ACP SRAM DMA descriptors */ set_acp_sysmem_dma_descriptors(acp_mmio, rtd->size, rtd->direction, rtd->pte_offset, - rtd->ch1, rtd->sram_bank, + ch_acp_sysmem, rtd->sram_bank, rtd->dma_dscr_idx_1, asic_type); /* Configure ACP SRAM <-> I2S DMA descriptors */ set_acp_to_i2s_dma_descriptors(acp_mmio, rtd->size, rtd->direction, rtd->sram_bank, - rtd->destination, rtd->ch2, + rtd->destination, ch_acp_i2s, rtd->dma_dscr_idx_2, asic_type); } @@ -995,16 +1005,24 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + u16 ch_acp_sysmem, ch_acp_i2s; if (!rtd) return -EINVAL; + if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + ch_acp_sysmem = rtd->ch1; + ch_acp_i2s = rtd->ch2; + } else { + ch_acp_i2s = rtd->ch1; + ch_acp_sysmem = rtd->ch2; + } config_acp_dma_channel(rtd->acp_mmio, - rtd->ch1, + ch_acp_sysmem, rtd->dma_dscr_idx_1, NUM_DSCRS_PER_CHANNEL, 0); config_acp_dma_channel(rtd->acp_mmio, - rtd->ch2, + ch_acp_i2s, rtd->dma_dscr_idx_2, NUM_DSCRS_PER_CHANNEL, 0); return 0; -- cgit v1.2.3 From 19e023e3befb4cb64b4a81b47a92a0c687672661 Mon Sep 17 00:00:00 2001 From: Agrawal, Akshu Date: Mon, 16 Jul 2018 15:02:41 +0800 Subject: ASoC: AMD: For capture have interrupts on I2S->ACP channel Having interrupts enabled for ACP<->SYSMEM DMA transfer, we are in for an interrupt storm. For both playback and capture interrupts should be enabled for I2S<->ACP DMA. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index eeb867767252..94bcf69008df 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -224,13 +224,11 @@ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio, switch (asic_type) { case CHIP_STONEY: dmadscr[i].xfer_val |= - BIT(22) | (ACP_DMA_ATTR_SHARED_MEM_TO_DAGB_GARLIC << 16) | (size / 2); break; default: dmadscr[i].xfer_val |= - BIT(22) | (ACP_DMA_ATTR_SHAREDMEM_TO_DAGB_ONION << 16) | (size / 2); } @@ -421,9 +419,9 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) switch (ch_num) { case ACP_TO_I2S_DMA_CH_NUM: - case ACP_TO_SYSRAM_CH_NUM: + case I2S_TO_ACP_DMA_CH_NUM: case ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM: - case ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM: + case I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM: dma_ctrl |= ACP_DMA_CNTL_0__DMAChIOCEn_MASK; break; default: @@ -705,18 +703,18 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) != 0) { + if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); - acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) << 16, + acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } - if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) { + if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & - BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16, + BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } -- cgit v1.2.3 From bb4b894addb09a069c072a0a032f644cc470d17f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 13 Jul 2018 16:36:28 +0100 Subject: ASoC: core: add support to card re-bind using component framework This patch aims at achieving dynamic behaviour of audio card when the dependent components disappear and reappear. With this patch the card is removed if any of the dependent component is removed and card is added back if the dependent component comes back. All this is done using component framework and matching based on component name. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- include/sound/soc.h | 7 ++++++ sound/soc/soc-core.c | 62 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 69 insertions(+) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index a4915148f739..a23ecdf3eff1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -1090,6 +1091,12 @@ struct snd_soc_card { struct work_struct deferred_resume_work; + /* component framework related */ + bool components_added; + /* set in machine driver to enable/disable auto re-binding */ + bool auto_bind; + struct component_match *match; + /* lists of probed devices belonging to this card */ struct list_head component_dev_list; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3be0310d5c81..08e189485009 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -279,11 +279,28 @@ static inline void snd_soc_debugfs_exit(void) #endif +static int snd_soc_card_comp_compare(struct device *dev, void *data) +{ + struct snd_soc_component *component; + + lockdep_assert_held(&client_mutex); + list_for_each_entry(component, &component_list, list) { + if (dev == component->dev) { + if (!strcmp(component->name, data)) + return 1; + break; + } + } + + return 0; +} + static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd, struct snd_soc_component *component) { struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_rtdcom_list *new_rtdcom; + char *cname; for_each_rtdcom(rtd, rtdcom) { /* already connected */ @@ -300,6 +317,13 @@ static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd, list_add_tail(&new_rtdcom->list, &rtd->component_list); + if (rtd->card->auto_bind && !rtd->card->components_added) { + cname = devm_kasprintf(rtd->card->dev, GFP_KERNEL, + "%s", component->name); + component_match_add(rtd->card->dev, &rtd->card->match, + snd_soc_card_comp_compare, cname); + } + return 0; } @@ -835,6 +859,25 @@ static bool soc_is_dai_link_bound(struct snd_soc_card *card, return false; } +static int snd_soc_card_comp_bind(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + + if (card->instantiated) + return 0; + + return snd_soc_register_card(card); +} + +static void snd_soc_card_comp_unbind(struct device *dev) +{ +} + +static const struct component_master_ops snd_soc_card_comp_ops = { + .bind = snd_soc_card_comp_bind, + .unbind = snd_soc_card_comp_unbind, +}; + static int soc_bind_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -2126,6 +2169,12 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->instantiated = 1; snd_soc_dapm_sync(&card->dapm); + if (card->auto_bind && !card->components_added) { + component_master_add_with_match(card->dev, + &snd_soc_card_comp_ops, + card->match); + card->components_added = true; + } mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); @@ -2749,6 +2798,9 @@ int snd_soc_unregister_card(struct snd_soc_card *card) dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); } + if (!card->auto_bind && card->components_added) + component_master_del(card->dev, &snd_soc_card_comp_ops); + return 0; } EXPORT_SYMBOL_GPL(snd_soc_unregister_card); @@ -3161,8 +3213,17 @@ int snd_soc_add_component(struct device *dev, snd_soc_component_add(component); + ret = component_add(dev, NULL); + if (ret < 0) { + dev_err(dev, "ASoC: Failed to add Component: %d\n", ret); + goto err_comp; + } + return 0; +err_comp: + soc_remove_component(component); + snd_soc_unregister_dais(component); err_cleanup: snd_soc_component_cleanup(component); err_free: @@ -3210,6 +3271,7 @@ static int __snd_soc_unregister_component(struct device *dev) mutex_unlock(&client_mutex); if (found) { + component_del(dev, NULL); snd_soc_component_cleanup(component); } -- cgit v1.2.3 From 605fcb69918528e1a448cba4d358cbd8ed532146 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 13 Jul 2018 16:36:29 +0100 Subject: ASoC: qdsp6: q6afe-dai: remove component fw related code Now that the component framework is integrated into the ASoC core, remove any redundant code in this driver. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 34 ++++------------------------------ 1 file changed, 4 insertions(+), 30 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 074582afda85..e988692a3ced 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -4,7 +4,6 @@ #include #include -#include #include #include #include @@ -1395,11 +1394,12 @@ static void of_q6afe_parse_dai_data(struct device *dev, } } -static int q6afe_dai_bind(struct device *dev, struct device *master, void *data) +static int q6afe_dai_dev_probe(struct platform_device *pdev) { struct q6afe_dai_data *dai_data; + struct device *dev = &pdev->dev; - dai_data = kzalloc(sizeof(*dai_data), GFP_KERNEL); + dai_data = devm_kzalloc(dev, sizeof(*dai_data), GFP_KERNEL); if (!dai_data) return -ENOMEM; @@ -1407,35 +1407,10 @@ static int q6afe_dai_bind(struct device *dev, struct device *master, void *data) of_q6afe_parse_dai_data(dev, dai_data); - return snd_soc_register_component(dev, &q6afe_dai_component, + return devm_snd_soc_register_component(dev, &q6afe_dai_component, q6afe_dais, ARRAY_SIZE(q6afe_dais)); } -static void q6afe_dai_unbind(struct device *dev, struct device *master, - void *data) -{ - struct q6afe_dai_data *dai_data = dev_get_drvdata(dev); - - snd_soc_unregister_component(dev); - kfree(dai_data); -} - -static const struct component_ops q6afe_dai_comp_ops = { - .bind = q6afe_dai_bind, - .unbind = q6afe_dai_unbind, -}; - -static int q6afe_dai_dev_probe(struct platform_device *pdev) -{ - return component_add(&pdev->dev, &q6afe_dai_comp_ops); -} - -static int q6afe_dai_dev_remove(struct platform_device *pdev) -{ - component_del(&pdev->dev, &q6afe_dai_comp_ops); - return 0; -} - static const struct of_device_id q6afe_dai_device_id[] = { { .compatible = "qcom,q6afe-dais" }, {}, @@ -1448,7 +1423,6 @@ static struct platform_driver q6afe_dai_platform_driver = { .of_match_table = of_match_ptr(q6afe_dai_device_id), }, .probe = q6afe_dai_dev_probe, - .remove = q6afe_dai_dev_remove, }; module_platform_driver(q6afe_dai_platform_driver); -- cgit v1.2.3 From f924e4fd89659908107a5529f02c21edb4770dba Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 13 Jul 2018 16:36:30 +0100 Subject: ASoC: qdsp6: q6asm-dai: remove component framework related code Now that the component framework is integrated into the ASoC core, remove any redundant code in this driver. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 35 ++++------------------------------- 1 file changed, 4 insertions(+), 31 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 1196dc7483d2..acf96c6549fc 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -7,7 +7,6 @@ #include #include #include -#include #include #include #include @@ -561,14 +560,15 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = { Q6ASM_FEDAI_DRIVER(8), }; -static int q6asm_dai_bind(struct device *dev, struct device *master, void *data) +static int q6asm_dai_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; struct device_node *node = dev->of_node; struct of_phandle_args args; struct q6asm_dai_data *pdata; int rc; - pdata = kzalloc(sizeof(struct q6asm_dai_data), GFP_KERNEL); + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); if (!pdata) return -ENOMEM; @@ -580,36 +580,10 @@ static int q6asm_dai_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(dev, pdata); - return snd_soc_register_component(dev, &q6asm_fe_dai_component, + return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component, q6asm_fe_dais, ARRAY_SIZE(q6asm_fe_dais)); } -static void q6asm_dai_unbind(struct device *dev, struct device *master, - void *data) -{ - struct q6asm_dai_data *pdata = dev_get_drvdata(dev); - - snd_soc_unregister_component(dev); - - kfree(pdata); - -} - -static const struct component_ops q6asm_dai_comp_ops = { - .bind = q6asm_dai_bind, - .unbind = q6asm_dai_unbind, -}; - -static int q6asm_dai_probe(struct platform_device *pdev) -{ - return component_add(&pdev->dev, &q6asm_dai_comp_ops); -} - -static int q6asm_dai_dev_remove(struct platform_device *pdev) -{ - component_del(&pdev->dev, &q6asm_dai_comp_ops); - return 0; -} static const struct of_device_id q6asm_dai_device_id[] = { { .compatible = "qcom,q6asm-dais" }, @@ -623,7 +597,6 @@ static struct platform_driver q6asm_dai_platform_driver = { .of_match_table = of_match_ptr(q6asm_dai_device_id), }, .probe = q6asm_dai_probe, - .remove = q6asm_dai_dev_remove, }; module_platform_driver(q6asm_dai_platform_driver); -- cgit v1.2.3 From 791940779d651c2219e97702d2245b5420b0c8ae Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 13 Jul 2018 16:36:31 +0100 Subject: ASoC: qdsp6: q6routing: remove component framework related code Now that the component framework is integrated into the ASoC core, remove any redundant code in this driver. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 32 ++++++-------------------------- 1 file changed, 6 insertions(+), 26 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 35269b492761..1d33b00e5b44 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -8,7 +8,6 @@ #include #include #include -#include #include #include #include @@ -977,9 +976,10 @@ static const struct snd_soc_component_driver msm_soc_routing_component = { .num_dapm_routes = ARRAY_SIZE(intercon), }; -static int q6routing_dai_bind(struct device *dev, struct device *master, - void *data) +static int q6pcm_routing_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; + routing_data = kzalloc(sizeof(*routing_data), GFP_KERNEL); if (!routing_data) return -ENOMEM; @@ -989,35 +989,15 @@ static int q6routing_dai_bind(struct device *dev, struct device *master, mutex_init(&routing_data->lock); dev_set_drvdata(dev, routing_data); - return snd_soc_register_component(dev, &msm_soc_routing_component, + return devm_snd_soc_register_component(dev, &msm_soc_routing_component, NULL, 0); } -static void q6routing_dai_unbind(struct device *dev, struct device *master, - void *d) +static int q6pcm_routing_remove(struct platform_device *pdev) { - struct msm_routing_data *data = dev_get_drvdata(dev); - - snd_soc_unregister_component(dev); - - kfree(data); - + kfree(routing_data); routing_data = NULL; -} -static const struct component_ops q6routing_dai_comp_ops = { - .bind = q6routing_dai_bind, - .unbind = q6routing_dai_unbind, -}; - -static int q6pcm_routing_probe(struct platform_device *pdev) -{ - return component_add(&pdev->dev, &q6routing_dai_comp_ops); -} - -static int q6pcm_routing_remove(struct platform_device *pdev) -{ - component_del(&pdev->dev, &q6routing_dai_comp_ops); return 0; } -- cgit v1.2.3 From 90ae7105eaf19342bb11e554059d62b84e01da12 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 13 Jul 2018 16:36:32 +0100 Subject: ASoC: qcom: apq8096: remove component framework related code Now that the component framework is integrated into the ASoC core, remove any redundant code in this driver. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/qcom/apq8096.c | 75 ++++-------------------------------------------- 1 file changed, 6 insertions(+), 69 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index cab8c4ff7c00..a56156281c8d 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -129,17 +129,18 @@ err: return ret; } -static int apq8096_bind(struct device *dev) +static int apq8096_platform_probe(struct platform_device *pdev) { struct snd_soc_card *card; + struct device *dev = &pdev->dev; int ret; card = kzalloc(sizeof(*card), GFP_KERNEL); if (!card) return -ENOMEM; - component_bind_all(dev, card); card->dev = dev; + card->auto_bind = true; dev_set_drvdata(dev, card); ret = apq8096_sbc_parse_of(card); if (ret) { @@ -154,82 +155,18 @@ static int apq8096_bind(struct device *dev) return 0; err: - component_unbind_all(dev, card); kfree(card); return ret; } -static void apq8096_unbind(struct device *dev) +static int apq8096_platform_remove(struct platform_device *pdev) { - struct snd_soc_card *card = dev_get_drvdata(dev); + struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); + card->auto_bind = false; snd_soc_unregister_card(card); - component_unbind_all(dev, card); kfree(card->dai_link); kfree(card); -} - -static const struct component_master_ops apq8096_ops = { - .bind = apq8096_bind, - .unbind = apq8096_unbind, -}; - -static int apq8016_compare_of(struct device *dev, void *data) -{ - return dev->of_node == data; -} - -static void apq8016_release_of(struct device *dev, void *data) -{ - of_node_put(data); -} - -static int add_audio_components(struct device *dev, - struct component_match **matchptr) -{ - struct device_node *np, *platform, *cpu, *node, *dai_node; - - node = dev->of_node; - - for_each_child_of_node(node, np) { - cpu = of_get_child_by_name(np, "cpu"); - if (cpu) { - dai_node = of_parse_phandle(cpu, "sound-dai", 0); - of_node_get(dai_node); - component_match_add_release(dev, matchptr, - apq8016_release_of, - apq8016_compare_of, - dai_node); - } - - platform = of_get_child_by_name(np, "platform"); - if (platform) { - dai_node = of_parse_phandle(platform, "sound-dai", 0); - component_match_add_release(dev, matchptr, - apq8016_release_of, - apq8016_compare_of, - dai_node); - } - } - - return 0; -} - -static int apq8096_platform_probe(struct platform_device *pdev) -{ - struct component_match *match = NULL; - int ret; - - ret = add_audio_components(&pdev->dev, &match); - if (ret) - return ret; - - return component_master_add_with_match(&pdev->dev, &apq8096_ops, match); -} - -static int apq8096_platform_remove(struct platform_device *pdev) -{ - component_master_del(&pdev->dev, &apq8096_ops); return 0; } -- cgit v1.2.3 From bf270262b7b8bb7b48a846c613f74e800abba392 Mon Sep 17 00:00:00 2001 From: Sriram Periyasamy Date: Mon, 16 Jul 2018 15:32:34 +0530 Subject: ASoC: hdac_hdmi: Add documentation for power management Add documentation for power management of HDAC HDMI codec device for various scenarios such as S0/S3, probe and playback use case. Signed-off-by: Sriram Periyasamy Signed-off-by: Sanyog Kale Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hdmi.c | 69 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 69 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 84f7a7a36e4b..30ccc902e5cf 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2103,6 +2103,75 @@ static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) } #ifdef CONFIG_PM +/* + * Power management sequences + * ========================== + * + * The following explains the PM handling of HDAC HDMI with its parent + * device SKL and display power usage + * + * Probe + * ----- + * In SKL probe, + * 1. skl_probe_work() powers up the display (refcount++ -> 1) + * 2. enumerates the codecs on the link + * 3. powers down the display (refcount-- -> 0) + * + * In HDAC HDMI probe, + * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1) + * 2. probe the codec + * 3. put the HDAC HDMI device to runtime suspend + * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * + * Once children are runtime suspended, SKL device also goes to runtime + * suspend + * + * HDMI Playback + * ------------- + * Open HDMI device, + * 1. skl_runtime_resume() invoked + * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) + * + * Close HDMI device, + * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * 2. skl_runtime_suspend() invoked + * + * S0/S3 Cycle with playback in progress + * ------------------------------------- + * When the device is opened for playback, the device is runtime active + * already and the display refcount is 1 as explained above. + * + * Entering to S3, + * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just + * increments the PM runtime usage count of the codec since the device + * is in use already + * 2. skl_suspend() powers down the display (refcount-- -> 0) + * + * Wakeup from S3, + * 1. skl_resume() powers up the display (refcount++ -> 1) + * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just + * decrements the PM runtime usage count of the codec since the device + * is in use already + * + * Once playback is stopped, the display refcount is set to 0 as explained + * above in the HDMI playback sequence. The PM handlings are designed in + * such way that to balance the refcount of display power when the codec + * device put to S3 while playback is going on. + * + * S0/S3 Cycle without playback in progress + * ---------------------------------------- + * Entering to S3, + * 1. hdmi_codec_prepare() invoke the runtime resume of codec + * 2. skl_runtime_resume() invoked + * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) + * 4. skl_suspend() powers down the display (refcount-- -> 0) + * + * Wakeup from S3, + * 1. skl_resume() powers up the display (refcount++ -> 1) + * 2. hdmi_codec_complete() invokes the runtime suspend of codec + * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) + * 4. skl_runtime_suspend() invoked + */ static int hdac_hdmi_runtime_suspend(struct device *dev) { struct hdac_ext_device *edev = to_hda_ext_device(dev); -- cgit v1.2.3 From 6dc4fa179fb86d2c986b2bc8a8377fe4d8c0428d Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:51 +0200 Subject: ASoC: meson: add axg fifo base driver Amlogic's axg SoCs have two types of fifos which are the memory interfaces of the audio subsystem. FRDDR provides the playback interface while TODDR provides the capture interface. The way these fifos operate is very similar. Only a few settings are specific to each. They implement the same pcm driver here and the specifics of each will be dealt with the related DAI driver. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/meson/Kconfig | 8 ++ sound/soc/meson/Makefile | 5 + sound/soc/meson/axg-fifo.c | 341 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/meson/axg-fifo.h | 80 +++++++++++ 6 files changed, 436 insertions(+) create mode 100644 sound/soc/meson/Kconfig create mode 100644 sound/soc/meson/Makefile create mode 100644 sound/soc/meson/axg-fifo.c create mode 100644 sound/soc/meson/axg-fifo.h (limited to 'sound/soc') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 41af6b9cc350..1cf11cf51e1d 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -57,6 +57,7 @@ source "sound/soc/kirkwood/Kconfig" source "sound/soc/img/Kconfig" source "sound/soc/intel/Kconfig" source "sound/soc/mediatek/Kconfig" +source "sound/soc/meson/Kconfig" source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/qcom/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 06389a5385d7..62a5f87c3cfc 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -38,6 +38,7 @@ obj-$(CONFIG_SND_SOC) += jz4740/ obj-$(CONFIG_SND_SOC) += img/ obj-$(CONFIG_SND_SOC) += intel/ obj-$(CONFIG_SND_SOC) += mediatek/ +obj-$(CONFIG_SND_SOC) += meson/ obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig new file mode 100644 index 000000000000..c3eb5e050308 --- /dev/null +++ b/sound/soc/meson/Kconfig @@ -0,0 +1,8 @@ +menu "ASoC support for Amlogic platforms" + depends on ARCH_MESON || COMPILE_TEST + +config SND_MESON_AXG_FIFO + tristate + select REGMAP_MMIO + +endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile new file mode 100644 index 000000000000..75289b6b3ade --- /dev/null +++ b/sound/soc/meson/Makefile @@ -0,0 +1,5 @@ +# SPDX-License-Identifier: (GPL-2.0 OR MIT) + +snd-soc-meson-axg-fifo-objs := axg-fifo.o + +obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c new file mode 100644 index 000000000000..db367d85290f --- /dev/null +++ b/sound/soc/meson/axg-fifo.c @@ -0,0 +1,341 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "axg-fifo.h" + +/* + * This file implements the platform operations common to the playback and + * capture frontend DAI. The logic behind this two types of fifo is very + * similar but some difference exist. + * These differences the respective DAI drivers + */ + +static struct snd_pcm_hardware axg_fifo_hw = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE), + + .formats = AXG_FIFO_FORMATS, + .rate_min = 5512, + .rate_max = 192000, + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .period_bytes_min = AXG_FIFO_MIN_DEPTH, + .period_bytes_max = UINT_MAX, + .periods_min = 2, + .periods_max = UINT_MAX, + + /* No real justification for this */ + .buffer_bytes_max = 1 * 1024 * 1024, +}; + +static struct snd_soc_dai *axg_fifo_dai(struct snd_pcm_substream *ss) +{ + struct snd_soc_pcm_runtime *rtd = ss->private_data; + + return rtd->cpu_dai; +} + +static struct axg_fifo *axg_fifo_data(struct snd_pcm_substream *ss) +{ + struct snd_soc_dai *dai = axg_fifo_dai(ss); + + return snd_soc_dai_get_drvdata(dai); +} + +static struct device *axg_fifo_dev(struct snd_pcm_substream *ss) +{ + struct snd_soc_dai *dai = axg_fifo_dai(ss); + + return dai->dev; +} + +static void __dma_enable(struct axg_fifo *fifo, bool enable) +{ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_DMA_EN, + enable ? CTRL0_DMA_EN : 0); +} + +static int axg_fifo_pcm_trigger(struct snd_pcm_substream *ss, int cmd) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + __dma_enable(fifo, true); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_STOP: + __dma_enable(fifo, false); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t axg_fifo_pcm_pointer(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + struct snd_pcm_runtime *runtime = ss->runtime; + unsigned int addr; + + regmap_read(fifo->map, FIFO_STATUS2, &addr); + + return bytes_to_frames(runtime, addr - (unsigned int)runtime->dma_addr); +} + +static int axg_fifo_pcm_hw_params(struct snd_pcm_substream *ss, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = ss->runtime; + struct axg_fifo *fifo = axg_fifo_data(ss); + dma_addr_t end_ptr; + unsigned int burst_num; + int ret; + + ret = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(params)); + if (ret < 0) + return ret; + + /* Setup dma memory pointers */ + end_ptr = runtime->dma_addr + runtime->dma_bytes - AXG_FIFO_BURST; + regmap_write(fifo->map, FIFO_START_ADDR, runtime->dma_addr); + regmap_write(fifo->map, FIFO_FINISH_ADDR, end_ptr); + + /* Setup interrupt periodicity */ + burst_num = params_period_bytes(params) / AXG_FIFO_BURST; + regmap_write(fifo->map, FIFO_INT_ADDR, burst_num); + + /* Enable block count irq */ + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), + CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT)); + + return 0; +} + +static int axg_fifo_pcm_hw_free(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + + /* Disable the block count irq */ + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), 0); + + return snd_pcm_lib_free_pages(ss); +} + +static void axg_fifo_ack_irq(struct axg_fifo *fifo, u8 mask) +{ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_INT_CLR(FIFO_INT_MASK), + CTRL1_INT_CLR(mask)); + + /* Clear must also be cleared */ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_INT_CLR(FIFO_INT_MASK), + 0); +} + +static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) +{ + struct snd_pcm_substream *ss = dev_id; + struct axg_fifo *fifo = axg_fifo_data(ss); + unsigned int status; + + regmap_read(fifo->map, FIFO_STATUS1, &status); + + status = STATUS1_INT_STS(status) & FIFO_INT_MASK; + if (status & FIFO_INT_COUNT_REPEAT) + snd_pcm_period_elapsed(ss); + else + dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", + status); + + /* Ack irqs */ + axg_fifo_ack_irq(fifo, status); + + return !status ? IRQ_NONE : IRQ_HANDLED; +} + +static int axg_fifo_pcm_open(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + struct device *dev = axg_fifo_dev(ss); + int ret; + + snd_soc_set_runtime_hwparams(ss, &axg_fifo_hw); + + /* + * Make sure the buffer and period size are multiple of the FIFO + * minimum depth size + */ + ret = snd_pcm_hw_constraint_step(ss->runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + AXG_FIFO_MIN_DEPTH); + if (ret) + return ret; + + ret = snd_pcm_hw_constraint_step(ss->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + AXG_FIFO_MIN_DEPTH); + if (ret) + return ret; + + ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0, + dev_name(dev), ss); + + /* Enable pclk to access registers and clock the fifo ip */ + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + /* Setup status2 so it reports the memory pointer */ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_STATUS2_SEL_MASK, + CTRL1_STATUS2_SEL(STATUS2_SEL_DDR_READ)); + + /* Make sure the dma is initially disabled */ + __dma_enable(fifo, false); + + /* Disable irqs until params are ready */ + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_INT_EN(FIFO_INT_MASK), 0); + + /* Clear any pending interrupt */ + axg_fifo_ack_irq(fifo, FIFO_INT_MASK); + + /* Take memory arbitror out of reset */ + ret = reset_control_deassert(fifo->arb); + if (ret) + clk_disable_unprepare(fifo->pclk); + + return ret; +} + +static int axg_fifo_pcm_close(struct snd_pcm_substream *ss) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + int ret; + + /* Put the memory arbitror back in reset */ + ret = reset_control_assert(fifo->arb); + + /* Disable fifo ip and register access */ + clk_disable_unprepare(fifo->pclk); + + /* remove IRQ */ + free_irq(fifo->irq, ss); + + return ret; +} + +const struct snd_pcm_ops axg_fifo_pcm_ops = { + .open = axg_fifo_pcm_open, + .close = axg_fifo_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = axg_fifo_pcm_hw_params, + .hw_free = axg_fifo_pcm_hw_free, + .pointer = axg_fifo_pcm_pointer, + .trigger = axg_fifo_pcm_trigger, +}; +EXPORT_SYMBOL_GPL(axg_fifo_pcm_ops); + +int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = axg_fifo_hw.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages(rtd->pcm->streams[type].substream, + SNDRV_DMA_TYPE_DEV, card->dev, + size, size); +} +EXPORT_SYMBOL_GPL(axg_fifo_pcm_new); + +static const struct regmap_config axg_fifo_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = FIFO_STATUS2, +}; + +int axg_fifo_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + const struct axg_fifo_match_data *data; + struct axg_fifo *fifo; + struct resource *res; + void __iomem *regs; + + data = of_device_get_match_data(dev); + if (!data) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + fifo = devm_kzalloc(dev, sizeof(*fifo), GFP_KERNEL); + if (!fifo) + return -ENOMEM; + platform_set_drvdata(pdev, fifo); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + fifo->map = devm_regmap_init_mmio(dev, regs, &axg_fifo_regmap_cfg); + if (IS_ERR(fifo->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(fifo->map)); + return PTR_ERR(fifo->map); + } + + fifo->pclk = devm_clk_get(dev, NULL); + if (IS_ERR(fifo->pclk)) { + if (PTR_ERR(fifo->pclk) != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %ld\n", + PTR_ERR(fifo->pclk)); + return PTR_ERR(fifo->pclk); + } + + fifo->arb = devm_reset_control_get_exclusive(dev, NULL); + if (IS_ERR(fifo->arb)) { + if (PTR_ERR(fifo->arb) != -EPROBE_DEFER) + dev_err(dev, "failed to get arb reset: %ld\n", + PTR_ERR(fifo->arb)); + return PTR_ERR(fifo->arb); + } + + fifo->irq = of_irq_get(dev->of_node, 0); + if (fifo->irq <= 0) { + dev_err(dev, "failed to get irq: %d\n", fifo->irq); + return fifo->irq; + } + + return devm_snd_soc_register_component(dev, data->component_drv, + data->dai_drv, 1); +} +EXPORT_SYMBOL_GPL(axg_fifo_probe); + +MODULE_DESCRIPTION("Amlogic AXG fifo driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h new file mode 100644 index 000000000000..cb6c4013ca33 --- /dev/null +++ b/sound/soc/meson/axg-fifo.h @@ -0,0 +1,80 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ +/* + * Copyright (c) 2018 BayLibre, SAS. + * Author: Jerome Brunet + */ + +#ifndef _MESON_AXG_FIFO_H +#define _MESON_AXG_FIFO_H + +struct clk; +struct platform_device; +struct regmap; +struct reset_control; + +struct snd_soc_component_driver; +struct snd_soc_dai; +struct snd_soc_dai_driver; +struct snd_pcm_ops; +struct snd_soc_pcm_runtime; + +#define AXG_FIFO_CH_MAX 128 +#define AXG_FIFO_RATES (SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000) +#define AXG_FIFO_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define AXG_FIFO_BURST 8 +#define AXG_FIFO_MIN_CNT 64 +#define AXG_FIFO_MIN_DEPTH (AXG_FIFO_BURST * AXG_FIFO_MIN_CNT) + +#define FIFO_INT_ADDR_FINISH BIT(0) +#define FIFO_INT_ADDR_INT BIT(1) +#define FIFO_INT_COUNT_REPEAT BIT(2) +#define FIFO_INT_COUNT_ONCE BIT(3) +#define FIFO_INT_FIFO_ZERO BIT(4) +#define FIFO_INT_FIFO_DEPTH BIT(5) +#define FIFO_INT_MASK GENMASK(7, 0) + +#define FIFO_CTRL0 0x00 +#define CTRL0_DMA_EN BIT(31) +#define CTRL0_INT_EN(x) ((x) << 16) +#define CTRL0_SEL_MASK GENMASK(2, 0) +#define CTRL0_SEL_SHIFT 0 +#define FIFO_CTRL1 0x04 +#define CTRL1_INT_CLR(x) ((x) << 0) +#define CTRL1_STATUS2_SEL_MASK GENMASK(11, 8) +#define CTRL1_STATUS2_SEL(x) ((x) << 8) +#define STATUS2_SEL_DDR_READ 0 +#define CTRL1_THRESHOLD_MASK GENMASK(23, 16) +#define CTRL1_THRESHOLD(x) ((x) << 16) +#define CTRL1_FRDDR_DEPTH_MASK GENMASK(31, 24) +#define CTRL1_FRDDR_DEPTH(x) ((x) << 24) +#define FIFO_START_ADDR 0x08 +#define FIFO_FINISH_ADDR 0x0c +#define FIFO_INT_ADDR 0x10 +#define FIFO_STATUS1 0x14 +#define STATUS1_INT_STS(x) ((x) << 0) +#define FIFO_STATUS2 0x18 + +struct axg_fifo { + struct regmap *map; + struct clk *pclk; + struct reset_control *arb; + int irq; +}; + +struct axg_fifo_match_data { + const struct snd_soc_component_driver *component_drv; + struct snd_soc_dai_driver *dai_drv; +}; + +extern const struct snd_pcm_ops axg_fifo_pcm_ops; + +int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type); +int axg_fifo_probe(struct platform_device *pdev); + +#endif /* _MESON_AXG_FIFO_H */ -- cgit v1.2.3 From 57d552e3ea76003643b2e771042659ce71bac7c2 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:52 +0200 Subject: ASoC: meson: add axg frddr driver Add the playback memory interface of Amlogic's axg SoCs. This device pulls data from DDR to an internal FIFO. This FIFO is then used to feed TDM and SPDIF Output devices. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 7 +++ sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-frddr.c | 141 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 150 insertions(+) create mode 100644 sound/soc/meson/axg-frddr.c (limited to 'sound/soc') diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index c3eb5e050308..cdd78f62e8d7 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -5,4 +5,11 @@ config SND_MESON_AXG_FIFO tristate select REGMAP_MMIO +config SND_MESON_AXG_FRDDR + tristate "Amlogic AXG Playback FIFO support" + select SND_MESON_AXG_FIFO + help + Select Y or M to add support for the frontend playback interfaces + embedded in the Amlogic AXG SoC family + endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 75289b6b3ade..9c5d7d4a8e33 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -1,5 +1,7 @@ # SPDX-License-Identifier: (GPL-2.0 OR MIT) snd-soc-meson-axg-fifo-objs := axg-fifo.o +snd-soc-meson-axg-frddr-objs := axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o +obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c new file mode 100644 index 000000000000..a6f6f6a2eca8 --- /dev/null +++ b/sound/soc/meson/axg-frddr.c @@ -0,0 +1,141 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +/* This driver implements the frontend playback DAI of AXG based SoCs */ + +#include +#include +#include +#include +#include +#include + +#include "axg-fifo.h" + +#define CTRL0_FRDDR_PP_MODE BIT(30) + +static int axg_frddr_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int fifo_depth, fifo_threshold; + int ret; + + /* Enable pclk to access registers and clock the fifo ip */ + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + /* Apply single buffer mode to the interface */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_FRDDR_PP_MODE, 0); + + /* + * TODO: We could adapt the fifo depth and the fifo threshold + * depending on the expected memory throughput and lantencies + * For now, we'll just use the same values as the vendor kernel + * Depth and threshold are zero based. + */ + fifo_depth = AXG_FIFO_MIN_CNT - 1; + fifo_threshold = (AXG_FIFO_MIN_CNT / 2) - 1; + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_FRDDR_DEPTH_MASK | CTRL1_THRESHOLD_MASK, + CTRL1_FRDDR_DEPTH(fifo_depth) | + CTRL1_THRESHOLD(fifo_threshold)); + + return 0; +} + +static void axg_frddr_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(fifo->pclk); +} + +static int axg_frddr_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_PLAYBACK); +} + +static const struct snd_soc_dai_ops axg_frddr_ops = { + .startup = axg_frddr_dai_startup, + .shutdown = axg_frddr_dai_shutdown, +}; + +static struct snd_soc_dai_driver axg_frddr_dai_drv = { + .name = "FRDDR", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .rates = AXG_FIFO_RATES, + .formats = AXG_FIFO_FORMATS, + }, + .ops = &axg_frddr_ops, + .pcm_new = axg_frddr_pcm_new, +}; + +static const char * const axg_frddr_sel_texts[] = { + "OUT 0", "OUT 1", "OUT 2", "OUT 3" +}; + +static SOC_ENUM_SINGLE_DECL(axg_frddr_sel_enum, FIFO_CTRL0, CTRL0_SEL_SHIFT, + axg_frddr_sel_texts); + +static const struct snd_kcontrol_new axg_frddr_out_demux = + SOC_DAPM_ENUM("Output Sink", axg_frddr_sel_enum); + +static const struct snd_soc_dapm_widget axg_frddr_dapm_widgets[] = { + SND_SOC_DAPM_DEMUX("SINK SEL", SND_SOC_NOPM, 0, 0, + &axg_frddr_out_demux), + SND_SOC_DAPM_AIF_OUT("OUT 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 3", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_frddr_dapm_routes[] = { + { "SINK SEL", NULL, "Playback" }, + { "OUT 0", "OUT 0", "SINK SEL" }, + { "OUT 1", "OUT 1", "SINK SEL" }, + { "OUT 2", "OUT 2", "SINK SEL" }, + { "OUT 3", "OUT 3", "SINK SEL" }, +}; + +static const struct snd_soc_component_driver axg_frddr_component_drv = { + .dapm_widgets = axg_frddr_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_frddr_dapm_widgets), + .dapm_routes = axg_frddr_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_frddr_dapm_routes), + .ops = &axg_fifo_pcm_ops +}; + +static const struct axg_fifo_match_data axg_frddr_match_data = { + .component_drv = &axg_frddr_component_drv, + .dai_drv = &axg_frddr_dai_drv +}; + +static const struct of_device_id axg_frddr_of_match[] = { + { + .compatible = "amlogic,axg-frddr", + .data = &axg_frddr_match_data, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_frddr_of_match); + +static struct platform_driver axg_frddr_pdrv = { + .probe = axg_fifo_probe, + .driver = { + .name = "axg-frddr", + .of_match_table = axg_frddr_of_match, + }, +}; +module_platform_driver(axg_frddr_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG playback fifo driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 7ed4877b403c9343a8e2c7581d9bcfceef0f40cf Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:53 +0200 Subject: ASoC: meson: add axg toddr driver Add the capture memory interface of Amlogic's axg SoCs. TDM, SPDIF or PDM input devices place audio samples inside this FIFO. The FIFO content is then pushed to DDR Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 7 ++ sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-toddr.c | 199 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 208 insertions(+) create mode 100644 sound/soc/meson/axg-toddr.c (limited to 'sound/soc') diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index cdd78f62e8d7..3916060edcae 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -12,4 +12,11 @@ config SND_MESON_AXG_FRDDR Select Y or M to add support for the frontend playback interfaces embedded in the Amlogic AXG SoC family +config SND_MESON_AXG_TODDR + tristate "Amlogic AXG Capture FIFO support" + select SND_MESON_AXG_FIFO + help + Select Y or M to add support for the frontend capture interfaces + embedded in the Amlogic AXG SoC family + endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 9c5d7d4a8e33..12edf2db2a24 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -2,6 +2,8 @@ snd-soc-meson-axg-fifo-objs := axg-fifo.o snd-soc-meson-axg-frddr-objs := axg-frddr.o +snd-soc-meson-axg-toddr-objs := axg-toddr.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o +obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c new file mode 100644 index 000000000000..c2c9bb312586 --- /dev/null +++ b/sound/soc/meson/axg-toddr.c @@ -0,0 +1,199 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +/* This driver implements the frontend capture DAI of AXG based SoCs */ + +#include +#include +#include +#include +#include +#include +#include + +#include "axg-fifo.h" + +#define CTRL0_TODDR_SEL_RESAMPLE BIT(30) +#define CTRL0_TODDR_EXT_SIGNED BIT(29) +#define CTRL0_TODDR_PP_MODE BIT(28) +#define CTRL0_TODDR_TYPE_MASK GENMASK(15, 13) +#define CTRL0_TODDR_TYPE(x) ((x) << 13) +#define CTRL0_TODDR_MSB_POS_MASK GENMASK(12, 8) +#define CTRL0_TODDR_MSB_POS(x) ((x) << 8) +#define CTRL0_TODDR_LSB_POS_MASK GENMASK(7, 3) +#define CTRL0_TODDR_LSB_POS(x) ((x) << 3) + +static int axg_toddr_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_CAPTURE); +} + +static int axg_toddr_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int type, width, msb = 31; + + /* + * NOTE: + * Almost all backend will place the MSB at bit 31, except SPDIF Input + * which will put it at index 28. When adding support for the SPDIF + * Input, we'll need to find which type of backend we are connected to. + */ + + switch (params_physical_width(params)) { + case 8: + type = 0; /* 8 samples of 8 bits */ + break; + case 16: + type = 2; /* 4 samples of 16 bits - right justified */ + break; + case 32: + type = 4; /* 2 samples of 32 bits - right justified */ + break; + default: + return -EINVAL; + } + + width = params_width(params); + + regmap_update_bits(fifo->map, FIFO_CTRL0, + CTRL0_TODDR_TYPE_MASK | + CTRL0_TODDR_MSB_POS_MASK | + CTRL0_TODDR_LSB_POS_MASK, + CTRL0_TODDR_TYPE(type) | + CTRL0_TODDR_MSB_POS(msb) | + CTRL0_TODDR_LSB_POS(msb - (width - 1))); + + return 0; +} + +static int axg_toddr_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int fifo_threshold; + int ret; + + /* Enable pclk to access registers and clock the fifo ip */ + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + /* Select orginal data - resampling not supported ATM */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SEL_RESAMPLE, 0); + + /* Only signed format are supported ATM */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_EXT_SIGNED, + CTRL0_TODDR_EXT_SIGNED); + + /* Apply single buffer mode to the interface */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_PP_MODE, 0); + + /* TODDR does not have a configurable fifo depth */ + fifo_threshold = AXG_FIFO_MIN_CNT - 1; + regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_THRESHOLD_MASK, + CTRL1_THRESHOLD(fifo_threshold)); + + return 0; +} + +static void axg_toddr_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(fifo->pclk); +} + +static const struct snd_soc_dai_ops axg_toddr_ops = { + .hw_params = axg_toddr_dai_hw_params, + .startup = axg_toddr_dai_startup, + .shutdown = axg_toddr_dai_shutdown, +}; + +static struct snd_soc_dai_driver axg_toddr_dai_drv = { + .name = "TODDR", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .rates = AXG_FIFO_RATES, + .formats = AXG_FIFO_FORMATS, + }, + .ops = &axg_toddr_ops, + .pcm_new = axg_toddr_pcm_new, +}; + +static const char * const axg_toddr_sel_texts[] = { + "IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 6" +}; + +static const unsigned int axg_toddr_sel_values[] = { + 0, 1, 2, 3, 4, 6 +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(axg_toddr_sel_enum, FIFO_CTRL0, + CTRL0_SEL_SHIFT, CTRL0_SEL_MASK, + axg_toddr_sel_texts, axg_toddr_sel_values); + +static const struct snd_kcontrol_new axg_toddr_in_mux = + SOC_DAPM_ENUM("Input Source", axg_toddr_sel_enum); + +static const struct snd_soc_dapm_widget axg_toddr_dapm_widgets[] = { + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_toddr_in_mux), + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 6", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_toddr_dapm_routes[] = { + { "Capture", NULL, "SRC SEL" }, + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "SRC SEL", "IN 3", "IN 3" }, + { "SRC SEL", "IN 4", "IN 4" }, + { "SRC SEL", "IN 6", "IN 6" }, +}; + +static const struct snd_soc_component_driver axg_toddr_component_drv = { + .dapm_widgets = axg_toddr_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_toddr_dapm_widgets), + .dapm_routes = axg_toddr_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_toddr_dapm_routes), + .ops = &axg_fifo_pcm_ops +}; + +static const struct axg_fifo_match_data axg_toddr_match_data = { + .component_drv = &axg_toddr_component_drv, + .dai_drv = &axg_toddr_dai_drv +}; + +static const struct of_device_id axg_toddr_of_match[] = { + { + .compatible = "amlogic,axg-toddr", + .data = &axg_toddr_match_data, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_toddr_of_match); + +static struct platform_driver axg_toddr_pdrv = { + .probe = axg_fifo_probe, + .driver = { + .name = "axg-toddr", + .of_match_table = axg_toddr_of_match, + }, +}; +module_platform_driver(axg_toddr_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG capture fifo driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 53eb4b7aaa045e23b6e8edb0ae0d047a4a3612ef Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:55 +0200 Subject: ASoC: meson: add axg spdif output Add support for the spdif output serializer of the axg SoC family Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 7 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-spdifout.c | 456 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 465 insertions(+) create mode 100644 sound/soc/meson/axg-spdifout.c (limited to 'sound/soc') diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 3916060edcae..9408214b5854 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -19,4 +19,11 @@ config SND_MESON_AXG_TODDR Select Y or M to add support for the frontend capture interfaces embedded in the Amlogic AXG SoC family +config SND_MESON_AXG_SPDIFOUT + tristate "Amlogic AXG SPDIF Output Support" + imply SND_SOC_SPDIF + help + Select Y or M to add support for SPDIF output serializer embedded + in the Amlogic AXG SoC family + endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 12edf2db2a24..d51ae045ef08 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -3,7 +3,9 @@ snd-soc-meson-axg-fifo-objs := axg-fifo.o snd-soc-meson-axg-frddr-objs := axg-frddr.o snd-soc-meson-axg-toddr-objs := axg-toddr.o +snd-soc-meson-axg-spdifout-objs := axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o +obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c new file mode 100644 index 000000000000..9dea528053ad --- /dev/null +++ b/sound/soc/meson/axg-spdifout.c @@ -0,0 +1,456 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include +#include +#include +#include + +/* + * NOTE: + * The meaning of bits SPDIFOUT_CTRL0_XXX_SEL is actually the opposite + * of what the documentation says. Manual control on V, U and C bits is + * applied when the related sel bits are cleared + */ + +#define SPDIFOUT_STAT 0x00 +#define SPDIFOUT_GAIN0 0x04 +#define SPDIFOUT_GAIN1 0x08 +#define SPDIFOUT_CTRL0 0x0c +#define SPDIFOUT_CTRL0_EN BIT(31) +#define SPDIFOUT_CTRL0_RST_OUT BIT(29) +#define SPDIFOUT_CTRL0_RST_IN BIT(28) +#define SPDIFOUT_CTRL0_USEL BIT(26) +#define SPDIFOUT_CTRL0_USET BIT(25) +#define SPDIFOUT_CTRL0_CHSTS_SEL BIT(24) +#define SPDIFOUT_CTRL0_DATA_SEL BIT(20) +#define SPDIFOUT_CTRL0_MSB_FIRST BIT(19) +#define SPDIFOUT_CTRL0_VSEL BIT(18) +#define SPDIFOUT_CTRL0_VSET BIT(17) +#define SPDIFOUT_CTRL0_MASK_MASK GENMASK(11, 4) +#define SPDIFOUT_CTRL0_MASK(x) ((x) << 4) +#define SPDIFOUT_CTRL1 0x10 +#define SPDIFOUT_CTRL1_MSB_POS_MASK GENMASK(12, 8) +#define SPDIFOUT_CTRL1_MSB_POS(x) ((x) << 8) +#define SPDIFOUT_CTRL1_TYPE_MASK GENMASK(6, 4) +#define SPDIFOUT_CTRL1_TYPE(x) ((x) << 4) +#define SPDIFOUT_PREAMB 0x14 +#define SPDIFOUT_SWAP 0x18 +#define SPDIFOUT_CHSTS0 0x1c +#define SPDIFOUT_CHSTS1 0x20 +#define SPDIFOUT_CHSTS2 0x24 +#define SPDIFOUT_CHSTS3 0x28 +#define SPDIFOUT_CHSTS4 0x2c +#define SPDIFOUT_CHSTS5 0x30 +#define SPDIFOUT_CHSTS6 0x34 +#define SPDIFOUT_CHSTS7 0x38 +#define SPDIFOUT_CHSTS8 0x3c +#define SPDIFOUT_CHSTS9 0x40 +#define SPDIFOUT_CHSTSA 0x44 +#define SPDIFOUT_CHSTSB 0x48 +#define SPDIFOUT_MUTE_VAL 0x4c + +struct axg_spdifout { + struct regmap *map; + struct clk *mclk; + struct clk *pclk; +}; + +static void axg_spdifout_enable(struct regmap *map) +{ + /* Apply both reset */ + regmap_update_bits(map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_RST_OUT | SPDIFOUT_CTRL0_RST_IN, + 0); + + /* Clear out reset before in reset */ + regmap_update_bits(map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_RST_OUT, SPDIFOUT_CTRL0_RST_OUT); + regmap_update_bits(map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_RST_IN, SPDIFOUT_CTRL0_RST_IN); + + /* Enable spdifout */ + regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN, + SPDIFOUT_CTRL0_EN); +} + +static void axg_spdifout_disable(struct regmap *map) +{ + regmap_update_bits(map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_EN, 0); +} + +static int axg_spdifout_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + axg_spdifout_enable(priv->map); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + axg_spdifout_disable(priv->map); + return 0; + + default: + return -EINVAL; + } +} + +static int axg_spdifout_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + + /* Use spdif valid bit to perform digital mute */ + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_VSET, + mute ? SPDIFOUT_CTRL0_VSET : 0); + + return 0; +} + +static int axg_spdifout_sample_fmt(struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + unsigned int val; + + /* Set the samples spdifout will pull from the FIFO */ + switch (params_channels(params)) { + case 1: + val = SPDIFOUT_CTRL0_MASK(0x1); + break; + case 2: + val = SPDIFOUT_CTRL0_MASK(0x3); + break; + default: + dev_err(dai->dev, "too many channels for spdif dai: %u\n", + params_channels(params)); + return -EINVAL; + } + + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_MASK_MASK, val); + + /* FIFO data are arranged in chunks of 64bits */ + switch (params_physical_width(params)) { + case 8: + /* 8 samples of 8 bits */ + val = SPDIFOUT_CTRL1_TYPE(0); + break; + case 16: + /* 4 samples of 16 bits - right justified */ + val = SPDIFOUT_CTRL1_TYPE(2); + break; + case 32: + /* 2 samples of 32 bits - right justified */ + val = SPDIFOUT_CTRL1_TYPE(4); + break; + default: + dev_err(dai->dev, "Unsupported physical width: %u\n", + params_physical_width(params)); + return -EINVAL; + } + + /* Position of the MSB in FIFO samples */ + val |= SPDIFOUT_CTRL1_MSB_POS(params_width(params) - 1); + + regmap_update_bits(priv->map, SPDIFOUT_CTRL1, + SPDIFOUT_CTRL1_MSB_POS_MASK | + SPDIFOUT_CTRL1_TYPE_MASK, val); + + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL, + 0); + + return 0; +} + +static int axg_spdifout_set_chsts(struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + unsigned int offset; + int ret; + u8 cs[4]; + u32 val; + + ret = snd_pcm_create_iec958_consumer_hw_params(params, cs, 4); + if (ret < 0) { + dev_err(dai->dev, "Creating IEC958 channel status failed %d\n", + ret); + return ret; + } + val = cs[0] | cs[1] << 8 | cs[2] << 16 | cs[3] << 24; + + /* Setup channel status A bits [31 - 0]*/ + regmap_write(priv->map, SPDIFOUT_CHSTS0, val); + + /* Clear channel status A bits [191 - 32] */ + for (offset = SPDIFOUT_CHSTS1; offset <= SPDIFOUT_CHSTS5; + offset += regmap_get_reg_stride(priv->map)) + regmap_write(priv->map, offset, 0); + + /* Setup channel status B bits [31 - 0]*/ + regmap_write(priv->map, SPDIFOUT_CHSTS6, val); + + /* Clear channel status B bits [191 - 32] */ + for (offset = SPDIFOUT_CHSTS7; offset <= SPDIFOUT_CHSTSB; + offset += regmap_get_reg_stride(priv->map)) + regmap_write(priv->map, offset, 0); + + return 0; +} + +static int axg_spdifout_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + unsigned int rate = params_rate(params); + int ret; + + /* 2 * 32bits per subframe * 2 channels = 128 */ + ret = clk_set_rate(priv->mclk, rate * 128); + if (ret) { + dev_err(dai->dev, "failed to set spdif clock\n"); + return ret; + } + + ret = axg_spdifout_sample_fmt(params, dai); + if (ret) { + dev_err(dai->dev, "failed to setup sample format\n"); + return ret; + } + + ret = axg_spdifout_set_chsts(params, dai); + if (ret) { + dev_err(dai->dev, "failed to setup channel status words\n"); + return ret; + } + + return 0; +} + +static int axg_spdifout_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + int ret; + + /* Clock the spdif output block */ + ret = clk_prepare_enable(priv->pclk); + if (ret) { + dev_err(dai->dev, "failed to enable pclk\n"); + return ret; + } + + /* Make sure the block is initially stopped */ + axg_spdifout_disable(priv->map); + + /* Insert data from bit 27 lsb first */ + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_MSB_FIRST | SPDIFOUT_CTRL0_DATA_SEL, + 0); + + /* Manual control of V, C and U, U = 0 */ + regmap_update_bits(priv->map, SPDIFOUT_CTRL0, + SPDIFOUT_CTRL0_CHSTS_SEL | SPDIFOUT_CTRL0_VSEL | + SPDIFOUT_CTRL0_USEL | SPDIFOUT_CTRL0_USET, + 0); + + /* Static SWAP configuration ATM */ + regmap_write(priv->map, SPDIFOUT_SWAP, 0x10); + + return 0; +} + +static void axg_spdifout_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(priv->pclk); +} + +static const struct snd_soc_dai_ops axg_spdifout_ops = { + .trigger = axg_spdifout_trigger, + .digital_mute = axg_spdifout_digital_mute, + .hw_params = axg_spdifout_hw_params, + .startup = axg_spdifout_startup, + .shutdown = axg_spdifout_shutdown, +}; + +static struct snd_soc_dai_driver axg_spdifout_dai_drv[] = { + { + .name = "SPDIF Output", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000), + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &axg_spdifout_ops, + }, +}; + +static const char * const spdifout_sel_texts[] = { + "IN 0", "IN 1", "IN 2", +}; + +static SOC_ENUM_SINGLE_DECL(axg_spdifout_sel_enum, SPDIFOUT_CTRL1, 24, + spdifout_sel_texts); + +static const struct snd_kcontrol_new axg_spdifout_in_mux = + SOC_DAPM_ENUM("Input Source", axg_spdifout_sel_enum); + +static const struct snd_soc_dapm_widget axg_spdifout_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_spdifout_in_mux), +}; + +static const struct snd_soc_dapm_route axg_spdifout_dapm_routes[] = { + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "Playback", NULL, "SRC SEL" }, +}; + +static const struct snd_kcontrol_new axg_spdifout_controls[] = { + SOC_DOUBLE("Playback Volume", SPDIFOUT_GAIN0, 0, 8, 255, 0), + SOC_DOUBLE("Playback Switch", SPDIFOUT_CTRL0, 22, 21, 1, 1), + SOC_SINGLE("Playback Gain Enable Switch", + SPDIFOUT_CTRL1, 26, 1, 0), + SOC_SINGLE("Playback Channels Mix Switch", + SPDIFOUT_CTRL0, 23, 1, 0), +}; + +static int axg_spdifout_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct axg_spdifout *priv = snd_soc_component_get_drvdata(component); + enum snd_soc_bias_level now = + snd_soc_component_get_bias_level(component); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (now == SND_SOC_BIAS_STANDBY) + ret = clk_prepare_enable(priv->mclk); + break; + + case SND_SOC_BIAS_STANDBY: + if (now == SND_SOC_BIAS_PREPARE) + clk_disable_unprepare(priv->mclk); + break; + + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_ON: + break; + } + + return ret; +} + +static const struct snd_soc_component_driver axg_spdifout_component_drv = { + .controls = axg_spdifout_controls, + .num_controls = ARRAY_SIZE(axg_spdifout_controls), + .dapm_widgets = axg_spdifout_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_spdifout_dapm_widgets), + .dapm_routes = axg_spdifout_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_spdifout_dapm_routes), + .set_bias_level = axg_spdifout_set_bias_level, +}; + +static const struct regmap_config axg_spdifout_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = SPDIFOUT_MUTE_VAL, +}; + +static const struct of_device_id axg_spdifout_of_match[] = { + { .compatible = "amlogic,axg-spdifout", }, + {} +}; +MODULE_DEVICE_TABLE(of, axg_spdifout_of_match); + +static int axg_spdifout_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct axg_spdifout *priv; + struct resource *res; + void __iomem *regs; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->map = devm_regmap_init_mmio(dev, regs, &axg_spdifout_regmap_cfg); + if (IS_ERR(priv->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(priv->map)); + return PTR_ERR(priv->map); + } + + priv->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(priv->pclk)) { + ret = PTR_ERR(priv->pclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %d\n", ret); + return ret; + } + + priv->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(priv->mclk)) { + ret = PTR_ERR(priv->mclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get mclk: %d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(dev, &axg_spdifout_component_drv, + axg_spdifout_dai_drv, ARRAY_SIZE(axg_spdifout_dai_drv)); +} + +static struct platform_driver axg_spdifout_pdrv = { + .probe = axg_spdifout_probe, + .driver = { + .name = "axg-spdifout", + .of_match_table = axg_spdifout_of_match, + }, +}; +module_platform_driver(axg_spdifout_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG SPDIF Output driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 868e49a4a00afaca07d2c450a02e49581eaece6c Mon Sep 17 00:00:00 2001 From: Stuart Henderson Date: Thu, 19 Jul 2018 11:50:37 +0100 Subject: ASoC: wm_adsp: Ensure DSP boot work complete before preloader_put return All controls derived from the loaded firmware should be created prior to returning from the preloader's put function, such that they are immediately available to user-space. Signed-off-by: Stuart Henderson Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b7b914963c62..4e7f8525449e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2672,6 +2672,8 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, snd_soc_dapm_sync(dapm); + flush_work(&dsp->boot_work); + return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put); -- cgit v1.2.3 From 517ee74e1b3124b696f293aa4e220418f8125b4c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 Jul 2018 11:50:35 +0100 Subject: ASoC: wm_adsp: Correct algorithm list allocation size Commit 6396bb221514 ("treewide: kzalloc() -> kcalloc()") was overlooked when doing some refactoring to the algorithm list handling, which lead to twice as much buffer being allocated as required for reading the algorithm list. A kcalloc is no longer appropriate since the allocation size is now in bytes not registers, as such change back to kzalloc. Fixes: 7f7cca08abf4 ("ASoC: wm_adsp: Simplify handling of alg offset and length") Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 07c17acc8a4f..99108d18de8d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1906,7 +1906,7 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, /* Convert length from DSP words to bytes */ len *= sizeof(u32); - alg = kcalloc(len, 2, GFP_KERNEL | GFP_DMA); + alg = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!alg) return ERR_PTR(-ENOMEM); -- cgit v1.2.3 From b7ede5af62ab6bfad0980fad58e82d3fb56866df Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 19 Jul 2018 11:50:36 +0100 Subject: ASoC: wm_adsp: Take prefix into account in control name length Currently when creating ALSA control names for the DSP the length of any prefix applied to the CODEC is not taken into account. Whilst this is mostly harmless it does result in ALSA doing the truncation of the control names and printing a warning. It is better to have the driver do the truncation so it can truncate from the start of parameter name itself to give a greater chance of the result maintain a unique name. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 99108d18de8d..eec73c98a141 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1343,6 +1343,9 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, int avail = SNDRV_CTL_ELEM_ID_NAME_MAXLEN - ret - 2; int skip = 0; + if (dsp->component->name_prefix) + avail -= strlen(dsp->component->name_prefix) + 1; + if (subname_len > avail) skip = subname_len - avail; -- cgit v1.2.3 From 3bbc2705a3d132b9a86a0e4083f82a2b3c9bfdfd Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 19 Jul 2018 11:50:38 +0100 Subject: ASoC: wm_adsp: Allow up to 8 channels for voice control Newer voice control firmwares can capture multiple audio channels. Allow up to 8 channels for future-proofing. Signed-off-by: Richard Fitzgerald Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index eec73c98a141..aeb1b8c27670 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -418,7 +418,7 @@ static const struct wm_adsp_fw_caps ctrl_caps[] = { { .id = SND_AUDIOCODEC_BESPOKE, .desc = { - .max_ch = 1, + .max_ch = 8, .sample_rates = { 16000 }, .num_sample_rates = 1, .formats = SNDRV_PCM_FMTBIT_S16_LE, -- cgit v1.2.3 From d52ed4b0bc73c1c7816f5b7a36229a95acfc76c8 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 19 Jul 2018 11:50:39 +0100 Subject: ASoC: wm_adsp: Parse HOST_BUFFER controls Currently the compressed streams in DSP firmwares are identified essentially by looking at a fixed location inside the firmware. This is fragile and also limits things to a single compressed stream. Here a new form of firmware parameter is added, the HOST_BUFFER which identifies a compressed stream from meta-data in the firmware file. This is more robust and allows for the possiblity of using multiple streams per core in the future. Currently the implementation is still limited to a single stream and will use the first HOST_BUFFER parameter encountered. If there aren't any HOST_BUFFER parameters it will fall back to the legacy way of finding the host buffer. Signed-off-by: Richard Fitzgerald Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 66 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wmfw.h | 1 + 2 files changed, 66 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index aeb1b8c27670..e39b0e0b04df 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1607,6 +1607,15 @@ static int wm_adsp_parse_coeff(struct wm_adsp *dsp, if (ret) return -EINVAL; break; + case WMFW_CTL_TYPE_HOST_BUFFER: + ret = wm_adsp_check_coeff_flags(dsp, &coeff_blk, + WMFW_CTL_FLAG_SYS | + WMFW_CTL_FLAG_VOLATILE | + WMFW_CTL_FLAG_READABLE, + 0); + if (ret) + return -EINVAL; + break; default: adsp_err(dsp, "Unknown control type: %d\n", coeff_blk.ctl_type); @@ -3200,7 +3209,7 @@ static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf, buf->host_buf_ptr + field_offset, data); } -static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf) +static int wm_adsp_legacy_host_buf_addr(struct wm_adsp_compr_buf *buf) { struct wm_adsp_alg_region *alg_region; struct wm_adsp *dsp = buf->dsp; @@ -3239,6 +3248,61 @@ static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf) return 0; } +static struct wm_coeff_ctl * +wm_adsp_find_host_buffer_ctrl(struct wm_adsp_compr_buf *buf) +{ + struct wm_adsp *dsp = buf->dsp; + struct wm_coeff_ctl *ctl; + + list_for_each_entry(ctl, &dsp->ctl_list, list) { + if (ctl->type != WMFW_CTL_TYPE_HOST_BUFFER) + continue; + + if (!ctl->enabled) + continue; + + return ctl; + } + + return NULL; +} + +static int wm_adsp_buffer_locate(struct wm_adsp_compr_buf *buf) +{ + struct wm_adsp *dsp = buf->dsp; + struct wm_coeff_ctl *ctl; + unsigned int reg; + u32 val; + int i, ret; + + ctl = wm_adsp_find_host_buffer_ctrl(buf); + if (!ctl) + return wm_adsp_legacy_host_buf_addr(buf); + + ret = wm_coeff_base_reg(ctl, ®); + if (ret) + return ret; + + for (i = 0; i < 5; ++i) { + ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val)); + if (ret < 0) + return ret; + + if (val) + break; + + usleep_range(1000, 2000); + } + + if (!val) + return -EIO; + + buf->host_buf_ptr = be32_to_cpu(val); + adsp_dbg(dsp, "host_buf_ptr=%x\n", buf->host_buf_ptr); + + return 0; +} + static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) { const struct wm_adsp_fw_caps *caps = wm_adsp_fw[buf->dsp->fw].caps; diff --git a/sound/soc/codecs/wmfw.h b/sound/soc/codecs/wmfw.h index ec78b9da020f..0c3f50acb8b1 100644 --- a/sound/soc/codecs/wmfw.h +++ b/sound/soc/codecs/wmfw.h @@ -29,6 +29,7 @@ /* Non-ALSA coefficient types start at 0x1000 */ #define WMFW_CTL_TYPE_ACKED 0x1000 /* acked control */ #define WMFW_CTL_TYPE_HOSTEVENT 0x1001 /* event control */ +#define WMFW_CTL_TYPE_HOST_BUFFER 0x1002 /* host buffer pointer */ struct wmfw_header { char magic[4]; -- cgit v1.2.3 From eea1662525bd4a158a67ac836b2a1fd9cf77cc81 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 18 Jul 2018 22:55:37 +0200 Subject: ASoC: rt5651: Add IN3 Boost volume control Add a mixer control for the IN3 Boost volume, IN3 is used for the headset mic on most devices, so this is necessary to control the headset mic volume. Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/codecs/rt5651.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 0462049e739c..985852fd9723 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -331,11 +331,13 @@ static const struct snd_kcontrol_new rt5651_snd_controls[] = { SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5651_DAC2_DIG_VOL, RT5651_L_VOL_SFT, RT5651_R_VOL_SFT, 175, 0, dac_vol_tlv), - /* IN1/IN2 Control */ + /* IN1/IN2/IN3 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5651_IN1_IN2, RT5651_BST_SFT1, 8, 0, bst_tlv), SOC_SINGLE_TLV("IN2 Boost", RT5651_IN1_IN2, RT5651_BST_SFT2, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN3 Boost", RT5651_IN3, + RT5651_BST_SFT1, 8, 0, bst_tlv), /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5651_INL1_INR1_VOL, RT5651_INL_VOL_SFT, RT5651_INR_VOL_SFT, -- cgit v1.2.3 From 0a3badd141f78535315cca9ff5062a7ebf414281 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 18 Jul 2018 22:55:38 +0200 Subject: ASoC: Intel: bytcr_rt5651: Fix using the wrong GPIO for the ext-amp on some boards Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the GpioIo one for the ext-amp-enable-gpio. So far we've been assuming that the GpioIo one always comes first, this commit adds code to detect which one comes first and to add the right gpio-mapping. This fixes sound not working on the Vios LTH17 laptop. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 69 +++++++++++++++++++++++++++++++++-- 1 file changed, 65 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index b687043c8425..601e47c33ba8 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -747,13 +747,74 @@ static const struct x86_cpu_id cherrytrail_cpu_ids[] = { {} }; -static const struct acpi_gpio_params ext_amp_enable_gpios = { 0, 0, false }; +static const struct acpi_gpio_params first_gpio = { 0, 0, false }; +static const struct acpi_gpio_params second_gpio = { 1, 0, false }; -static const struct acpi_gpio_mapping byt_rt5651_gpios[] = { - { "ext-amp-enable-gpios", &ext_amp_enable_gpios, 1 }, +static const struct acpi_gpio_mapping byt_rt5651_amp_en_first[] = { + { "ext-amp-enable-gpios", &first_gpio, 1 }, { }, }; +static const struct acpi_gpio_mapping byt_rt5651_amp_en_second[] = { + { "ext-amp-enable-gpios", &second_gpio, 1 }, + { }, +}; + +/* + * Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other + * boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the + * GpioIo one for the ext-amp-enable-gpio and both count for the index in + * acpi_gpio_params index. So we have 2 different mappings and the code + * below figures out which one to use. + */ +struct byt_rt5651_acpi_resource_data { + int gpio_count; + int gpio_int_idx; +}; + +static int snd_byt_rt5651_acpi_resource(struct acpi_resource *ares, void *arg) +{ + struct byt_rt5651_acpi_resource_data *data = arg; + + if (ares->type != ACPI_RESOURCE_TYPE_GPIO) + return 0; + + if (ares->data.gpio.connection_type == ACPI_RESOURCE_GPIO_TYPE_INT) + data->gpio_int_idx = data->gpio_count; + + data->gpio_count++; + return 0; +} + +static void snd_byt_rt5651_mc_add_amp_en_gpio_mapping(struct device *codec) +{ + struct byt_rt5651_acpi_resource_data data = { 0, -1 }; + LIST_HEAD(resources); + int ret; + + ret = acpi_dev_get_resources(ACPI_COMPANION(codec), &resources, + snd_byt_rt5651_acpi_resource, &data); + if (ret < 0) { + dev_warn(codec, "Failed to get ACPI resources, not adding external amplifier GPIO mapping\n"); + return; + } + + /* All info we need is gathered during the walk */ + acpi_dev_free_resource_list(&resources); + + switch (data.gpio_int_idx) { + case 0: + devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_second); + break; + case 1: + devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_first); + break; + default: + dev_warn(codec, "Unknown GpioInt index %d, not adding external amplifier GPIO mapping\n", + data.gpio_int_idx); + } +} + struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ u64 aif_value; /* 1: AIF1, 2: AIF2 */ u64 mclock_value; /* usually 25MHz (0x17d7940), ignored */ @@ -876,7 +937,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* Cherry Trail devices use an external amplifier enable gpio */ if (x86_match_cpu(cherrytrail_cpu_ids)) { - devm_acpi_dev_add_driver_gpios(codec_dev, byt_rt5651_gpios); + snd_byt_rt5651_mc_add_amp_en_gpio_mapping(codec_dev); priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child( &pdev->dev, "ext-amp-enable", 0, codec_dev->fwnode, -- cgit v1.2.3 From 8627fb257e1673d2c2277494545642921097da86 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 18 Jul 2018 22:55:39 +0200 Subject: ASoC: Intel: bytcr_rt5651: Set OVCD limit for VIOS LTH17 to 2000uA With the default over current detect limit of 1500uA headsets on often get detected as headphones on the VIOS LTH17 and even when detected as headset the OVCD current triggers often while plugged in, resulting in false-positive button press detection. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 601e47c33ba8..53ac97c15fc6 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -414,8 +414,11 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "VIOS"), DMI_MATCH(DMI_PRODUCT_NAME, "LTH17"), }, - .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_IN2_MAP), + .driver_data = (void *)(BYT_RT5651_IN1_IN2_MAP | + BYT_RT5651_JD1_1 | + BYT_RT5651_OVCD_TH_2000UA | + BYT_RT5651_OVCD_SF_1P0 | + BYT_RT5651_MCLK_EN), }, {} }; -- cgit v1.2.3 From ac275ee5aa67abe9b65d66071ee333c6b0905b93 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 18 Jul 2018 22:55:40 +0200 Subject: ASoC: Intel: bytcr_rt5651: Add IN2 input mapping During the recent cleanup series 3 of the 6 input mappings where removed from the bytcr_rt5651 machine driver because testing showed that none of them were used. However some devices do actually have their internal mic on IN2 (and only IN2, not IN1 and IN2), this did not show during previous tests due to a bug in the userspace UCM input device switching code. This commit re-adds the IN2 mapping for devices with the internal mic. on IN2 and the headser mic on IN3 and enables this mapping on devices with their internal mic on IN2. This commit also changes the default internal mic input to IN2, because all my 7 test devices have their mic there. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 53ac97c15fc6..d85530b1cc8e 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -44,6 +44,7 @@ enum { BYT_RT5651_DMIC_MAP, BYT_RT5651_IN1_MAP, + BYT_RT5651_IN2_MAP, BYT_RT5651_IN1_IN2_MAP, }; @@ -93,9 +94,9 @@ struct byt_rt5651_private { struct snd_soc_jack jack; }; -/* Default: jack-detect on JD1_1, internal mic on in1, headsetmic on in3 */ +/* Default: jack-detect on JD1_1, internal mic on in2, headsetmic on in3 */ static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_MAP; + BYT_RT5651_IN2_MAP; static void log_quirks(struct device *dev) { @@ -103,6 +104,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk DMIC_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_MAP) dev_info(dev, "quirk IN1_MAP enabled"); + if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP) + dev_info(dev, "quirk IN2_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN1_IN2_MAP) dev_info(dev, "quirk IN1_IN2_MAP enabled"); if (BYT_RT5651_JDSRC(byt_rt5651_quirk)) { @@ -270,6 +273,12 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = { {"IN3P", NULL, "Headset Mic"}, }; +static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = { + {"Internal Mic", NULL, "micbias1"}, + {"IN2P", NULL, "Internal Mic"}, + {"IN3P", NULL, "Headset Mic"}, +}; + static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { {"Internal Mic", NULL, "micbias1"}, {"IN1P", NULL, "Internal Mic"}, @@ -364,7 +373,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "X1D3_C806N"), }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_MAP | + BYT_RT5651_IN2_MAP | BYT_RT5651_HP_LR_SWAPPED), }, { @@ -375,7 +384,7 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "D2D3_Vi8A1"), }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | - BYT_RT5651_IN1_MAP | + BYT_RT5651_IN2_MAP | BYT_RT5651_HP_LR_SWAPPED), }, { @@ -468,6 +477,10 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in1_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_map); break; + case BYT_RT5651_IN2_MAP: + custom_map = byt_rt5651_intmic_in2_map; + num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_map); + break; case BYT_RT5651_IN1_IN2_MAP: custom_map = byt_rt5651_intmic_in1_in2_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_in2_map); @@ -825,7 +838,7 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { - const char * const mic_name[] = { "dmic", "in1", "in12" }; + const char * const mic_name[] = { "dmic", "in1", "in2", "in12" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; struct device *codec_dev; -- cgit v1.2.3 From a0d1d867c262f4ad5d8e4925e2212711ebdbf2b7 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 18 Jul 2018 22:55:41 +0200 Subject: ASoC: Intel: bytcr_rt5651: Add mono speaker quirk During my initial round of bytcr_rt5651 long-name patches I did not include a difference for mono vs stereo speaker setups in the longname because it seems that all 5651 devices with only a single speaker do some mixing of left + right on the PCB. However further testing has shown that while this works great when only playing audio on the left or right channel, the output becomes garbled when using both channels at once. Something which does not happen when using the Stereo DAC MIXL / MIXR switches to mix the channels together inside the codec and then only outputting on a single channel. So we need to have separate UCM profiles and thus separate long-names for devices with a mono speaker vs stereo speakers. Just as we already have for the bytcr_rt5640 case. This commit adds a new BYT_RT5651_MONO_SPEAKER quirk and adds "stereo-spk" or "mono-spk" to the long-name based on this and enables this mapping on devices with a mono speaker. Changing the long-name like this is ok for now, since I'm still working on the UCM profiles, so they are not in upstream alsa-lib yet. This brings the long-name naming scheme fully in sync with the bytcr_rt5640 case, which is good from a consistency pov. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index d85530b1cc8e..8374e633796d 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -79,6 +79,7 @@ enum { #define BYT_RT5651_SSP0_AIF1 BIT(20) #define BYT_RT5651_SSP0_AIF2 BIT(21) #define BYT_RT5651_HP_LR_SWAPPED BIT(22) +#define BYT_RT5651_MONO_SPEAKER BIT(23) #define BYT_RT5651_DEFAULT_QUIRKS (BYT_RT5651_MCLK_EN | \ BYT_RT5651_JD1_1 | \ @@ -128,6 +129,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0_AIF1 enabled\n"); if (byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2) dev_info(dev, "quirk SSP0_AIF2 enabled\n"); + if (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) + dev_info(dev, "quirk MONO_SPEAKER enabled\n"); } #define BYT_CODEC_DAI1 "rt5651-aif1" @@ -374,7 +377,8 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN2_MAP | - BYT_RT5651_HP_LR_SWAPPED), + BYT_RT5651_HP_LR_SWAPPED | + BYT_RT5651_MONO_SPEAKER), }, { /* Chuwi Vi8 Plus (CWI519) */ @@ -385,7 +389,8 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN2_MAP | - BYT_RT5651_HP_LR_SWAPPED), + BYT_RT5651_HP_LR_SWAPPED | + BYT_RT5651_MONO_SPEAKER), }, { /* KIANO SlimNote 14.2 */ @@ -700,7 +705,7 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */ static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ -static char byt_rt5651_long_name[40]; /* = "bytcr-rt5651-*-mic[-swapped-hp]" */ +static char byt_rt5651_long_name[50]; /* = "bytcr-rt5651-*-spk-*-mic[-swapped-hp]" */ static int byt_rt5651_suspend(struct snd_soc_card *card) { @@ -1025,7 +1030,9 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) hp_swapped = ""; snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name), - "bytcr-rt5651-%s-mic%s", + "bytcr-rt5651-%s-spk-%s-mic%s", + (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ? + "mono" : "stereo", mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], hp_swapped); byt_rt5651_card.long_name = byt_rt5651_long_name; -- cgit v1.2.3 From 06aa6e51273c9ac458af0bb9be95603cfbad14ec Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 18 Jul 2018 22:55:42 +0200 Subject: ASoC: Intel: bytcr_rt5651: Add quirk table entries for various devices Add quirk table entries for the following tablets: ITWorks TW701 Ployer Momo7w Trekstor win7 Yours 8" These all use the default settings, except that they only have a single speaker and thus need the mono-speaker quirk. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 8374e633796d..f8a68bdb3885 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -392,6 +392,21 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_HP_LR_SWAPPED | BYT_RT5651_MONO_SPEAKER), }, + { + /* I.T.Works TW701, Ployer Momo7w and Trekstor ST70416-6 + * (these all use the same mainboard) */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."), + /* Partial match for all of itWORKS.G.WI71C.JGBMRBA, + * TREK.G.WI71C.JGBMRBA0x and MOMO.G.WI71C.MABMRBA02 */ + DMI_MATCH(DMI_BIOS_VERSION, ".G.WI71C."), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_SSP0_AIF1 | + BYT_RT5651_MONO_SPEAKER), + }, { /* KIANO SlimNote 14.2 */ .callback = byt_rt5651_quirk_cb, @@ -434,6 +449,19 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_OVCD_SF_1P0 | BYT_RT5651_MCLK_EN), }, + { + /* Yours Y8W81 (and others using the same mainboard) */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_BIOS_VENDOR, "INSYDE Corp."), + /* Partial match for all devs with a W86C mainboard */ + DMI_MATCH(DMI_BIOS_VERSION, ".F.W86C."), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_SSP0_AIF1 | + BYT_RT5651_MONO_SPEAKER), + }, {} }; -- cgit v1.2.3 From 9ee6f8a8cbbd203fb0a844f937007a9525422697 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 1 Jul 2018 11:30:23 +0200 Subject: ASoC: Intel: bytcr_rt5640: Add quirk for the "Connect Tablet 9" tablet Add a quirk for the "Connect Tablet 9" tablet, this tablet has a mono-speaker. Otherwise it works fine with the defaults. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 657910a08261..d32844f94d74 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -486,6 +486,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP), }, + { /* Connect Tablet 9 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Connect"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "Tablet 9"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Dell Inc."), -- cgit v1.2.3 From 2ec42486358f63c4a426514c395d13f4b9de5da8 Mon Sep 17 00:00:00 2001 From: Marcel Ziswiler Date: Fri, 20 Jul 2018 10:04:23 +0200 Subject: ASoC: tegra: improve goto error label While the two error labels "err" and "err_clk_put" goto the same place it is rather confusing that the earlier one is certainly used later again. Signed-off-by: Marcel Ziswiler Signed-off-by: Mark Brown --- sound/soc/tegra/tegra20_ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index affad46bf188..682ef33afb5f 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -377,7 +377,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ret = clk_prepare_enable(ac97->clk_ac97); if (ret) { dev_err(&pdev->dev, "clk_enable failed: %d\n", ret); - goto err; + goto err_clk_put; } ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); -- cgit v1.2.3 From 1a11d88f499ceb69e9b4098ddc36866820335a54 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:58 +0200 Subject: ASoC: meson: add tdm formatter base driver Add Amlogic's axg TDM core driver. On this SoC, tdm is bit more complex than usual, mainly because the different TDM input decoders can be attached to any of TDM pad interface, including the output pads. For the this, TDM on this SoC is modeled like this: - TDM interface provides the DAIs the codecs will be attached to. The main responsibility of this driver is to manage the pad format and the TDM clock rates. - TDM Formatters: These are the entities which are actually dealing with the TDM signal. TDMOUT produce a TDM signal from the audio sample provided by FRDDR using the clocks provided the TDM interface. TDMIN feeds TODDR with audio sample using the clocks and TDM signal provided by the TDM Interface. - TDM Streams: This provides the link between 1 DAI stream of the TDM interface and one (or more) TDM formatters. This driver provides the TDM formatter and TDM stream operations. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 4 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-tdm-formatter.c | 381 ++++++++++++++++++++++++++++++++++++ sound/soc/meson/axg-tdm-formatter.h | 39 ++++ sound/soc/meson/axg-tdm.h | 74 +++++++ 5 files changed, 500 insertions(+) create mode 100644 sound/soc/meson/axg-tdm-formatter.c create mode 100644 sound/soc/meson/axg-tdm-formatter.h create mode 100644 sound/soc/meson/axg-tdm.h (limited to 'sound/soc') diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 9408214b5854..80a88689491f 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -19,6 +19,10 @@ config SND_MESON_AXG_TODDR Select Y or M to add support for the frontend capture interfaces embedded in the Amlogic AXG SoC family +config SND_MESON_AXG_TDM_FORMATTER + tristate + select REGMAP_MMIO + config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" imply SND_SOC_SPDIF diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index d51ae045ef08..a06b56a1c995 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -3,9 +3,11 @@ snd-soc-meson-axg-fifo-objs := axg-fifo.o snd-soc-meson-axg-frddr-objs := axg-frddr.o snd-soc-meson-axg-toddr-objs := axg-toddr.o +snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o +obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c new file mode 100644 index 000000000000..43e390f9358a --- /dev/null +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -0,0 +1,381 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include + +#include "axg-tdm-formatter.h" + +struct axg_tdm_formatter { + struct list_head list; + struct axg_tdm_stream *stream; + const struct axg_tdm_formatter_driver *drv; + struct clk *pclk; + struct clk *sclk; + struct clk *lrclk; + struct clk *sclk_sel; + struct clk *lrclk_sel; + bool enabled; + struct regmap *map; +}; + +int axg_tdm_formatter_set_channel_masks(struct regmap *map, + struct axg_tdm_stream *ts, + unsigned int offset) +{ + unsigned int val, ch = ts->channels; + unsigned long mask; + int i, j; + + /* + * Distribute the channels of the stream over the available slots + * of each TDM lane + */ + for (i = 0; i < AXG_TDM_NUM_LANES; i++) { + val = 0; + mask = ts->mask[i]; + + for (j = find_first_bit(&mask, 32); + (j < 32) && ch; + j = find_next_bit(&mask, 32, j + 1)) { + val |= 1 << j; + ch -= 1; + } + + regmap_write(map, offset, val); + offset += regmap_get_reg_stride(map); + } + + /* + * If we still have channel left at the end of the process, it means + * the stream has more channels than we can accommodate and we should + * have caught this earlier. + */ + if (WARN_ON(ch != 0)) { + pr_err("channel mask error\n"); + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks); + +static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter) +{ + struct axg_tdm_stream *ts = formatter->stream; + bool invert = formatter->drv->invert_sclk; + int ret; + + /* Do nothing if the formatter is already enabled */ + if (formatter->enabled) + return 0; + + /* + * If sclk is inverted, invert it back and provide the inversion + * required by the formatter + */ + invert ^= axg_tdm_sclk_invert(ts->iface->fmt); + ret = clk_set_phase(formatter->sclk, invert ? 180 : 0); + if (ret) + return ret; + + /* Setup the stream parameter in the formatter */ + ret = formatter->drv->ops->prepare(formatter->map, formatter->stream); + if (ret) + return ret; + + /* Enable the signal clocks feeding the formatter */ + ret = clk_prepare_enable(formatter->sclk); + if (ret) + return ret; + + ret = clk_prepare_enable(formatter->lrclk); + if (ret) { + clk_disable_unprepare(formatter->sclk); + return ret; + } + + /* Finally, actually enable the formatter */ + formatter->drv->ops->enable(formatter->map); + formatter->enabled = true; + + return 0; +} + +static void axg_tdm_formatter_disable(struct axg_tdm_formatter *formatter) +{ + /* Do nothing if the formatter is already disabled */ + if (!formatter->enabled) + return; + + formatter->drv->ops->disable(formatter->map); + clk_disable_unprepare(formatter->lrclk); + clk_disable_unprepare(formatter->sclk); + formatter->enabled = false; +} + +static int axg_tdm_formatter_attach(struct axg_tdm_formatter *formatter) +{ + struct axg_tdm_stream *ts = formatter->stream; + int ret = 0; + + mutex_lock(&ts->lock); + + /* Catch up if the stream is already running when we attach */ + if (ts->ready) { + ret = axg_tdm_formatter_enable(formatter); + if (ret) { + pr_err("failed to enable formatter\n"); + goto out; + } + } + + list_add_tail(&formatter->list, &ts->formatter_list); +out: + mutex_unlock(&ts->lock); + return ret; +} + +static void axg_tdm_formatter_dettach(struct axg_tdm_formatter *formatter) +{ + struct axg_tdm_stream *ts = formatter->stream; + + mutex_lock(&ts->lock); + list_del(&formatter->list); + mutex_unlock(&ts->lock); + + axg_tdm_formatter_disable(formatter); +} + +static int axg_tdm_formatter_power_up(struct axg_tdm_formatter *formatter, + struct snd_soc_dapm_widget *w) +{ + struct axg_tdm_stream *ts = formatter->drv->ops->get_stream(w); + int ret; + + /* + * If we don't get a stream at this stage, it would mean that the + * widget is powering up but is not attached to any backend DAI. + * It should not happen, ever ! + */ + if (WARN_ON(!ts)) + return -ENODEV; + + /* Clock our device */ + ret = clk_prepare_enable(formatter->pclk); + if (ret) + return ret; + + /* Reparent the bit clock to the TDM interface */ + ret = clk_set_parent(formatter->sclk_sel, ts->iface->sclk); + if (ret) + goto disable_pclk; + + /* Reparent the sample clock to the TDM interface */ + ret = clk_set_parent(formatter->lrclk_sel, ts->iface->lrclk); + if (ret) + goto disable_pclk; + + formatter->stream = ts; + ret = axg_tdm_formatter_attach(formatter); + if (ret) + goto disable_pclk; + + return 0; + +disable_pclk: + clk_disable_unprepare(formatter->pclk); + return ret; +} + +static void axg_tdm_formatter_power_down(struct axg_tdm_formatter *formatter) +{ + axg_tdm_formatter_dettach(formatter); + clk_disable_unprepare(formatter->pclk); + formatter->stream = NULL; +} + +int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, + int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct axg_tdm_formatter *formatter = snd_soc_component_get_drvdata(c); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = axg_tdm_formatter_power_up(formatter, w); + break; + + case SND_SOC_DAPM_PRE_PMD: + axg_tdm_formatter_power_down(formatter); + break; + + default: + dev_err(c->dev, "Unexpected event %d\n", event); + return -EINVAL; + } + + return ret; +} +EXPORT_SYMBOL_GPL(axg_tdm_formatter_event); + +int axg_tdm_formatter_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + const struct axg_tdm_formatter_driver *drv; + struct axg_tdm_formatter *formatter; + struct resource *res; + void __iomem *regs; + int ret; + + drv = of_device_get_match_data(dev); + if (!drv) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + formatter = devm_kzalloc(dev, sizeof(*formatter), GFP_KERNEL); + if (!formatter) + return -ENOMEM; + platform_set_drvdata(pdev, formatter); + formatter->drv = drv; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + formatter->map = devm_regmap_init_mmio(dev, regs, drv->regmap_cfg); + if (IS_ERR(formatter->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(formatter->map)); + return PTR_ERR(formatter->map); + } + + /* Peripharal clock */ + formatter->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(formatter->pclk)) { + ret = PTR_ERR(formatter->pclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %d\n", ret); + return ret; + } + + /* Formatter bit clock */ + formatter->sclk = devm_clk_get(dev, "sclk"); + if (IS_ERR(formatter->sclk)) { + ret = PTR_ERR(formatter->sclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get sclk: %d\n", ret); + return ret; + } + + /* Formatter sample clock */ + formatter->lrclk = devm_clk_get(dev, "lrclk"); + if (IS_ERR(formatter->lrclk)) { + ret = PTR_ERR(formatter->lrclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get lrclk: %d\n", ret); + return ret; + } + + /* Formatter bit clock input multiplexer */ + formatter->sclk_sel = devm_clk_get(dev, "sclk_sel"); + if (IS_ERR(formatter->sclk_sel)) { + ret = PTR_ERR(formatter->sclk_sel); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get sclk_sel: %d\n", ret); + return ret; + } + + /* Formatter sample clock input multiplexer */ + formatter->lrclk_sel = devm_clk_get(dev, "lrclk_sel"); + if (IS_ERR(formatter->lrclk_sel)) { + ret = PTR_ERR(formatter->lrclk_sel); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get lrclk_sel: %d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(dev, drv->component_drv, + NULL, 0); +} +EXPORT_SYMBOL_GPL(axg_tdm_formatter_probe); + +int axg_tdm_stream_start(struct axg_tdm_stream *ts) +{ + struct axg_tdm_formatter *formatter; + int ret = 0; + + mutex_lock(&ts->lock); + ts->ready = true; + + /* Start all the formatters attached to the stream */ + list_for_each_entry(formatter, &ts->formatter_list, list) { + ret = axg_tdm_formatter_enable(formatter); + if (ret) { + pr_err("failed to start tdm stream\n"); + goto out; + } + } + +out: + mutex_unlock(&ts->lock); + return ret; +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_start); + +void axg_tdm_stream_stop(struct axg_tdm_stream *ts) +{ + struct axg_tdm_formatter *formatter; + + mutex_lock(&ts->lock); + ts->ready = false; + + /* Stop all the formatters attached to the stream */ + list_for_each_entry(formatter, &ts->formatter_list, list) { + axg_tdm_formatter_disable(formatter); + } + + mutex_unlock(&ts->lock); +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_stop); + +struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface) +{ + struct axg_tdm_stream *ts; + + ts = kzalloc(sizeof(*ts), GFP_KERNEL); + if (ts) { + INIT_LIST_HEAD(&ts->formatter_list); + mutex_init(&ts->lock); + ts->iface = iface; + } + + return ts; +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_alloc); + +void axg_tdm_stream_free(struct axg_tdm_stream *ts) +{ + /* + * If the list is not empty, it would mean that one of the formatter + * widget is still powered and attached to the interface while we + * we are removing the TDM DAI. It should not be possible + */ + WARN_ON(!list_empty(&ts->formatter_list)); + mutex_destroy(&ts->lock); + kfree(ts); +} +EXPORT_SYMBOL_GPL(axg_tdm_stream_free); + +MODULE_DESCRIPTION("Amlogic AXG TDM formatter driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h new file mode 100644 index 000000000000..cf947caf3cb1 --- /dev/null +++ b/sound/soc/meson/axg-tdm-formatter.h @@ -0,0 +1,39 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) + * + * Copyright (c) 2018 Baylibre SAS. + * Author: Jerome Brunet + */ + +#ifndef _MESON_AXG_TDM_FORMATTER_H +#define _MESON_AXG_TDM_FORMATTER_H + +#include "axg-tdm.h" + +struct platform_device; +struct regmap; +struct snd_soc_dapm_widget; +struct snd_kcontrol; + +struct axg_tdm_formatter_ops { + struct axg_tdm_stream *(*get_stream)(struct snd_soc_dapm_widget *w); + void (*enable)(struct regmap *map); + void (*disable)(struct regmap *map); + int (*prepare)(struct regmap *map, struct axg_tdm_stream *ts); +}; + +struct axg_tdm_formatter_driver { + const struct snd_soc_component_driver *component_drv; + const struct regmap_config *regmap_cfg; + const struct axg_tdm_formatter_ops *ops; + bool invert_sclk; +}; + +int axg_tdm_formatter_set_channel_masks(struct regmap *map, + struct axg_tdm_stream *ts, + unsigned int offset); +int axg_tdm_formatter_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, + int event); +int axg_tdm_formatter_probe(struct platform_device *pdev); + +#endif /* _MESON_AXG_TDM_FORMATTER_H */ diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h new file mode 100644 index 000000000000..435d95b86457 --- /dev/null +++ b/sound/soc/meson/axg-tdm.h @@ -0,0 +1,74 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) + * + * Copyright (c) 2018 Baylibre SAS. + * Author: Jerome Brunet + */ + +#ifndef _MESON_AXG_TDM_H +#define _MESON_AXG_TDM_H + +#include +#include +#include +#include +#include + +#define AXG_TDM_NUM_LANES 4 +#define AXG_TDM_CHANNEL_MAX 128 +#define AXG_TDM_RATES (SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000) +#define AXG_TDM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +struct axg_tdm_iface { + struct clk *sclk; + struct clk *lrclk; + struct clk *mclk; + unsigned long mclk_rate; + + /* format is common to all the DAIs of the iface */ + unsigned int fmt; + unsigned int slots; + unsigned int slot_width; + + /* For component wide symmetry */ + int rate; +}; + +static inline bool axg_tdm_lrclk_invert(unsigned int fmt) +{ + return (fmt & SND_SOC_DAIFMT_I2S) ^ + !!(fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_NB_IF)); +} + +static inline bool axg_tdm_sclk_invert(unsigned int fmt) +{ + return fmt & (SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_IB_NF); +} + +struct axg_tdm_stream { + struct axg_tdm_iface *iface; + struct list_head formatter_list; + struct mutex lock; + unsigned int channels; + unsigned int width; + unsigned int physical_width; + u32 *mask; + bool ready; +}; + +struct axg_tdm_stream *axg_tdm_stream_alloc(struct axg_tdm_iface *iface); +void axg_tdm_stream_free(struct axg_tdm_stream *ts); +int axg_tdm_stream_start(struct axg_tdm_stream *ts); +void axg_tdm_stream_stop(struct axg_tdm_stream *ts); + +static inline int axg_tdm_stream_reset(struct axg_tdm_stream *ts) +{ + axg_tdm_stream_stop(ts); + return axg_tdm_stream_start(ts); +} + +#endif /* _MESON_AXG_TDM_H */ -- cgit v1.2.3 From d60e4f1e4be5e2dfb55fb084b119aed094227a35 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:42:59 +0200 Subject: ASoC: meson: add tdm interface driver Add Amlogic's axg TDM interface driver. This driver manages the format and clocks provided on the pads. On this SoC, each stream direction provides 4 serial lanes. This makes a maximum of 8 channels in i2s modes and 128 channels in DSP modes. While each lanes operate on the same slot number (same bit clock), they may have different TDM masks. This requires to provide a function to let the card set the 4 masks, in lieu of the usual set_tdm_slots() callback of the dai driver. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 4 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-tdm-interface.c | 542 ++++++++++++++++++++++++++++++++++++ sound/soc/meson/axg-tdm.h | 4 + 4 files changed, 552 insertions(+) create mode 100644 sound/soc/meson/axg-tdm-interface.c (limited to 'sound/soc') diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 80a88689491f..869b359c2ce7 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -23,6 +23,10 @@ config SND_MESON_AXG_TDM_FORMATTER tristate select REGMAP_MMIO +config SND_MESON_AXG_TDM_INTERFACE + tristate + select SND_MESON_AXG_TDM_FORMATTER + config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" imply SND_SOC_SPDIF diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index a06b56a1c995..1a8eb77402e3 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -4,10 +4,12 @@ snd-soc-meson-axg-fifo-objs := axg-fifo.o snd-soc-meson-axg-frddr-objs := axg-frddr.o snd-soc-meson-axg-toddr-objs := axg-toddr.o snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o +snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o +obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c new file mode 100644 index 000000000000..7b8baf46d968 --- /dev/null +++ b/sound/soc/meson/axg-tdm-interface.c @@ -0,0 +1,542 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include +#include + +#include "axg-tdm.h" + +enum { + TDM_IFACE_PAD, + TDM_IFACE_LOOPBACK, +}; + +static unsigned int axg_tdm_slots_total(u32 *mask) +{ + unsigned int slots = 0; + int i; + + if (!mask) + return 0; + + /* Count the total number of slots provided by all 4 lanes */ + for (i = 0; i < AXG_TDM_NUM_LANES; i++) + slots += hweight32(mask[i]); + + return slots; +} + +int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, + u32 *rx_mask, unsigned int slots, + unsigned int slot_width) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + struct axg_tdm_stream *tx = (struct axg_tdm_stream *) + dai->playback_dma_data; + struct axg_tdm_stream *rx = (struct axg_tdm_stream *) + dai->capture_dma_data; + unsigned int tx_slots, rx_slots; + + tx_slots = axg_tdm_slots_total(tx_mask); + rx_slots = axg_tdm_slots_total(rx_mask); + + /* We should at least have a slot for a valid interface */ + if (!tx_slots && !rx_slots) { + dev_err(dai->dev, "interface has no slot\n"); + return -EINVAL; + } + + /* + * Amend the dai driver channel number and let dpcm channel merge do + * its job + */ + if (tx) { + tx->mask = tx_mask; + dai->driver->playback.channels_max = tx_slots; + } + + if (rx) { + rx->mask = rx_mask; + dai->driver->capture.channels_max = rx_slots; + } + + iface->slots = slots; + + switch (slot_width) { + case 0: + /* defaults width to 32 if not provided */ + iface->slot_width = 32; + break; + case 8: + case 16: + case 24: + case 32: + iface->slot_width = slot_width; + break; + default: + dev_err(dai->dev, "unsupported slot width: %d\n", slot_width); + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots); + +static int axg_tdm_iface_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + int ret = -ENOTSUPP; + + if (dir == SND_SOC_CLOCK_OUT && clk_id == 0) { + if (!iface->mclk) { + dev_warn(dai->dev, "master clock not provided\n"); + } else { + ret = clk_set_rate(iface->mclk, freq); + if (!ret) + iface->mclk_rate = freq; + } + } + + return ret; +} + +static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + + /* These modes are not supported */ + if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) { + dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n"); + return -EINVAL; + } + + /* If the TDM interface is the clock master, it requires mclk */ + if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) { + dev_err(dai->dev, "cpu clock master: mclk missing\n"); + return -ENODEV; + } + + iface->fmt = fmt; + return 0; +} + +static int axg_tdm_iface_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + struct axg_tdm_stream *ts = + snd_soc_dai_get_dma_data(dai, substream); + int ret; + + if (!axg_tdm_slots_total(ts->mask)) { + dev_err(dai->dev, "interface has not slots\n"); + return -EINVAL; + } + + /* Apply component wide rate symmetry */ + if (dai->component->active) { + ret = snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + iface->rate); + if (ret < 0) { + dev_err(dai->dev, + "can't set iface rate constraint\n"); + return ret; + } + } + + return 0; +} + +static int axg_tdm_iface_set_stream(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + unsigned int channels = params_channels(params); + unsigned int width = params_width(params); + + /* Save rate and sample_bits for component symmetry */ + iface->rate = params_rate(params); + + /* Make sure this interface can cope with the stream */ + if (axg_tdm_slots_total(ts->mask) < channels) { + dev_err(dai->dev, "not enough slots for channels\n"); + return -EINVAL; + } + + if (iface->slot_width < width) { + dev_err(dai->dev, "incompatible slots width for stream\n"); + return -EINVAL; + } + + /* Save the parameter for tdmout/tdmin widgets */ + ts->physical_width = params_physical_width(params); + ts->width = params_width(params); + ts->channels = params_channels(params); + + return 0; +} + +static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + unsigned int ratio_num; + int ret; + + ret = clk_set_rate(iface->lrclk, params_rate(params)); + if (ret) { + dev_err(dai->dev, "setting sample clock failed: %d\n", ret); + return ret; + } + + switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + /* 50% duty cycle ratio */ + ratio_num = 1; + break; + + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* + * A zero duty cycle ratio will result in setting the mininum + * ratio possible which, for this clock, is 1 cycle of the + * parent bclk clock high and the rest low, This is exactly + * what we want here. + */ + ratio_num = 0; + break; + + default: + return -EINVAL; + } + + ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); + if (ret) { + dev_err(dai->dev, + "setting sample clock duty cycle failed: %d\n", ret); + return ret; + } + + /* Set sample clock inversion */ + ret = clk_set_phase(iface->lrclk, + axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0); + if (ret) { + dev_err(dai->dev, + "setting sample clock phase failed: %d\n", ret); + return ret; + } + + return 0; +} + +static int axg_tdm_iface_set_sclk(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + unsigned long srate; + int ret; + + srate = iface->slots * iface->slot_width * params_rate(params); + + if (!iface->mclk_rate) { + /* If no specific mclk is requested, default to bit clock * 4 */ + clk_set_rate(iface->mclk, 4 * srate); + } else { + /* Check if we can actually get the bit clock from mclk */ + if (iface->mclk_rate % srate) { + dev_err(dai->dev, + "can't derive sclk %lu from mclk %lu\n", + srate, iface->mclk_rate); + return -EINVAL; + } + } + + ret = clk_set_rate(iface->sclk, srate); + if (ret) { + dev_err(dai->dev, "setting bit clock failed: %d\n", ret); + return ret; + } + + /* Set the bit clock inversion */ + ret = clk_set_phase(iface->sclk, + axg_tdm_sclk_invert(iface->fmt) ? 0 : 180); + if (ret) { + dev_err(dai->dev, "setting bit clock phase failed: %d\n", ret); + return ret; + } + + return ret; +} + +static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + int ret; + + switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + if (iface->slots > 2) { + dev_err(dai->dev, "bad slot number for format: %d\n", + iface->slots); + return -EINVAL; + } + break; + + case SND_SOC_DAI_FORMAT_DSP_A: + case SND_SOC_DAI_FORMAT_DSP_B: + break; + + default: + dev_err(dai->dev, "unsupported dai format\n"); + return -EINVAL; + } + + ret = axg_tdm_iface_set_stream(substream, params, dai); + if (ret) + return ret; + + if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) { + ret = axg_tdm_iface_set_sclk(dai, params); + if (ret) + return ret; + + ret = axg_tdm_iface_set_lrclk(dai, params); + if (ret) + return ret; + } + + return 0; +} + +static int axg_tdm_iface_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + + /* Stop all attached formatters */ + axg_tdm_stream_stop(ts); + + return 0; +} + +static int axg_tdm_iface_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); + + /* Force all attached formatters to update */ + return axg_tdm_stream_reset(ts); +} + +static int axg_tdm_iface_remove_dai(struct snd_soc_dai *dai) +{ + if (dai->capture_dma_data) + axg_tdm_stream_free(dai->capture_dma_data); + + if (dai->playback_dma_data) + axg_tdm_stream_free(dai->playback_dma_data); + + return 0; +} + +static int axg_tdm_iface_probe_dai(struct snd_soc_dai *dai) +{ + struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); + + if (dai->capture_widget) { + dai->capture_dma_data = axg_tdm_stream_alloc(iface); + if (!dai->capture_dma_data) + return -ENOMEM; + } + + if (dai->playback_widget) { + dai->playback_dma_data = axg_tdm_stream_alloc(iface); + if (!dai->playback_dma_data) { + axg_tdm_iface_remove_dai(dai); + return -ENOMEM; + } + } + + return 0; +} + +static const struct snd_soc_dai_ops axg_tdm_iface_ops = { + .set_sysclk = axg_tdm_iface_set_sysclk, + .set_fmt = axg_tdm_iface_set_fmt, + .startup = axg_tdm_iface_startup, + .hw_params = axg_tdm_iface_hw_params, + .prepare = axg_tdm_iface_prepare, + .hw_free = axg_tdm_iface_hw_free, +}; + +/* TDM Backend DAIs */ +static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { + [TDM_IFACE_PAD] = { + .name = "TDM Pad", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = AXG_TDM_CHANNEL_MAX, + .rates = AXG_TDM_RATES, + .formats = AXG_TDM_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = AXG_TDM_CHANNEL_MAX, + .rates = AXG_TDM_RATES, + .formats = AXG_TDM_FORMATS, + }, + .id = TDM_IFACE_PAD, + .ops = &axg_tdm_iface_ops, + .probe = axg_tdm_iface_probe_dai, + .remove = axg_tdm_iface_remove_dai, + }, + [TDM_IFACE_LOOPBACK] = { + .name = "TDM Loopback", + .capture = { + .stream_name = "Loopback", + .channels_min = 1, + .channels_max = AXG_TDM_CHANNEL_MAX, + .rates = AXG_TDM_RATES, + .formats = AXG_TDM_FORMATS, + }, + .id = TDM_IFACE_LOOPBACK, + .ops = &axg_tdm_iface_ops, + .probe = axg_tdm_iface_probe_dai, + .remove = axg_tdm_iface_remove_dai, + }, +}; + +static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct axg_tdm_iface *iface = snd_soc_component_get_drvdata(component); + enum snd_soc_bias_level now = + snd_soc_component_get_bias_level(component); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (now == SND_SOC_BIAS_STANDBY) + ret = clk_prepare_enable(iface->mclk); + break; + + case SND_SOC_BIAS_STANDBY: + if (now == SND_SOC_BIAS_PREPARE) + clk_disable_unprepare(iface->mclk); + break; + + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_ON: + break; + } + + return ret; +} + +static const struct snd_soc_component_driver axg_tdm_iface_component_drv = { + .set_bias_level = axg_tdm_iface_set_bias_level, +}; + +static const struct of_device_id axg_tdm_iface_of_match[] = { + { .compatible = "amlogic,axg-tdm-iface", }, + {} +}; +MODULE_DEVICE_TABLE(of, axg_tdm_iface_of_match); + +static int axg_tdm_iface_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct snd_soc_dai_driver *dai_drv; + struct axg_tdm_iface *iface; + int ret, i; + + iface = devm_kzalloc(dev, sizeof(*iface), GFP_KERNEL); + if (!iface) + return -ENOMEM; + platform_set_drvdata(pdev, iface); + + /* + * Duplicate dai driver: depending on the slot masks configuration + * We'll change the number of channel provided by DAI stream, so dpcm + * channel merge can be done properly + */ + dai_drv = devm_kcalloc(dev, ARRAY_SIZE(axg_tdm_iface_dai_drv), + sizeof(*dai_drv), GFP_KERNEL); + if (!dai_drv) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(axg_tdm_iface_dai_drv); i++) + memcpy(&dai_drv[i], &axg_tdm_iface_dai_drv[i], + sizeof(*dai_drv)); + + /* Bit clock provided on the pad */ + iface->sclk = devm_clk_get(dev, "sclk"); + if (IS_ERR(iface->sclk)) { + ret = PTR_ERR(iface->sclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get sclk: %d\n", ret); + return ret; + } + + /* Sample clock provided on the pad */ + iface->lrclk = devm_clk_get(dev, "lrclk"); + if (IS_ERR(iface->lrclk)) { + ret = PTR_ERR(iface->lrclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get lrclk: %d\n", ret); + return ret; + } + + /* + * mclk maybe be missing when the cpu dai is in slave mode and + * the codec does not require it to provide a master clock. + * At this point, ignore the error if mclk is missing. We'll + * throw an error if the cpu dai is master and mclk is missing + */ + iface->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(iface->mclk)) { + ret = PTR_ERR(iface->mclk); + if (ret == -ENOENT) { + iface->mclk = NULL; + } else { + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get mclk: %d\n", ret); + return ret; + } + } + + return devm_snd_soc_register_component(dev, + &axg_tdm_iface_component_drv, dai_drv, + ARRAY_SIZE(axg_tdm_iface_dai_drv)); +} + +static struct platform_driver axg_tdm_iface_pdrv = { + .probe = axg_tdm_iface_probe, + .driver = { + .name = "axg-tdm-iface", + .of_match_table = axg_tdm_iface_of_match, + }, +}; +module_platform_driver(axg_tdm_iface_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG TDM interface driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h index 435d95b86457..e578b6f40a07 100644 --- a/sound/soc/meson/axg-tdm.h +++ b/sound/soc/meson/axg-tdm.h @@ -71,4 +71,8 @@ static inline int axg_tdm_stream_reset(struct axg_tdm_stream *ts) return axg_tdm_stream_start(ts); } +int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, + u32 *rx_mask, unsigned int slots, + unsigned int slot_width); + #endif /* _MESON_AXG_TDM_H */ -- cgit v1.2.3 From c41c2a355b86368608377eaf3df442ec0f342f1e Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:43:00 +0200 Subject: ASoC: meson: add tdm output driver Add Amlogic's axg tdm output driver which pulls data from FRDDR fifo and produce the TDM signals for 4 output lanes. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 8 ++ sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-tdmout.c | 259 +++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 269 insertions(+) create mode 100644 sound/soc/meson/axg-tdmout.c (limited to 'sound/soc') diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 869b359c2ce7..08a522b77749 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -27,6 +27,14 @@ config SND_MESON_AXG_TDM_INTERFACE tristate select SND_MESON_AXG_TDM_FORMATTER +config SND_MESON_AXG_TDMOUT + tristate "Amlogic AXG TDM Output Support" + select SND_MESON_AXG_TDM_FORMATTER + select SND_MESON_AXG_TDM_INTERFACE + help + Select Y or M to add support for TDM output formatter embedded + in the Amlogic AXG SoC family + config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" imply SND_SOC_SPDIF diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 1a8eb77402e3..a665c6b76a95 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -5,6 +5,7 @@ snd-soc-meson-axg-frddr-objs := axg-frddr.o snd-soc-meson-axg-toddr-objs := axg-toddr.o snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o +snd-soc-meson-axg-tdmout-objs := axg-tdmout.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o @@ -12,4 +13,5 @@ obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o +obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c new file mode 100644 index 000000000000..f73368ee1088 --- /dev/null +++ b/sound/soc/meson/axg-tdmout.c @@ -0,0 +1,259 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include + +#include "axg-tdm-formatter.h" + +#define TDMOUT_CTRL0 0x00 +#define TDMOUT_CTRL0_BITNUM_MASK GENMASK(4, 0) +#define TDMOUT_CTRL0_BITNUM(x) ((x) << 0) +#define TDMOUT_CTRL0_SLOTNUM_MASK GENMASK(9, 5) +#define TDMOUT_CTRL0_SLOTNUM(x) ((x) << 5) +#define TDMOUT_CTRL0_INIT_BITNUM_MASK GENMASK(19, 15) +#define TDMOUT_CTRL0_INIT_BITNUM(x) ((x) << 15) +#define TDMOUT_CTRL0_ENABLE BIT(31) +#define TDMOUT_CTRL0_RST_OUT BIT(29) +#define TDMOUT_CTRL0_RST_IN BIT(28) +#define TDMOUT_CTRL1 0x04 +#define TDMOUT_CTRL1_TYPE_MASK GENMASK(6, 4) +#define TDMOUT_CTRL1_TYPE(x) ((x) << 4) +#define TDMOUT_CTRL1_MSB_POS_MASK GENMASK(12, 8) +#define TDMOUT_CTRL1_MSB_POS(x) ((x) << 8) +#define TDMOUT_CTRL1_SEL_SHIFT 24 +#define TDMOUT_CTRL1_GAIN_EN 26 +#define TDMOUT_CTRL1_WS_INV BIT(28) +#define TDMOUT_SWAP 0x08 +#define TDMOUT_MASK0 0x0c +#define TDMOUT_MASK1 0x10 +#define TDMOUT_MASK2 0x14 +#define TDMOUT_MASK3 0x18 +#define TDMOUT_STAT 0x1c +#define TDMOUT_GAIN0 0x20 +#define TDMOUT_GAIN1 0x24 +#define TDMOUT_MUTE_VAL 0x28 +#define TDMOUT_MUTE0 0x2c +#define TDMOUT_MUTE1 0x30 +#define TDMOUT_MUTE2 0x34 +#define TDMOUT_MUTE3 0x38 +#define TDMOUT_MASK_VAL 0x3c + +static const struct regmap_config axg_tdmout_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = TDMOUT_MASK_VAL, +}; + +static const struct snd_kcontrol_new axg_tdmout_controls[] = { + SOC_DOUBLE("Lane 0 Volume", TDMOUT_GAIN0, 0, 8, 255, 0), + SOC_DOUBLE("Lane 1 Volume", TDMOUT_GAIN0, 16, 24, 255, 0), + SOC_DOUBLE("Lane 2 Volume", TDMOUT_GAIN1, 0, 8, 255, 0), + SOC_DOUBLE("Lane 3 Volume", TDMOUT_GAIN1, 16, 24, 255, 0), + SOC_SINGLE("Gain Enable Switch", TDMOUT_CTRL1, + TDMOUT_CTRL1_GAIN_EN, 1, 0), +}; + +static const char * const tdmout_sel_texts[] = { + "IN 0", "IN 1", "IN 2", +}; + +static SOC_ENUM_SINGLE_DECL(axg_tdmout_sel_enum, TDMOUT_CTRL1, + TDMOUT_CTRL1_SEL_SHIFT, tdmout_sel_texts); + +static const struct snd_kcontrol_new axg_tdmout_in_mux = + SOC_DAPM_ENUM("Input Source", axg_tdmout_sel_enum); + +static struct snd_soc_dai * +axg_tdmout_get_be(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dai *be; + + snd_soc_dapm_widget_for_each_sink_path(w, p) { + if (!p->connect) + continue; + + if (p->sink->id == snd_soc_dapm_dai_in) + return (struct snd_soc_dai *)p->sink->priv; + + be = axg_tdmout_get_be(p->sink); + if (be) + return be; + } + + return NULL; +} + +static struct axg_tdm_stream * +axg_tdmout_get_tdm_stream(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dai *be = axg_tdmout_get_be(w); + + if (!be) + return NULL; + + return be->playback_dma_data; +} + +static void axg_tdmout_enable(struct regmap *map) +{ + /* Apply both reset */ + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_RST_OUT | TDMOUT_CTRL0_RST_IN, 0); + + /* Clear out reset before in reset */ + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_RST_OUT, TDMOUT_CTRL0_RST_OUT); + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_RST_IN, TDMOUT_CTRL0_RST_IN); + + /* Actually enable tdmout */ + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_ENABLE, TDMOUT_CTRL0_ENABLE); +} + +static void axg_tdmout_disable(struct regmap *map) +{ + regmap_update_bits(map, TDMOUT_CTRL0, TDMOUT_CTRL0_ENABLE, 0); +} + +static int axg_tdmout_prepare(struct regmap *map, struct axg_tdm_stream *ts) +{ + unsigned int val = 0; + + /* Set the stream skew */ + switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + val |= TDMOUT_CTRL0_INIT_BITNUM(1); + break; + + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_DSP_B: + val |= TDMOUT_CTRL0_INIT_BITNUM(2); + break; + + default: + pr_err("Unsupported format: %u\n", + ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + /* Set the slot width */ + val |= TDMOUT_CTRL0_BITNUM(ts->iface->slot_width - 1); + + /* Set the slot number */ + val |= TDMOUT_CTRL0_SLOTNUM(ts->iface->slots - 1); + + regmap_update_bits(map, TDMOUT_CTRL0, + TDMOUT_CTRL0_INIT_BITNUM_MASK | + TDMOUT_CTRL0_BITNUM_MASK | + TDMOUT_CTRL0_SLOTNUM_MASK, val); + + /* Set the sample width */ + val = TDMOUT_CTRL1_MSB_POS(ts->width - 1); + + /* FIFO data are arranged in chunks of 64bits */ + switch (ts->physical_width) { + case 8: + /* 8 samples of 8 bits */ + val |= TDMOUT_CTRL1_TYPE(0); + break; + case 16: + /* 4 samples of 16 bits - right justified */ + val |= TDMOUT_CTRL1_TYPE(2); + break; + case 32: + /* 2 samples of 32 bits - right justified */ + val |= TDMOUT_CTRL1_TYPE(4); + break; + default: + pr_err("Unsupported physical width: %u\n", + ts->physical_width); + return -EINVAL; + } + + /* If the sample clock is inverted, invert it back for the formatter */ + if (axg_tdm_lrclk_invert(ts->iface->fmt)) + val |= TDMOUT_CTRL1_WS_INV; + + regmap_update_bits(map, TDMOUT_CTRL1, + (TDMOUT_CTRL1_TYPE_MASK | TDMOUT_CTRL1_MSB_POS_MASK | + TDMOUT_CTRL1_WS_INV), val); + + /* Set static swap mask configuration */ + regmap_write(map, TDMOUT_SWAP, 0x76543210); + + return axg_tdm_formatter_set_channel_masks(map, ts, TDMOUT_MASK0); +} + +static const struct snd_soc_dapm_widget axg_tdmout_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmout_in_mux), + SND_SOC_DAPM_PGA_E("ENC", SND_SOC_NOPM, 0, 0, NULL, 0, + axg_tdm_formatter_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)), + SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_tdmout_dapm_routes[] = { + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "ENC", NULL, "SRC SEL" }, + { "OUT", NULL, "ENC" }, +}; + +static const struct snd_soc_component_driver axg_tdmout_component_drv = { + .controls = axg_tdmout_controls, + .num_controls = ARRAY_SIZE(axg_tdmout_controls), + .dapm_widgets = axg_tdmout_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdmout_dapm_widgets), + .dapm_routes = axg_tdmout_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdmout_dapm_routes), +}; + +static const struct axg_tdm_formatter_ops axg_tdmout_ops = { + .get_stream = axg_tdmout_get_tdm_stream, + .prepare = axg_tdmout_prepare, + .enable = axg_tdmout_enable, + .disable = axg_tdmout_disable, +}; + +static const struct axg_tdm_formatter_driver axg_tdmout_drv = { + .component_drv = &axg_tdmout_component_drv, + .regmap_cfg = &axg_tdmout_regmap_cfg, + .ops = &axg_tdmout_ops, + .invert_sclk = true, +}; + +static const struct of_device_id axg_tdmout_of_match[] = { + { + .compatible = "amlogic,axg-tdmout", + .data = &axg_tdmout_drv, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_tdmout_of_match); + +static struct platform_driver axg_tdmout_pdrv = { + .probe = axg_tdm_formatter_probe, + .driver = { + .name = "axg-tdmout", + .of_match_table = axg_tdmout_of_match, + }, +}; +module_platform_driver(axg_tdmout_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG TDM output formatter driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 13a22e6a98f8b47d61948fcd095d862377b3b143 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:43:01 +0200 Subject: ASoC: meson: add tdm input driver Add Amlogic's axg TDM input driver which take the TDM signal of 4 input lanes and push the decoded audio samples to TODDR fifo Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 8 ++ sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-tdmin.c | 229 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 239 insertions(+) create mode 100644 sound/soc/meson/axg-tdmin.c (limited to 'sound/soc') diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 08a522b77749..00d05df67b52 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -27,6 +27,14 @@ config SND_MESON_AXG_TDM_INTERFACE tristate select SND_MESON_AXG_TDM_FORMATTER +config SND_MESON_AXG_TDMIN + tristate "Amlogic AXG TDM Input Support" + select SND_MESON_AXG_TDM_FORMATTER + select SND_MESON_AXG_TDM_INTERFACE + help + Select Y or M to add support for TDM input formatter embedded + in the Amlogic AXG SoC family + config SND_MESON_AXG_TDMOUT tristate "Amlogic AXG TDM Output Support" select SND_MESON_AXG_TDM_FORMATTER diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index a665c6b76a95..f62833fb44d8 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -5,6 +5,7 @@ snd-soc-meson-axg-frddr-objs := axg-frddr.o snd-soc-meson-axg-toddr-objs := axg-toddr.o snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o +snd-soc-meson-axg-tdmin-objs := axg-tdmin.o snd-soc-meson-axg-tdmout-objs := axg-tdmout.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o @@ -13,5 +14,6 @@ obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o +obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c new file mode 100644 index 000000000000..bbac44c81688 --- /dev/null +++ b/sound/soc/meson/axg-tdmin.c @@ -0,0 +1,229 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include +#include + +#include "axg-tdm-formatter.h" + +#define TDMIN_CTRL 0x00 +#define TDMIN_CTRL_ENABLE BIT(31) +#define TDMIN_CTRL_I2S_MODE BIT(30) +#define TDMIN_CTRL_RST_OUT BIT(29) +#define TDMIN_CTRL_RST_IN BIT(28) +#define TDMIN_CTRL_WS_INV BIT(25) +#define TDMIN_CTRL_SEL_SHIFT 20 +#define TDMIN_CTRL_IN_BIT_SKEW_MASK GENMASK(18, 16) +#define TDMIN_CTRL_IN_BIT_SKEW(x) ((x) << 16) +#define TDMIN_CTRL_LSB_FIRST BIT(5) +#define TDMIN_CTRL_BITNUM_MASK GENMASK(4, 0) +#define TDMIN_CTRL_BITNUM(x) ((x) << 0) +#define TDMIN_SWAP 0x04 +#define TDMIN_MASK0 0x08 +#define TDMIN_MASK1 0x0c +#define TDMIN_MASK2 0x10 +#define TDMIN_MASK3 0x14 +#define TDMIN_STAT 0x18 +#define TDMIN_MUTE_VAL 0x1c +#define TDMIN_MUTE0 0x20 +#define TDMIN_MUTE1 0x24 +#define TDMIN_MUTE2 0x28 +#define TDMIN_MUTE3 0x2c + +static const struct regmap_config axg_tdmin_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = TDMIN_MUTE3, +}; + +static const char * const axg_tdmin_sel_texts[] = { + "IN 0", "IN 1", "IN 2", "IN 3", "IN 4", "IN 5", +}; + +/* Change to special mux control to reset dapm */ +static SOC_ENUM_SINGLE_DECL(axg_tdmin_sel_enum, TDMIN_CTRL, + TDMIN_CTRL_SEL_SHIFT, axg_tdmin_sel_texts); + +static const struct snd_kcontrol_new axg_tdmin_in_mux = + SOC_DAPM_ENUM("Input Source", axg_tdmin_sel_enum); + +static struct snd_soc_dai * +axg_tdmin_get_be(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dai *be; + + snd_soc_dapm_widget_for_each_source_path(w, p) { + if (!p->connect) + continue; + + if (p->source->id == snd_soc_dapm_dai_out) + return (struct snd_soc_dai *)p->source->priv; + + be = axg_tdmin_get_be(p->source); + if (be) + return be; + } + + return NULL; +} + +static struct axg_tdm_stream * +axg_tdmin_get_tdm_stream(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dai *be = axg_tdmin_get_be(w); + + if (!be) + return NULL; + + return be->capture_dma_data; +} + +static void axg_tdmin_enable(struct regmap *map) +{ + /* Apply both reset */ + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_RST_OUT | TDMIN_CTRL_RST_IN, 0); + + /* Clear out reset before in reset */ + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_RST_OUT, TDMIN_CTRL_RST_OUT); + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_RST_IN, TDMIN_CTRL_RST_IN); + + /* Actually enable tdmin */ + regmap_update_bits(map, TDMIN_CTRL, + TDMIN_CTRL_ENABLE, TDMIN_CTRL_ENABLE); +} + +static void axg_tdmin_disable(struct regmap *map) +{ + regmap_update_bits(map, TDMIN_CTRL, TDMIN_CTRL_ENABLE, 0); +} + +static int axg_tdmin_prepare(struct regmap *map, struct axg_tdm_stream *ts) +{ + unsigned int val = 0; + + /* Set stream skew */ + switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + val |= TDMIN_CTRL_IN_BIT_SKEW(3); + break; + + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_DSP_B: + val = TDMIN_CTRL_IN_BIT_SKEW(2); + break; + + default: + pr_err("Unsupported format: %u\n", + ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + /* Set stream format mode */ + switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + val |= TDMIN_CTRL_I2S_MODE; + break; + } + + /* If the sample clock is inverted, invert it back for the formatter */ + if (axg_tdm_lrclk_invert(ts->iface->fmt)) + val |= TDMIN_CTRL_WS_INV; + + /* Set the slot width */ + val |= TDMIN_CTRL_BITNUM(ts->iface->slot_width - 1); + + /* + * The following also reset LSB_FIRST which result in the formatter + * placing the first bit received at bit 31 + */ + regmap_update_bits(map, TDMIN_CTRL, + (TDMIN_CTRL_IN_BIT_SKEW_MASK | TDMIN_CTRL_WS_INV | + TDMIN_CTRL_I2S_MODE | TDMIN_CTRL_LSB_FIRST | + TDMIN_CTRL_BITNUM_MASK), val); + + /* Set static swap mask configuration */ + regmap_write(map, TDMIN_SWAP, 0x76543210); + + return axg_tdm_formatter_set_channel_masks(map, ts, TDMIN_MASK0); +} + +static const struct snd_soc_dapm_widget axg_tdmin_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("IN 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("IN 5", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("SRC SEL", SND_SOC_NOPM, 0, 0, &axg_tdmin_in_mux), + SND_SOC_DAPM_PGA_E("DEC", SND_SOC_NOPM, 0, 0, NULL, 0, + axg_tdm_formatter_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD)), + SND_SOC_DAPM_AIF_OUT("OUT", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route axg_tdmin_dapm_routes[] = { + { "SRC SEL", "IN 0", "IN 0" }, + { "SRC SEL", "IN 1", "IN 1" }, + { "SRC SEL", "IN 2", "IN 2" }, + { "SRC SEL", "IN 3", "IN 3" }, + { "SRC SEL", "IN 4", "IN 4" }, + { "SRC SEL", "IN 5", "IN 5" }, + { "DEC", NULL, "SRC SEL" }, + { "OUT", NULL, "DEC" }, +}; + +static const struct snd_soc_component_driver axg_tdmin_component_drv = { + .dapm_widgets = axg_tdmin_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdmin_dapm_widgets), + .dapm_routes = axg_tdmin_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdmin_dapm_routes), +}; + +static const struct axg_tdm_formatter_ops axg_tdmin_ops = { + .get_stream = axg_tdmin_get_tdm_stream, + .prepare = axg_tdmin_prepare, + .enable = axg_tdmin_enable, + .disable = axg_tdmin_disable, +}; + +static const struct axg_tdm_formatter_driver axg_tdmin_drv = { + .component_drv = &axg_tdmin_component_drv, + .regmap_cfg = &axg_tdmin_regmap_cfg, + .ops = &axg_tdmin_ops, + .invert_sclk = false, +}; + +static const struct of_device_id axg_tdmin_of_match[] = { + { + .compatible = "amlogic,axg-tdmin", + .data = &axg_tdmin_drv, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_tdmin_of_match); + +static struct platform_driver axg_tdmin_pdrv = { + .probe = axg_tdm_formatter_probe, + .driver = { + .name = "axg-tdmin", + .of_match_table = axg_tdmin_of_match, + }, +}; +module_platform_driver(axg_tdmin_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG TDM input formatter driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From cbdfab3b675f4c34258b0ec9e4707de44e1f6989 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:43:02 +0200 Subject: ASoC: export snd_soc_of_get_slot_mask Amlogic's axg card driver can't use snd_soc_of_parse_tdm_slot() directly because it needs to handle 4 mask for each direction. Yet the parsing of each mask is the same, so export snd_soc_of_get_slot_mask() to reuse the the existing code. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 7 ++++--- 2 files changed, 7 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index a23ecdf3eff1..ace474e8649e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1433,6 +1433,9 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, const char *propname); +int snd_soc_of_get_slot_mask(struct device_node *np, + const char *prop_name, + unsigned int *mask); int snd_soc_of_parse_tdm_slot(struct device_node *np, unsigned int *tx_mask, unsigned int *rx_mask, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 08e189485009..ad5b0ef16d82 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3428,9 +3428,9 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets); -static int snd_soc_of_get_slot_mask(struct device_node *np, - const char *prop_name, - unsigned int *mask) +int snd_soc_of_get_slot_mask(struct device_node *np, + const char *prop_name, + unsigned int *mask) { u32 val; const __be32 *of_slot_mask = of_get_property(np, prop_name, &val); @@ -3445,6 +3445,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np, return val; } +EXPORT_SYMBOL_GPL(snd_soc_of_get_slot_mask); int snd_soc_of_parse_tdm_slot(struct device_node *np, unsigned int *tx_mask, -- cgit v1.2.3 From 7864a79f37b55769b817d5e6c5ae0ca4bfdba93b Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Tue, 17 Jul 2018 17:43:04 +0200 Subject: ASoC: meson: add axg sound card support Add the axg sound card to handle the specifities of the axg audio sub system. This card is required to: * setup the dpcm links specific to the AXG (with a cpu sound dai) * handle the 4 lanes masks of the tdm interfaces * add the loopback link when a tdm pad interface has a playback stream * handle multi-codec links Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 11 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-card.c | 671 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 684 insertions(+) create mode 100644 sound/soc/meson/axg-card.c (limited to 'sound/soc') diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 00d05df67b52..4cf93c05a982 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -43,6 +43,17 @@ config SND_MESON_AXG_TDMOUT Select Y or M to add support for TDM output formatter embedded in the Amlogic AXG SoC family +config SND_MESON_AXG_SOUND_CARD + tristate "Amlogic AXG Sound Card Support" + select SND_MESON_AXG_TDM_INTERFACE + imply SND_MESON_AXG_FRDDR + imply SND_MESON_AXG_TODDR + imply SND_MESON_AXG_TDMIN + imply SND_MESON_AXG_TDMOUT + imply SND_MESON_AXG_SPDIFOUT + help + Select Y or M to add support for the AXG SoC sound card + config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" imply SND_SOC_SPDIF diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index f62833fb44d8..c5e003b093db 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -7,6 +7,7 @@ snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o snd-soc-meson-axg-tdmin-objs := axg-tdmin.o snd-soc-meson-axg-tdmout-objs := axg-tdmout.o +snd-soc-meson-axg-sound-card-objs := axg-card.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o @@ -16,4 +17,5 @@ obj-$(CONFIG_SND_MESON_AXG_TDM_FORMATTER) += snd-soc-meson-axg-tdm-formatter.o obj-$(CONFIG_SND_MESON_AXG_TDM_INTERFACE) += snd-soc-meson-axg-tdm-interface.o obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o +obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c new file mode 100644 index 000000000000..d6d1081d94ad --- /dev/null +++ b/sound/soc/meson/axg-card.c @@ -0,0 +1,671 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet + +#include +#include +#include +#include + +#include "axg-tdm.h" + +struct axg_card { + struct snd_soc_card card; + void **link_data; +}; + +struct axg_dai_link_tdm_mask { + u32 tx; + u32 rx; +}; + +struct axg_dai_link_tdm_data { + unsigned int mclk_fs; + unsigned int slots; + unsigned int slot_width; + u32 *tx_mask; + u32 *rx_mask; + struct axg_dai_link_tdm_mask *codec_masks; +}; + +#define PREFIX "amlogic," + +static int axg_card_reallocate_links(struct axg_card *priv, + unsigned int num_links) +{ + struct snd_soc_dai_link *links; + void **ldata; + + links = krealloc(priv->card.dai_link, + num_links * sizeof(*priv->card.dai_link), + GFP_KERNEL | __GFP_ZERO); + ldata = krealloc(priv->link_data, + num_links * sizeof(*priv->link_data), + GFP_KERNEL | __GFP_ZERO); + + if (!links || !ldata) { + dev_err(priv->card.dev, "failed to allocate links\n"); + return -ENOMEM; + } + + priv->card.dai_link = links; + priv->link_data = ldata; + priv->card.num_links = num_links; + return 0; +} + +static int axg_card_parse_dai(struct snd_soc_card *card, + struct device_node *node, + struct device_node **dai_of_node, + const char **dai_name) +{ + struct of_phandle_args args; + int ret; + + if (!dai_name || !dai_of_node || !node) + return -EINVAL; + + ret = of_parse_phandle_with_args(node, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(card->dev, "can't parse dai %d\n", ret); + return ret; + } + *dai_of_node = args.np; + + return snd_soc_get_dai_name(&args, dai_name); +} + +static int axg_card_set_link_name(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + const char *prefix) +{ + char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s", + prefix, link->cpu_of_node->full_name); + if (!name) + return -ENOMEM; + + link->name = name; + link->stream_name = name; + + return 0; +} + +static void axg_card_clean_references(struct axg_card *priv) +{ + struct snd_soc_card *card = &priv->card; + struct snd_soc_dai_link *link; + int i, j; + + if (card->dai_link) { + for (i = 0; i < card->num_links; i++) { + link = &card->dai_link[i]; + of_node_put(link->cpu_of_node); + for (j = 0; j < link->num_codecs; j++) + of_node_put(link->codecs[j].of_node); + } + } + + if (card->aux_dev) { + for (i = 0; i < card->num_aux_devs; i++) + of_node_put(card->aux_dev[i].codec_of_node); + } + + kfree(card->dai_link); + kfree(priv->link_data); +} + +static int axg_card_add_aux_devices(struct snd_soc_card *card) +{ + struct device_node *node = card->dev->of_node; + struct snd_soc_aux_dev *aux; + int num, i; + + num = of_count_phandle_with_args(node, PREFIX "aux-devs", NULL); + if (num == -ENOENT) { + /* + * It is ok to have no auxiliary devices but for this card it + * is a strange situtation. Let's warn the about it. + */ + dev_warn(card->dev, "card has no auxiliary devices\n"); + return 0; + } else if (num < 0) { + dev_err(card->dev, "error getting auxiliary devices: %d\n", + num); + return num; + } + + aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL); + if (!aux) + return -ENOMEM; + card->aux_dev = aux; + card->num_aux_devs = num; + + for (i = 0; i < card->num_aux_devs; i++, aux++) { + aux->codec_of_node = of_parse_phandle(node, + PREFIX "aux-devs", i); + if (!aux->codec_of_node) + return -EINVAL; + } + + return 0; +} + +static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct axg_dai_link_tdm_data *be = + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + struct snd_soc_dai *codec_dai; + unsigned int mclk; + int ret, i; + + if (be->mclk_fs) { + mclk = params_rate(params) * be->mclk_fs; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret && ret != -ENOTSUPP) + return ret; + } + + return 0; +} + +static const struct snd_soc_ops axg_card_tdm_be_ops = { + .hw_params = axg_card_tdm_be_hw_params, +}; + +static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct axg_dai_link_tdm_data *be = + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + struct snd_soc_dai *codec_dai; + int ret, i; + + for (i = 0; i < rtd->num_codecs; i++) { + codec_dai = rtd->codec_dais[i]; + ret = snd_soc_dai_set_tdm_slot(codec_dai, + be->codec_masks[i].tx, + be->codec_masks[i].rx, + be->slots, be->slot_width); + if (ret && ret != -ENOTSUPP) { + dev_err(codec_dai->dev, + "setting tdm link slots failed\n"); + return ret; + } + } + + ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, be->tx_mask, be->rx_mask, + be->slots, be->slot_width); + if (ret) { + dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n"); + return ret; + } + + return 0; +} + +static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct axg_dai_link_tdm_data *be = + (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; + int ret; + + /* The loopback rx_mask is the pad tx_mask */ + ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, NULL, be->tx_mask, + be->slots, be->slot_width); + if (ret) { + dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n"); + return ret; + } + + return 0; +} + +static int axg_card_add_tdm_loopback(struct snd_soc_card *card, + int *index) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link *pad = &card->dai_link[*index]; + struct snd_soc_dai_link *lb; + int ret; + + /* extend links */ + ret = axg_card_reallocate_links(priv, card->num_links + 1); + if (ret) + return ret; + + lb = &card->dai_link[*index + 1]; + + lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name); + if (!lb->name) + return -ENOMEM; + + lb->stream_name = lb->name; + lb->cpu_of_node = pad->cpu_of_node; + lb->cpu_dai_name = "TDM Loopback"; + lb->codec_name = "snd-soc-dummy"; + lb->codec_dai_name = "snd-soc-dummy-dai"; + lb->dpcm_capture = 1; + lb->no_pcm = 1; + lb->ops = &axg_card_tdm_be_ops; + lb->init = axg_card_tdm_dai_lb_init; + + /* Provide the same link data to the loopback */ + priv->link_data[*index + 1] = priv->link_data[*index]; + + /* + * axg_card_clean_references() will iterate over this link, + * make sure the node count is balanced + */ + of_node_get(lb->cpu_of_node); + + /* Let add_links continue where it should */ + *index += 1; + + return 0; +} + +static unsigned int axg_card_parse_daifmt(struct device_node *node, + struct device_node *cpu_node) +{ + struct device_node *bitclkmaster = NULL; + struct device_node *framemaster = NULL; + unsigned int daifmt; + + daifmt = snd_soc_of_parse_daifmt(node, PREFIX, + &bitclkmaster, &framemaster); + daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + + /* If no master is provided, default to cpu master */ + if (!bitclkmaster || bitclkmaster == cpu_node) { + daifmt |= (!framemaster || framemaster == cpu_node) ? + SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM; + } else { + daifmt |= (!framemaster || framemaster == cpu_node) ? + SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM; + } + + of_node_put(bitclkmaster); + of_node_put(framemaster); + + return daifmt; +} + +static int axg_card_parse_cpu_tdm_slots(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + struct axg_dai_link_tdm_data *be) +{ + char propname[32]; + u32 tx, rx; + int i; + + be->tx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES, + sizeof(*be->tx_mask), GFP_KERNEL); + be->rx_mask = devm_kcalloc(card->dev, AXG_TDM_NUM_LANES, + sizeof(*be->rx_mask), GFP_KERNEL); + if (!be->tx_mask || !be->rx_mask) + return -ENOMEM; + + for (i = 0, tx = 0; i < AXG_TDM_NUM_LANES; i++) { + snprintf(propname, 32, "dai-tdm-slot-tx-mask-%d", i); + snd_soc_of_get_slot_mask(node, propname, &be->tx_mask[i]); + tx = max(tx, be->tx_mask[i]); + } + + /* Disable playback is the interface has no tx slots */ + if (!tx) + link->dpcm_playback = 0; + + for (i = 0, rx = 0; i < AXG_TDM_NUM_LANES; i++) { + snprintf(propname, 32, "dai-tdm-slot-rx-mask-%d", i); + snd_soc_of_get_slot_mask(node, propname, &be->rx_mask[i]); + rx = max(rx, be->rx_mask[i]); + } + + /* Disable capture is the interface has no rx slots */ + if (!rx) + link->dpcm_capture = 0; + + /* ... but the interface should at least have one of them */ + if (!tx && !rx) { + dev_err(card->dev, "tdm link has no cpu slots\n"); + return -EINVAL; + } + + of_property_read_u32(node, "dai-tdm-slot-num", &be->slots); + if (!be->slots) { + /* + * If the slot number is not provided, set it such as it + * accommodates the largest mask + */ + be->slots = fls(max(tx, rx)); + } else if (be->slots < fls(max(tx, rx)) || be->slots > 32) { + /* + * Error if the slots can't accommodate the largest mask or + * if it is just too big + */ + dev_err(card->dev, "bad slot number\n"); + return -EINVAL; + } + + of_property_read_u32(node, "dai-tdm-slot-width", &be->slot_width); + + return 0; +} + +static int axg_card_parse_codecs_masks(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + struct axg_dai_link_tdm_data *be) +{ + struct axg_dai_link_tdm_mask *codec_mask; + struct device_node *np; + + codec_mask = devm_kcalloc(card->dev, link->num_codecs, + sizeof(*codec_mask), GFP_KERNEL); + if (!codec_mask) + return -ENOMEM; + + be->codec_masks = codec_mask; + + for_each_child_of_node(node, np) { + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask", + &codec_mask->rx); + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask", + &codec_mask->tx); + + codec_mask++; + } + + return 0; +} + +static int axg_card_parse_tdm(struct snd_soc_card *card, + struct device_node *node, + int *index) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link *link = &card->dai_link[*index]; + struct axg_dai_link_tdm_data *be; + int ret; + + /* Allocate tdm link parameters */ + be = devm_kzalloc(card->dev, sizeof(*be), GFP_KERNEL); + if (!be) + return -ENOMEM; + priv->link_data[*index] = be; + + /* Setup tdm link */ + link->ops = &axg_card_tdm_be_ops; + link->init = axg_card_tdm_dai_init; + link->dai_fmt = axg_card_parse_daifmt(node, link->cpu_of_node); + + of_property_read_u32(node, "mclk-fs", &be->mclk_fs); + + ret = axg_card_parse_cpu_tdm_slots(card, link, node, be); + if (ret) { + dev_err(card->dev, "error parsing tdm link slots\n"); + return ret; + } + + ret = axg_card_parse_codecs_masks(card, link, node, be); + if (ret) + return ret; + + /* Add loopback if the pad dai has playback */ + if (link->dpcm_playback) { + ret = axg_card_add_tdm_loopback(card, index); + if (ret) + return ret; + } + + return 0; +} + +static int axg_card_set_be_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node) +{ + struct snd_soc_dai_link_component *codec; + struct device_node *np; + int ret, num_codecs; + + link->no_pcm = 1; + link->dpcm_playback = 1; + link->dpcm_capture = 1; + + num_codecs = of_get_child_count(node); + if (!num_codecs) { + dev_err(card->dev, "be link %s has no codec\n", + node->full_name); + return -EINVAL; + } + + codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL); + if (!codec) + return -ENOMEM; + + link->codecs = codec; + link->num_codecs = num_codecs; + + for_each_child_of_node(node, np) { + ret = axg_card_parse_dai(card, np, &codec->of_node, + &codec->dai_name); + if (ret) { + of_node_put(np); + return ret; + } + + codec++; + } + + ret = axg_card_set_link_name(card, link, "be"); + if (ret) + dev_err(card->dev, "error setting %s link name\n", np->name); + + return ret; +} + +static int axg_card_set_fe_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + bool is_playback) +{ + link->dynamic = 1; + link->dpcm_merged_format = 1; + link->dpcm_merged_chan = 1; + link->dpcm_merged_rate = 1; + link->codec_dai_name = "snd-soc-dummy-dai"; + link->codec_name = "snd-soc-dummy"; + + if (is_playback) + link->dpcm_playback = 1; + else + link->dpcm_capture = 1; + + return axg_card_set_link_name(card, link, "fe"); +} + +static int axg_card_cpu_is_capture_fe(struct device_node *np) +{ + return of_device_is_compatible(np, PREFIX "axg-toddr"); +} + +static int axg_card_cpu_is_playback_fe(struct device_node *np) +{ + return of_device_is_compatible(np, PREFIX "axg-frddr"); +} + +static int axg_card_cpu_is_tdm_iface(struct device_node *np) +{ + return of_device_is_compatible(np, PREFIX "axg-tdm-iface"); +} + +static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, + int *index) +{ + struct snd_soc_dai_link *dai_link = &card->dai_link[*index]; + int ret; + + ret = axg_card_parse_dai(card, np, &dai_link->cpu_of_node, + &dai_link->cpu_dai_name); + if (ret) + return ret; + + if (axg_card_cpu_is_playback_fe(dai_link->cpu_of_node)) + ret = axg_card_set_fe_link(card, dai_link, true); + else if (axg_card_cpu_is_capture_fe(dai_link->cpu_of_node)) + ret = axg_card_set_fe_link(card, dai_link, false); + else + ret = axg_card_set_be_link(card, dai_link, np); + + if (ret) + return ret; + + if (axg_card_cpu_is_tdm_iface(dai_link->cpu_of_node)) + ret = axg_card_parse_tdm(card, np, index); + + return ret; +} + +static int axg_card_add_links(struct snd_soc_card *card) +{ + struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct device_node *node = card->dev->of_node; + struct device_node *np; + int num, i, ret; + + num = of_get_child_count(node); + if (!num) { + dev_err(card->dev, "card has no links\n"); + return -EINVAL; + } + + ret = axg_card_reallocate_links(priv, num); + if (ret) + return ret; + + i = 0; + for_each_child_of_node(node, np) { + ret = axg_card_add_link(card, np, &i); + if (ret) { + of_node_put(np); + return ret; + } + + i++; + } + + return 0; +} + +static int axg_card_parse_of_optional(struct snd_soc_card *card, + const char *propname, + int (*func)(struct snd_soc_card *c, + const char *p)) +{ + /* If property is not provided, don't fail ... */ + if (!of_property_read_bool(card->dev->of_node, propname)) + return 0; + + /* ... but do fail if it is provided and the parsing fails */ + return func(card, propname); +} + +static const struct of_device_id axg_card_of_match[] = { + { .compatible = "amlogic,axg-sound-card", }, + {} +}; +MODULE_DEVICE_TABLE(of, axg_card_of_match); + +static int axg_card_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct axg_card *priv; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + platform_set_drvdata(pdev, priv); + snd_soc_card_set_drvdata(&priv->card, priv); + + priv->card.owner = THIS_MODULE; + priv->card.dev = dev; + + ret = snd_soc_of_parse_card_name(&priv->card, PREFIX "name"); + if (ret < 0) + return ret; + + ret = axg_card_parse_of_optional(&priv->card, PREFIX "routing", + snd_soc_of_parse_audio_routing); + if (ret) { + dev_err(dev, "error while parsing routing\n"); + return ret; + } + + ret = axg_card_parse_of_optional(&priv->card, PREFIX "widgets", + snd_soc_of_parse_audio_simple_widgets); + if (ret) { + dev_err(dev, "error while parsing widgets\n"); + return ret; + } + + ret = axg_card_add_links(&priv->card); + if (ret) + goto out_err; + + ret = axg_card_add_aux_devices(&priv->card); + if (ret) + goto out_err; + + ret = devm_snd_soc_register_card(dev, &priv->card); + if (ret) + goto out_err; + + return 0; + +out_err: + axg_card_clean_references(priv); + return ret; +} + +static int axg_card_remove(struct platform_device *pdev) +{ + struct axg_card *priv = platform_get_drvdata(pdev); + + axg_card_clean_references(priv); + + return 0; +} + +static struct platform_driver axg_card_pdrv = { + .probe = axg_card_probe, + .remove = axg_card_remove, + .driver = { + .name = "axg-sound-card", + .of_match_table = axg_card_of_match, + }, +}; +module_platform_driver(axg_card_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG ALSA machine driver"); +MODULE_AUTHOR("Jerome Brunet "); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From a8e43c21a8a32a3af4abc605b6ebcab039f28e00 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 16 Jul 2018 08:24:45 +0200 Subject: ASoC: pxa: remove clock divider and pll setup from zylonite and magician The SSP DAI now handles the clocking setup itself, all it needs is the master clock frequency. Remove the code from Zylonite and Magician platforms. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/magician.c | 106 +---------------------------------------------- sound/soc/pxa/zylonite.c | 9 ---- 2 files changed, 2 insertions(+), 113 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 2fc012b06c43..935a248e5bf6 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -90,95 +90,9 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int acps, acds, width; - unsigned int div4 = PXA_SSP_CLK_SCDB_4; + unsigned int width; int ret = 0; - width = snd_pcm_format_physical_width(params_format(params)); - - /* - * rate = SSPSCLK / (2 * width(16 or 32)) - * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) - */ - switch (params_rate(params)) { - case 8000: - /* off by a factor of 2: bug in the PXA27x audio clock? */ - acps = 32842000; - switch (width) { - case 16: - /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_16; - break; - default: /* 32 */ - /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_8; - } - break; - case 11025: - acps = 5622000; - switch (width) { - case 16: - /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_4; - break; - default: /* 32 */ - /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - } - break; - case 22050: - acps = 5622000; - switch (width) { - case 16: - /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 44100: - acps = 5622000; - switch (width) { - case 16: - /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 48000: - acps = 12235000; - switch (width) { - case 16: - /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 96000: - default: - acps = 12235000; - switch (width) { - case 16: - /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - break; - default: /* 32 */ - /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - div4 = PXA_SSP_CLK_SCDB_1; - break; - } - break; - } - /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -191,6 +105,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + width = snd_pcm_format_physical_width(params_format(params)); ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; @@ -201,23 +116,6 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* set the SSP audio system clock ACDS divider */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_ACDS, acds); - if (ret < 0) - return ret; - - /* set the SSP audio system clock SCDB divider4 */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_SCDB, div4); - if (ret < 0) - return ret; - - /* set SSP audio pll clock */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps); - if (ret < 0) - return ret; - return 0; } diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ba468e560dd2..230eee450f45 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -83,11 +83,9 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int pll_out = 0; unsigned int wm9713_div = 0; int ret = 0; int rate = params_rate(params); - int width = snd_pcm_format_physical_width(params_format(params)); /* Only support ratios that we can generate neatly from the AC97 * based master clock - in particular, this excludes 44.1kHz. @@ -109,17 +107,10 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Add 1 to the width for the leading clock cycle */ - pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out); - if (ret < 0) - return ret; - if (clk_pout) ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, WM9713_PCMDIV(wm9713_div)); -- cgit v1.2.3 From a241c3d95b8b985834d218eedde3923ed683e862 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 24 Jul 2018 11:36:45 +0200 Subject: ASoC: meson: axg-spdifout: select SND_PCM_IEC958 When CONFIG_SND_PCM_IEC958 is disabled, we get a link error for the new driver: sound/soc/meson/axg-spdifout.o: In function `axg_spdifout_hw_params': axg-spdifout.c:(.text+0x650): undefined reference to `snd_pcm_create_iec958_consumer_hw_params' The other users use 'select', so we should do the same here. Fixes: 53eb4b7aaa04 ("ASoC: meson: add axg spdif output") Signed-off-by: Arnd Bergmann Acked-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 4cf93c05a982..8af8bc358a90 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -56,6 +56,7 @@ config SND_MESON_AXG_SOUND_CARD config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" + select SND_PCM_IEC958 imply SND_SOC_SPDIF help Select Y or M to add support for SPDIF output serializer embedded -- cgit v1.2.3 From d96f8bd28cd0bae3e6702ae90df593628ef6906f Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 24 Jul 2018 15:49:23 +0800 Subject: ASoC: rt5514: Fix the issue of the delay volume applied The patch fixes the issue of the delay volume applied. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5514.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 1570b91bf018..dca82dd6e3bf 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, }; static const struct reg_default rt5514_reg[] = { @@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, -- cgit v1.2.3 From 467b061f1ac80f323bedb56d20a24f7bc0d2cec5 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 23 Jul 2018 16:54:03 +0100 Subject: ASoC: core: add support to snd_soc_dai_get_channel_map() On Qualcomm platforms, specifically with SLIMbus interfaced codecs, the codec slim channel numbers are passed to DSP while configuring the slim audio path. Having get_channel_map() would allow dais to share such information across multiple dais. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 8 ++++++++ sound/soc/soc-core.c | 22 ++++++++++++++++++++++ 2 files changed, 30 insertions(+) (limited to 'sound/soc') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index a14bc0608ae9..f5d70041108f 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -138,6 +138,11 @@ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction); + +int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot); + int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); struct snd_soc_dai_ops { @@ -165,6 +170,9 @@ struct snd_soc_dai_ops { int (*set_channel_map)(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot); + int (*get_channel_map)(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); int (*set_sdw_stream)(struct snd_soc_dai *dai, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad5b0ef16d82..81b27923303d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2679,6 +2679,28 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); +/** + * snd_soc_dai_get_channel_map - Get DAI audio channel map + * @dai: DAI + * @tx_num: how many TX channels + * @tx_slot: pointer to an array which imply the TX slot number channel + * 0~num-1 uses + * @rx_num: how many RX channels + * @rx_slot: pointer to an array which imply the RX slot number channel + * 0~num-1 uses + */ +int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot) +{ + if (dai->driver->ops->get_channel_map) + return dai->driver->ops->get_channel_map(dai, tx_num, tx_slot, + rx_num, rx_slot); + else + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_get_channel_map); + /** * snd_soc_dai_set_tristate - configure DAI system or master clock. * @dai: DAI -- cgit v1.2.3 From aa624a0a9243460b9037889a05663d3b9a9e8f99 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 24 Jul 2018 09:48:30 -0300 Subject: ASoC: fsl-asoc-card: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 18 +++++++----------- 1 file changed, 7 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4a6750aa3637..07808c6d5461 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -1,14 +1,10 @@ -/* - * Freescale Generic ASoC Sound Card driver with ASRC - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale Generic ASoC Sound Card driver with ASRC +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen #include #include -- cgit v1.2.3 From 2ba28053686334075c8c8bff6c4f6e3fc9ac51ae Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 24 Jul 2018 09:48:31 -0300 Subject: ASoC: fsl_asrc: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc.c | 18 +++++++----------- sound/soc/fsl/fsl_asrc.h | 5 +---- sound/soc/fsl/fsl_asrc_dma.c | 18 +++++++----------- 3 files changed, 15 insertions(+), 26 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index adfb8135d739..528e8b108422 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -1,14 +1,10 @@ -/* - * Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen #include #include diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index d558dd5499a5..c60075112570 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -1,13 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* * fsl_asrc.h - Freescale ASRC ALSA SoC header file * * Copyright (C) 2014 Freescale Semiconductor, Inc. * * Author: Nicolin Chen - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. */ #ifndef _FSL_ASRC_H diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 565e16d8fe85..1033ac6631b0 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -1,14 +1,10 @@ -/* - * Freescale ASRC ALSA SoC Platform (DMA) driver - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale ASRC ALSA SoC Platform (DMA) driver +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen #include #include -- cgit v1.2.3 From ad47191a72ca0a95327e8bb16155e5e216f1d97d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 24 Jul 2018 09:48:32 -0300 Subject: ASoC: fsl_utils: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_utils.c | 18 +++++++----------- sound/soc/fsl/fsl_utils.h | 7 ++----- 2 files changed, 9 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 7592b0406370..7f0fa4b52223 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -1,14 +1,10 @@ -/** - * Freescale ALSA SoC Machine driver utility - * - * Author: Timur Tabi - * - * Copyright 2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale ALSA SoC Machine driver utility +// +// Author: Timur Tabi +// +// Copyright 2010 Freescale Semiconductor, Inc. #include #include diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h index 1687b66ef18e..c5dc2a14b492 100644 --- a/sound/soc/fsl/fsl_utils.h +++ b/sound/soc/fsl/fsl_utils.h @@ -1,13 +1,10 @@ -/** +/* SPDX-License-Identifier: GPL-2.0 */ +/* * Freescale ALSA SoC Machine driver utility * * Author: Timur Tabi * * Copyright 2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. */ #ifndef _FSL_UTILS_H -- cgit v1.2.3 From 6b24e03ecf1ce188071d142115233b10ed8767b2 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 24 Jul 2018 09:48:33 -0300 Subject: ASoC: imx-sgtl5000: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-sgtl5000.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index b99e0b5e00e9..c29200cf755a 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -1,14 +1,7 @@ -/* - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * - * The code contained herein is licensed under the GNU General Public - * License. You may obtain a copy of the GNU General Public License - * Version 2 or later at the following locations: - * - * http://www.opensource.org/licenses/gpl-license.html - * http://www.gnu.org/copyleft/gpl.html - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright 2012 Freescale Semiconductor, Inc. +// Copyright 2012 Linaro Ltd. #include #include -- cgit v1.2.3 From d77a760842751eb620a7e74f74e5a5ad6b3d6cef Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 20 Jul 2018 09:48:20 +0800 Subject: ASoC: rt5631: add Volume to the name of volume control add Volume to the name of volume control. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index cf6dce69eb2a..e52e4670cf65 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -229,10 +229,10 @@ static SOC_ENUM_SINGLE_DECL(rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG, static const struct snd_kcontrol_new rt5631_snd_controls[] = { /* MIC */ SOC_ENUM("MIC1 Mode Control", rt5631_mic1_mode_enum), - SOC_SINGLE_TLV("MIC1 Boost", RT5631_MIC_CTRL_2, + SOC_SINGLE_TLV("MIC1 Boost Volume", RT5631_MIC_CTRL_2, RT5631_MIC1_BOOST_SHIFT, 8, 0, mic_bst_tlv), SOC_ENUM("MIC2 Mode Control", rt5631_mic2_mode_enum), - SOC_SINGLE_TLV("MIC2 Boost", RT5631_MIC_CTRL_2, + SOC_SINGLE_TLV("MIC2 Boost Volume", RT5631_MIC_CTRL_2, RT5631_MIC2_BOOST_SHIFT, 8, 0, mic_bst_tlv), /* MONO IN */ SOC_ENUM("MONOIN Mode Control", rt5631_monoin_mode_enum), -- cgit v1.2.3 From 92beb0a26975c6459794d885d27f48357c1aefd0 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 24 Jul 2018 16:12:44 -0500 Subject: ASoC: Intel: Haswell: fix endianness handling Make all Sparse warnings go away by using le16/32_to_cpu. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/haswell/sst-haswell-dsp.c | 53 +++++++++++++++++-------------- 1 file changed, 29 insertions(+), 24 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c index b2bec36d074c..a28220e67cdf 100644 --- a/sound/soc/intel/haswell/sst-haswell-dsp.c +++ b/sound/soc/intel/haswell/sst-haswell-dsp.c @@ -93,29 +93,31 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, struct sst_module_template template; int count, ret; void __iomem *ram; + int type = le16_to_cpu(module->type); + int entry_point = le32_to_cpu(module->entry_point); /* TODO: allowed module types need to be configurable */ - if (module->type != SST_HSW_MODULE_BASE_FW - && module->type != SST_HSW_MODULE_PCM_SYSTEM - && module->type != SST_HSW_MODULE_PCM - && module->type != SST_HSW_MODULE_PCM_REFERENCE - && module->type != SST_HSW_MODULE_PCM_CAPTURE - && module->type != SST_HSW_MODULE_WAVES - && module->type != SST_HSW_MODULE_LPAL) + if (type != SST_HSW_MODULE_BASE_FW && + type != SST_HSW_MODULE_PCM_SYSTEM && + type != SST_HSW_MODULE_PCM && + type != SST_HSW_MODULE_PCM_REFERENCE && + type != SST_HSW_MODULE_PCM_CAPTURE && + type != SST_HSW_MODULE_WAVES && + type != SST_HSW_MODULE_LPAL) return 0; dev_dbg(dsp->dev, "new module sign 0x%s size 0x%x blocks 0x%x type 0x%x\n", module->signature, module->mod_size, - module->blocks, module->type); - dev_dbg(dsp->dev, " entrypoint 0x%x\n", module->entry_point); + module->blocks, type); + dev_dbg(dsp->dev, " entrypoint 0x%x\n", entry_point); dev_dbg(dsp->dev, " persistent 0x%x scratch 0x%x\n", module->info.persistent_size, module->info.scratch_size); memset(&template, 0, sizeof(template)); - template.id = module->type; - template.entry = module->entry_point - 4; - template.persistent_size = module->info.persistent_size; - template.scratch_size = module->info.scratch_size; + template.id = type; + template.entry = entry_point - 4; + template.persistent_size = le32_to_cpu(module->info.persistent_size); + template.scratch_size = le32_to_cpu(module->info.scratch_size); mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -123,26 +125,26 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, block = (void *)module + sizeof(*module); - for (count = 0; count < module->blocks; count++) { + for (count = 0; count < le32_to_cpu(module->blocks); count++) { - if (block->size <= 0) { + if (le32_to_cpu(block->size) <= 0) { dev_err(dsp->dev, "error: block %d size invalid\n", count); sst_module_free(mod); return -EINVAL; } - switch (block->type) { + switch (le32_to_cpu(block->type)) { case SST_HSW_IRAM: ram = dsp->addr.lpe; - mod->offset = - block->ram_offset + dsp->addr.iram_offset; + mod->offset = le32_to_cpu(block->ram_offset) + + dsp->addr.iram_offset; mod->type = SST_MEM_IRAM; break; case SST_HSW_DRAM: case SST_HSW_REGS: ram = dsp->addr.lpe; - mod->offset = block->ram_offset; + mod->offset = le32_to_cpu(block->ram_offset); mod->type = SST_MEM_DRAM; break; default: @@ -152,7 +154,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - mod->size = block->size; + mod->size = le32_to_cpu(block->size); mod->data = (void *)block + sizeof(*block); mod->data_offset = mod->data - fw->dma_buf; @@ -169,7 +171,8 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return ret; } - block = (void *)block + sizeof(*block) + block->size; + block = (void *)block + sizeof(*block) + + le32_to_cpu(block->size); } mod->state = SST_MODULE_STATE_LOADED; @@ -188,7 +191,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) /* verify FW */ if ((strncmp(header->signature, SST_HSW_FW_SIGN, 4) != 0) || - (sst_fw->size != header->file_size + sizeof(*header))) { + (sst_fw->size != + le32_to_cpu(header->file_size) + sizeof(*header))) { dev_err(dsp->dev, "error: invalid fw sign/filesize mismatch\n"); return -EINVAL; } @@ -199,7 +203,7 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) /* parse each module */ module = (void *)sst_fw->dma_buf + sizeof(*header); - for (count = 0; count < header->modules; count++) { + for (count = 0; count < le32_to_cpu(header->modules); count++) { /* module */ ret = hsw_parse_module(dsp, sst_fw, module); @@ -207,7 +211,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) dev_err(dsp->dev, "error: invalid module %d\n", count); return ret; } - module = (void *)module + sizeof(*module) + module->mod_size; + module = (void *)module + sizeof(*module) + + le32_to_cpu(module->mod_size); } return 0; -- cgit v1.2.3 From 86efd35ec1e34094914494dacb83077e5df2d1b8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 24 Jul 2018 16:12:45 -0500 Subject: ASoC: Intel: Skylake: BDL definitions should be __le32 Make sure definitions are consistent with usage. Detected with Sparse. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-sst-cldma.c | 8 ++++---- sound/soc/intel/skylake/skl-sst-cldma.h | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/skylake/skl-sst-cldma.c b/sound/soc/intel/skylake/skl-sst-cldma.c index d2b1d60fec02..5bc0d38da7e3 100644 --- a/sound/soc/intel/skylake/skl-sst-cldma.c +++ b/sound/soc/intel/skylake/skl-sst-cldma.c @@ -83,9 +83,9 @@ static void skl_cldma_stream_clear(struct sst_dsp *ctx) /* Code loader helper APIs */ static void skl_cldma_setup_bdle(struct sst_dsp *ctx, struct snd_dma_buffer *dmab_data, - u32 **bdlp, int size, int with_ioc) + __le32 **bdlp, int size, int with_ioc) { - u32 *bdl = *bdlp; + __le32 *bdl = *bdlp; ctx->cl_dev.frags = 0; while (size > 0) { @@ -330,7 +330,7 @@ void skl_cldma_process_intr(struct sst_dsp *ctx) int skl_cldma_prepare(struct sst_dsp *ctx) { int ret; - u32 *bdl; + __le32 *bdl; ctx->cl_dev.bufsize = SKL_MAX_BUFFER_SIZE; @@ -359,7 +359,7 @@ int skl_cldma_prepare(struct sst_dsp *ctx) ctx->dsp_ops.free_dma_buf(ctx->dev, &ctx->cl_dev.dmab_data); return ret; } - bdl = (u32 *)ctx->cl_dev.dmab_bdl.area; + bdl = (__le32 *)ctx->cl_dev.dmab_bdl.area; /* Allocate BDLs */ ctx->cl_dev.ops.cl_setup_bdle(ctx, &ctx->cl_dev.dmab_data, diff --git a/sound/soc/intel/skylake/skl-sst-cldma.h b/sound/soc/intel/skylake/skl-sst-cldma.h index 5b730a1a0ae4..ec736921a083 100644 --- a/sound/soc/intel/skylake/skl-sst-cldma.h +++ b/sound/soc/intel/skylake/skl-sst-cldma.h @@ -203,7 +203,7 @@ struct sst_dsp; struct skl_cl_dev_ops { void (*cl_setup_bdle)(struct sst_dsp *ctx, struct snd_dma_buffer *dmab_data, - u32 **bdlp, int size, int with_ioc); + __le32 **bdlp, int size, int with_ioc); void (*cl_setup_controller)(struct sst_dsp *ctx, struct snd_dma_buffer *dmab_bdl, unsigned int max_size, u32 page_count); -- cgit v1.2.3 From ef3cb7423358ff0cadacb76f26e3e1f4da8be4ce Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 24 Jul 2018 16:12:46 -0500 Subject: ASoC: Intel: common: make sst_dma functions static sst_dma_new and sst_dma_free are not used in any other file and don't have a prototype. Move to static functions and remove EXPORT_SYMBOL_GPL statement. Reported by sparse warnings. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/sst-firmware.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 657afc02f1c4..11041aedea31 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -270,7 +270,7 @@ void sst_dsp_dma_put_channel(struct sst_dsp *dsp) } EXPORT_SYMBOL_GPL(sst_dsp_dma_put_channel); -int sst_dma_new(struct sst_dsp *sst) +static int sst_dma_new(struct sst_dsp *sst) { struct sst_pdata *sst_pdata = sst->pdata; struct sst_dma *dma; @@ -320,9 +320,8 @@ err_dma_dev: devm_kfree(sst->dev, dma); return ret; } -EXPORT_SYMBOL(sst_dma_new); -void sst_dma_free(struct sst_dma *dma) +static void sst_dma_free(struct sst_dma *dma) { if (dma == NULL) @@ -335,7 +334,6 @@ void sst_dma_free(struct sst_dma *dma) dw_remove(dma->chip); } -EXPORT_SYMBOL(sst_dma_free); /* create new generic firmware object */ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, -- cgit v1.2.3 From ce1cfe295abaa7436f9049bcb2562c843089abfc Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 24 Jul 2018 16:12:47 -0500 Subject: ASoC: Intel: Atom: simplify iomem address and casts Simplify code and add relevant casts to make Sparse warnings go away Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_drv_interface.c | 29 ++++++++++++++-------------- 1 file changed, 15 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 6a8b253c58d2..5455d6e0ab53 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -266,17 +266,15 @@ static int sst_cdev_ack(struct device *dev, unsigned int str_id, stream->cumm_bytes += bytes; dev_dbg(dev, "bytes copied %d inc by %ld\n", stream->cumm_bytes, bytes); - memcpy_fromio(&fw_tstamp, - ((void *)(ctx->mailbox + ctx->tstamp) - +(str_id * sizeof(fw_tstamp))), - sizeof(fw_tstamp)); + addr = ((void __iomem *)(ctx->mailbox + ctx->tstamp)) + + (str_id * sizeof(fw_tstamp)); + + memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp)); fw_tstamp.bytes_copied = stream->cumm_bytes; dev_dbg(dev, "bytes sent to fw %llu inc by %ld\n", fw_tstamp.bytes_copied, bytes); - addr = ((void *)(ctx->mailbox + ctx->tstamp)) + - (str_id * sizeof(fw_tstamp)); offset = offsetof(struct snd_sst_tstamp, bytes_copied); sst_shim_write(addr, offset, fw_tstamp.bytes_copied); return 0; @@ -360,11 +358,12 @@ static int sst_cdev_tstamp(struct device *dev, unsigned int str_id, struct snd_sst_tstamp fw_tstamp = {0,}; struct stream_info *stream; struct intel_sst_drv *ctx = dev_get_drvdata(dev); + void __iomem *addr; + + addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) + + (str_id * sizeof(fw_tstamp)); - memcpy_fromio(&fw_tstamp, - ((void *)(ctx->mailbox + ctx->tstamp) - +(str_id * sizeof(fw_tstamp))), - sizeof(fw_tstamp)); + memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp)); stream = get_stream_info(ctx, str_id); if (!stream) @@ -530,6 +529,7 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info) struct snd_sst_tstamp fw_tstamp; unsigned int str_id; struct intel_sst_drv *ctx = dev_get_drvdata(dev); + void __iomem *addr; str_id = info->str_id; stream = get_stream_info(ctx, str_id); @@ -540,10 +540,11 @@ static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info) return -EINVAL; substream = stream->pcm_substream; - memcpy_fromio(&fw_tstamp, - ((void *)(ctx->mailbox + ctx->tstamp) - + (str_id * sizeof(fw_tstamp))), - sizeof(fw_tstamp)); + addr = (void __iomem *)(ctx->mailbox + ctx->tstamp) + + (str_id * sizeof(fw_tstamp)); + + memcpy_fromio(&fw_tstamp, addr, sizeof(fw_tstamp)); + return sst_calc_tstamp(ctx, info, substream, &fw_tstamp); } -- cgit v1.2.3 From 9a0daaab31e9e39047ced79409313c34dae4635a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 24 Jul 2018 16:12:48 -0500 Subject: ASoC: Intel: Atom: fix inversion between __iowrite32 and __ioread32 This looks like a copy/paste issue, but clearly there is an inversion that is obvious when checking the arguments. Detected with Sparse - now that we have fewer warnings this one was easy to find. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_loader.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index a686eef2cf7f..27413ebae956 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -44,15 +44,15 @@ void memcpy32_toio(void __iomem *dst, const void *src, int count) /* __iowrite32_copy uses 32-bit count values so divide by 4 for * right count in words */ - __iowrite32_copy(dst, src, count/4); + __iowrite32_copy(dst, src, count / 4); } void memcpy32_fromio(void *dst, const void __iomem *src, int count) { - /* __iowrite32_copy uses 32-bit count values so divide by 4 for + /* __ioread32_copy uses 32-bit count values so divide by 4 for * right count in words */ - __iowrite32_copy(dst, src, count/4); + __ioread32_copy(dst, src, count / 4); } /** -- cgit v1.2.3 From fe65324e3f5205072a2d55ac9c63ec77155fa528 Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja Date: Tue, 24 Jul 2018 19:50:48 -0500 Subject: ASoC: Intel: Skylake: fix widget handling include DAPM Mux and output widgets into the list. Signed-off-by: Rakesh Ughreja Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 647e52aecdc3..9845d853644c 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -108,6 +108,9 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w, case snd_soc_dapm_aif_out: case snd_soc_dapm_dai_out: case snd_soc_dapm_switch: + case snd_soc_dapm_output: + case snd_soc_dapm_mux: + return false; default: return true; -- cgit v1.2.3 From c183fec10ae6b3d1dfb77471d4158e576f6b43ae Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Wed, 25 Jul 2018 17:00:59 +0800 Subject: ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002 DA7219 for our platform need to be configured for 1.8V. Hence, we add a volatge regulator with supplies of 1.8V in the machine driver. Signed-off-by: Adam Thomson Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp-da7219-max98357a.c | 43 ++++++++++++++++++++++++++++++++++++ 2 files changed, 44 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6cbf9cf4d1a4..58c1dcb4d255 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -8,6 +8,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_ADAU7002 + select REGULATOR depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f42606e5879e..cd3cf6e691a9 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -32,6 +32,8 @@ #include #include #include +#include +#include #include #include #include @@ -320,11 +322,52 @@ static struct snd_soc_card cz_card = { .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct regulator_consumer_supply acp_da7219_supplies[] = { + REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("IOVDD", "ADAU7002:00"), +}; + +static struct regulator_init_data acp_da7219_data = { + .constraints = { + .always_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(acp_da7219_supplies), + .consumer_supplies = acp_da7219_supplies, +}; + +static struct regulator_config acp_da7219_cfg = { + .init_data = &acp_da7219_data, +}; + +static struct regulator_ops acp_da7219_ops = { +}; + +static struct regulator_desc acp_da7219_desc = { + .name = "reg-fixed-1.8V", + .type = REGULATOR_VOLTAGE, + .owner = THIS_MODULE, + .ops = &acp_da7219_ops, + .fixed_uV = 1800000, /* 1.8V */ + .n_voltages = 1, +}; + static int cz_probe(struct platform_device *pdev) { int ret; struct snd_soc_card *card; struct acp_platform_info *machine; + struct regulator_dev *rdev; + + acp_da7219_cfg.dev = &pdev->dev; + rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, + &acp_da7219_cfg); + if (IS_ERR(rdev)) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + ret); + return -EINVAL; + } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); -- cgit v1.2.3 From 7699676081ded2e149e798f9205f1f089aa3a6bc Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Thu, 26 Jul 2018 14:04:08 +0800 Subject: ASoC: AMD: Fix build warning Fixes sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_probe': sound/soc/amd/acp-da7219-max98357a.c:367:3: warning: 'ret' may be used uninitialized in this function [-Wmaybe-uninitialized] dev_err(&pdev->dev, "Failed to register regulator: %d\n", ret); Reported-by: Stephen Rothwell Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index cd3cf6e691a9..8e3275a96a82 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -365,7 +365,7 @@ static int cz_probe(struct platform_device *pdev) &acp_da7219_cfg); if (IS_ERR(rdev)) { dev_err(&pdev->dev, "Failed to register regulator: %d\n", - ret); + (int)PTR_ERR(rdev)); return -EINVAL; } -- cgit v1.2.3 From 036e4963bfb2d4513c30efd80e4bd50ff6c79e3e Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 26 Jul 2018 14:45:42 +0200 Subject: ASoC: meson: use IRQ_RETVAL in the fifo irq handler Use IRQ_RETVAL instead of the open coded ternary operation. Suggested-by: Takashi Iwai Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/axg-fifo.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index db367d85290f..30262550e37b 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -174,7 +174,7 @@ static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) /* Ack irqs */ axg_fifo_ack_irq(fifo, status); - return !status ? IRQ_NONE : IRQ_HANDLED; + return IRQ_RETVAL(status); } static int axg_fifo_pcm_open(struct snd_pcm_substream *ss) -- cgit v1.2.3 From 435857e015dc7b337c5f21d195ac2e4ffd694283 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 26 Jul 2018 14:45:44 +0200 Subject: ASoC: meson: align axg card driver with DT bindings documentation Drop amlogic prefix in front of the generic DT properties and change property "name" to "model". Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/axg-card.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index d6d1081d94ad..2914ba0d965b 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -123,7 +123,7 @@ static int axg_card_add_aux_devices(struct snd_soc_card *card) struct snd_soc_aux_dev *aux; int num, i; - num = of_count_phandle_with_args(node, PREFIX "aux-devs", NULL); + num = of_count_phandle_with_args(node, "audio-aux-devs", NULL); if (num == -ENOENT) { /* * It is ok to have no auxiliary devices but for this card it @@ -144,8 +144,8 @@ static int axg_card_add_aux_devices(struct snd_soc_card *card) card->num_aux_devs = num; for (i = 0; i < card->num_aux_devs; i++, aux++) { - aux->codec_of_node = of_parse_phandle(node, - PREFIX "aux-devs", i); + aux->codec_of_node = + of_parse_phandle(node, "audio-aux-devs", i); if (!aux->codec_of_node) return -EINVAL; } @@ -610,18 +610,18 @@ static int axg_card_probe(struct platform_device *pdev) priv->card.owner = THIS_MODULE; priv->card.dev = dev; - ret = snd_soc_of_parse_card_name(&priv->card, PREFIX "name"); + ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret < 0) return ret; - ret = axg_card_parse_of_optional(&priv->card, PREFIX "routing", + ret = axg_card_parse_of_optional(&priv->card, "audio-routing", snd_soc_of_parse_audio_routing); if (ret) { dev_err(dev, "error while parsing routing\n"); return ret; } - ret = axg_card_parse_of_optional(&priv->card, PREFIX "widgets", + ret = axg_card_parse_of_optional(&priv->card, "audio-widgets", snd_soc_of_parse_audio_simple_widgets); if (ret) { dev_err(dev, "error while parsing widgets\n"); -- cgit v1.2.3 From c889a45d229938a94b50aadb819def8bb11a6a54 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 22:40:49 +0200 Subject: ASoC: zte: Fix incorrect PCM format bit usages zx-tdm driver sets the DAI driver definitions with the format bits wrongly set with SNDRV_PCM_FORMAT_*, instead of SNDRV_PCM_FMTBIT_*. This patch corrects the definitions. Spotted by a sparse warning: sound/soc/zte/zx-tdm.c:363:35: warning: restricted snd_pcm_format_t degrades to integer Fixes: 870e0ddc4345 ("ASoC: zx-tdm: add zte's tdm controller driver") Cc: Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/zte/zx-tdm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c index dc955272f58b..389272eeba9a 100644 --- a/sound/soc/zte/zx-tdm.c +++ b/sound/soc/zte/zx-tdm.c @@ -144,8 +144,8 @@ static void zx_tdm_rx_dma_en(struct zx_tdm_info *tdm, bool on) #define ZX_TDM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) #define ZX_TDM_FMTBIT \ - (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_MU_LAW | \ - SNDRV_PCM_FORMAT_A_LAW) + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_MU_LAW | \ + SNDRV_PCM_FMTBIT_A_LAW) static int zx_tdm_dai_probe(struct snd_soc_dai *dai) { -- cgit v1.2.3 From 40d1299f87bf915931970c8e6ea3852acacd1889 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 22:42:08 +0200 Subject: ASoC: dmaengine: Fix missing __user prefix in copy_user callback It seems that __user prefix was forgotten to be added to dmaengine_copy_user callback while we refactored the user-copy PCM core. This patch adds the missing prefix, remove the superfluous cast, and add the needed cast (__force is needed for downgrading from user pointer to kernel pointer), too. Spotted by a sparse warning like: sound/soc/soc-generic-dmaengine-pcm.c:397:27: warning: incorrect type in initializer (incompatible argument 4 (different address spaces)) Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 13bdca6e41c5..120f7b39e256 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -334,7 +334,7 @@ static snd_pcm_uframes_t dmaengine_pcm_pointer( static int dmaengine_copy_user(struct snd_pcm_substream *substream, int channel, unsigned long hwoff, - void *buf, unsigned long bytes) + void __user *buf, unsigned long bytes) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component = @@ -350,18 +350,17 @@ static int dmaengine_copy_user(struct snd_pcm_substream *substream, int ret; if (is_playback) - if (copy_from_user(dma_ptr, (void __user *)buf, bytes)) + if (copy_from_user(dma_ptr, buf, bytes)) return -EFAULT; if (process) { - ret = process(substream, channel, hwoff, - (void __user *)buf, bytes); + ret = process(substream, channel, hwoff, (__force void *)buf, bytes); if (ret < 0) return ret; } if (!is_playback) - if (copy_to_user((void __user *)buf, dma_ptr, bytes)) + if (copy_to_user(buf, dma_ptr, bytes)) return -EFAULT; return 0; -- cgit v1.2.3 From 3ba66feb59810e2ce616da0c4f1a5230c74768a8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 22:43:26 +0200 Subject: ASoC: dapm: Use int for format bit position fmt in snd_soc_dai_link_event() contains the format bit position, not the format bit itself. Hence it can be a simple integer instead of the explicit u64. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0602b2888d52..7e96793050c9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3656,7 +3656,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; - u64 fmt; + unsigned int fmt; int ret; if (WARN_ON(!config) || -- cgit v1.2.3 From b5453e8ca311fdb6003c6583ad101d2b9131b994 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:17:19 +0200 Subject: ASoC: intel: Fix snd_pcm_format_t handling As sparse warns, the PCM format type can't be dealt as integer as found in Intel SST driver codes. Fix them in the following two ways: - The open code with snd_mask_set() and params->masks reference is replaced with params_set_format() - The rest codes with snd_mask_set(fmt, SNDRV_PCM_FORMAT_XXX) are replaced with the new helper, snd_mask_set_format(). Reported-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw-rt5677.c | 4 +--- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 ++-- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- 9 files changed, 11 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 6ea360f33575..efcfd906c856 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -154,9 +154,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 3aba5bcf806a..be6e4b40bf03 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -160,7 +160,7 @@ static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index b68c289558a8..27308337ab12 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -221,7 +221,7 @@ static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP5 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 7961f1fd18bd..38f6ab74709d 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -152,7 +152,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 3a61252fe450..21a6490746a6 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -434,14 +434,14 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, rate->min = rate->max = 48000; channels->min = channels->max = 2; snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); } /* * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 92f5fb2ae0a3..a892b37eab7c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -307,7 +307,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, rate->min = rate->max = 48000; channels->min = channels->max = 2; snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) @@ -320,7 +320,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * thus changing the mask here */ if (!strcmp(be_dai_link->name, "SSP0-Codec")) - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 3ff6646cfa21..d31482b8c9bb 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -157,7 +157,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index b0610bba3cfa..e877bb60beb1 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -346,7 +346,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 38a1495c29cf..0e1818dd4cc6 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -229,7 +229,7 @@ static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 24 bit */ snd_mask_none(fmt); - snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); return 0; } -- cgit v1.2.3 From ebc22af0c9268dc4032326c20b02f2f227203330 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:17:20 +0200 Subject: ASoC: fsl: Use snd_mask_set_format() Use the new helper function snd_mask_set_format() for avoiding the ugly cast with __force prefix. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 07808c6d5461..44433b20435c 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -195,7 +195,7 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); snd_mask_none(mask); - snd_mask_set(mask, (__force int)priv->asrc_format); + snd_mask_set_format(mask, priv->asrc_format); return 0; } -- cgit v1.2.3 From 79b8a50813a80ac15fdcdc96674098dee15eaaf5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:17:21 +0200 Subject: ASoC: pcm186x: Declare PCM format with snd_pcm_format_t The PCM format type is with __bitwise, so we should use the dedicated snd_pcm_format_t instead of int. This fixes the sparse warning like: sound/soc/codecs/pcm186x.c:268:44: warning: incorrect type in initializer (different base types) Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/pcm186x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index 88fde70b1e9e..690c26e7389e 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -265,7 +265,7 @@ static int pcm186x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct pcm186x_priv *priv = snd_soc_component_get_drvdata(component); unsigned int rate = params_rate(params); - unsigned int format = params_format(params); + snd_pcm_format_t format = params_format(params); unsigned int width = params_width(params); unsigned int channels = params_channels(params); unsigned int div_lrck; -- cgit v1.2.3 From 8adf3df4156345f1edcdfa8c7f7beeb0de351ce2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jul 2018 23:17:22 +0200 Subject: ASoC: dmaengine: Use standard pcm_format_to_bits() macro The conversion from PCM format type to bits needs an explicit cast, and it'll be uglier. Since we have a standard macro for that, let's use it instead. This patch fixes the sparse warning: sound/soc/soc-generic-dmaengine-pcm.c:200:63: warning: restricted snd_pcm_format_t degrades to integer Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 120f7b39e256..52fd7af952a5 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -188,7 +188,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea case 32: case 64: if (addr_widths & (1 << (bits / 8))) - hw.formats |= (1LL << i); + hw.formats |= pcm_format_to_bits(i); break; default: /* Unsupported types */ -- cgit v1.2.3 From 279fef50b607f9cee94f10bae84f6730e97ccd7c Mon Sep 17 00:00:00 2001 From: Edward Cragg Date: Fri, 27 Jul 2018 13:59:28 +0100 Subject: ASoC: tegra: i2s: Fix typo/broken macro Fix typo in macro TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK. Signed-off-by: Edward Cragg Signed-off-by: Jorge Sanjuan Reviewed-by: Jon Hunter Signed-off-by: Mark Brown --- sound/soc/tegra/tegra30_i2s.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index 774fc6ad2026..2e561e946de2 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -173,7 +173,7 @@ /* Number of slots in frame, minus 1 */ #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT 16 #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US 7 -#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK (TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_SHIFT) +#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK (TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT) /* TDM mode slot enable bitmask */ #define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT 8 -- cgit v1.2.3 From c9c9780d8fa526a124933134a7e4041fa09662f6 Mon Sep 17 00:00:00 2001 From: Ladislav Michl Date: Sat, 28 Jul 2018 14:30:17 +0200 Subject: ASoC: wm8988: fix typo in rate constraints Remove duplicated entry and add missing zero in rate constraints. Signed-off-by: Ladislav Michl Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 62200117444b..6e52c6a8bab3 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -522,7 +522,7 @@ static inline int get_coeff(int mclk, int rate) /* The set of rates we can generate from the above for each SYSCLK */ static const unsigned int rates_12288[] = { - 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000, + 8000, 12000, 16000, 24000, 32000, 48000, 96000, }; static const struct snd_pcm_hw_constraint_list constraints_12288 = { @@ -540,7 +540,7 @@ static const struct snd_pcm_hw_constraint_list constraints_112896 = { }; static const unsigned int rates_12[] = { - 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000, + 8000, 11025, 12000, 16000, 22050, 24000, 32000, 41100, 48000, 48000, 88235, 96000, }; -- cgit v1.2.3 From fe209b97184f577b70ae0799c2b149be36092f84 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 30 Jul 2018 07:51:41 +0000 Subject: ASoC: ak4642: convert to SPDX identifiers As original license mentioned, it is GPL-2.0 in SPDX. Then, MODULE_LICENSE() should be "GPL v2" instead of "GPL". See ${LINUX}/include/linux/module.h "GPL" [GNU Public License v2 or later] "GPL v2" [GNU Public License v2] Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 26 +++++++++++--------------- 1 file changed, 11 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 605055964529..353237025514 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -1,17 +1,13 @@ -/* - * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto - * - * Based on wm8731.c by Richard Purdie - * Based on ak4535.c by Richard Purdie - * Based on wm8753.c by Liam Girdwood - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Kuninori Morimoto +// +// Based on wm8731.c by Richard Purdie +// Based on ak4535.c by Richard Purdie +// Based on wm8753.c by Liam Girdwood /* ** CAUTION ** * @@ -709,4 +705,4 @@ module_i2c_driver(ak4642_i2c_driver); MODULE_DESCRIPTION("Soc AK4642 driver"); MODULE_AUTHOR("Kuninori Morimoto "); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 7a968dc66aecaaea3424ca2525b758687b39bc0e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 30 Jul 2018 07:52:15 +0000 Subject: ASoC: ak4554: convert to SPDX identifiers As original license mentioned, it is GPL-2.0 in SPDX. Then, MODULE_LICENSE() should be "GPL v2" instead of "GPL". See ${LINUX}/include/linux/module.h "GPL" [GNU Public License v2 or later] "GPL v2" [GNU Public License v2] Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4554.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index b7ee13406d93..2fa83a1a84cf 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -1,13 +1,8 @@ -/* - * ak4554.c - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// ak4554.c +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto #include #include @@ -97,6 +92,6 @@ static struct platform_driver ak4554_driver = { }; module_platform_driver(ak4554_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SoC AK4554 driver"); MODULE_AUTHOR("Kuninori Morimoto "); -- cgit v1.2.3 From c0ca5604d43233b0f5cfb793be9638af21f5535d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 30 Jul 2018 07:52:46 +0000 Subject: ASoC: da7210: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 27 +++++++++++---------------- 1 file changed, 11 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a664111b7184..e172913d04a4 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1,19 +1,14 @@ -/* - * DA7210 ALSA Soc codec driver - * - * Copyright (c) 2009 Dialog Semiconductor - * Written by David Chen - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Cleanups by Kuninori Morimoto - * - * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// DA7210 ALSA Soc codec driver +// +// Copyright (c) 2009 Dialog Semiconductor +// Written by David Chen +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Cleanups by Kuninori Morimoto +// +// Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S #include #include -- cgit v1.2.3 From e028937c77f44bb38cdbc464afce2bf41d96006b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 30 Jul 2018 07:53:08 +0000 Subject: ASoC: ak4613: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4613.c | 26 +++++++++++--------------- 1 file changed, 11 insertions(+), 15 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 8523ff9351cf..c1181a20714d 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -1,18 +1,14 @@ -/* - * ak4613.c -- Asahi Kasei ALSA Soc Audio driver - * - * Copyright (C) 2015 Renesas Electronics Corporation - * Kuninori Morimoto - * - * Based on ak4642.c by Kuninori Morimoto - * Based on wm8731.c by Richard Purdie - * Based on ak4535.c by Richard Purdie - * Based on wm8753.c by Liam Girdwood - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ak4613.c -- Asahi Kasei ALSA Soc Audio driver +// +// Copyright (C) 2015 Renesas Electronics Corporation +// Kuninori Morimoto +// +// Based on ak4642.c by Kuninori Morimoto +// Based on wm8731.c by Richard Purdie +// Based on ak4535.c by Richard Purdie +// Based on wm8753.c by Liam Girdwood #include #include -- cgit v1.2.3 From 7464d3faf62a5adcb99f4f8ea9baabda78aed4aa Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 30 Jul 2018 08:00:22 +0000 Subject: ASoC: sh: Kconfig: convert to SPDX identifiers By default all files without license information are under the default license of the kernel, which is GPL version 2. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 0ae0800bf3a8..dc20f0f7080a 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -1,3 +1,4 @@ +# SPDX-License-Identifier: GPL-2.0 menu "SoC Audio support for Renesas SoCs" depends on SUPERH || ARCH_RENESAS || COMPILE_TEST -- cgit v1.2.3 From 4321723648b0abb456f7a9af51bb09a4ec60799d Mon Sep 17 00:00:00 2001 From: Alexey Khoroshilov Date: Sat, 28 Jul 2018 00:06:59 +0300 Subject: ASoC: tegra_alc5632: fix device_node refcounting tegra_alc5632_probe() increments reference count of device nodes with of_parse_phandle(), but there is no code decrementing them in the driver. The patch adds of_node_put() to tegra_alc5632_remove() and to error handling paths in the probe. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov Acked-by: Jon Hunter Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 5197d6b18cb6..98d87801d57a 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -190,14 +190,14 @@ static int tegra_alc5632_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node; ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -210,6 +210,13 @@ static int tegra_alc5632_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&alc5632->util_data); +err_put_cpu_of_node: + of_node_put(tegra_alc5632_dai.cpu_of_node); + tegra_alc5632_dai.cpu_of_node = NULL; + tegra_alc5632_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_alc5632_dai.codec_of_node); + tegra_alc5632_dai.codec_of_node = NULL; err: return ret; } @@ -223,6 +230,12 @@ static int tegra_alc5632_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_alc5632_dai.cpu_of_node); + tegra_alc5632_dai.cpu_of_node = NULL; + tegra_alc5632_dai.platform_of_node = NULL; + of_node_put(tegra_alc5632_dai.codec_of_node); + tegra_alc5632_dai.codec_of_node = NULL; + return 0; } -- cgit v1.2.3 From ae1c696a480c67c45fb23b35162183f72c6be0e1 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Thu, 26 Jul 2018 15:49:10 -0500 Subject: ASoC: sirf: Fix potential NULL pointer dereference There is a potential execution path in which function platform_get_resource() returns NULL. If this happens, we will end up having a NULL pointer dereference. Fix this by replacing devm_ioremap with devm_ioremap_resource, which has the NULL check and the memory region request. This code was detected with the help of Coccinelle. Cc: stable@vger.kernel.org Fixes: 2bd8d1d5cf89 ("ASoC: sirf: Add audio usp interface driver") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/sirf/sirf-usp.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c index 77e7dcf969d0..d70fcd4a1adf 100644 --- a/sound/soc/sirf/sirf-usp.c +++ b/sound/soc/sirf/sirf-usp.c @@ -370,10 +370,9 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev) platform_set_drvdata(pdev, usp); mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap(&pdev->dev, mem_res->start, - resource_size(mem_res)); - if (base == NULL) - return -ENOMEM; + base = devm_ioremap_resource(&pdev->dev, mem_res); + if (IS_ERR(base)) + return PTR_ERR(base); usp->regmap = devm_regmap_init_mmio(&pdev->dev, base, &sirf_usp_regmap_config); if (IS_ERR(usp->regmap)) -- cgit v1.2.3 From 8fc9983db199bb397d48e32a6400765b70f1995a Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Fri, 27 Jul 2018 11:37:28 +0900 Subject: ASoC: uniphier: add support for multichannel output This patch adds multichannel PCM output support for LD11/LD20. Currently driver tested and supported only 2ch, 6ch, and 8ch. Signed-off-by: Katsuhiro Suzuki Signed-off-by: Mark Brown --- sound/soc/uniphier/aio-core.c | 78 ++++++++++++++++++++++++++++++++++++++++--- sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-reg.h | 1 + sound/soc/uniphier/aio.h | 3 ++ 4 files changed, 78 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 638cb3fc5f7b..8b09bbb0f8d0 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -264,6 +264,57 @@ void aio_port_reset(struct uniphier_aio_sub *sub) } } +/** + * aio_port_set_ch - set channels of LPCM + * @sub: the AIO substream pointer, PCM substream only + * @ch : count of channels + * + * Set suitable slot selecting to input/output port block of AIO. + * + * This function may return error if non-PCM substream. + * + * Return: Zero if successful, otherwise a negative value on error. + */ +static int aio_port_set_ch(struct uniphier_aio_sub *sub) +{ + struct regmap *r = sub->aio->chip->regmap; + u32 slotsel_2ch[] = { + 0, 0, 0, 0, 0, + }; + u32 slotsel_multi[] = { + OPORTMXTYSLOTCTR_SLOTSEL_SLOT0, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT1, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT2, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT3, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT4, + }; + u32 mode, *slotsel; + int i; + + switch (params_channels(&sub->params)) { + case 8: + case 6: + mode = OPORTMXTYSLOTCTR_MODE; + slotsel = slotsel_multi; + break; + case 2: + mode = 0; + slotsel = slotsel_2ch; + break; + default: + return -EINVAL; + } + + for (i = 0; i < AUD_MAX_SLOTSEL; i++) { + regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i), + OPORTMXTYSLOTCTR_MODE, mode); + regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i), + OPORTMXTYSLOTCTR_SLOTSEL_MASK, slotsel[i]); + } + + return 0; +} + /** * aio_port_set_rate - set sampling rate of LPCM * @sub: the AIO substream pointer, PCM substream only @@ -575,6 +626,10 @@ int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, rate = params_rate(params); } + ret = aio_port_set_ch(sub); + if (ret) + return ret; + ret = aio_port_set_rate(sub, rate); if (ret) return ret; @@ -731,15 +786,28 @@ void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol) int aio_if_set_param(struct uniphier_aio_sub *sub, int pass_through) { struct regmap *r = sub->aio->chip->regmap; - u32 v; + u32 memfmt, v; if (sub->swm->dir == PORT_DIR_OUTPUT) { - if (pass_through) + if (pass_through) { v = PBOUTMXCTR0_ENDIAN_0123 | PBOUTMXCTR0_MEMFMT_STREAM; - else - v = PBOUTMXCTR0_ENDIAN_3210 | - PBOUTMXCTR0_MEMFMT_2CH; + } else { + switch (params_channels(&sub->params)) { + case 2: + memfmt = PBOUTMXCTR0_MEMFMT_2CH; + break; + case 6: + memfmt = PBOUTMXCTR0_MEMFMT_6CH; + break; + case 8: + memfmt = PBOUTMXCTR0_MEMFMT_8CH; + break; + default: + return -EINVAL; + } + v = PBOUTMXCTR0_ENDIAN_3210 | memfmt; + } regmap_write(r, PBOUTMXCTR0(sub->swm->oif.map), v); regmap_write(r, PBOUTMXCTR1(sub->swm->oif.map), 0); diff --git a/sound/soc/uniphier/aio-ld11.c b/sound/soc/uniphier/aio-ld11.c index ab04d3331be9..de962df245ba 100644 --- a/sound/soc/uniphier/aio-ld11.c +++ b/sound/soc/uniphier/aio-ld11.c @@ -286,7 +286,7 @@ static struct snd_soc_dai_driver uniphier_aio_dai_ld11[] = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_48000, .channels_min = 2, - .channels_max = 2, + .channels_max = 8, }, .ops = &uniphier_aio_i2s_ops, }, diff --git a/sound/soc/uniphier/aio-reg.h b/sound/soc/uniphier/aio-reg.h index 45fdc6ae358a..734395dbcffb 100644 --- a/sound/soc/uniphier/aio-reg.h +++ b/sound/soc/uniphier/aio-reg.h @@ -374,6 +374,7 @@ #define OPORTMXTYVOLGAINSTATUS(n, m) (0x42108 + 0x400 * (n) + 0x20 * (m)) #define OPORTMXTYVOLGAINSTATUS_CUR_MASK GENMASK(15, 0) #define OPORTMXTYSLOTCTR(n, m) (0x42114 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYSLOTCTR_MODE BIT(15) #define OPORTMXTYSLOTCTR_SLOTSEL_MASK GENMASK(11, 8) #define OPORTMXTYSLOTCTR_SLOTSEL_SLOT0 (0x8 << 8) #define OPORTMXTYSLOTCTR_SLOTSEL_SLOT1 (0x9 << 8) diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index aa89c2f6fa24..23a5c3c68658 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -141,6 +141,9 @@ enum IEC61937_PC { #define AUD_MIN_FRAGMENT_SIZE (4 * 1024) #define AUD_MAX_FRAGMENT_SIZE (16 * 1024) +/* max 5 slots, 10 channels, 2 channel in 1 slot */ +#define AUD_MAX_SLOTSEL 5 + /* * This is a selector for virtual register map of AIO. * -- cgit v1.2.3 From d8504acca759ae672c6d34d49a33d54ace094cbb Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki Date: Fri, 27 Jul 2018 11:37:44 +0900 Subject: ASoC: uniphier: change functions to static This patch changes some functions that are not used by other objects to static. Signed-off-by: Katsuhiro Suzuki Signed-off-by: Mark Brown --- sound/soc/uniphier/aio-core.c | 6 +++--- sound/soc/uniphier/aio.h | 3 --- 2 files changed, 3 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 8b09bbb0f8d0..9bcba06ba52e 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -327,7 +327,7 @@ static int aio_port_set_ch(struct uniphier_aio_sub *sub) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) +static int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) { struct regmap *r = sub->aio->chip->regmap; struct device *dev = &sub->aio->chip->pdev->dev; @@ -446,7 +446,7 @@ int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_fmt(struct uniphier_aio_sub *sub) +static int aio_port_set_fmt(struct uniphier_aio_sub *sub) { struct regmap *r = sub->aio->chip->regmap; struct device *dev = &sub->aio->chip->pdev->dev; @@ -511,7 +511,7 @@ int aio_port_set_fmt(struct uniphier_aio_sub *sub) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_clk(struct uniphier_aio_sub *sub) +static int aio_port_set_clk(struct uniphier_aio_sub *sub) { struct uniphier_aio_chip *chip = sub->aio->chip; struct device *dev = &sub->aio->chip->pdev->dev; diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index 23a5c3c68658..ca6ccbae0ee8 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -325,9 +325,6 @@ int aio_chip_set_pll(struct uniphier_aio_chip *chip, int pll_id, void aio_chip_init(struct uniphier_aio_chip *chip); int aio_init(struct uniphier_aio_sub *sub); void aio_port_reset(struct uniphier_aio_sub *sub); -int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate); -int aio_port_set_fmt(struct uniphier_aio_sub *sub); -int aio_port_set_clk(struct uniphier_aio_sub *sub); int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, const struct snd_pcm_hw_params *params); void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable); -- cgit v1.2.3 From e57d4ca882e289a2ddc844e82fa33ad1453e9871 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 27 Jul 2018 13:18:00 +0100 Subject: ASoC: wcd9335: add support to wcd9335 codec Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC, It supports both I2S/I2C and SLIMbus audio interfaces. On slimbus interface it supports two data lanes; 16 Tx ports and 8 Rx ports. It has Seven DACs and nine dedicated interpolators, Seven (six audio ADCs, and one VBAT ADC), Multibutton headset control (MBHC), Active noise cancellation and Sidetone paths and processing. This patchset adds very basic support for playback and capture via the 9 interpolators and ADC respectively. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd9335.c | 1154 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1161 insertions(+) create mode 100644 sound/soc/codecs/wcd9335.c (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index efb095dbcd71..cb09abf18dde 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1066,6 +1066,11 @@ config SND_SOC_UDA1380 tristate depends on I2C +config SND_SOC_WCD9335 + tristate "WCD9335 Codec" + depends on MFD_WCD9335 + tristate + config SND_SOC_WL1273 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7ae7c85e8219..01410b63daac 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -192,6 +192,7 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o +snd-soc-wcd9335-objs := wcd9335.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm-adsp-objs := wm_adsp.o snd-soc-wm0010-objs := wm0010.o @@ -451,6 +452,7 @@ obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o +obj-$(CONFIG_SND_SOC_WCD9335) += snd-soc-wcd9335.o obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c new file mode 100644 index 000000000000..bd9de5d45fa9 --- /dev/null +++ b/sound/soc/codecs/wcd9335.c @@ -0,0 +1,1154 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. +// Copyright (c) 2017-2018, Linaro Limited + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#define WCD9335_RATES_MASK (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) +/* Fractional Rates */ +#define WCD9335_FRAC_RATES_MASK (SNDRV_PCM_RATE_44100) +#define WCD9335_FORMATS_S16_S24_LE (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/* slave port water mark level + * (0: 6bytes, 1: 9bytes, 2: 12 bytes, 3: 15 bytes) + */ +#define SLAVE_PORT_WATER_MARK_6BYTES 0 +#define SLAVE_PORT_WATER_MARK_9BYTES 1 +#define SLAVE_PORT_WATER_MARK_12BYTES 2 +#define SLAVE_PORT_WATER_MARK_15BYTES 3 +#define SLAVE_PORT_WATER_MARK_SHIFT 1 +#define SLAVE_PORT_ENABLE 1 +#define SLAVE_PORT_DISABLE 0 +#define WCD9335_SLIM_WATER_MARK_VAL \ + ((SLAVE_PORT_WATER_MARK_12BYTES << SLAVE_PORT_WATER_MARK_SHIFT) | \ + (SLAVE_PORT_ENABLE)) + +#define WCD9335_SLIM_NUM_PORT_REG 3 +#define WCD9335_SLIM_PGD_PORT_INT_TX_EN0 (WCD9335_SLIM_PGD_PORT_INT_EN0 + 2) + +#define WCD9335_MCLK_CLK_12P288MHZ 12288000 +#define WCD9335_MCLK_CLK_9P6MHZ 9600000 + +#define WCD9335_SLIM_CLOSE_TIMEOUT 1000 +#define WCD9335_SLIM_IRQ_OVERFLOW (1 << 0) +#define WCD9335_SLIM_IRQ_UNDERFLOW (1 << 1) +#define WCD9335_SLIM_IRQ_PORT_CLOSED (1 << 2) + +#define WCD9335_NUM_INTERPOLATORS 9 +#define WCD9335_RX_START 16 +#define WCD9335_SLIM_CH_START 128 + +#define WCD9335_SLIM_RX_CH(p) \ + {.port = p + WCD9335_RX_START, .shift = p,} + +/* vout step value */ +#define WCD9335_CALCULATE_VOUT_D(req_mv) (((req_mv - 650) * 10) / 25) + +enum { + WCD9335_RX0 = 0, + WCD9335_RX1, + WCD9335_RX2, + WCD9335_RX3, + WCD9335_RX4, + WCD9335_RX5, + WCD9335_RX6, + WCD9335_RX7, + WCD9335_RX8, + WCD9335_RX9, + WCD9335_RX10, + WCD9335_RX11, + WCD9335_RX12, + WCD9335_RX_MAX, +}; + +enum { + SIDO_SOURCE_INTERNAL = 0, + SIDO_SOURCE_RCO_BG, +}; + +enum wcd9335_sido_voltage { + SIDO_VOLTAGE_SVS_MV = 950, + SIDO_VOLTAGE_NOMINAL_MV = 1100, +}; + +enum { + AIF1_PB = 0, + AIF1_CAP, + AIF2_PB, + AIF2_CAP, + AIF3_PB, + AIF3_CAP, + AIF4_PB, + NUM_CODEC_DAIS, +}; + +enum { + INTn_2_INP_SEL_ZERO = 0, + INTn_2_INP_SEL_RX0, + INTn_2_INP_SEL_RX1, + INTn_2_INP_SEL_RX2, + INTn_2_INP_SEL_RX3, + INTn_2_INP_SEL_RX4, + INTn_2_INP_SEL_RX5, + INTn_2_INP_SEL_RX6, + INTn_2_INP_SEL_RX7, + INTn_2_INP_SEL_PROXIMITY, +}; + +enum { + INTn_1_MIX_INP_SEL_ZERO = 0, + INTn_1_MIX_INP_SEL_DEC0, + INTn_1_MIX_INP_SEL_DEC1, + INTn_1_MIX_INP_SEL_IIR0, + INTn_1_MIX_INP_SEL_IIR1, + INTn_1_MIX_INP_SEL_RX0, + INTn_1_MIX_INP_SEL_RX1, + INTn_1_MIX_INP_SEL_RX2, + INTn_1_MIX_INP_SEL_RX3, + INTn_1_MIX_INP_SEL_RX4, + INTn_1_MIX_INP_SEL_RX5, + INTn_1_MIX_INP_SEL_RX6, + INTn_1_MIX_INP_SEL_RX7, + +}; + +enum wcd_clock_type { + WCD_CLK_OFF, + WCD_CLK_RCO, + WCD_CLK_MCLK, +}; + +struct wcd9335_slim_ch { + u32 ch_num; + u16 port; + u16 shift; + struct list_head list; +}; + +struct wcd_slim_codec_dai_data { + struct list_head slim_ch_list; + struct slim_stream_config sconfig; + struct slim_stream_runtime *sruntime; +}; + +struct wcd9335_codec { + struct device *dev; + struct clk *mclk; + struct clk *native_clk; + u32 mclk_rate; + u8 intf_type; + u8 version; + + struct slim_device *slim; + struct slim_device *slim_ifd; + struct regmap *regmap; + struct regmap *if_regmap; + struct regmap_irq_chip_data *irq_data; + + struct wcd9335_slim_ch rx_chs[WCD9335_RX_MAX]; + u32 num_rx_port; + + int sido_input_src; + enum wcd9335_sido_voltage sido_voltage; + + struct wcd_slim_codec_dai_data dai[NUM_CODEC_DAIS]; + struct snd_soc_component *component; + + int master_bias_users; + int clk_mclk_users; + int clk_rco_users; + int sido_ccl_cnt; + enum wcd_clock_type clk_type; + + u32 hph_mode; +}; + +static const struct wcd9335_slim_ch wcd9335_rx_chs[WCD9335_RX_MAX] = { + WCD9335_SLIM_RX_CH(0), /* 16 */ + WCD9335_SLIM_RX_CH(1), /* 17 */ + WCD9335_SLIM_RX_CH(2), + WCD9335_SLIM_RX_CH(3), + WCD9335_SLIM_RX_CH(4), + WCD9335_SLIM_RX_CH(5), + WCD9335_SLIM_RX_CH(6), + WCD9335_SLIM_RX_CH(7), + WCD9335_SLIM_RX_CH(8), + WCD9335_SLIM_RX_CH(9), + WCD9335_SLIM_RX_CH(10), + WCD9335_SLIM_RX_CH(11), + WCD9335_SLIM_RX_CH(12), +}; + +struct interp_sample_rate { + int rate; + int rate_val; +}; + +static struct interp_sample_rate int_mix_rate_val[] = { + {48000, 0x4}, /* 48K */ + {96000, 0x5}, /* 96K */ + {192000, 0x6}, /* 192K */ +}; + +static struct interp_sample_rate int_prim_rate_val[] = { + {8000, 0x0}, /* 8K */ + {16000, 0x1}, /* 16K */ + {24000, -EINVAL},/* 24K */ + {32000, 0x3}, /* 32K */ + {48000, 0x4}, /* 48K */ + {96000, 0x5}, /* 96K */ + {192000, 0x6}, /* 192K */ + {384000, 0x7}, /* 384K */ + {44100, 0x8}, /* 44.1K */ +}; + +struct wcd9335_reg_mask_val { + u16 reg; + u8 mask; + u8 val; +}; + +static const struct wcd9335_reg_mask_val wcd9335_codec_reg_init_val_2_0[] = { + {WCD9335_RCO_CTRL_2, 0x0F, 0x08}, + {WCD9335_RX_BIAS_FLYB_MID_RST, 0xF0, 0x10}, + {WCD9335_FLYBACK_CTRL_1, 0x20, 0x20}, + {WCD9335_HPH_OCP_CTL, 0xFF, 0x5A}, + {WCD9335_HPH_L_TEST, 0x01, 0x01}, + {WCD9335_HPH_R_TEST, 0x01, 0x01}, + {WCD9335_CDC_BOOST0_BOOST_CFG1, 0x3F, 0x12}, + {WCD9335_CDC_BOOST0_BOOST_CFG2, 0x1C, 0x08}, + {WCD9335_CDC_COMPANDER7_CTL7, 0x1E, 0x18}, + {WCD9335_CDC_BOOST1_BOOST_CFG1, 0x3F, 0x12}, + {WCD9335_CDC_BOOST1_BOOST_CFG2, 0x1C, 0x08}, + {WCD9335_CDC_COMPANDER8_CTL7, 0x1E, 0x18}, + {WCD9335_CDC_TX0_TX_PATH_SEC7, 0xFF, 0x45}, + {WCD9335_CDC_RX0_RX_PATH_SEC0, 0xFC, 0xF4}, + {WCD9335_HPH_REFBUFF_LP_CTL, 0x08, 0x08}, + {WCD9335_HPH_REFBUFF_LP_CTL, 0x06, 0x02}, +}; + +static const struct wcd9335_reg_mask_val wcd9335_codec_reg_init_common_val[] = { + /* Rbuckfly/R_EAR(32) */ + {WCD9335_CDC_CLSH_K2_MSB, 0x0F, 0x00}, + {WCD9335_CDC_CLSH_K2_LSB, 0xFF, 0x60}, + {WCD9335_CPE_SS_DMIC_CFG, 0x80, 0x00}, + {WCD9335_CDC_BOOST0_BOOST_CTL, 0x70, 0x50}, + {WCD9335_CDC_BOOST1_BOOST_CTL, 0x70, 0x50}, + {WCD9335_CDC_RX7_RX_PATH_CFG1, 0x08, 0x08}, + {WCD9335_CDC_RX8_RX_PATH_CFG1, 0x08, 0x08}, + {WCD9335_ANA_LO_1_2, 0x3C, 0X3C}, + {WCD9335_DIFF_LO_COM_SWCAP_REFBUF_FREQ, 0x70, 0x00}, + {WCD9335_DIFF_LO_COM_PA_FREQ, 0x70, 0x40}, + {WCD9335_SOC_MAD_AUDIO_CTL_2, 0x03, 0x03}, + {WCD9335_CDC_TOP_TOP_CFG1, 0x02, 0x02}, + {WCD9335_CDC_TOP_TOP_CFG1, 0x01, 0x01}, + {WCD9335_EAR_CMBUFF, 0x08, 0x00}, + {WCD9335_CDC_TX9_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_TX10_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_TX11_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_TX12_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_COMPANDER7_CTL3, 0x80, 0x80}, + {WCD9335_CDC_COMPANDER8_CTL3, 0x80, 0x80}, + {WCD9335_CDC_COMPANDER7_CTL7, 0x01, 0x01}, + {WCD9335_CDC_COMPANDER8_CTL7, 0x01, 0x01}, + {WCD9335_CDC_RX0_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX1_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX2_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX3_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX4_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX5_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX6_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX7_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX8_RX_PATH_CFG0, 0x01, 0x01}, + {WCD9335_CDC_RX0_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX1_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX2_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX3_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX4_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX5_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX6_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX7_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_CDC_RX8_RX_PATH_MIX_CFG, 0x01, 0x01}, + {WCD9335_VBADC_IBIAS_FE, 0x0C, 0x08}, +}; + +static int wcd9335_set_mix_interpolator_rate(struct snd_soc_dai *dai, + int rate_val, + u32 rate) +{ + struct snd_soc_component *component = dai->component; + struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); + struct wcd9335_slim_ch *ch; + int val, j; + + list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) { + for (j = 0; j < WCD9335_NUM_INTERPOLATORS; j++) { + val = snd_soc_component_read32(component, + WCD9335_CDC_RX_INP_MUX_RX_INT_CFG1(j)) & + WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; + + if (val == (ch->shift + INTn_2_INP_SEL_RX0)) + snd_soc_component_update_bits(component, + WCD9335_CDC_RX_PATH_MIX_CTL(j), + WCD9335_CDC_MIX_PCM_RATE_MASK, + rate_val); + } + } + + return 0; +} + +static int wcd9335_set_prim_interpolator_rate(struct snd_soc_dai *dai, + u8 rate_val, + u32 rate) +{ + struct snd_soc_component *comp = dai->component; + struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); + struct wcd9335_slim_ch *ch; + u8 cfg0, cfg1, inp0_sel, inp1_sel, inp2_sel; + int inp, j; + + list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) { + inp = ch->shift + INTn_1_MIX_INP_SEL_RX0; + /* + * Loop through all interpolator MUX inputs and find out + * to which interpolator input, the slim rx port + * is connected + */ + for (j = 0; j < WCD9335_NUM_INTERPOLATORS; j++) { + cfg0 = snd_soc_component_read32(comp, + WCD9335_CDC_RX_INP_MUX_RX_INT_CFG0(j)); + cfg1 = snd_soc_component_read32(comp, + WCD9335_CDC_RX_INP_MUX_RX_INT_CFG1(j)); + + inp0_sel = cfg0 & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; + inp1_sel = (cfg0 >> 4) & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; + inp2_sel = (cfg1 >> 4) & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; + + if ((inp0_sel == inp) || (inp1_sel == inp) || + (inp2_sel == inp)) { + /* rate is in Hz */ + if ((j == 0) && (rate == 44100)) + dev_info(wcd->dev, + "Cannot set 44.1KHz on INT0\n"); + else + snd_soc_component_update_bits(comp, + WCD9335_CDC_RX_PATH_CTL(j), + WCD9335_CDC_MIX_PCM_RATE_MASK, + rate_val); + } + } + } + + return 0; +} + +static int wcd9335_set_interpolator_rate(struct snd_soc_dai *dai, u32 rate) +{ + int i; + + /* set mixing path rate */ + for (i = 0; i < ARRAY_SIZE(int_mix_rate_val); i++) { + if (rate == int_mix_rate_val[i].rate) { + wcd9335_set_mix_interpolator_rate(dai, + int_mix_rate_val[i].rate_val, rate); + break; + } + } + + /* set primary path sample rate */ + for (i = 0; i < ARRAY_SIZE(int_prim_rate_val); i++) { + if (rate == int_prim_rate_val[i].rate) { + wcd9335_set_prim_interpolator_rate(dai, + int_prim_rate_val[i].rate_val, rate); + break; + } + } + + return 0; +} + +static int wcd9335_slim_set_hw_params(struct wcd9335_codec *wcd, + struct wcd_slim_codec_dai_data *dai_data, + int direction) +{ + struct list_head *slim_ch_list = &dai_data->slim_ch_list; + struct slim_stream_config *cfg = &dai_data->sconfig; + struct wcd9335_slim_ch *ch; + u16 payload = 0; + int ret, i; + + cfg->ch_count = 0; + cfg->direction = direction; + cfg->port_mask = 0; + + /* Configure slave interface device */ + list_for_each_entry(ch, slim_ch_list, list) { + cfg->ch_count++; + payload |= 1 << ch->shift; + cfg->port_mask |= BIT(ch->port); + } + + cfg->chs = kcalloc(cfg->ch_count, sizeof(unsigned int), GFP_KERNEL); + if (!cfg->chs) + return -ENOMEM; + + i = 0; + list_for_each_entry(ch, slim_ch_list, list) { + cfg->chs[i++] = ch->ch_num; + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + /* write to interface device */ + ret = regmap_write(wcd->if_regmap, + WCD9335_SLIM_PGD_RX_PORT_MULTI_CHNL_0(ch->port), + payload); + + if (ret < 0) + goto err; + + /* configure the slave port for water mark and enable*/ + ret = regmap_write(wcd->if_regmap, + WCD9335_SLIM_PGD_RX_PORT_CFG(ch->port), + WCD9335_SLIM_WATER_MARK_VAL); + if (ret < 0) + goto err; + } + } + + dai_data->sruntime = slim_stream_allocate(wcd->slim, "WCD9335-SLIM"); + slim_stream_prepare(dai_data->sruntime, cfg); + + return 0; + +err: + dev_err(wcd->dev, "Error Setting slim hw params\n"); + kfree(cfg->chs); + cfg->chs = NULL; + + return ret; +} + +static int wcd9335_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct wcd9335_codec *wcd; + int ret; + + wcd = snd_soc_component_get_drvdata(dai->component); + + switch (substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + ret = wcd9335_set_interpolator_rate(dai, params_rate(params)); + if (ret) { + dev_err(wcd->dev, "cannot set sample rate: %u\n", + params_rate(params)); + return ret; + } + switch (params_width(params)) { + case 16 ... 24: + wcd->dai[dai->id].sconfig.bps = params_width(params); + break; + default: + dev_err(wcd->dev, "%s: Invalid format 0x%x\n", + __func__, params_width(params)); + return -EINVAL; + } + break; + default: + dev_err(wcd->dev, "Invalid stream type %d\n", + substream->stream); + return -EINVAL; + }; + + wcd->dai[dai->id].sconfig.rate = params_rate(params); + wcd9335_slim_set_hw_params(wcd, &wcd->dai[dai->id], substream->stream); + + return 0; +} + +static int wcd9335_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct wcd_slim_codec_dai_data *dai_data; + struct wcd9335_codec *wcd; + + wcd = snd_soc_component_get_drvdata(dai->component); + dai_data = &wcd->dai[dai->id]; + slim_stream_enable(dai_data->sruntime); + + return 0; +} + +static int wcd9335_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct wcd9335_codec *wcd; + int i; + + wcd = snd_soc_component_get_drvdata(dai->component); + + if (!tx_slot || !rx_slot) { + dev_err(wcd->dev, "Invalid tx_slot=%p, rx_slot=%p\n", + tx_slot, rx_slot); + return -EINVAL; + } + + if (wcd->rx_chs) { + wcd->num_rx_port = rx_num; + for (i = 0; i < rx_num; i++) { + wcd->rx_chs[i].ch_num = rx_slot[i]; + INIT_LIST_HEAD(&wcd->rx_chs[i].list); + } + } + + return 0; +} + +static int wcd9335_get_channel_map(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot) +{ + struct wcd9335_slim_ch *ch; + struct wcd9335_codec *wcd; + int i = 0; + + wcd = snd_soc_component_get_drvdata(dai->component); + + switch (dai->id) { + case AIF1_PB: + case AIF2_PB: + case AIF3_PB: + case AIF4_PB: + if (!rx_slot || !rx_num) { + dev_err(wcd->dev, "Invalid rx_slot %p or rx_num %p\n", + rx_slot, rx_num); + return -EINVAL; + } + + list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) + rx_slot[i++] = ch->ch_num; + + *rx_num = i; + break; + default: + dev_err(wcd->dev, "Invalid DAI ID %x\n", dai->id); + break; + } + + return 0; +} + +static struct snd_soc_dai_ops wcd9335_dai_ops = { + .hw_params = wcd9335_hw_params, + .prepare = wcd9335_prepare, + .set_channel_map = wcd9335_set_channel_map, + .get_channel_map = wcd9335_get_channel_map, +}; + +static struct snd_soc_dai_driver wcd9335_slim_dais[] = { + [0] = { + .name = "wcd9335_rx1", + .id = AIF1_PB, + .playback = { + .stream_name = "AIF1 Playback", + .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, + .formats = WCD9335_FORMATS_S16_S24_LE, + .rate_max = 192000, + .rate_min = 8000, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &wcd9335_dai_ops, + }, + [1] = { + .name = "wcd9335_tx1", + .id = AIF1_CAP, + .capture = { + .stream_name = "AIF1 Capture", + .rates = WCD9335_RATES_MASK, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 4, + }, + .ops = &wcd9335_dai_ops, + }, + [2] = { + .name = "wcd9335_rx2", + .id = AIF2_PB, + .playback = { + .stream_name = "AIF2 Playback", + .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, + .formats = WCD9335_FORMATS_S16_S24_LE, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &wcd9335_dai_ops, + }, + [3] = { + .name = "wcd9335_tx2", + .id = AIF2_CAP, + .capture = { + .stream_name = "AIF2 Capture", + .rates = WCD9335_RATES_MASK, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 4, + }, + .ops = &wcd9335_dai_ops, + }, + [4] = { + .name = "wcd9335_rx3", + .id = AIF3_PB, + .playback = { + .stream_name = "AIF3 Playback", + .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, + .formats = WCD9335_FORMATS_S16_S24_LE, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &wcd9335_dai_ops, + }, + [5] = { + .name = "wcd9335_tx3", + .id = AIF3_CAP, + .capture = { + .stream_name = "AIF3 Capture", + .rates = WCD9335_RATES_MASK, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 4, + }, + .ops = &wcd9335_dai_ops, + }, + [6] = { + .name = "wcd9335_rx4", + .id = AIF4_PB, + .playback = { + .stream_name = "AIF4 Playback", + .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, + .formats = WCD9335_FORMATS_S16_S24_LE, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &wcd9335_dai_ops, + }, +}; + +static irqreturn_t wcd9335_slimbus_irq(int irq, void *data) +{ + struct wcd9335_codec *wcd = data; + unsigned long status = 0; + int i, j, port_id; + unsigned int val, int_val = 0; + bool tx; + unsigned short reg = 0; + + for (i = WCD9335_SLIM_PGD_PORT_INT_STATUS_RX_0, j = 0; + i <= WCD9335_SLIM_PGD_PORT_INT_STATUS_TX_1; i++, j++) { + regmap_read(wcd->if_regmap, i, &val); + status |= ((u32)val << (8 * j)); + } + + for_each_set_bit(j, &status, 32) { + tx = (j >= 16 ? true : false); + port_id = (tx ? j - 16 : j); + regmap_read(wcd->if_regmap, + WCD9335_SLIM_PGD_PORT_INT_RX_SOURCE0 + j, &val); + if (val) { + if (!tx) + reg = WCD9335_SLIM_PGD_PORT_INT_EN0 + + (port_id / 8); + else + reg = WCD9335_SLIM_PGD_PORT_INT_TX_EN0 + + (port_id / 8); + regmap_read( + wcd->if_regmap, reg, &int_val); + /* + * Ignore interrupts for ports for which the + * interrupts are not specifically enabled. + */ + if (!(int_val & (1 << (port_id % 8)))) + continue; + } + if (val & WCD9335_SLIM_IRQ_OVERFLOW) + dev_err_ratelimited(wcd->dev, + "%s: overflow error on %s port %d, value %x\n", + __func__, (tx ? "TX" : "RX"), port_id, val); + if (val & WCD9335_SLIM_IRQ_UNDERFLOW) + dev_err_ratelimited(wcd->dev, + "%s: underflow error on %s port %d, value %x\n", + __func__, (tx ? "TX" : "RX"), port_id, val); + if ((val & WCD9335_SLIM_IRQ_OVERFLOW) || + (val & WCD9335_SLIM_IRQ_UNDERFLOW)) { + if (!tx) + reg = WCD9335_SLIM_PGD_PORT_INT_EN0 + + (port_id / 8); + else + reg = WCD9335_SLIM_PGD_PORT_INT_TX_EN0 + + (port_id / 8); + regmap_read( + wcd->if_regmap, reg, &int_val); + if (int_val & (1 << (port_id % 8))) { + int_val = int_val ^ (1 << (port_id % 8)); + regmap_write(wcd->if_regmap, + reg, int_val); + } + } + + regmap_write(wcd->if_regmap, + WCD9335_SLIM_PGD_PORT_INT_CLR_RX_0 + (j / 8), + BIT(j % 8)); + } + + return IRQ_HANDLED; +} + +static int wcd9335_setup_irqs(struct wcd9335_codec *wcd) +{ + int slim_irq = regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS); + int i, ret = 0; + + ret = request_threaded_irq(slim_irq, NULL, wcd9335_slimbus_irq, + IRQF_TRIGGER_RISING, "SLIMBus Slave", wcd); + if (ret) { + dev_err(wcd->dev, "Failed to request SLIMBUS irq\n"); + return ret; + } + + /* enable interrupts on all slave ports */ + for (i = 0; i < WCD9335_SLIM_NUM_PORT_REG; i++) + regmap_write(wcd->if_regmap, WCD9335_SLIM_PGD_PORT_INT_EN0 + i, + 0xFF); + + return ret; +} + +static void wcd9335_cdc_sido_ccl_enable(struct wcd9335_codec *wcd, + bool ccl_flag) +{ + struct snd_soc_component *comp = wcd->component; + + if (ccl_flag) { + if (++wcd->sido_ccl_cnt == 1) + snd_soc_component_write(comp, WCD9335_SIDO_SIDO_CCL_10, + WCD9335_SIDO_SIDO_CCL_DEF_VALUE); + } else { + if (wcd->sido_ccl_cnt == 0) { + dev_err(wcd->dev, "sido_ccl already disabled\n"); + return; + } + if (--wcd->sido_ccl_cnt == 0) + snd_soc_component_write(comp, WCD9335_SIDO_SIDO_CCL_10, + WCD9335_SIDO_SIDO_CCL_10_ICHARG_PWR_SEL_C320FF); + } +} + +static int wcd9335_enable_master_bias(struct wcd9335_codec *wcd) +{ + wcd->master_bias_users++; + if (wcd->master_bias_users == 1) { + regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, + WCD9335_ANA_BIAS_EN_MASK, + WCD9335_ANA_BIAS_ENABLE); + regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, + WCD9335_ANA_BIAS_PRECHRG_EN_MASK, + WCD9335_ANA_BIAS_PRECHRG_ENABLE); + /* + * 1ms delay is required after pre-charge is enabled + * as per HW requirement + */ + usleep_range(1000, 1100); + regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, + WCD9335_ANA_BIAS_PRECHRG_EN_MASK, + WCD9335_ANA_BIAS_PRECHRG_DISABLE); + regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, + WCD9335_ANA_BIAS_PRECHRG_CTL_MODE, + WCD9335_ANA_BIAS_PRECHRG_CTL_MODE_MANUAL); + } + + return 0; +} + +static int wcd9335_enable_mclk(struct wcd9335_codec *wcd) +{ + /* Enable mclk requires master bias to be enabled first */ + if (wcd->master_bias_users <= 0) + return -EINVAL; + + if (((wcd->clk_mclk_users == 0) && (wcd->clk_type == WCD_CLK_MCLK)) || + ((wcd->clk_mclk_users > 0) && (wcd->clk_type != WCD_CLK_MCLK))) { + dev_err(wcd->dev, "Error enabling MCLK, clk_type: %d\n", + wcd->clk_type); + return -EINVAL; + } + + if (++wcd->clk_mclk_users == 1) { + regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, + WCD9335_ANA_CLK_EXT_CLKBUF_EN_MASK, + WCD9335_ANA_CLK_EXT_CLKBUF_ENABLE); + regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, + WCD9335_ANA_CLK_MCLK_SRC_MASK, + WCD9335_ANA_CLK_MCLK_SRC_EXTERNAL); + regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, + WCD9335_ANA_CLK_MCLK_EN_MASK, + WCD9335_ANA_CLK_MCLK_ENABLE); + regmap_update_bits(wcd->regmap, + WCD9335_CDC_CLK_RST_CTRL_FS_CNT_CONTROL, + WCD9335_CDC_CLK_RST_CTRL_FS_CNT_EN_MASK, + WCD9335_CDC_CLK_RST_CTRL_FS_CNT_ENABLE); + regmap_update_bits(wcd->regmap, + WCD9335_CDC_CLK_RST_CTRL_MCLK_CONTROL, + WCD9335_CDC_CLK_RST_CTRL_MCLK_EN_MASK, + WCD9335_CDC_CLK_RST_CTRL_MCLK_ENABLE); + /* + * 10us sleep is required after clock is enabled + * as per HW requirement + */ + usleep_range(10, 15); + } + + wcd->clk_type = WCD_CLK_MCLK; + + return 0; +} + +static int wcd9335_disable_mclk(struct wcd9335_codec *wcd) +{ + if (wcd->clk_mclk_users <= 0) + return -EINVAL; + + if (--wcd->clk_mclk_users == 0) { + if (wcd->clk_rco_users > 0) { + /* MCLK to RCO switch */ + regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, + WCD9335_ANA_CLK_MCLK_SRC_MASK, + WCD9335_ANA_CLK_MCLK_SRC_RCO); + wcd->clk_type = WCD_CLK_RCO; + } else { + regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, + WCD9335_ANA_CLK_MCLK_EN_MASK, + WCD9335_ANA_CLK_MCLK_DISABLE); + wcd->clk_type = WCD_CLK_OFF; + } + + regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, + WCD9335_ANA_CLK_EXT_CLKBUF_EN_MASK, + WCD9335_ANA_CLK_EXT_CLKBUF_DISABLE); + } + + return 0; +} + +static int wcd9335_disable_master_bias(struct wcd9335_codec *wcd) +{ + if (wcd->master_bias_users <= 0) + return -EINVAL; + + wcd->master_bias_users--; + if (wcd->master_bias_users == 0) { + regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, + WCD9335_ANA_BIAS_EN_MASK, + WCD9335_ANA_BIAS_DISABLE); + regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, + WCD9335_ANA_BIAS_PRECHRG_CTL_MODE, + WCD9335_ANA_BIAS_PRECHRG_CTL_MODE_MANUAL); + } + return 0; +} + +static int wcd9335_cdc_req_mclk_enable(struct wcd9335_codec *wcd, + bool enable) +{ + int ret = 0; + + if (enable) { + wcd9335_cdc_sido_ccl_enable(wcd, true); + ret = clk_prepare_enable(wcd->mclk); + if (ret) { + dev_err(wcd->dev, "%s: ext clk enable failed\n", + __func__); + goto err; + } + /* get BG */ + wcd9335_enable_master_bias(wcd); + /* get MCLK */ + wcd9335_enable_mclk(wcd); + + } else { + /* put MCLK */ + wcd9335_disable_mclk(wcd); + /* put BG */ + wcd9335_disable_master_bias(wcd); + clk_disable_unprepare(wcd->mclk); + wcd9335_cdc_sido_ccl_enable(wcd, false); + } +err: + return ret; +} + +static void wcd9335_codec_apply_sido_voltage(struct wcd9335_codec *wcd, + enum wcd9335_sido_voltage req_mv) +{ + struct snd_soc_component *comp = wcd->component; + int vout_d_val; + + if (req_mv == wcd->sido_voltage) + return; + + /* compute the vout_d step value */ + vout_d_val = WCD9335_CALCULATE_VOUT_D(req_mv) & + WCD9335_ANA_BUCK_VOUT_MASK; + snd_soc_component_write(comp, WCD9335_ANA_BUCK_VOUT_D, vout_d_val); + snd_soc_component_update_bits(comp, WCD9335_ANA_BUCK_CTL, + WCD9335_ANA_BUCK_CTL_RAMP_START_MASK, + WCD9335_ANA_BUCK_CTL_RAMP_START_ENABLE); + + /* 1 msec sleep required after SIDO Vout_D voltage change */ + usleep_range(1000, 1100); + wcd->sido_voltage = req_mv; + snd_soc_component_update_bits(comp, WCD9335_ANA_BUCK_CTL, + WCD9335_ANA_BUCK_CTL_RAMP_START_MASK, + WCD9335_ANA_BUCK_CTL_RAMP_START_DISABLE); +} + +static int wcd9335_codec_update_sido_voltage(struct wcd9335_codec *wcd, + enum wcd9335_sido_voltage req_mv) +{ + int ret = 0; + + /* enable mclk before setting SIDO voltage */ + ret = wcd9335_cdc_req_mclk_enable(wcd, true); + if (ret) { + dev_err(wcd->dev, "Ext clk enable failed\n"); + goto err; + } + + wcd9335_codec_apply_sido_voltage(wcd, req_mv); + wcd9335_cdc_req_mclk_enable(wcd, false); + +err: + return ret; +} + +static int _wcd9335_codec_enable_mclk(struct snd_soc_component *component, + int enable) +{ + struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); + int ret; + + if (enable) { + ret = wcd9335_cdc_req_mclk_enable(wcd, true); + if (ret) + return ret; + + wcd9335_codec_apply_sido_voltage(wcd, + SIDO_VOLTAGE_NOMINAL_MV); + } else { + wcd9335_codec_update_sido_voltage(wcd, + wcd->sido_voltage); + wcd9335_cdc_req_mclk_enable(wcd, false); + } + + return 0; +} + +static void wcd9335_enable_sido_buck(struct snd_soc_component *component) +{ + struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); + + snd_soc_component_update_bits(component, WCD9335_ANA_RCO, + WCD9335_ANA_RCO_BG_EN_MASK, + WCD9335_ANA_RCO_BG_ENABLE); + snd_soc_component_update_bits(component, WCD9335_ANA_BUCK_CTL, + WCD9335_ANA_BUCK_CTL_VOUT_D_IREF_MASK, + WCD9335_ANA_BUCK_CTL_VOUT_D_IREF_EXT); + /* 100us sleep needed after IREF settings */ + usleep_range(100, 110); + snd_soc_component_update_bits(component, WCD9335_ANA_BUCK_CTL, + WCD9335_ANA_BUCK_CTL_VOUT_D_VREF_MASK, + WCD9335_ANA_BUCK_CTL_VOUT_D_VREF_EXT); + /* 100us sleep needed after VREF settings */ + usleep_range(100, 110); + wcd->sido_input_src = SIDO_SOURCE_RCO_BG; +} + +static int wcd9335_enable_efuse_sensing(struct snd_soc_component *comp) +{ + _wcd9335_codec_enable_mclk(comp, true); + snd_soc_component_update_bits(comp, + WCD9335_CHIP_TIER_CTRL_EFUSE_CTL, + WCD9335_CHIP_TIER_CTRL_EFUSE_EN_MASK, + WCD9335_CHIP_TIER_CTRL_EFUSE_ENABLE); + /* + * 5ms sleep required after enabling efuse control + * before checking the status. + */ + usleep_range(5000, 5500); + + if (!(snd_soc_component_read32(comp, + WCD9335_CHIP_TIER_CTRL_EFUSE_STATUS) & + WCD9335_CHIP_TIER_CTRL_EFUSE_EN_MASK)) + WARN(1, "%s: Efuse sense is not complete\n", __func__); + + wcd9335_enable_sido_buck(comp); + _wcd9335_codec_enable_mclk(comp, false); + + return 0; +} + +static void wcd9335_codec_init(struct snd_soc_component *component) +{ + struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); + int i; + + /* ungate MCLK and set clk rate */ + regmap_update_bits(wcd->regmap, WCD9335_CODEC_RPM_CLK_GATE, + WCD9335_CODEC_RPM_CLK_GATE_MCLK_GATE_MASK, 0); + + regmap_update_bits(wcd->regmap, WCD9335_CODEC_RPM_CLK_MCLK_CFG, + WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, + WCD9335_CODEC_RPM_CLK_MCLK_CFG_9P6MHZ); + + for (i = 0; i < ARRAY_SIZE(wcd9335_codec_reg_init_common_val); i++) + snd_soc_component_update_bits(component, + wcd9335_codec_reg_init_common_val[i].reg, + wcd9335_codec_reg_init_common_val[i].mask, + wcd9335_codec_reg_init_common_val[i].val); + + if (WCD9335_IS_2_0(wcd->version)) { + for (i = 0; i < ARRAY_SIZE(wcd9335_codec_reg_init_val_2_0); i++) + snd_soc_component_update_bits(component, + wcd9335_codec_reg_init_val_2_0[i].reg, + wcd9335_codec_reg_init_val_2_0[i].mask, + wcd9335_codec_reg_init_val_2_0[i].val); + } + + wcd9335_enable_efuse_sensing(component); +} + +static int wcd9335_codec_probe(struct snd_soc_component *component) +{ + struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); + int i; + + snd_soc_component_init_regmap(component, wcd->regmap); + wcd->component = component; + + wcd9335_codec_init(component); + + for (i = 0; i < NUM_CODEC_DAIS; i++) + INIT_LIST_HEAD(&wcd->dai[i].slim_ch_list); + + return wcd9335_setup_irqs(wcd); +} + +static void wcd9335_codec_remove(struct snd_soc_component *comp) +{ + struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); + + free_irq(regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS), wcd); +} + +static int wcd9335_codec_set_sysclk(struct snd_soc_component *comp, + int clk_id, int source, + unsigned int freq, int dir) +{ + struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); + + wcd->mclk_rate = freq; + + if (wcd->mclk_rate == WCD9335_MCLK_CLK_12P288MHZ) + snd_soc_component_update_bits(comp, + WCD9335_CODEC_RPM_CLK_MCLK_CFG, + WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, + WCD9335_CODEC_RPM_CLK_MCLK_CFG_12P288MHZ); + else if (wcd->mclk_rate == WCD9335_MCLK_CLK_9P6MHZ) + snd_soc_component_update_bits(comp, + WCD9335_CODEC_RPM_CLK_MCLK_CFG, + WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, + WCD9335_CODEC_RPM_CLK_MCLK_CFG_9P6MHZ); + + return clk_set_rate(wcd->mclk, freq); +} + +static const struct snd_soc_component_driver wcd9335_component_drv = { + .probe = wcd9335_codec_probe, + .remove = wcd9335_codec_remove, + .set_sysclk = wcd9335_codec_set_sysclk, +}; + +static int wcd9335_probe(struct platform_device *pdev) +{ + struct wcd9335 *pdata = dev_get_drvdata(pdev->dev.parent); + struct device *dev = &pdev->dev; + struct wcd9335_codec *wcd; + + wcd = devm_kzalloc(dev, sizeof(*wcd), GFP_KERNEL); + if (!wcd) + return -ENOMEM; + + dev_set_drvdata(dev, wcd); + + memcpy(wcd->rx_chs, wcd9335_rx_chs, sizeof(wcd9335_rx_chs)); + + wcd->regmap = pdata->regmap; + wcd->if_regmap = pdata->ifd_regmap; + wcd->slim = pdata->slim; + wcd->slim_ifd = pdata->slim_ifd; + wcd->irq_data = pdata->irq_data; + wcd->version = pdata->version; + wcd->intf_type = pdata->intf_type; + wcd->dev = dev; + wcd->mclk = pdata->mclk; + wcd->native_clk = pdata->native_clk; + wcd->sido_input_src = SIDO_SOURCE_INTERNAL; + wcd->sido_voltage = SIDO_VOLTAGE_NOMINAL_MV; + + return devm_snd_soc_register_component(dev, &wcd9335_component_drv, + wcd9335_slim_dais, + ARRAY_SIZE(wcd9335_slim_dais)); +} + +static struct platform_driver wcd9335_codec_driver = { + .probe = wcd9335_probe, + .driver = { + .name = "wcd9335-codec", + }, +}; +module_platform_driver(wcd9335_codec_driver); +MODULE_DESCRIPTION("WCD9335 Codec driver"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From c8cb5f775c8dacb605e628a320ded42be3bd9453 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 27 Jul 2018 13:18:01 +0100 Subject: ASoC: wcd9335: add CLASS-H Controller support CLASS-H controller/Amplifier is common accorss Qualcomm WCD codec series. This patchset adds basic CLASS-H controller apis for WCD codecs after wcd9335 to use. Signed-off-by: Srinivas Kandagatla Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/wcd-clsh.c | 605 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wcd-clsh.h | 49 ++++ sound/soc/codecs/wcd9335.c | 10 + 4 files changed, 665 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/wcd-clsh.c create mode 100644 sound/soc/codecs/wcd-clsh.h (limited to 'sound/soc') diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 01410b63daac..e3a3d4694e15 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -192,7 +192,7 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o -snd-soc-wcd9335-objs := wcd9335.o +snd-soc-wcd9335-objs := wcd-clsh.o wcd9335.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm-adsp-objs := wm_adsp.o snd-soc-wm0010-objs := wm0010.o diff --git a/sound/soc/codecs/wcd-clsh.c b/sound/soc/codecs/wcd-clsh.c new file mode 100644 index 000000000000..2393456cbd97 --- /dev/null +++ b/sound/soc/codecs/wcd-clsh.c @@ -0,0 +1,605 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. +// Copyright (c) 2017-2018, Linaro Limited + +#include +#include +#include +#include +#include +#include +#include "wcd-clsh.h" + +struct wcd_clsh_ctrl { + int state; + int mode; + int flyback_users; + int buck_users; + int clsh_users; + int codec_version; + struct snd_soc_component *comp; +}; + +/* Class-H registers for codecs from and above WCD9335 */ +#define WCD9XXX_A_CDC_RX0_RX_PATH_CFG0 WCD9335_REG(0xB, 0x42) +#define WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK BIT(6) +#define WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE BIT(6) +#define WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE 0 +#define WCD9XXX_A_CDC_RX1_RX_PATH_CFG0 WCD9335_REG(0xB, 0x56) +#define WCD9XXX_A_CDC_RX2_RX_PATH_CFG0 WCD9335_REG(0xB, 0x6A) +#define WCD9XXX_A_CDC_CLSH_K1_MSB WCD9335_REG(0xC, 0x08) +#define WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK GENMASK(3, 0) +#define WCD9XXX_A_CDC_CLSH_K1_LSB WCD9335_REG(0xC, 0x09) +#define WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK GENMASK(7, 0) +#define WCD9XXX_A_ANA_RX_SUPPLIES WCD9335_REG(0x6, 0x08) +#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK BIT(1) +#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_H 0 +#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_AB BIT(1) +#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK BIT(2) +#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_UHQA BIT(2) +#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_DEFAULT 0 +#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK BIT(3) +#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_UHQA BIT(3) +#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_DEFAULT 0 +#define WCD9XXX_A_ANA_RX_VNEG_EN_MASK BIT(6) +#define WCD9XXX_A_ANA_RX_VNEG_EN_SHIFT 6 +#define WCD9XXX_A_ANA_RX_VNEG_ENABLE BIT(6) +#define WCD9XXX_A_ANA_RX_VNEG_DISABLE 0 +#define WCD9XXX_A_ANA_RX_VPOS_EN_MASK BIT(7) +#define WCD9XXX_A_ANA_RX_VPOS_EN_SHIFT 7 +#define WCD9XXX_A_ANA_RX_VPOS_ENABLE BIT(7) +#define WCD9XXX_A_ANA_RX_VPOS_DISABLE 0 +#define WCD9XXX_A_ANA_HPH WCD9335_REG(0x6, 0x09) +#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_MASK GENMASK(3, 2) +#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_UHQA 0x08 +#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_LP 0x04 +#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL 0x0 +#define WCD9XXX_A_CDC_CLSH_CRC WCD9335_REG(0xC, 0x01) +#define WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK BIT(0) +#define WCD9XXX_A_CDC_CLSH_CRC_CLK_ENABLE BIT(0) +#define WCD9XXX_A_CDC_CLSH_CRC_CLK_DISABLE 0 +#define WCD9XXX_FLYBACK_EN WCD9335_REG(0x6, 0xA4) +#define WCD9XXX_FLYBACK_EN_DELAY_SEL_MASK GENMASK(6, 5) +#define WCD9XXX_FLYBACK_EN_DELAY_26P25_US 0x40 +#define WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK BIT(4) +#define WCD9XXX_FLYBACK_EN_PWDN_WITHOUT_DELAY BIT(4) +#define WCD9XXX_FLYBACK_EN_PWDN_WITH_DELAY 0 +#define WCD9XXX_RX_BIAS_FLYB_BUFF WCD9335_REG(0x6, 0xC7) +#define WCD9XXX_RX_BIAS_FLYB_VNEG_5_UA_MASK GENMASK(7, 4) +#define WCD9XXX_RX_BIAS_FLYB_VPOS_5_UA_MASK GENMASK(0, 3) +#define WCD9XXX_HPH_L_EN WCD9335_REG(0x6, 0xD3) +#define WCD9XXX_HPH_CONST_SEL_L_MASK GENMASK(7, 3) +#define WCD9XXX_HPH_CONST_SEL_BYPASS 0 +#define WCD9XXX_HPH_CONST_SEL_LP_PATH 0x40 +#define WCD9XXX_HPH_CONST_SEL_HQ_PATH 0x80 +#define WCD9XXX_HPH_R_EN WCD9335_REG(0x6, 0xD6) +#define WCD9XXX_HPH_REFBUFF_UHQA_CTL WCD9335_REG(0x6, 0xDD) +#define WCD9XXX_HPH_REFBUFF_UHQA_GAIN_MASK GENMASK(2, 0) +#define WCD9XXX_CLASSH_CTRL_VCL_2 WCD9335_REG(0x6, 0x9B) +#define WCD9XXX_CLASSH_CTRL_VCL_2_VREF_FILT_1_MASK GENMASK(5, 4) +#define WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_50KOHM 0x20 +#define WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_0KOHM 0x0 +#define WCD9XXX_CDC_RX1_RX_PATH_CTL WCD9335_REG(0xB, 0x55) +#define WCD9XXX_CDC_RX2_RX_PATH_CTL WCD9335_REG(0xB, 0x69) +#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_CONTROL WCD9335_REG(0xD, 0x41) +#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_EN_MASK BIT(0) +#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_11P3_EN_MASK BIT(1) +#define WCD9XXX_CLASSH_CTRL_CCL_1 WCD9335_REG(0x6, 0x9C) +#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_MASK GENMASK(7, 4) +#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA 0x50 +#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_30MA 0x30 + +#define CLSH_REQ_ENABLE true +#define CLSH_REQ_DISABLE false +#define WCD_USLEEP_RANGE 50 + +enum { + DAC_GAIN_0DB = 0, + DAC_GAIN_0P2DB, + DAC_GAIN_0P4DB, + DAC_GAIN_0P6DB, + DAC_GAIN_0P8DB, + DAC_GAIN_M0P2DB, + DAC_GAIN_M0P4DB, + DAC_GAIN_M0P6DB, +}; + +static inline void wcd_enable_clsh_block(struct wcd_clsh_ctrl *ctrl, + bool enable) +{ + struct snd_soc_component *comp = ctrl->comp; + + if ((enable && ++ctrl->clsh_users == 1) || + (!enable && --ctrl->clsh_users == 0)) + snd_soc_component_update_bits(comp, WCD9XXX_A_CDC_CLSH_CRC, + WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK, + enable); + if (ctrl->clsh_users < 0) + ctrl->clsh_users = 0; +} + +static inline bool wcd_clsh_enable_status(struct snd_soc_component *comp) +{ + return snd_soc_component_read32(comp, WCD9XXX_A_CDC_CLSH_CRC) & + WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK; +} + +static inline void wcd_clsh_set_buck_mode(struct snd_soc_component *comp, + int mode) +{ + /* set to HIFI */ + if (mode == CLS_H_HIFI) + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK, + WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_UHQA); + else + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK, + WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_DEFAULT); +} + +static inline void wcd_clsh_set_flyback_mode(struct snd_soc_component *comp, + int mode) +{ + /* set to HIFI */ + if (mode == CLS_H_HIFI) + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK, + WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_UHQA); + else + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK, + WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_DEFAULT); +} + +static void wcd_clsh_buck_ctrl(struct wcd_clsh_ctrl *ctrl, + int mode, + bool enable) +{ + struct snd_soc_component *comp = ctrl->comp; + + /* enable/disable buck */ + if ((enable && (++ctrl->buck_users == 1)) || + (!enable && (--ctrl->buck_users == 0))) + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_VPOS_EN_MASK, + enable << WCD9XXX_A_ANA_RX_VPOS_EN_SHIFT); + /* + * 500us sleep is required after buck enable/disable + * as per HW requirement + */ + usleep_range(500, 500 + WCD_USLEEP_RANGE); +} + +static void wcd_clsh_flyback_ctrl(struct wcd_clsh_ctrl *ctrl, + int mode, + bool enable) +{ + struct snd_soc_component *comp = ctrl->comp; + + int vneg[] = {0x00, 0x40}; + + /* enable/disable flyback */ + if ((enable && (++ctrl->flyback_users == 1)) || + (!enable && (--ctrl->flyback_users == 0))) { + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_VNEG_EN_MASK, + enable << WCD9XXX_A_ANA_RX_VNEG_EN_SHIFT); + /* 100usec delay is needed as per HW requirement */ + usleep_range(100, 110); + + if (enable && (WCD9335_IS_1_1(ctrl->codec_version))) { + wcd_clsh_set_flyback_mode(comp, CLS_H_HIFI); + snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, + WCD9XXX_FLYBACK_EN_DELAY_SEL_MASK, + WCD9XXX_FLYBACK_EN_DELAY_26P25_US); + snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, + WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK, + WCD9XXX_FLYBACK_EN_PWDN_WITHOUT_DELAY); + vneg[0] = snd_soc_component_read32(comp, + WCD9XXX_A_ANA_RX_SUPPLIES); + vneg[0] &= ~(0x40); + vneg[1] = vneg[0] | 0x40; + + snd_soc_component_write(comp, + WCD9XXX_A_ANA_RX_SUPPLIES, vneg[0]); + snd_soc_component_write(comp, + WCD9XXX_A_ANA_RX_SUPPLIES, vneg[1]); + /* 500usec delay is needed as per HW requirement */ + usleep_range(500, 510); + snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, + WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK, + WCD9XXX_FLYBACK_EN_PWDN_WITH_DELAY); + wcd_clsh_set_flyback_mode(comp, mode); + } + + } + /* + * 500us sleep is required after flyback enable/disable + * as per HW requirement + */ + usleep_range(500, 500 + WCD_USLEEP_RANGE); +} + +static void wcd_clsh_set_gain_path(struct wcd_clsh_ctrl *ctrl, int mode) +{ + struct snd_soc_component *comp = ctrl->comp; + int val = 0; + + switch (mode) { + case CLS_H_NORMAL: + case CLS_AB: + val = WCD9XXX_HPH_CONST_SEL_BYPASS; + break; + case CLS_H_HIFI: + val = WCD9XXX_HPH_CONST_SEL_HQ_PATH; + break; + case CLS_H_LP: + val = WCD9XXX_HPH_CONST_SEL_LP_PATH; + break; + }; + + snd_soc_component_update_bits(comp, WCD9XXX_HPH_L_EN, + WCD9XXX_HPH_CONST_SEL_L_MASK, + val); + + snd_soc_component_update_bits(comp, WCD9XXX_HPH_R_EN, + WCD9XXX_HPH_CONST_SEL_L_MASK, + val); +} + +static void wcd_clsh_set_hph_mode(struct snd_soc_component *comp, + int mode) +{ + int val = 0, gain = 0, res_val; + int ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; + + res_val = WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_0KOHM; + switch (mode) { + case CLS_H_NORMAL: + res_val = WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_50KOHM; + val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL; + gain = DAC_GAIN_0DB; + ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; + break; + case CLS_AB: + val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL; + gain = DAC_GAIN_0DB; + ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; + break; + case CLS_H_HIFI: + val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_UHQA; + gain = DAC_GAIN_M0P2DB; + ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; + break; + case CLS_H_LP: + val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_LP; + ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_30MA; + break; + }; + + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_HPH, + WCD9XXX_A_ANA_HPH_PWR_LEVEL_MASK, val); + snd_soc_component_update_bits(comp, WCD9XXX_CLASSH_CTRL_VCL_2, + WCD9XXX_CLASSH_CTRL_VCL_2_VREF_FILT_1_MASK, + res_val); + if (mode != CLS_H_LP) + snd_soc_component_update_bits(comp, + WCD9XXX_HPH_REFBUFF_UHQA_CTL, + WCD9XXX_HPH_REFBUFF_UHQA_GAIN_MASK, + gain); + snd_soc_component_update_bits(comp, WCD9XXX_CLASSH_CTRL_CCL_1, + WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_MASK, + ipeak); +} + +static void wcd_clsh_set_flyback_current(struct snd_soc_component *comp, + int mode) +{ + + snd_soc_component_update_bits(comp, WCD9XXX_RX_BIAS_FLYB_BUFF, + WCD9XXX_RX_BIAS_FLYB_VPOS_5_UA_MASK, 0x0A); + snd_soc_component_update_bits(comp, WCD9XXX_RX_BIAS_FLYB_BUFF, + WCD9XXX_RX_BIAS_FLYB_VNEG_5_UA_MASK, 0x0A); + /* Sleep needed to avoid click and pop as per HW requirement */ + usleep_range(100, 110); +} + +static void wcd_clsh_set_buck_regulator_mode(struct snd_soc_component *comp, + int mode) +{ + if (mode == CLS_AB) + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK, + WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_AB); + else + snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, + WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK, + WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_H); +} + +static void wcd_clsh_state_lo(struct wcd_clsh_ctrl *ctrl, int req_state, + bool is_enable, int mode) +{ + struct snd_soc_component *comp = ctrl->comp; + + if (mode != CLS_AB) { + dev_err(comp->dev, "%s: LO cannot be in this mode: %d\n", + __func__, mode); + return; + } + + if (is_enable) { + wcd_clsh_set_buck_regulator_mode(comp, mode); + wcd_clsh_set_buck_mode(comp, mode); + wcd_clsh_set_flyback_mode(comp, mode); + wcd_clsh_flyback_ctrl(ctrl, mode, true); + wcd_clsh_set_flyback_current(comp, mode); + wcd_clsh_buck_ctrl(ctrl, mode, true); + } else { + wcd_clsh_buck_ctrl(ctrl, mode, false); + wcd_clsh_flyback_ctrl(ctrl, mode, false); + wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); + } +} + +static void wcd_clsh_state_hph_r(struct wcd_clsh_ctrl *ctrl, int req_state, + bool is_enable, int mode) +{ + struct snd_soc_component *comp = ctrl->comp; + + if (mode == CLS_H_NORMAL) { + dev_err(comp->dev, "%s: Normal mode not applicable for hph_r\n", + __func__); + return; + } + + if (is_enable) { + if (mode != CLS_AB) { + wcd_enable_clsh_block(ctrl, true); + /* + * These K1 values depend on the Headphone Impedance + * For now it is assumed to be 16 ohm + */ + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_CLSH_K1_MSB, + WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK, + 0x00); + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_CLSH_K1_LSB, + WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK, + 0xC0); + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_RX2_RX_PATH_CFG0, + WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, + WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); + } + wcd_clsh_set_buck_regulator_mode(comp, mode); + wcd_clsh_set_flyback_mode(comp, mode); + wcd_clsh_flyback_ctrl(ctrl, mode, true); + wcd_clsh_set_flyback_current(comp, mode); + wcd_clsh_set_buck_mode(comp, mode); + wcd_clsh_buck_ctrl(ctrl, mode, true); + wcd_clsh_set_hph_mode(comp, mode); + wcd_clsh_set_gain_path(ctrl, mode); + } else { + wcd_clsh_set_hph_mode(comp, CLS_H_NORMAL); + + if (mode != CLS_AB) { + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_RX2_RX_PATH_CFG0, + WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, + WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); + wcd_enable_clsh_block(ctrl, false); + } + /* buck and flyback set to default mode and disable */ + wcd_clsh_buck_ctrl(ctrl, CLS_H_NORMAL, false); + wcd_clsh_flyback_ctrl(ctrl, CLS_H_NORMAL, false); + wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); + } +} + +static void wcd_clsh_state_hph_l(struct wcd_clsh_ctrl *ctrl, int req_state, + bool is_enable, int mode) +{ + struct snd_soc_component *comp = ctrl->comp; + + if (mode == CLS_H_NORMAL) { + dev_err(comp->dev, "%s: Normal mode not applicable for hph_l\n", + __func__); + return; + } + + if (is_enable) { + if (mode != CLS_AB) { + wcd_enable_clsh_block(ctrl, true); + /* + * These K1 values depend on the Headphone Impedance + * For now it is assumed to be 16 ohm + */ + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_CLSH_K1_MSB, + WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK, + 0x00); + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_CLSH_K1_LSB, + WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK, + 0xC0); + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_RX1_RX_PATH_CFG0, + WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, + WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); + } + wcd_clsh_set_buck_regulator_mode(comp, mode); + wcd_clsh_set_flyback_mode(comp, mode); + wcd_clsh_flyback_ctrl(ctrl, mode, true); + wcd_clsh_set_flyback_current(comp, mode); + wcd_clsh_set_buck_mode(comp, mode); + wcd_clsh_buck_ctrl(ctrl, mode, true); + wcd_clsh_set_hph_mode(comp, mode); + wcd_clsh_set_gain_path(ctrl, mode); + } else { + wcd_clsh_set_hph_mode(comp, CLS_H_NORMAL); + + if (mode != CLS_AB) { + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_RX1_RX_PATH_CFG0, + WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, + WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); + wcd_enable_clsh_block(ctrl, false); + } + /* set buck and flyback to Default Mode */ + wcd_clsh_buck_ctrl(ctrl, CLS_H_NORMAL, false); + wcd_clsh_flyback_ctrl(ctrl, CLS_H_NORMAL, false); + wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); + } +} + +static void wcd_clsh_state_ear(struct wcd_clsh_ctrl *ctrl, int req_state, + bool is_enable, int mode) +{ + struct snd_soc_component *comp = ctrl->comp; + + if (mode != CLS_H_NORMAL) { + dev_err(comp->dev, "%s: mode: %d cannot be used for EAR\n", + __func__, mode); + return; + } + + if (is_enable) { + wcd_enable_clsh_block(ctrl, true); + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_RX0_RX_PATH_CFG0, + WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, + WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); + wcd_clsh_set_buck_mode(comp, mode); + wcd_clsh_set_flyback_mode(comp, mode); + wcd_clsh_flyback_ctrl(ctrl, mode, true); + wcd_clsh_set_flyback_current(comp, mode); + wcd_clsh_buck_ctrl(ctrl, mode, true); + } else { + snd_soc_component_update_bits(comp, + WCD9XXX_A_CDC_RX0_RX_PATH_CFG0, + WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, + WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); + wcd_enable_clsh_block(ctrl, false); + wcd_clsh_buck_ctrl(ctrl, mode, false); + wcd_clsh_flyback_ctrl(ctrl, mode, false); + wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); + wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); + } +} + +static int _wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, int req_state, + bool is_enable, int mode) +{ + switch (req_state) { + case WCD_CLSH_STATE_EAR: + wcd_clsh_state_ear(ctrl, req_state, is_enable, mode); + break; + case WCD_CLSH_STATE_HPHL: + wcd_clsh_state_hph_l(ctrl, req_state, is_enable, mode); + break; + case WCD_CLSH_STATE_HPHR: + wcd_clsh_state_hph_r(ctrl, req_state, is_enable, mode); + break; + break; + case WCD_CLSH_STATE_LO: + wcd_clsh_state_lo(ctrl, req_state, is_enable, mode); + break; + default: + break; + } + + return 0; +} + +/* + * Function: wcd_clsh_is_state_valid + * Params: state + * Description: + * Provides information on valid states of Class H configuration + */ +static bool wcd_clsh_is_state_valid(int state) +{ + switch (state) { + case WCD_CLSH_STATE_IDLE: + case WCD_CLSH_STATE_EAR: + case WCD_CLSH_STATE_HPHL: + case WCD_CLSH_STATE_HPHR: + case WCD_CLSH_STATE_LO: + return true; + default: + return false; + }; +} + +/* + * Function: wcd_clsh_fsm + * Params: ctrl, req_state, req_type, clsh_event + * Description: + * This function handles PRE DAC and POST DAC conditions of different devices + * and updates class H configuration of different combination of devices + * based on validity of their states. ctrl will contain current + * class h state information + */ +int wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, + enum wcd_clsh_event clsh_event, + int nstate, + enum wcd_clsh_mode mode) +{ + struct snd_soc_component *comp = ctrl->comp; + + if (nstate == ctrl->state) + return 0; + + if (!wcd_clsh_is_state_valid(nstate)) { + dev_err(comp->dev, "Class-H not a valid new state:\n"); + return -EINVAL; + } + + switch (clsh_event) { + case WCD_CLSH_EVENT_PRE_DAC: + _wcd_clsh_ctrl_set_state(ctrl, nstate, CLSH_REQ_ENABLE, mode); + break; + case WCD_CLSH_EVENT_POST_PA: + _wcd_clsh_ctrl_set_state(ctrl, nstate, CLSH_REQ_DISABLE, mode); + break; + }; + + ctrl->state = nstate; + ctrl->mode = mode; + + return 0; +} + +int wcd_clsh_ctrl_get_state(struct wcd_clsh_ctrl *ctrl) +{ + return ctrl->state; +} + +struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc(struct snd_soc_component *comp, + int version) +{ + struct wcd_clsh_ctrl *ctrl; + + ctrl = kzalloc(sizeof(*ctrl), GFP_KERNEL); + if (!ctrl) + return ERR_PTR(-ENOMEM); + + ctrl->state = WCD_CLSH_STATE_IDLE; + ctrl->comp = comp; + + return ctrl; +} + +void wcd_clsh_ctrl_free(struct wcd_clsh_ctrl *ctrl) +{ + kfree(ctrl); +} diff --git a/sound/soc/codecs/wcd-clsh.h b/sound/soc/codecs/wcd-clsh.h new file mode 100644 index 000000000000..a902f9893467 --- /dev/null +++ b/sound/soc/codecs/wcd-clsh.h @@ -0,0 +1,49 @@ +/* SPDX-License-Identifier: GPL-2.0 */ + +#ifndef _WCD_CLSH_V2_H_ +#define _WCD_CLSH_V2_H_ +#include + +enum wcd_clsh_event { + WCD_CLSH_EVENT_PRE_DAC = 1, + WCD_CLSH_EVENT_POST_PA, +}; + +/* + * Basic states for Class H state machine. + * represented as a bit mask within a u8 data type + * bit 0: EAR mode + * bit 1: HPH Left mode + * bit 2: HPH Right mode + * bit 3: Lineout mode + */ +#define WCD_CLSH_STATE_IDLE 0 +#define WCD_CLSH_STATE_EAR BIT(0) +#define WCD_CLSH_STATE_HPHL BIT(1) +#define WCD_CLSH_STATE_HPHR BIT(2) +#define WCD_CLSH_STATE_LO BIT(3) +#define WCD_CLSH_STATE_MAX 4 +#define NUM_CLSH_STATES_V2 BIT(WCD_CLSH_STATE_MAX) + +enum wcd_clsh_mode { + CLS_H_NORMAL = 0, /* Class-H Default */ + CLS_H_HIFI, /* Class-H HiFi */ + CLS_H_LP, /* Class-H Low Power */ + CLS_AB, /* Class-AB */ + CLS_H_LOHIFI, /* LoHIFI */ + CLS_NONE, /* None of the above modes */ +}; + +struct wcd_clsh_ctrl; + +extern struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc( + struct snd_soc_component *component, + int version); +extern void wcd_clsh_ctrl_free(struct wcd_clsh_ctrl *ctrl); +extern int wcd_clsh_ctrl_get_state(struct wcd_clsh_ctrl *ctrl); +extern int wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, + enum wcd_clsh_event event, + int state, + enum wcd_clsh_mode mode); + +#endif /* _WCD_CLSH_V2_H_ */ diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index bd9de5d45fa9..06c73699f16f 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -21,6 +21,7 @@ #include #include #include +#include "wcd-clsh.h" #define WCD9335_RATES_MASK (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ @@ -181,6 +182,7 @@ struct wcd9335_codec { int sido_ccl_cnt; enum wcd_clock_type clk_type; + struct wcd_clsh_ctrl *clsh_ctrl; u32 hph_mode; }; @@ -1066,6 +1068,13 @@ static int wcd9335_codec_probe(struct snd_soc_component *component) int i; snd_soc_component_init_regmap(component, wcd->regmap); + /* Class-H Init*/ + wcd->clsh_ctrl = wcd_clsh_ctrl_alloc(component, wcd->version); + if (IS_ERR(wcd->clsh_ctrl)) + return PTR_ERR(wcd->clsh_ctrl); + + /* Default HPH Mode to Class-H HiFi */ + wcd->hph_mode = CLS_H_HIFI; wcd->component = component; wcd9335_codec_init(component); @@ -1080,6 +1089,7 @@ static void wcd9335_codec_remove(struct snd_soc_component *comp) { struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); + wcd_clsh_ctrl_free(wcd->clsh_ctrl); free_irq(regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS), wcd); } -- cgit v1.2.3 From c25e295cd77b37903ddc9ee27384e17aad08f27c Mon Sep 17 00:00:00 2001 From: Rohit kumar Date: Wed, 1 Aug 2018 14:31:08 +0530 Subject: ASoC: qcom: Add support to parse common audio device nodes This adds support to parse cpu, platform and codec device nodes and add them in dai-links. Also, update apq8096 machine driver to use the common API. Acked-by: Srinivas Kandagatla Signed-off-by: Rohit kumar Signed-off-by: Mark Brown --- sound/soc/qcom/Makefile | 2 +- sound/soc/qcom/apq8096.c | 111 +++++----------------------------------------- sound/soc/qcom/common.c | 112 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/common.h | 12 +++++ 4 files changed, 136 insertions(+), 101 deletions(-) create mode 100644 sound/soc/qcom/common.c create mode 100644 sound/soc/qcom/common.h (limited to 'sound/soc') diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index 206945bb9ba1..fefecc072265 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -13,7 +13,7 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o # Machine snd-soc-storm-objs := storm.o snd-soc-apq8016-sbc-objs := apq8016_sbc.o -snd-soc-apq8096-objs := apq8096.o +snd-soc-apq8096-objs := apq8096.o common.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index a56156281c8d..1e4a90d59228 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -9,6 +9,7 @@ #include #include #include +#include "common.h" static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) @@ -24,109 +25,16 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int apq8096_sbc_parse_of(struct snd_soc_card *card) +static void apq8096_add_be_ops(struct snd_soc_card *card) { - struct device_node *np; - struct device_node *codec = NULL; - struct device_node *platform = NULL; - struct device_node *cpu = NULL; - struct device *dev = card->dev; - struct snd_soc_dai_link *link; - int ret, num_links; - - ret = snd_soc_of_parse_card_name(card, "qcom,model"); - if (ret) { - dev_err(dev, "Error parsing card name: %d\n", ret); - return ret; - } - - /* DAPM routes */ - if (of_property_read_bool(dev->of_node, "qcom,audio-routing")) { - ret = snd_soc_of_parse_audio_routing(card, - "qcom,audio-routing"); - if (ret) - return ret; - } - - /* Populate links */ - num_links = of_get_child_count(dev->of_node); - - /* Allocate the DAI link array */ - card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL); - if (!card->dai_link) - return -ENOMEM; + struct snd_soc_dai_link *link = card->dai_link; + int i, num_links = card->num_links; - card->num_links = num_links; - link = card->dai_link; - - for_each_child_of_node(dev->of_node, np) { - cpu = of_get_child_by_name(np, "cpu"); - if (!cpu) { - dev_err(dev, "Can't find cpu DT node\n"); - ret = -EINVAL; - goto err; - } - - link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); - if (!link->cpu_of_node) { - dev_err(card->dev, "error getting cpu phandle\n"); - ret = -EINVAL; - goto err; - } - - ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); - if (ret) { - dev_err(card->dev, "error getting cpu dai name\n"); - goto err; - } - - platform = of_get_child_by_name(np, "platform"); - codec = of_get_child_by_name(np, "codec"); - if (codec && platform) { - link->platform_of_node = of_parse_phandle(platform, - "sound-dai", - 0); - if (!link->platform_of_node) { - dev_err(card->dev, "platform dai not found\n"); - ret = -EINVAL; - goto err; - } - - ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); - if (ret < 0) { - dev_err(card->dev, "codec dai not found\n"); - goto err; - } - link->no_pcm = 1; - link->ignore_pmdown_time = 1; + for (i = 0; i < num_links; i++) { + if (link->no_pcm == 1) link->be_hw_params_fixup = apq8096_be_hw_params_fixup; - } else { - link->platform_of_node = link->cpu_of_node; - link->codec_dai_name = "snd-soc-dummy-dai"; - link->codec_name = "snd-soc-dummy"; - link->dynamic = 1; - } - - link->ignore_suspend = 1; - ret = of_property_read_string(np, "link-name", &link->name); - if (ret) { - dev_err(card->dev, "error getting codec dai_link name\n"); - goto err; - } - - link->dpcm_playback = 1; - link->dpcm_capture = 1; - link->stream_name = link->name; link++; } - - return 0; -err: - of_node_put(cpu); - of_node_put(codec); - of_node_put(platform); - kfree(card->dai_link); - return ret; } static int apq8096_platform_probe(struct platform_device *pdev) @@ -142,18 +50,21 @@ static int apq8096_platform_probe(struct platform_device *pdev) card->dev = dev; card->auto_bind = true; dev_set_drvdata(dev, card); - ret = apq8096_sbc_parse_of(card); + ret = qcom_snd_parse_of(card); if (ret) { dev_err(dev, "Error parsing OF data\n"); goto err; } + apq8096_add_be_ops(card); ret = snd_soc_register_card(card); if (ret) - goto err; + goto err_card_register; return 0; +err_card_register: + kfree(card->dai_link); err: kfree(card); return ret; diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c new file mode 100644 index 000000000000..eb1b9da05dd4 --- /dev/null +++ b/sound/soc/qcom/common.c @@ -0,0 +1,112 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2018, Linaro Limited. +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#include +#include "common.h" + +int qcom_snd_parse_of(struct snd_soc_card *card) +{ + struct device_node *np; + struct device_node *codec = NULL; + struct device_node *platform = NULL; + struct device_node *cpu = NULL; + struct device *dev = card->dev; + struct snd_soc_dai_link *link; + int ret, num_links; + + ret = snd_soc_of_parse_card_name(card, "model"); + if (ret) { + dev_err(dev, "Error parsing card name: %d\n", ret); + return ret; + } + + /* DAPM routes */ + if (of_property_read_bool(dev->of_node, "audio-routing")) { + ret = snd_soc_of_parse_audio_routing(card, + "audio-routing"); + if (ret) + return ret; + } + + /* Populate links */ + num_links = of_get_child_count(dev->of_node); + + /* Allocate the DAI link array */ + card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL); + if (!card->dai_link) + return -ENOMEM; + + card->num_links = num_links; + link = card->dai_link; + for_each_child_of_node(dev->of_node, np) { + cpu = of_get_child_by_name(np, "cpu"); + if (!cpu) { + dev_err(dev, "Can't find cpu DT node\n"); + ret = -EINVAL; + goto err; + } + + link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); + if (!link->cpu_of_node) { + dev_err(card->dev, "error getting cpu phandle\n"); + ret = -EINVAL; + goto err; + } + + ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); + if (ret) { + dev_err(card->dev, "error getting cpu dai name\n"); + goto err; + } + + platform = of_get_child_by_name(np, "platform"); + codec = of_get_child_by_name(np, "codec"); + if (codec && platform) { + link->platform_of_node = of_parse_phandle(platform, + "sound-dai", + 0); + if (!link->platform_of_node) { + dev_err(card->dev, "platform dai not found\n"); + ret = -EINVAL; + goto err; + } + + ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); + if (ret < 0) { + dev_err(card->dev, "codec dai not found\n"); + goto err; + } + link->no_pcm = 1; + link->ignore_pmdown_time = 1; + } else { + link->platform_of_node = link->cpu_of_node; + link->codec_dai_name = "snd-soc-dummy-dai"; + link->codec_name = "snd-soc-dummy"; + link->dynamic = 1; + } + + link->ignore_suspend = 1; + ret = of_property_read_string(np, "link-name", &link->name); + if (ret) { + dev_err(card->dev, "error getting codec dai_link name\n"); + goto err; + } + + link->dpcm_playback = 1; + link->dpcm_capture = 1; + link->stream_name = link->name; + link++; + } + + return 0; +err: + of_node_put(cpu); + of_node_put(codec); + of_node_put(platform); + kfree(card->dai_link); + return ret; +} +EXPORT_SYMBOL(qcom_snd_parse_of); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/common.h b/sound/soc/qcom/common.h new file mode 100644 index 000000000000..ad5d2cf27459 --- /dev/null +++ b/sound/soc/qcom/common.h @@ -0,0 +1,12 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#ifndef __QCOM_SND_COMMON_H__ +#define __QCOM_SND_COMMON_H__ + +#include +#include + +int qcom_snd_parse_of(struct snd_soc_card *card); + +#endif -- cgit v1.2.3 From 6b1687bf76ef84cb1e31386c4871a01fe66937bf Mon Sep 17 00:00:00 2001 From: Rohit kumar Date: Wed, 1 Aug 2018 14:31:09 +0530 Subject: ASoC: qcom: add sdm845 sound card support This patch adds sdm845 audio machine driver support. Acked-by: Srinivas Kandagatla Signed-off-by: Rohit kumar Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 8 ++ sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sdm845.c | 286 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 296 insertions(+) create mode 100644 sound/soc/qcom/sdm845.c (limited to 'sound/soc') diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 87838fa27997..350730839c6f 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -90,3 +90,11 @@ config SND_SOC_MSM8996 Support for Qualcomm Technologies LPASS audio block in APQ8096 SoC-based systems. Say Y if you want to use audio device on this SoCs + +config SND_SOC_SDM845 + tristate "SoC Machine driver for SDM845 boards" + select SND_SOC_QDSP6 + help + To add support for audio on Qualcomm Technologies Inc. + SDM845 SoC-based systems. + Say Y if you want to use audio device on this SoCs. diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index fefecc072265..f0e94d48ba98 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -14,10 +14,12 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o snd-soc-storm-objs := storm.o snd-soc-apq8016-sbc-objs := apq8016_sbc.o snd-soc-apq8096-objs := apq8096.o common.o +snd-soc-sdm845-objs := sdm845.o common.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o +obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o #DSP lib obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/ diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c new file mode 100644 index 000000000000..bf4ec4646906 --- /dev/null +++ b/sound/soc/qcom/sdm845.c @@ -0,0 +1,286 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2018, The Linux Foundation. All rights reserved. + */ + +#include +#include +#include +#include +#include +#include +#include +#include "common.h" +#include "qdsp6/q6afe.h" + +#define DEFAULT_SAMPLE_RATE_48K 48000 +#define DEFAULT_MCLK_RATE 24576000 +#define DEFAULT_BCLK_RATE 12288000 + +struct sdm845_snd_data { + struct snd_soc_card *card; + uint32_t pri_mi2s_clk_count; + uint32_t quat_tdm_clk_count; +}; + +static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28}; + +static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret = 0; + int channels, slot_width; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + slot_width = 32; + break; + default: + dev_err(rtd->dev, "%s: invalid param format 0x%x\n", + __func__, params_format(params)); + return -EINVAL; + } + + channels = params_channels(params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0x3, + 8, slot_width); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n", + __func__, ret); + goto end; + } + + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, + channels, tdm_slot_offset); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n", + __func__, ret); + goto end; + } + } else { + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xf, 0, + 8, slot_width); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n", + __func__, ret); + goto end; + } + + ret = snd_soc_dai_set_channel_map(cpu_dai, channels, + tdm_slot_offset, 0, NULL); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n", + __func__, ret); + goto end; + } + } +end: + return ret; +} + +static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret = 0; + + switch (cpu_dai->id) { + case QUATERNARY_TDM_RX_0: + case QUATERNARY_TDM_TX_0: + ret = sdm845_tdm_snd_hw_params(substream, params); + break; + default: + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); + break; + } + return ret; +} + +static int sdm845_snd_startup(struct snd_pcm_substream *substream) +{ + unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + case PRIMARY_MI2S_TX: + if (++(data->pri_mi2s_clk_count) == 1) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_MCLK_1, + DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, + DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + } + snd_soc_dai_set_fmt(cpu_dai, fmt); + break; + + case QUATERNARY_TDM_RX_0: + case QUATERNARY_TDM_TX_0: + if (++(data->quat_tdm_clk_count) == 1) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT, + DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + } + break; + + default: + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); + break; + } + return 0; +} + +static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + case PRIMARY_MI2S_TX: + if (--(data->pri_mi2s_clk_count) == 0) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_MCLK_1, + 0, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, + 0, SNDRV_PCM_STREAM_PLAYBACK); + }; + break; + + case QUATERNARY_TDM_RX_0: + case QUATERNARY_TDM_TX_0: + if (--(data->quat_tdm_clk_count) == 0) { + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT, + 0, SNDRV_PCM_STREAM_PLAYBACK); + } + break; + + default: + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id); + break; + } +} + +static struct snd_soc_ops sdm845_be_ops = { + .hw_params = sdm845_snd_hw_params, + .startup = sdm845_snd_startup, + .shutdown = sdm845_snd_shutdown, +}; + +static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + rate->min = rate->max = DEFAULT_SAMPLE_RATE_48K; + channels->min = channels->max = 2; + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static void sdm845_add_be_ops(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *link = card->dai_link; + int i, num_links = card->num_links; + + for (i = 0; i < num_links; i++) { + if (link->no_pcm == 1) { + link->ops = &sdm845_be_ops; + link->be_hw_params_fixup = sdm845_be_hw_params_fixup; + } + link++; + } +} + +static int sdm845_snd_platform_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card; + struct sdm845_snd_data *data; + struct device *dev = &pdev->dev; + int ret; + + card = kzalloc(sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + /* Allocate the private data */ + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + card->dev = dev; + card->auto_bind = true; + dev_set_drvdata(dev, card); + ret = qcom_snd_parse_of(card); + if (ret) { + dev_err(dev, "Error parsing OF data\n"); + goto parse_dt_fail; + } + + data->card = card; + snd_soc_card_set_drvdata(card, data); + + sdm845_add_be_ops(card); + ret = snd_soc_register_card(card); + if (ret) { + dev_err(dev, "Sound card registration failed\n"); + goto register_card_fail; + } + return ret; + +register_card_fail: + kfree(card->dai_link); +parse_dt_fail: + kfree(data); + kfree(card); + return ret; +} + +static int sdm845_snd_platform_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); + + card->auto_bind = false; + snd_soc_unregister_card(card); + kfree(card->dai_link); + kfree(data); + kfree(card); + return 0; +} + +static const struct of_device_id sdm845_snd_device_id[] = { + { .compatible = "qcom,sdm845-sndcard" }, + {}, +}; +MODULE_DEVICE_TABLE(of, sdm845_snd_device_id); + +static struct platform_driver sdm845_snd_driver = { + .probe = sdm845_snd_platform_probe, + .remove = sdm845_snd_platform_remove, + .driver = { + .name = "msm-snd-sdm845", + .of_match_table = sdm845_snd_device_id, + }, +}; +module_platform_driver(sdm845_snd_driver); + +MODULE_DESCRIPTION("sdm845 ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From d101f9b96ee08f0454989bc3adb10e6cf7f3f953 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 1 Aug 2018 11:47:39 +0100 Subject: ASoC: nau8540: remove redundant variable osrate Variable osrate is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'osrate' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Mark Brown --- sound/soc/codecs/nau8540.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index 17104f8dc1a9..e3c8cd17daf2 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -362,11 +362,8 @@ static const struct snd_soc_dapm_route nau8540_dapm_routes[] = { static int nau8540_clock_check(struct nau8540 *nau8540, int rate, int osr) { - int osrate; - if (osr >= ARRAY_SIZE(osr_adc_sel)) return -EINVAL; - osrate = osr_adc_sel[osr].osr; if (rate * osr > CLK_ADC_MAX) { dev_err(nau8540->dev, "exceed the maximum frequency of CLK_ADC\n"); -- cgit v1.2.3 From 18127744cf446f113ca33f07e5cea893388f781a Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 1 Aug 2018 11:47:40 +0100 Subject: ASoC: stm32: remove redundant pointers 'priv' and 'rtd' Pointer 'priv' is assigned and not used, removing this allows the removal of pointer 'rtd'. Cleans up clang warning: warning: variable 'priv' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King Signed-off-by: Mark Brown --- sound/soc/stm/stm32_adfsdm.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 0e9373064032..706ff005234f 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -269,16 +269,10 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_pcm_runtime *rtd) static void stm32_adfsdm_pcm_free(struct snd_pcm *pcm) { struct snd_pcm_substream *substream; - struct snd_soc_pcm_runtime *rtd; - struct stm32_adfsdm_priv *priv; substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; - if (substream) { - rtd = substream->private_data; - priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); - + if (substream) snd_pcm_lib_preallocate_free_for_all(pcm); - } } static struct snd_soc_component_driver stm32_adfsdm_soc_platform = { -- cgit v1.2.3 From b0a39d356ae1478a5876bb02ba08af155a3a5554 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Aug 2018 14:18:50 +0100 Subject: ASoC: wcd9335: Fix build due to CLASS-H Controller support This reverts commit c8cb5f775c8dac (ASoC: vert "ASoC: wcd9335: add CLASS-H Controller support) due to missing dependencies. Signed-off-by: Mark Brown --- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/wcd-clsh.c | 605 -------------------------------------------- sound/soc/codecs/wcd-clsh.h | 49 ---- sound/soc/codecs/wcd9335.c | 10 - 4 files changed, 1 insertion(+), 665 deletions(-) delete mode 100644 sound/soc/codecs/wcd-clsh.c delete mode 100644 sound/soc/codecs/wcd-clsh.h (limited to 'sound/soc') diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e3a3d4694e15..01410b63daac 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -192,7 +192,7 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o -snd-soc-wcd9335-objs := wcd-clsh.o wcd9335.o +snd-soc-wcd9335-objs := wcd9335.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm-adsp-objs := wm_adsp.o snd-soc-wm0010-objs := wm0010.o diff --git a/sound/soc/codecs/wcd-clsh.c b/sound/soc/codecs/wcd-clsh.c deleted file mode 100644 index 2393456cbd97..000000000000 --- a/sound/soc/codecs/wcd-clsh.c +++ /dev/null @@ -1,605 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. -// Copyright (c) 2017-2018, Linaro Limited - -#include -#include -#include -#include -#include -#include -#include "wcd-clsh.h" - -struct wcd_clsh_ctrl { - int state; - int mode; - int flyback_users; - int buck_users; - int clsh_users; - int codec_version; - struct snd_soc_component *comp; -}; - -/* Class-H registers for codecs from and above WCD9335 */ -#define WCD9XXX_A_CDC_RX0_RX_PATH_CFG0 WCD9335_REG(0xB, 0x42) -#define WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK BIT(6) -#define WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE BIT(6) -#define WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE 0 -#define WCD9XXX_A_CDC_RX1_RX_PATH_CFG0 WCD9335_REG(0xB, 0x56) -#define WCD9XXX_A_CDC_RX2_RX_PATH_CFG0 WCD9335_REG(0xB, 0x6A) -#define WCD9XXX_A_CDC_CLSH_K1_MSB WCD9335_REG(0xC, 0x08) -#define WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK GENMASK(3, 0) -#define WCD9XXX_A_CDC_CLSH_K1_LSB WCD9335_REG(0xC, 0x09) -#define WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK GENMASK(7, 0) -#define WCD9XXX_A_ANA_RX_SUPPLIES WCD9335_REG(0x6, 0x08) -#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK BIT(1) -#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_H 0 -#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_AB BIT(1) -#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK BIT(2) -#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_UHQA BIT(2) -#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_DEFAULT 0 -#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK BIT(3) -#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_UHQA BIT(3) -#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_DEFAULT 0 -#define WCD9XXX_A_ANA_RX_VNEG_EN_MASK BIT(6) -#define WCD9XXX_A_ANA_RX_VNEG_EN_SHIFT 6 -#define WCD9XXX_A_ANA_RX_VNEG_ENABLE BIT(6) -#define WCD9XXX_A_ANA_RX_VNEG_DISABLE 0 -#define WCD9XXX_A_ANA_RX_VPOS_EN_MASK BIT(7) -#define WCD9XXX_A_ANA_RX_VPOS_EN_SHIFT 7 -#define WCD9XXX_A_ANA_RX_VPOS_ENABLE BIT(7) -#define WCD9XXX_A_ANA_RX_VPOS_DISABLE 0 -#define WCD9XXX_A_ANA_HPH WCD9335_REG(0x6, 0x09) -#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_MASK GENMASK(3, 2) -#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_UHQA 0x08 -#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_LP 0x04 -#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL 0x0 -#define WCD9XXX_A_CDC_CLSH_CRC WCD9335_REG(0xC, 0x01) -#define WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK BIT(0) -#define WCD9XXX_A_CDC_CLSH_CRC_CLK_ENABLE BIT(0) -#define WCD9XXX_A_CDC_CLSH_CRC_CLK_DISABLE 0 -#define WCD9XXX_FLYBACK_EN WCD9335_REG(0x6, 0xA4) -#define WCD9XXX_FLYBACK_EN_DELAY_SEL_MASK GENMASK(6, 5) -#define WCD9XXX_FLYBACK_EN_DELAY_26P25_US 0x40 -#define WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK BIT(4) -#define WCD9XXX_FLYBACK_EN_PWDN_WITHOUT_DELAY BIT(4) -#define WCD9XXX_FLYBACK_EN_PWDN_WITH_DELAY 0 -#define WCD9XXX_RX_BIAS_FLYB_BUFF WCD9335_REG(0x6, 0xC7) -#define WCD9XXX_RX_BIAS_FLYB_VNEG_5_UA_MASK GENMASK(7, 4) -#define WCD9XXX_RX_BIAS_FLYB_VPOS_5_UA_MASK GENMASK(0, 3) -#define WCD9XXX_HPH_L_EN WCD9335_REG(0x6, 0xD3) -#define WCD9XXX_HPH_CONST_SEL_L_MASK GENMASK(7, 3) -#define WCD9XXX_HPH_CONST_SEL_BYPASS 0 -#define WCD9XXX_HPH_CONST_SEL_LP_PATH 0x40 -#define WCD9XXX_HPH_CONST_SEL_HQ_PATH 0x80 -#define WCD9XXX_HPH_R_EN WCD9335_REG(0x6, 0xD6) -#define WCD9XXX_HPH_REFBUFF_UHQA_CTL WCD9335_REG(0x6, 0xDD) -#define WCD9XXX_HPH_REFBUFF_UHQA_GAIN_MASK GENMASK(2, 0) -#define WCD9XXX_CLASSH_CTRL_VCL_2 WCD9335_REG(0x6, 0x9B) -#define WCD9XXX_CLASSH_CTRL_VCL_2_VREF_FILT_1_MASK GENMASK(5, 4) -#define WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_50KOHM 0x20 -#define WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_0KOHM 0x0 -#define WCD9XXX_CDC_RX1_RX_PATH_CTL WCD9335_REG(0xB, 0x55) -#define WCD9XXX_CDC_RX2_RX_PATH_CTL WCD9335_REG(0xB, 0x69) -#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_CONTROL WCD9335_REG(0xD, 0x41) -#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_EN_MASK BIT(0) -#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_11P3_EN_MASK BIT(1) -#define WCD9XXX_CLASSH_CTRL_CCL_1 WCD9335_REG(0x6, 0x9C) -#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_MASK GENMASK(7, 4) -#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA 0x50 -#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_30MA 0x30 - -#define CLSH_REQ_ENABLE true -#define CLSH_REQ_DISABLE false -#define WCD_USLEEP_RANGE 50 - -enum { - DAC_GAIN_0DB = 0, - DAC_GAIN_0P2DB, - DAC_GAIN_0P4DB, - DAC_GAIN_0P6DB, - DAC_GAIN_0P8DB, - DAC_GAIN_M0P2DB, - DAC_GAIN_M0P4DB, - DAC_GAIN_M0P6DB, -}; - -static inline void wcd_enable_clsh_block(struct wcd_clsh_ctrl *ctrl, - bool enable) -{ - struct snd_soc_component *comp = ctrl->comp; - - if ((enable && ++ctrl->clsh_users == 1) || - (!enable && --ctrl->clsh_users == 0)) - snd_soc_component_update_bits(comp, WCD9XXX_A_CDC_CLSH_CRC, - WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK, - enable); - if (ctrl->clsh_users < 0) - ctrl->clsh_users = 0; -} - -static inline bool wcd_clsh_enable_status(struct snd_soc_component *comp) -{ - return snd_soc_component_read32(comp, WCD9XXX_A_CDC_CLSH_CRC) & - WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK; -} - -static inline void wcd_clsh_set_buck_mode(struct snd_soc_component *comp, - int mode) -{ - /* set to HIFI */ - if (mode == CLS_H_HIFI) - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK, - WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_UHQA); - else - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK, - WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_DEFAULT); -} - -static inline void wcd_clsh_set_flyback_mode(struct snd_soc_component *comp, - int mode) -{ - /* set to HIFI */ - if (mode == CLS_H_HIFI) - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK, - WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_UHQA); - else - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK, - WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_DEFAULT); -} - -static void wcd_clsh_buck_ctrl(struct wcd_clsh_ctrl *ctrl, - int mode, - bool enable) -{ - struct snd_soc_component *comp = ctrl->comp; - - /* enable/disable buck */ - if ((enable && (++ctrl->buck_users == 1)) || - (!enable && (--ctrl->buck_users == 0))) - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VPOS_EN_MASK, - enable << WCD9XXX_A_ANA_RX_VPOS_EN_SHIFT); - /* - * 500us sleep is required after buck enable/disable - * as per HW requirement - */ - usleep_range(500, 500 + WCD_USLEEP_RANGE); -} - -static void wcd_clsh_flyback_ctrl(struct wcd_clsh_ctrl *ctrl, - int mode, - bool enable) -{ - struct snd_soc_component *comp = ctrl->comp; - - int vneg[] = {0x00, 0x40}; - - /* enable/disable flyback */ - if ((enable && (++ctrl->flyback_users == 1)) || - (!enable && (--ctrl->flyback_users == 0))) { - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VNEG_EN_MASK, - enable << WCD9XXX_A_ANA_RX_VNEG_EN_SHIFT); - /* 100usec delay is needed as per HW requirement */ - usleep_range(100, 110); - - if (enable && (WCD9335_IS_1_1(ctrl->codec_version))) { - wcd_clsh_set_flyback_mode(comp, CLS_H_HIFI); - snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, - WCD9XXX_FLYBACK_EN_DELAY_SEL_MASK, - WCD9XXX_FLYBACK_EN_DELAY_26P25_US); - snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, - WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK, - WCD9XXX_FLYBACK_EN_PWDN_WITHOUT_DELAY); - vneg[0] = snd_soc_component_read32(comp, - WCD9XXX_A_ANA_RX_SUPPLIES); - vneg[0] &= ~(0x40); - vneg[1] = vneg[0] | 0x40; - - snd_soc_component_write(comp, - WCD9XXX_A_ANA_RX_SUPPLIES, vneg[0]); - snd_soc_component_write(comp, - WCD9XXX_A_ANA_RX_SUPPLIES, vneg[1]); - /* 500usec delay is needed as per HW requirement */ - usleep_range(500, 510); - snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, - WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK, - WCD9XXX_FLYBACK_EN_PWDN_WITH_DELAY); - wcd_clsh_set_flyback_mode(comp, mode); - } - - } - /* - * 500us sleep is required after flyback enable/disable - * as per HW requirement - */ - usleep_range(500, 500 + WCD_USLEEP_RANGE); -} - -static void wcd_clsh_set_gain_path(struct wcd_clsh_ctrl *ctrl, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - int val = 0; - - switch (mode) { - case CLS_H_NORMAL: - case CLS_AB: - val = WCD9XXX_HPH_CONST_SEL_BYPASS; - break; - case CLS_H_HIFI: - val = WCD9XXX_HPH_CONST_SEL_HQ_PATH; - break; - case CLS_H_LP: - val = WCD9XXX_HPH_CONST_SEL_LP_PATH; - break; - }; - - snd_soc_component_update_bits(comp, WCD9XXX_HPH_L_EN, - WCD9XXX_HPH_CONST_SEL_L_MASK, - val); - - snd_soc_component_update_bits(comp, WCD9XXX_HPH_R_EN, - WCD9XXX_HPH_CONST_SEL_L_MASK, - val); -} - -static void wcd_clsh_set_hph_mode(struct snd_soc_component *comp, - int mode) -{ - int val = 0, gain = 0, res_val; - int ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; - - res_val = WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_0KOHM; - switch (mode) { - case CLS_H_NORMAL: - res_val = WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_50KOHM; - val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL; - gain = DAC_GAIN_0DB; - ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; - break; - case CLS_AB: - val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL; - gain = DAC_GAIN_0DB; - ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; - break; - case CLS_H_HIFI: - val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_UHQA; - gain = DAC_GAIN_M0P2DB; - ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; - break; - case CLS_H_LP: - val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_LP; - ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_30MA; - break; - }; - - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_HPH, - WCD9XXX_A_ANA_HPH_PWR_LEVEL_MASK, val); - snd_soc_component_update_bits(comp, WCD9XXX_CLASSH_CTRL_VCL_2, - WCD9XXX_CLASSH_CTRL_VCL_2_VREF_FILT_1_MASK, - res_val); - if (mode != CLS_H_LP) - snd_soc_component_update_bits(comp, - WCD9XXX_HPH_REFBUFF_UHQA_CTL, - WCD9XXX_HPH_REFBUFF_UHQA_GAIN_MASK, - gain); - snd_soc_component_update_bits(comp, WCD9XXX_CLASSH_CTRL_CCL_1, - WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_MASK, - ipeak); -} - -static void wcd_clsh_set_flyback_current(struct snd_soc_component *comp, - int mode) -{ - - snd_soc_component_update_bits(comp, WCD9XXX_RX_BIAS_FLYB_BUFF, - WCD9XXX_RX_BIAS_FLYB_VPOS_5_UA_MASK, 0x0A); - snd_soc_component_update_bits(comp, WCD9XXX_RX_BIAS_FLYB_BUFF, - WCD9XXX_RX_BIAS_FLYB_VNEG_5_UA_MASK, 0x0A); - /* Sleep needed to avoid click and pop as per HW requirement */ - usleep_range(100, 110); -} - -static void wcd_clsh_set_buck_regulator_mode(struct snd_soc_component *comp, - int mode) -{ - if (mode == CLS_AB) - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK, - WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_AB); - else - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK, - WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_H); -} - -static void wcd_clsh_state_lo(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (mode != CLS_AB) { - dev_err(comp->dev, "%s: LO cannot be in this mode: %d\n", - __func__, mode); - return; - } - - if (is_enable) { - wcd_clsh_set_buck_regulator_mode(comp, mode); - wcd_clsh_set_buck_mode(comp, mode); - wcd_clsh_set_flyback_mode(comp, mode); - wcd_clsh_flyback_ctrl(ctrl, mode, true); - wcd_clsh_set_flyback_current(comp, mode); - wcd_clsh_buck_ctrl(ctrl, mode, true); - } else { - wcd_clsh_buck_ctrl(ctrl, mode, false); - wcd_clsh_flyback_ctrl(ctrl, mode, false); - wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); - } -} - -static void wcd_clsh_state_hph_r(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (mode == CLS_H_NORMAL) { - dev_err(comp->dev, "%s: Normal mode not applicable for hph_r\n", - __func__); - return; - } - - if (is_enable) { - if (mode != CLS_AB) { - wcd_enable_clsh_block(ctrl, true); - /* - * These K1 values depend on the Headphone Impedance - * For now it is assumed to be 16 ohm - */ - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_CLSH_K1_MSB, - WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK, - 0x00); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_CLSH_K1_LSB, - WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK, - 0xC0); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX2_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); - } - wcd_clsh_set_buck_regulator_mode(comp, mode); - wcd_clsh_set_flyback_mode(comp, mode); - wcd_clsh_flyback_ctrl(ctrl, mode, true); - wcd_clsh_set_flyback_current(comp, mode); - wcd_clsh_set_buck_mode(comp, mode); - wcd_clsh_buck_ctrl(ctrl, mode, true); - wcd_clsh_set_hph_mode(comp, mode); - wcd_clsh_set_gain_path(ctrl, mode); - } else { - wcd_clsh_set_hph_mode(comp, CLS_H_NORMAL); - - if (mode != CLS_AB) { - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX2_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); - wcd_enable_clsh_block(ctrl, false); - } - /* buck and flyback set to default mode and disable */ - wcd_clsh_buck_ctrl(ctrl, CLS_H_NORMAL, false); - wcd_clsh_flyback_ctrl(ctrl, CLS_H_NORMAL, false); - wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); - } -} - -static void wcd_clsh_state_hph_l(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (mode == CLS_H_NORMAL) { - dev_err(comp->dev, "%s: Normal mode not applicable for hph_l\n", - __func__); - return; - } - - if (is_enable) { - if (mode != CLS_AB) { - wcd_enable_clsh_block(ctrl, true); - /* - * These K1 values depend on the Headphone Impedance - * For now it is assumed to be 16 ohm - */ - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_CLSH_K1_MSB, - WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK, - 0x00); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_CLSH_K1_LSB, - WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK, - 0xC0); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX1_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); - } - wcd_clsh_set_buck_regulator_mode(comp, mode); - wcd_clsh_set_flyback_mode(comp, mode); - wcd_clsh_flyback_ctrl(ctrl, mode, true); - wcd_clsh_set_flyback_current(comp, mode); - wcd_clsh_set_buck_mode(comp, mode); - wcd_clsh_buck_ctrl(ctrl, mode, true); - wcd_clsh_set_hph_mode(comp, mode); - wcd_clsh_set_gain_path(ctrl, mode); - } else { - wcd_clsh_set_hph_mode(comp, CLS_H_NORMAL); - - if (mode != CLS_AB) { - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX1_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); - wcd_enable_clsh_block(ctrl, false); - } - /* set buck and flyback to Default Mode */ - wcd_clsh_buck_ctrl(ctrl, CLS_H_NORMAL, false); - wcd_clsh_flyback_ctrl(ctrl, CLS_H_NORMAL, false); - wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); - } -} - -static void wcd_clsh_state_ear(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (mode != CLS_H_NORMAL) { - dev_err(comp->dev, "%s: mode: %d cannot be used for EAR\n", - __func__, mode); - return; - } - - if (is_enable) { - wcd_enable_clsh_block(ctrl, true); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX0_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); - wcd_clsh_set_buck_mode(comp, mode); - wcd_clsh_set_flyback_mode(comp, mode); - wcd_clsh_flyback_ctrl(ctrl, mode, true); - wcd_clsh_set_flyback_current(comp, mode); - wcd_clsh_buck_ctrl(ctrl, mode, true); - } else { - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX0_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); - wcd_enable_clsh_block(ctrl, false); - wcd_clsh_buck_ctrl(ctrl, mode, false); - wcd_clsh_flyback_ctrl(ctrl, mode, false); - wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); - } -} - -static int _wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - switch (req_state) { - case WCD_CLSH_STATE_EAR: - wcd_clsh_state_ear(ctrl, req_state, is_enable, mode); - break; - case WCD_CLSH_STATE_HPHL: - wcd_clsh_state_hph_l(ctrl, req_state, is_enable, mode); - break; - case WCD_CLSH_STATE_HPHR: - wcd_clsh_state_hph_r(ctrl, req_state, is_enable, mode); - break; - break; - case WCD_CLSH_STATE_LO: - wcd_clsh_state_lo(ctrl, req_state, is_enable, mode); - break; - default: - break; - } - - return 0; -} - -/* - * Function: wcd_clsh_is_state_valid - * Params: state - * Description: - * Provides information on valid states of Class H configuration - */ -static bool wcd_clsh_is_state_valid(int state) -{ - switch (state) { - case WCD_CLSH_STATE_IDLE: - case WCD_CLSH_STATE_EAR: - case WCD_CLSH_STATE_HPHL: - case WCD_CLSH_STATE_HPHR: - case WCD_CLSH_STATE_LO: - return true; - default: - return false; - }; -} - -/* - * Function: wcd_clsh_fsm - * Params: ctrl, req_state, req_type, clsh_event - * Description: - * This function handles PRE DAC and POST DAC conditions of different devices - * and updates class H configuration of different combination of devices - * based on validity of their states. ctrl will contain current - * class h state information - */ -int wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, - enum wcd_clsh_event clsh_event, - int nstate, - enum wcd_clsh_mode mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (nstate == ctrl->state) - return 0; - - if (!wcd_clsh_is_state_valid(nstate)) { - dev_err(comp->dev, "Class-H not a valid new state:\n"); - return -EINVAL; - } - - switch (clsh_event) { - case WCD_CLSH_EVENT_PRE_DAC: - _wcd_clsh_ctrl_set_state(ctrl, nstate, CLSH_REQ_ENABLE, mode); - break; - case WCD_CLSH_EVENT_POST_PA: - _wcd_clsh_ctrl_set_state(ctrl, nstate, CLSH_REQ_DISABLE, mode); - break; - }; - - ctrl->state = nstate; - ctrl->mode = mode; - - return 0; -} - -int wcd_clsh_ctrl_get_state(struct wcd_clsh_ctrl *ctrl) -{ - return ctrl->state; -} - -struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc(struct snd_soc_component *comp, - int version) -{ - struct wcd_clsh_ctrl *ctrl; - - ctrl = kzalloc(sizeof(*ctrl), GFP_KERNEL); - if (!ctrl) - return ERR_PTR(-ENOMEM); - - ctrl->state = WCD_CLSH_STATE_IDLE; - ctrl->comp = comp; - - return ctrl; -} - -void wcd_clsh_ctrl_free(struct wcd_clsh_ctrl *ctrl) -{ - kfree(ctrl); -} diff --git a/sound/soc/codecs/wcd-clsh.h b/sound/soc/codecs/wcd-clsh.h deleted file mode 100644 index a902f9893467..000000000000 --- a/sound/soc/codecs/wcd-clsh.h +++ /dev/null @@ -1,49 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ - -#ifndef _WCD_CLSH_V2_H_ -#define _WCD_CLSH_V2_H_ -#include - -enum wcd_clsh_event { - WCD_CLSH_EVENT_PRE_DAC = 1, - WCD_CLSH_EVENT_POST_PA, -}; - -/* - * Basic states for Class H state machine. - * represented as a bit mask within a u8 data type - * bit 0: EAR mode - * bit 1: HPH Left mode - * bit 2: HPH Right mode - * bit 3: Lineout mode - */ -#define WCD_CLSH_STATE_IDLE 0 -#define WCD_CLSH_STATE_EAR BIT(0) -#define WCD_CLSH_STATE_HPHL BIT(1) -#define WCD_CLSH_STATE_HPHR BIT(2) -#define WCD_CLSH_STATE_LO BIT(3) -#define WCD_CLSH_STATE_MAX 4 -#define NUM_CLSH_STATES_V2 BIT(WCD_CLSH_STATE_MAX) - -enum wcd_clsh_mode { - CLS_H_NORMAL = 0, /* Class-H Default */ - CLS_H_HIFI, /* Class-H HiFi */ - CLS_H_LP, /* Class-H Low Power */ - CLS_AB, /* Class-AB */ - CLS_H_LOHIFI, /* LoHIFI */ - CLS_NONE, /* None of the above modes */ -}; - -struct wcd_clsh_ctrl; - -extern struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc( - struct snd_soc_component *component, - int version); -extern void wcd_clsh_ctrl_free(struct wcd_clsh_ctrl *ctrl); -extern int wcd_clsh_ctrl_get_state(struct wcd_clsh_ctrl *ctrl); -extern int wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, - enum wcd_clsh_event event, - int state, - enum wcd_clsh_mode mode); - -#endif /* _WCD_CLSH_V2_H_ */ diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 06c73699f16f..bd9de5d45fa9 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -21,7 +21,6 @@ #include #include #include -#include "wcd-clsh.h" #define WCD9335_RATES_MASK (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ @@ -182,7 +181,6 @@ struct wcd9335_codec { int sido_ccl_cnt; enum wcd_clock_type clk_type; - struct wcd_clsh_ctrl *clsh_ctrl; u32 hph_mode; }; @@ -1068,13 +1066,6 @@ static int wcd9335_codec_probe(struct snd_soc_component *component) int i; snd_soc_component_init_regmap(component, wcd->regmap); - /* Class-H Init*/ - wcd->clsh_ctrl = wcd_clsh_ctrl_alloc(component, wcd->version); - if (IS_ERR(wcd->clsh_ctrl)) - return PTR_ERR(wcd->clsh_ctrl); - - /* Default HPH Mode to Class-H HiFi */ - wcd->hph_mode = CLS_H_HIFI; wcd->component = component; wcd9335_codec_init(component); @@ -1089,7 +1080,6 @@ static void wcd9335_codec_remove(struct snd_soc_component *comp) { struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - wcd_clsh_ctrl_free(wcd->clsh_ctrl); free_irq(regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS), wcd); } -- cgit v1.2.3 From b74fd69043262003c5b7ce545e59f5a7234ab290 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Aug 2018 14:22:56 +0100 Subject: ASoC: wcd9335: Fix build This reverts commit e57d4ca882e28 (ASoC: wcd9335: add support to wcd9335 codec) due to build failures caused by missing dependencies. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 - sound/soc/codecs/Makefile | 2 - sound/soc/codecs/wcd9335.c | 1154 -------------------------------------------- 3 files changed, 1161 deletions(-) delete mode 100644 sound/soc/codecs/wcd9335.c (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb09abf18dde..efb095dbcd71 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1066,11 +1066,6 @@ config SND_SOC_UDA1380 tristate depends on I2C -config SND_SOC_WCD9335 - tristate "WCD9335 Codec" - depends on MFD_WCD9335 - tristate - config SND_SOC_WL1273 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 01410b63daac..7ae7c85e8219 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -192,7 +192,6 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o -snd-soc-wcd9335-objs := wcd9335.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm-adsp-objs := wm_adsp.o snd-soc-wm0010-objs := wm0010.o @@ -452,7 +451,6 @@ obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o -obj-$(CONFIG_SND_SOC_WCD9335) += snd-soc-wcd9335.o obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c deleted file mode 100644 index bd9de5d45fa9..000000000000 --- a/sound/soc/codecs/wcd9335.c +++ /dev/null @@ -1,1154 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. -// Copyright (c) 2017-2018, Linaro Limited - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#define WCD9335_RATES_MASK (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) -/* Fractional Rates */ -#define WCD9335_FRAC_RATES_MASK (SNDRV_PCM_RATE_44100) -#define WCD9335_FORMATS_S16_S24_LE (SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE) - -/* slave port water mark level - * (0: 6bytes, 1: 9bytes, 2: 12 bytes, 3: 15 bytes) - */ -#define SLAVE_PORT_WATER_MARK_6BYTES 0 -#define SLAVE_PORT_WATER_MARK_9BYTES 1 -#define SLAVE_PORT_WATER_MARK_12BYTES 2 -#define SLAVE_PORT_WATER_MARK_15BYTES 3 -#define SLAVE_PORT_WATER_MARK_SHIFT 1 -#define SLAVE_PORT_ENABLE 1 -#define SLAVE_PORT_DISABLE 0 -#define WCD9335_SLIM_WATER_MARK_VAL \ - ((SLAVE_PORT_WATER_MARK_12BYTES << SLAVE_PORT_WATER_MARK_SHIFT) | \ - (SLAVE_PORT_ENABLE)) - -#define WCD9335_SLIM_NUM_PORT_REG 3 -#define WCD9335_SLIM_PGD_PORT_INT_TX_EN0 (WCD9335_SLIM_PGD_PORT_INT_EN0 + 2) - -#define WCD9335_MCLK_CLK_12P288MHZ 12288000 -#define WCD9335_MCLK_CLK_9P6MHZ 9600000 - -#define WCD9335_SLIM_CLOSE_TIMEOUT 1000 -#define WCD9335_SLIM_IRQ_OVERFLOW (1 << 0) -#define WCD9335_SLIM_IRQ_UNDERFLOW (1 << 1) -#define WCD9335_SLIM_IRQ_PORT_CLOSED (1 << 2) - -#define WCD9335_NUM_INTERPOLATORS 9 -#define WCD9335_RX_START 16 -#define WCD9335_SLIM_CH_START 128 - -#define WCD9335_SLIM_RX_CH(p) \ - {.port = p + WCD9335_RX_START, .shift = p,} - -/* vout step value */ -#define WCD9335_CALCULATE_VOUT_D(req_mv) (((req_mv - 650) * 10) / 25) - -enum { - WCD9335_RX0 = 0, - WCD9335_RX1, - WCD9335_RX2, - WCD9335_RX3, - WCD9335_RX4, - WCD9335_RX5, - WCD9335_RX6, - WCD9335_RX7, - WCD9335_RX8, - WCD9335_RX9, - WCD9335_RX10, - WCD9335_RX11, - WCD9335_RX12, - WCD9335_RX_MAX, -}; - -enum { - SIDO_SOURCE_INTERNAL = 0, - SIDO_SOURCE_RCO_BG, -}; - -enum wcd9335_sido_voltage { - SIDO_VOLTAGE_SVS_MV = 950, - SIDO_VOLTAGE_NOMINAL_MV = 1100, -}; - -enum { - AIF1_PB = 0, - AIF1_CAP, - AIF2_PB, - AIF2_CAP, - AIF3_PB, - AIF3_CAP, - AIF4_PB, - NUM_CODEC_DAIS, -}; - -enum { - INTn_2_INP_SEL_ZERO = 0, - INTn_2_INP_SEL_RX0, - INTn_2_INP_SEL_RX1, - INTn_2_INP_SEL_RX2, - INTn_2_INP_SEL_RX3, - INTn_2_INP_SEL_RX4, - INTn_2_INP_SEL_RX5, - INTn_2_INP_SEL_RX6, - INTn_2_INP_SEL_RX7, - INTn_2_INP_SEL_PROXIMITY, -}; - -enum { - INTn_1_MIX_INP_SEL_ZERO = 0, - INTn_1_MIX_INP_SEL_DEC0, - INTn_1_MIX_INP_SEL_DEC1, - INTn_1_MIX_INP_SEL_IIR0, - INTn_1_MIX_INP_SEL_IIR1, - INTn_1_MIX_INP_SEL_RX0, - INTn_1_MIX_INP_SEL_RX1, - INTn_1_MIX_INP_SEL_RX2, - INTn_1_MIX_INP_SEL_RX3, - INTn_1_MIX_INP_SEL_RX4, - INTn_1_MIX_INP_SEL_RX5, - INTn_1_MIX_INP_SEL_RX6, - INTn_1_MIX_INP_SEL_RX7, - -}; - -enum wcd_clock_type { - WCD_CLK_OFF, - WCD_CLK_RCO, - WCD_CLK_MCLK, -}; - -struct wcd9335_slim_ch { - u32 ch_num; - u16 port; - u16 shift; - struct list_head list; -}; - -struct wcd_slim_codec_dai_data { - struct list_head slim_ch_list; - struct slim_stream_config sconfig; - struct slim_stream_runtime *sruntime; -}; - -struct wcd9335_codec { - struct device *dev; - struct clk *mclk; - struct clk *native_clk; - u32 mclk_rate; - u8 intf_type; - u8 version; - - struct slim_device *slim; - struct slim_device *slim_ifd; - struct regmap *regmap; - struct regmap *if_regmap; - struct regmap_irq_chip_data *irq_data; - - struct wcd9335_slim_ch rx_chs[WCD9335_RX_MAX]; - u32 num_rx_port; - - int sido_input_src; - enum wcd9335_sido_voltage sido_voltage; - - struct wcd_slim_codec_dai_data dai[NUM_CODEC_DAIS]; - struct snd_soc_component *component; - - int master_bias_users; - int clk_mclk_users; - int clk_rco_users; - int sido_ccl_cnt; - enum wcd_clock_type clk_type; - - u32 hph_mode; -}; - -static const struct wcd9335_slim_ch wcd9335_rx_chs[WCD9335_RX_MAX] = { - WCD9335_SLIM_RX_CH(0), /* 16 */ - WCD9335_SLIM_RX_CH(1), /* 17 */ - WCD9335_SLIM_RX_CH(2), - WCD9335_SLIM_RX_CH(3), - WCD9335_SLIM_RX_CH(4), - WCD9335_SLIM_RX_CH(5), - WCD9335_SLIM_RX_CH(6), - WCD9335_SLIM_RX_CH(7), - WCD9335_SLIM_RX_CH(8), - WCD9335_SLIM_RX_CH(9), - WCD9335_SLIM_RX_CH(10), - WCD9335_SLIM_RX_CH(11), - WCD9335_SLIM_RX_CH(12), -}; - -struct interp_sample_rate { - int rate; - int rate_val; -}; - -static struct interp_sample_rate int_mix_rate_val[] = { - {48000, 0x4}, /* 48K */ - {96000, 0x5}, /* 96K */ - {192000, 0x6}, /* 192K */ -}; - -static struct interp_sample_rate int_prim_rate_val[] = { - {8000, 0x0}, /* 8K */ - {16000, 0x1}, /* 16K */ - {24000, -EINVAL},/* 24K */ - {32000, 0x3}, /* 32K */ - {48000, 0x4}, /* 48K */ - {96000, 0x5}, /* 96K */ - {192000, 0x6}, /* 192K */ - {384000, 0x7}, /* 384K */ - {44100, 0x8}, /* 44.1K */ -}; - -struct wcd9335_reg_mask_val { - u16 reg; - u8 mask; - u8 val; -}; - -static const struct wcd9335_reg_mask_val wcd9335_codec_reg_init_val_2_0[] = { - {WCD9335_RCO_CTRL_2, 0x0F, 0x08}, - {WCD9335_RX_BIAS_FLYB_MID_RST, 0xF0, 0x10}, - {WCD9335_FLYBACK_CTRL_1, 0x20, 0x20}, - {WCD9335_HPH_OCP_CTL, 0xFF, 0x5A}, - {WCD9335_HPH_L_TEST, 0x01, 0x01}, - {WCD9335_HPH_R_TEST, 0x01, 0x01}, - {WCD9335_CDC_BOOST0_BOOST_CFG1, 0x3F, 0x12}, - {WCD9335_CDC_BOOST0_BOOST_CFG2, 0x1C, 0x08}, - {WCD9335_CDC_COMPANDER7_CTL7, 0x1E, 0x18}, - {WCD9335_CDC_BOOST1_BOOST_CFG1, 0x3F, 0x12}, - {WCD9335_CDC_BOOST1_BOOST_CFG2, 0x1C, 0x08}, - {WCD9335_CDC_COMPANDER8_CTL7, 0x1E, 0x18}, - {WCD9335_CDC_TX0_TX_PATH_SEC7, 0xFF, 0x45}, - {WCD9335_CDC_RX0_RX_PATH_SEC0, 0xFC, 0xF4}, - {WCD9335_HPH_REFBUFF_LP_CTL, 0x08, 0x08}, - {WCD9335_HPH_REFBUFF_LP_CTL, 0x06, 0x02}, -}; - -static const struct wcd9335_reg_mask_val wcd9335_codec_reg_init_common_val[] = { - /* Rbuckfly/R_EAR(32) */ - {WCD9335_CDC_CLSH_K2_MSB, 0x0F, 0x00}, - {WCD9335_CDC_CLSH_K2_LSB, 0xFF, 0x60}, - {WCD9335_CPE_SS_DMIC_CFG, 0x80, 0x00}, - {WCD9335_CDC_BOOST0_BOOST_CTL, 0x70, 0x50}, - {WCD9335_CDC_BOOST1_BOOST_CTL, 0x70, 0x50}, - {WCD9335_CDC_RX7_RX_PATH_CFG1, 0x08, 0x08}, - {WCD9335_CDC_RX8_RX_PATH_CFG1, 0x08, 0x08}, - {WCD9335_ANA_LO_1_2, 0x3C, 0X3C}, - {WCD9335_DIFF_LO_COM_SWCAP_REFBUF_FREQ, 0x70, 0x00}, - {WCD9335_DIFF_LO_COM_PA_FREQ, 0x70, 0x40}, - {WCD9335_SOC_MAD_AUDIO_CTL_2, 0x03, 0x03}, - {WCD9335_CDC_TOP_TOP_CFG1, 0x02, 0x02}, - {WCD9335_CDC_TOP_TOP_CFG1, 0x01, 0x01}, - {WCD9335_EAR_CMBUFF, 0x08, 0x00}, - {WCD9335_CDC_TX9_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_TX10_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_TX11_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_TX12_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_COMPANDER7_CTL3, 0x80, 0x80}, - {WCD9335_CDC_COMPANDER8_CTL3, 0x80, 0x80}, - {WCD9335_CDC_COMPANDER7_CTL7, 0x01, 0x01}, - {WCD9335_CDC_COMPANDER8_CTL7, 0x01, 0x01}, - {WCD9335_CDC_RX0_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX1_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX2_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX3_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX4_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX5_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX6_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX7_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX8_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX0_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX1_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX2_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX3_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX4_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX5_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX6_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX7_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX8_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_VBADC_IBIAS_FE, 0x0C, 0x08}, -}; - -static int wcd9335_set_mix_interpolator_rate(struct snd_soc_dai *dai, - int rate_val, - u32 rate) -{ - struct snd_soc_component *component = dai->component; - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - struct wcd9335_slim_ch *ch; - int val, j; - - list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) { - for (j = 0; j < WCD9335_NUM_INTERPOLATORS; j++) { - val = snd_soc_component_read32(component, - WCD9335_CDC_RX_INP_MUX_RX_INT_CFG1(j)) & - WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; - - if (val == (ch->shift + INTn_2_INP_SEL_RX0)) - snd_soc_component_update_bits(component, - WCD9335_CDC_RX_PATH_MIX_CTL(j), - WCD9335_CDC_MIX_PCM_RATE_MASK, - rate_val); - } - } - - return 0; -} - -static int wcd9335_set_prim_interpolator_rate(struct snd_soc_dai *dai, - u8 rate_val, - u32 rate) -{ - struct snd_soc_component *comp = dai->component; - struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - struct wcd9335_slim_ch *ch; - u8 cfg0, cfg1, inp0_sel, inp1_sel, inp2_sel; - int inp, j; - - list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) { - inp = ch->shift + INTn_1_MIX_INP_SEL_RX0; - /* - * Loop through all interpolator MUX inputs and find out - * to which interpolator input, the slim rx port - * is connected - */ - for (j = 0; j < WCD9335_NUM_INTERPOLATORS; j++) { - cfg0 = snd_soc_component_read32(comp, - WCD9335_CDC_RX_INP_MUX_RX_INT_CFG0(j)); - cfg1 = snd_soc_component_read32(comp, - WCD9335_CDC_RX_INP_MUX_RX_INT_CFG1(j)); - - inp0_sel = cfg0 & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; - inp1_sel = (cfg0 >> 4) & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; - inp2_sel = (cfg1 >> 4) & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; - - if ((inp0_sel == inp) || (inp1_sel == inp) || - (inp2_sel == inp)) { - /* rate is in Hz */ - if ((j == 0) && (rate == 44100)) - dev_info(wcd->dev, - "Cannot set 44.1KHz on INT0\n"); - else - snd_soc_component_update_bits(comp, - WCD9335_CDC_RX_PATH_CTL(j), - WCD9335_CDC_MIX_PCM_RATE_MASK, - rate_val); - } - } - } - - return 0; -} - -static int wcd9335_set_interpolator_rate(struct snd_soc_dai *dai, u32 rate) -{ - int i; - - /* set mixing path rate */ - for (i = 0; i < ARRAY_SIZE(int_mix_rate_val); i++) { - if (rate == int_mix_rate_val[i].rate) { - wcd9335_set_mix_interpolator_rate(dai, - int_mix_rate_val[i].rate_val, rate); - break; - } - } - - /* set primary path sample rate */ - for (i = 0; i < ARRAY_SIZE(int_prim_rate_val); i++) { - if (rate == int_prim_rate_val[i].rate) { - wcd9335_set_prim_interpolator_rate(dai, - int_prim_rate_val[i].rate_val, rate); - break; - } - } - - return 0; -} - -static int wcd9335_slim_set_hw_params(struct wcd9335_codec *wcd, - struct wcd_slim_codec_dai_data *dai_data, - int direction) -{ - struct list_head *slim_ch_list = &dai_data->slim_ch_list; - struct slim_stream_config *cfg = &dai_data->sconfig; - struct wcd9335_slim_ch *ch; - u16 payload = 0; - int ret, i; - - cfg->ch_count = 0; - cfg->direction = direction; - cfg->port_mask = 0; - - /* Configure slave interface device */ - list_for_each_entry(ch, slim_ch_list, list) { - cfg->ch_count++; - payload |= 1 << ch->shift; - cfg->port_mask |= BIT(ch->port); - } - - cfg->chs = kcalloc(cfg->ch_count, sizeof(unsigned int), GFP_KERNEL); - if (!cfg->chs) - return -ENOMEM; - - i = 0; - list_for_each_entry(ch, slim_ch_list, list) { - cfg->chs[i++] = ch->ch_num; - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { - /* write to interface device */ - ret = regmap_write(wcd->if_regmap, - WCD9335_SLIM_PGD_RX_PORT_MULTI_CHNL_0(ch->port), - payload); - - if (ret < 0) - goto err; - - /* configure the slave port for water mark and enable*/ - ret = regmap_write(wcd->if_regmap, - WCD9335_SLIM_PGD_RX_PORT_CFG(ch->port), - WCD9335_SLIM_WATER_MARK_VAL); - if (ret < 0) - goto err; - } - } - - dai_data->sruntime = slim_stream_allocate(wcd->slim, "WCD9335-SLIM"); - slim_stream_prepare(dai_data->sruntime, cfg); - - return 0; - -err: - dev_err(wcd->dev, "Error Setting slim hw params\n"); - kfree(cfg->chs); - cfg->chs = NULL; - - return ret; -} - -static int wcd9335_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct wcd9335_codec *wcd; - int ret; - - wcd = snd_soc_component_get_drvdata(dai->component); - - switch (substream->stream) { - case SNDRV_PCM_STREAM_PLAYBACK: - ret = wcd9335_set_interpolator_rate(dai, params_rate(params)); - if (ret) { - dev_err(wcd->dev, "cannot set sample rate: %u\n", - params_rate(params)); - return ret; - } - switch (params_width(params)) { - case 16 ... 24: - wcd->dai[dai->id].sconfig.bps = params_width(params); - break; - default: - dev_err(wcd->dev, "%s: Invalid format 0x%x\n", - __func__, params_width(params)); - return -EINVAL; - } - break; - default: - dev_err(wcd->dev, "Invalid stream type %d\n", - substream->stream); - return -EINVAL; - }; - - wcd->dai[dai->id].sconfig.rate = params_rate(params); - wcd9335_slim_set_hw_params(wcd, &wcd->dai[dai->id], substream->stream); - - return 0; -} - -static int wcd9335_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct wcd_slim_codec_dai_data *dai_data; - struct wcd9335_codec *wcd; - - wcd = snd_soc_component_get_drvdata(dai->component); - dai_data = &wcd->dai[dai->id]; - slim_stream_enable(dai_data->sruntime); - - return 0; -} - -static int wcd9335_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) -{ - struct wcd9335_codec *wcd; - int i; - - wcd = snd_soc_component_get_drvdata(dai->component); - - if (!tx_slot || !rx_slot) { - dev_err(wcd->dev, "Invalid tx_slot=%p, rx_slot=%p\n", - tx_slot, rx_slot); - return -EINVAL; - } - - if (wcd->rx_chs) { - wcd->num_rx_port = rx_num; - for (i = 0; i < rx_num; i++) { - wcd->rx_chs[i].ch_num = rx_slot[i]; - INIT_LIST_HEAD(&wcd->rx_chs[i].list); - } - } - - return 0; -} - -static int wcd9335_get_channel_map(struct snd_soc_dai *dai, - unsigned int *tx_num, unsigned int *tx_slot, - unsigned int *rx_num, unsigned int *rx_slot) -{ - struct wcd9335_slim_ch *ch; - struct wcd9335_codec *wcd; - int i = 0; - - wcd = snd_soc_component_get_drvdata(dai->component); - - switch (dai->id) { - case AIF1_PB: - case AIF2_PB: - case AIF3_PB: - case AIF4_PB: - if (!rx_slot || !rx_num) { - dev_err(wcd->dev, "Invalid rx_slot %p or rx_num %p\n", - rx_slot, rx_num); - return -EINVAL; - } - - list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) - rx_slot[i++] = ch->ch_num; - - *rx_num = i; - break; - default: - dev_err(wcd->dev, "Invalid DAI ID %x\n", dai->id); - break; - } - - return 0; -} - -static struct snd_soc_dai_ops wcd9335_dai_ops = { - .hw_params = wcd9335_hw_params, - .prepare = wcd9335_prepare, - .set_channel_map = wcd9335_set_channel_map, - .get_channel_map = wcd9335_get_channel_map, -}; - -static struct snd_soc_dai_driver wcd9335_slim_dais[] = { - [0] = { - .name = "wcd9335_rx1", - .id = AIF1_PB, - .playback = { - .stream_name = "AIF1 Playback", - .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, - .formats = WCD9335_FORMATS_S16_S24_LE, - .rate_max = 192000, - .rate_min = 8000, - .channels_min = 1, - .channels_max = 2, - }, - .ops = &wcd9335_dai_ops, - }, - [1] = { - .name = "wcd9335_tx1", - .id = AIF1_CAP, - .capture = { - .stream_name = "AIF1 Capture", - .rates = WCD9335_RATES_MASK, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 4, - }, - .ops = &wcd9335_dai_ops, - }, - [2] = { - .name = "wcd9335_rx2", - .id = AIF2_PB, - .playback = { - .stream_name = "AIF2 Playback", - .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, - .formats = WCD9335_FORMATS_S16_S24_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 2, - }, - .ops = &wcd9335_dai_ops, - }, - [3] = { - .name = "wcd9335_tx2", - .id = AIF2_CAP, - .capture = { - .stream_name = "AIF2 Capture", - .rates = WCD9335_RATES_MASK, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 4, - }, - .ops = &wcd9335_dai_ops, - }, - [4] = { - .name = "wcd9335_rx3", - .id = AIF3_PB, - .playback = { - .stream_name = "AIF3 Playback", - .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, - .formats = WCD9335_FORMATS_S16_S24_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 2, - }, - .ops = &wcd9335_dai_ops, - }, - [5] = { - .name = "wcd9335_tx3", - .id = AIF3_CAP, - .capture = { - .stream_name = "AIF3 Capture", - .rates = WCD9335_RATES_MASK, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 4, - }, - .ops = &wcd9335_dai_ops, - }, - [6] = { - .name = "wcd9335_rx4", - .id = AIF4_PB, - .playback = { - .stream_name = "AIF4 Playback", - .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, - .formats = WCD9335_FORMATS_S16_S24_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 2, - }, - .ops = &wcd9335_dai_ops, - }, -}; - -static irqreturn_t wcd9335_slimbus_irq(int irq, void *data) -{ - struct wcd9335_codec *wcd = data; - unsigned long status = 0; - int i, j, port_id; - unsigned int val, int_val = 0; - bool tx; - unsigned short reg = 0; - - for (i = WCD9335_SLIM_PGD_PORT_INT_STATUS_RX_0, j = 0; - i <= WCD9335_SLIM_PGD_PORT_INT_STATUS_TX_1; i++, j++) { - regmap_read(wcd->if_regmap, i, &val); - status |= ((u32)val << (8 * j)); - } - - for_each_set_bit(j, &status, 32) { - tx = (j >= 16 ? true : false); - port_id = (tx ? j - 16 : j); - regmap_read(wcd->if_regmap, - WCD9335_SLIM_PGD_PORT_INT_RX_SOURCE0 + j, &val); - if (val) { - if (!tx) - reg = WCD9335_SLIM_PGD_PORT_INT_EN0 + - (port_id / 8); - else - reg = WCD9335_SLIM_PGD_PORT_INT_TX_EN0 + - (port_id / 8); - regmap_read( - wcd->if_regmap, reg, &int_val); - /* - * Ignore interrupts for ports for which the - * interrupts are not specifically enabled. - */ - if (!(int_val & (1 << (port_id % 8)))) - continue; - } - if (val & WCD9335_SLIM_IRQ_OVERFLOW) - dev_err_ratelimited(wcd->dev, - "%s: overflow error on %s port %d, value %x\n", - __func__, (tx ? "TX" : "RX"), port_id, val); - if (val & WCD9335_SLIM_IRQ_UNDERFLOW) - dev_err_ratelimited(wcd->dev, - "%s: underflow error on %s port %d, value %x\n", - __func__, (tx ? "TX" : "RX"), port_id, val); - if ((val & WCD9335_SLIM_IRQ_OVERFLOW) || - (val & WCD9335_SLIM_IRQ_UNDERFLOW)) { - if (!tx) - reg = WCD9335_SLIM_PGD_PORT_INT_EN0 + - (port_id / 8); - else - reg = WCD9335_SLIM_PGD_PORT_INT_TX_EN0 + - (port_id / 8); - regmap_read( - wcd->if_regmap, reg, &int_val); - if (int_val & (1 << (port_id % 8))) { - int_val = int_val ^ (1 << (port_id % 8)); - regmap_write(wcd->if_regmap, - reg, int_val); - } - } - - regmap_write(wcd->if_regmap, - WCD9335_SLIM_PGD_PORT_INT_CLR_RX_0 + (j / 8), - BIT(j % 8)); - } - - return IRQ_HANDLED; -} - -static int wcd9335_setup_irqs(struct wcd9335_codec *wcd) -{ - int slim_irq = regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS); - int i, ret = 0; - - ret = request_threaded_irq(slim_irq, NULL, wcd9335_slimbus_irq, - IRQF_TRIGGER_RISING, "SLIMBus Slave", wcd); - if (ret) { - dev_err(wcd->dev, "Failed to request SLIMBUS irq\n"); - return ret; - } - - /* enable interrupts on all slave ports */ - for (i = 0; i < WCD9335_SLIM_NUM_PORT_REG; i++) - regmap_write(wcd->if_regmap, WCD9335_SLIM_PGD_PORT_INT_EN0 + i, - 0xFF); - - return ret; -} - -static void wcd9335_cdc_sido_ccl_enable(struct wcd9335_codec *wcd, - bool ccl_flag) -{ - struct snd_soc_component *comp = wcd->component; - - if (ccl_flag) { - if (++wcd->sido_ccl_cnt == 1) - snd_soc_component_write(comp, WCD9335_SIDO_SIDO_CCL_10, - WCD9335_SIDO_SIDO_CCL_DEF_VALUE); - } else { - if (wcd->sido_ccl_cnt == 0) { - dev_err(wcd->dev, "sido_ccl already disabled\n"); - return; - } - if (--wcd->sido_ccl_cnt == 0) - snd_soc_component_write(comp, WCD9335_SIDO_SIDO_CCL_10, - WCD9335_SIDO_SIDO_CCL_10_ICHARG_PWR_SEL_C320FF); - } -} - -static int wcd9335_enable_master_bias(struct wcd9335_codec *wcd) -{ - wcd->master_bias_users++; - if (wcd->master_bias_users == 1) { - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_EN_MASK, - WCD9335_ANA_BIAS_ENABLE); - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_PRECHRG_EN_MASK, - WCD9335_ANA_BIAS_PRECHRG_ENABLE); - /* - * 1ms delay is required after pre-charge is enabled - * as per HW requirement - */ - usleep_range(1000, 1100); - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_PRECHRG_EN_MASK, - WCD9335_ANA_BIAS_PRECHRG_DISABLE); - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_PRECHRG_CTL_MODE, - WCD9335_ANA_BIAS_PRECHRG_CTL_MODE_MANUAL); - } - - return 0; -} - -static int wcd9335_enable_mclk(struct wcd9335_codec *wcd) -{ - /* Enable mclk requires master bias to be enabled first */ - if (wcd->master_bias_users <= 0) - return -EINVAL; - - if (((wcd->clk_mclk_users == 0) && (wcd->clk_type == WCD_CLK_MCLK)) || - ((wcd->clk_mclk_users > 0) && (wcd->clk_type != WCD_CLK_MCLK))) { - dev_err(wcd->dev, "Error enabling MCLK, clk_type: %d\n", - wcd->clk_type); - return -EINVAL; - } - - if (++wcd->clk_mclk_users == 1) { - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_EXT_CLKBUF_EN_MASK, - WCD9335_ANA_CLK_EXT_CLKBUF_ENABLE); - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_MCLK_SRC_MASK, - WCD9335_ANA_CLK_MCLK_SRC_EXTERNAL); - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_MCLK_EN_MASK, - WCD9335_ANA_CLK_MCLK_ENABLE); - regmap_update_bits(wcd->regmap, - WCD9335_CDC_CLK_RST_CTRL_FS_CNT_CONTROL, - WCD9335_CDC_CLK_RST_CTRL_FS_CNT_EN_MASK, - WCD9335_CDC_CLK_RST_CTRL_FS_CNT_ENABLE); - regmap_update_bits(wcd->regmap, - WCD9335_CDC_CLK_RST_CTRL_MCLK_CONTROL, - WCD9335_CDC_CLK_RST_CTRL_MCLK_EN_MASK, - WCD9335_CDC_CLK_RST_CTRL_MCLK_ENABLE); - /* - * 10us sleep is required after clock is enabled - * as per HW requirement - */ - usleep_range(10, 15); - } - - wcd->clk_type = WCD_CLK_MCLK; - - return 0; -} - -static int wcd9335_disable_mclk(struct wcd9335_codec *wcd) -{ - if (wcd->clk_mclk_users <= 0) - return -EINVAL; - - if (--wcd->clk_mclk_users == 0) { - if (wcd->clk_rco_users > 0) { - /* MCLK to RCO switch */ - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_MCLK_SRC_MASK, - WCD9335_ANA_CLK_MCLK_SRC_RCO); - wcd->clk_type = WCD_CLK_RCO; - } else { - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_MCLK_EN_MASK, - WCD9335_ANA_CLK_MCLK_DISABLE); - wcd->clk_type = WCD_CLK_OFF; - } - - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_EXT_CLKBUF_EN_MASK, - WCD9335_ANA_CLK_EXT_CLKBUF_DISABLE); - } - - return 0; -} - -static int wcd9335_disable_master_bias(struct wcd9335_codec *wcd) -{ - if (wcd->master_bias_users <= 0) - return -EINVAL; - - wcd->master_bias_users--; - if (wcd->master_bias_users == 0) { - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_EN_MASK, - WCD9335_ANA_BIAS_DISABLE); - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_PRECHRG_CTL_MODE, - WCD9335_ANA_BIAS_PRECHRG_CTL_MODE_MANUAL); - } - return 0; -} - -static int wcd9335_cdc_req_mclk_enable(struct wcd9335_codec *wcd, - bool enable) -{ - int ret = 0; - - if (enable) { - wcd9335_cdc_sido_ccl_enable(wcd, true); - ret = clk_prepare_enable(wcd->mclk); - if (ret) { - dev_err(wcd->dev, "%s: ext clk enable failed\n", - __func__); - goto err; - } - /* get BG */ - wcd9335_enable_master_bias(wcd); - /* get MCLK */ - wcd9335_enable_mclk(wcd); - - } else { - /* put MCLK */ - wcd9335_disable_mclk(wcd); - /* put BG */ - wcd9335_disable_master_bias(wcd); - clk_disable_unprepare(wcd->mclk); - wcd9335_cdc_sido_ccl_enable(wcd, false); - } -err: - return ret; -} - -static void wcd9335_codec_apply_sido_voltage(struct wcd9335_codec *wcd, - enum wcd9335_sido_voltage req_mv) -{ - struct snd_soc_component *comp = wcd->component; - int vout_d_val; - - if (req_mv == wcd->sido_voltage) - return; - - /* compute the vout_d step value */ - vout_d_val = WCD9335_CALCULATE_VOUT_D(req_mv) & - WCD9335_ANA_BUCK_VOUT_MASK; - snd_soc_component_write(comp, WCD9335_ANA_BUCK_VOUT_D, vout_d_val); - snd_soc_component_update_bits(comp, WCD9335_ANA_BUCK_CTL, - WCD9335_ANA_BUCK_CTL_RAMP_START_MASK, - WCD9335_ANA_BUCK_CTL_RAMP_START_ENABLE); - - /* 1 msec sleep required after SIDO Vout_D voltage change */ - usleep_range(1000, 1100); - wcd->sido_voltage = req_mv; - snd_soc_component_update_bits(comp, WCD9335_ANA_BUCK_CTL, - WCD9335_ANA_BUCK_CTL_RAMP_START_MASK, - WCD9335_ANA_BUCK_CTL_RAMP_START_DISABLE); -} - -static int wcd9335_codec_update_sido_voltage(struct wcd9335_codec *wcd, - enum wcd9335_sido_voltage req_mv) -{ - int ret = 0; - - /* enable mclk before setting SIDO voltage */ - ret = wcd9335_cdc_req_mclk_enable(wcd, true); - if (ret) { - dev_err(wcd->dev, "Ext clk enable failed\n"); - goto err; - } - - wcd9335_codec_apply_sido_voltage(wcd, req_mv); - wcd9335_cdc_req_mclk_enable(wcd, false); - -err: - return ret; -} - -static int _wcd9335_codec_enable_mclk(struct snd_soc_component *component, - int enable) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - int ret; - - if (enable) { - ret = wcd9335_cdc_req_mclk_enable(wcd, true); - if (ret) - return ret; - - wcd9335_codec_apply_sido_voltage(wcd, - SIDO_VOLTAGE_NOMINAL_MV); - } else { - wcd9335_codec_update_sido_voltage(wcd, - wcd->sido_voltage); - wcd9335_cdc_req_mclk_enable(wcd, false); - } - - return 0; -} - -static void wcd9335_enable_sido_buck(struct snd_soc_component *component) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - - snd_soc_component_update_bits(component, WCD9335_ANA_RCO, - WCD9335_ANA_RCO_BG_EN_MASK, - WCD9335_ANA_RCO_BG_ENABLE); - snd_soc_component_update_bits(component, WCD9335_ANA_BUCK_CTL, - WCD9335_ANA_BUCK_CTL_VOUT_D_IREF_MASK, - WCD9335_ANA_BUCK_CTL_VOUT_D_IREF_EXT); - /* 100us sleep needed after IREF settings */ - usleep_range(100, 110); - snd_soc_component_update_bits(component, WCD9335_ANA_BUCK_CTL, - WCD9335_ANA_BUCK_CTL_VOUT_D_VREF_MASK, - WCD9335_ANA_BUCK_CTL_VOUT_D_VREF_EXT); - /* 100us sleep needed after VREF settings */ - usleep_range(100, 110); - wcd->sido_input_src = SIDO_SOURCE_RCO_BG; -} - -static int wcd9335_enable_efuse_sensing(struct snd_soc_component *comp) -{ - _wcd9335_codec_enable_mclk(comp, true); - snd_soc_component_update_bits(comp, - WCD9335_CHIP_TIER_CTRL_EFUSE_CTL, - WCD9335_CHIP_TIER_CTRL_EFUSE_EN_MASK, - WCD9335_CHIP_TIER_CTRL_EFUSE_ENABLE); - /* - * 5ms sleep required after enabling efuse control - * before checking the status. - */ - usleep_range(5000, 5500); - - if (!(snd_soc_component_read32(comp, - WCD9335_CHIP_TIER_CTRL_EFUSE_STATUS) & - WCD9335_CHIP_TIER_CTRL_EFUSE_EN_MASK)) - WARN(1, "%s: Efuse sense is not complete\n", __func__); - - wcd9335_enable_sido_buck(comp); - _wcd9335_codec_enable_mclk(comp, false); - - return 0; -} - -static void wcd9335_codec_init(struct snd_soc_component *component) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - int i; - - /* ungate MCLK and set clk rate */ - regmap_update_bits(wcd->regmap, WCD9335_CODEC_RPM_CLK_GATE, - WCD9335_CODEC_RPM_CLK_GATE_MCLK_GATE_MASK, 0); - - regmap_update_bits(wcd->regmap, WCD9335_CODEC_RPM_CLK_MCLK_CFG, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_9P6MHZ); - - for (i = 0; i < ARRAY_SIZE(wcd9335_codec_reg_init_common_val); i++) - snd_soc_component_update_bits(component, - wcd9335_codec_reg_init_common_val[i].reg, - wcd9335_codec_reg_init_common_val[i].mask, - wcd9335_codec_reg_init_common_val[i].val); - - if (WCD9335_IS_2_0(wcd->version)) { - for (i = 0; i < ARRAY_SIZE(wcd9335_codec_reg_init_val_2_0); i++) - snd_soc_component_update_bits(component, - wcd9335_codec_reg_init_val_2_0[i].reg, - wcd9335_codec_reg_init_val_2_0[i].mask, - wcd9335_codec_reg_init_val_2_0[i].val); - } - - wcd9335_enable_efuse_sensing(component); -} - -static int wcd9335_codec_probe(struct snd_soc_component *component) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - int i; - - snd_soc_component_init_regmap(component, wcd->regmap); - wcd->component = component; - - wcd9335_codec_init(component); - - for (i = 0; i < NUM_CODEC_DAIS; i++) - INIT_LIST_HEAD(&wcd->dai[i].slim_ch_list); - - return wcd9335_setup_irqs(wcd); -} - -static void wcd9335_codec_remove(struct snd_soc_component *comp) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - - free_irq(regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS), wcd); -} - -static int wcd9335_codec_set_sysclk(struct snd_soc_component *comp, - int clk_id, int source, - unsigned int freq, int dir) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - - wcd->mclk_rate = freq; - - if (wcd->mclk_rate == WCD9335_MCLK_CLK_12P288MHZ) - snd_soc_component_update_bits(comp, - WCD9335_CODEC_RPM_CLK_MCLK_CFG, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_12P288MHZ); - else if (wcd->mclk_rate == WCD9335_MCLK_CLK_9P6MHZ) - snd_soc_component_update_bits(comp, - WCD9335_CODEC_RPM_CLK_MCLK_CFG, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_9P6MHZ); - - return clk_set_rate(wcd->mclk, freq); -} - -static const struct snd_soc_component_driver wcd9335_component_drv = { - .probe = wcd9335_codec_probe, - .remove = wcd9335_codec_remove, - .set_sysclk = wcd9335_codec_set_sysclk, -}; - -static int wcd9335_probe(struct platform_device *pdev) -{ - struct wcd9335 *pdata = dev_get_drvdata(pdev->dev.parent); - struct device *dev = &pdev->dev; - struct wcd9335_codec *wcd; - - wcd = devm_kzalloc(dev, sizeof(*wcd), GFP_KERNEL); - if (!wcd) - return -ENOMEM; - - dev_set_drvdata(dev, wcd); - - memcpy(wcd->rx_chs, wcd9335_rx_chs, sizeof(wcd9335_rx_chs)); - - wcd->regmap = pdata->regmap; - wcd->if_regmap = pdata->ifd_regmap; - wcd->slim = pdata->slim; - wcd->slim_ifd = pdata->slim_ifd; - wcd->irq_data = pdata->irq_data; - wcd->version = pdata->version; - wcd->intf_type = pdata->intf_type; - wcd->dev = dev; - wcd->mclk = pdata->mclk; - wcd->native_clk = pdata->native_clk; - wcd->sido_input_src = SIDO_SOURCE_INTERNAL; - wcd->sido_voltage = SIDO_VOLTAGE_NOMINAL_MV; - - return devm_snd_soc_register_component(dev, &wcd9335_component_drv, - wcd9335_slim_dais, - ARRAY_SIZE(wcd9335_slim_dais)); -} - -static struct platform_driver wcd9335_codec_driver = { - .probe = wcd9335_probe, - .driver = { - .name = "wcd9335-codec", - }, -}; -module_platform_driver(wcd9335_codec_driver); -MODULE_DESCRIPTION("WCD9335 Codec driver"); -MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 9fb4c2bf130b922c77c16a8368732699799c40de Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Wed, 1 Aug 2018 15:37:33 +0530 Subject: ASoC: soc-pcm: Use delay set in component pointer function Take into account the base delay set in pointer callback. There are cases where a pointer function populates runtime->delay, such as: ./sound/pci/hda/hda_controller.c ./sound/soc/intel/atom/sst-mfld-platform-pcm.c This delay was getting lost and was overwritten by delays from codec or cpu dai delay function if exposed. Now, Total delay = base delay + cpu_dai delay + codec_dai delay Signed-off-by: Akshu Agrawal Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4019bc10897c..9833e53754cb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1179,6 +1179,9 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) snd_pcm_sframes_t codec_delay = 0; int i; + /* clearing the previous total delay */ + runtime->delay = 0; + for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -1190,6 +1193,8 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) offset = component->driver->ops->pointer(substream); break; } + /* base delay if assigned in pointer callback */ + delay = runtime->delay; if (cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); -- cgit v1.2.3 From 9a73f6a235c2351adc07471519d4980e2fb33bbc Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 1 Aug 2018 14:47:09 -0500 Subject: ASoC: wm8961: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1271173 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index f70f563d59f3..68b4cadc308f 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -653,6 +653,7 @@ static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_B: aif |= WM8961_LRP; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif |= 3; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { -- cgit v1.2.3 From 065dcc270af66acb4226b49f635b41a054b663d4 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 1 Aug 2018 14:49:13 -0500 Subject: ASoC: rt5640: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1056547 ("Missing break in switch") Addresses-Coverity-ID: 1056548 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 8bf8d360c25f..27770143ae8f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1665,6 +1665,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id) break; case RT5640_IF_113: ret |= RT5640_U_IF1; + /* fall through */ case RT5640_IF_312: case RT5640_IF_213: ret |= RT5640_U_IF2; @@ -1680,6 +1681,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id) break; case RT5640_IF_223: ret |= RT5640_U_IF1; + /* fall through */ case RT5640_IF_123: case RT5640_IF_321: ret |= RT5640_U_IF2; -- cgit v1.2.3 From 43a26bd026dab09a8b28c40e94ba534a52375b20 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 1 Aug 2018 14:50:20 -0500 Subject: ASoC: rt5677: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1271174 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8a0181a2db08..922becfee59b 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4417,6 +4417,7 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, break; case 25: slot_width_25 = 0x8080; + /* fall through */ case 24: val |= (2 << 8); break; -- cgit v1.2.3 From 85e7e77079f3201799467803f6c8533a9921e32d Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 1 Aug 2018 14:51:18 -0500 Subject: ASoC: wm8955: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115047 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8955.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index ba44e3d6c1e0..cd204f79647d 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -686,6 +686,7 @@ static int wm8955_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif |= WM8955_LRP; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif |= 0x3; break; -- cgit v1.2.3 From 3eb7dbc6d844d2033271434d02ff90ac9b19e393 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 1 Aug 2018 14:51:59 -0500 Subject: ASoC: wm8960: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115041 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index c30f5aa392c6..8dc1f3d6a988 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -839,6 +839,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, iface |= 0x000c; break; } + /* fall through */ default: dev_err(component->dev, "unsupported width %d\n", params_width(params)); -- cgit v1.2.3 From da41787b9f319a7a23fcc8cf65c6ad646803962e Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 1 Aug 2018 14:52:41 -0500 Subject: ASoC: wm8904: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115042 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9037a35b931d..1965635ec07c 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1455,6 +1455,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= 0x3 | WM8904_AIF_LRCLK_INV; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x3; break; -- cgit v1.2.3 From 42ef3c94ff6e8315377c04504f1bce50d1f21738 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 1 Aug 2018 14:53:19 -0500 Subject: ASoC: wm8996: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 146354 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index d9d206046f8c..78a408236cfb 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1858,6 +1858,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, case 24576000: ratediv = WM8996_SYSCLK_DIV; wm8996->sysclk /= 2; + /* fall through */ case 11289600: case 12288000: snd_soc_component_update_bits(component, WM8996_AIF_RATE, -- cgit v1.2.3 From a9531ab151117e976544e0dc74a9bc8cbd01145f Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 1 Aug 2018 14:54:03 -0500 Subject: ASoC: wm8962: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115043 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a11e9d6bf950..efd8910b1ff7 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2649,6 +2649,7 @@ static int wm8962_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif0 |= WM8962_LRCLK_INV | 3; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif0 |= 3; -- cgit v1.2.3 From af5d1d5d4ba71164711025e077511ad48d5690e9 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 1 Aug 2018 14:54:45 -0500 Subject: ASoC: wm8995: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115045 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 60e227832331..68c99fe37097 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1465,6 +1465,7 @@ static int wm8995_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif |= WM8995_AIF1_LRCLK_INV; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif |= (0x3 << WM8995_AIF1_FMT_SHIFT); break; -- cgit v1.2.3 From 7a2235ef507822679ac89007a6a37b23f30eed56 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 1 Aug 2018 14:55:26 -0500 Subject: ASoC: wm9081: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1357430 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 5a0ea7b3c149..399255d1f78a 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -933,6 +933,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif2 |= WM9081_AIF_LRCLK_INV; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: aif2 |= 0x3; break; -- cgit v1.2.3 From 2cea1542859bc812f1ec51ea71c06e927e5b922e Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 1 Aug 2018 14:56:16 -0500 Subject: ASoC: wm8994: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115050 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7fdfdf3f6e67..62f8c5b9ba92 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2432,6 +2432,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, snd_soc_component_update_bits(component, WM8994_POWER_MANAGEMENT_2, WM8994_OPCLK_ENA, 0); } + /* fall through */ default: return -EINVAL; -- cgit v1.2.3 From 73edbe42582207ff8f47168f05124376394aa643 Mon Sep 17 00:00:00 2001 From: Rohit kumar Date: Thu, 2 Aug 2018 12:32:59 +0530 Subject: ASoC: qcom: Fix unmet dependency warning for SND_SOC_SDM845 Add DEPENDS_ON QCOM_APR for SND_SOC_SDM845 to fix the warning: unmet direct dependencies detected for SND_SOC_QDSP6. Reported-by: Stephen Rothwell Signed-off-by: Rohit kumar Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 350730839c6f..5f03312ef007 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -93,6 +93,7 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" + depends on QCOM_APR select SND_SOC_QDSP6 help To add support for audio on Qualcomm Technologies Inc. -- cgit v1.2.3 From 1eb576881ff884dd6d10272b96cc336d156492c2 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 2 Aug 2018 16:03:36 +0100 Subject: ASoC: apq8096: remove auto rebinding Remove auto rebinding support, as component framework can deadlock in few usecases if we are trying to add new/remove component within a bind/unbind callbacks. Card rebinding is ASoC core feature so all the previous component framework stuff in q6dsp remains removed. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/apq8096.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 1e4a90d59228..6ee7e66cbaaa 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -48,7 +48,6 @@ static int apq8096_platform_probe(struct platform_device *pdev) return -ENOMEM; card->dev = dev; - card->auto_bind = true; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); if (ret) { @@ -74,7 +73,6 @@ static int apq8096_platform_remove(struct platform_device *pdev) { struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); - card->auto_bind = false; snd_soc_unregister_card(card); kfree(card->dai_link); kfree(card); -- cgit v1.2.3 From 62121debfb31a8700e387bd2987779b3a98bc520 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 2 Aug 2018 16:03:37 +0100 Subject: ASoC: smd845: remove auto rebinding Remove auto rebinding support, as component framework can deadlock in few usecases if we are trying to add new/remove component within a bind/unbind callbacks. Card rebinding is ASoC core feature so all the previous component framework stuff in q6dsp remains removed. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/sdm845.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index bf4ec4646906..be0cb1122036 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -226,7 +226,6 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) return -ENOMEM; card->dev = dev; - card->auto_bind = true; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); if (ret) { @@ -258,7 +257,6 @@ static int sdm845_snd_platform_remove(struct platform_device *pdev) struct snd_soc_card *card = dev_get_drvdata(&pdev->dev); struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); - card->auto_bind = false; snd_soc_unregister_card(card); kfree(card->dai_link); kfree(data); -- cgit v1.2.3 From 611cbc8799b672f41b6ba7afed758ad9efb959a7 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 2 Aug 2018 16:03:38 +0100 Subject: ASoC: core: remove support for card rebind using component framework DRM based audio components get registered inside the component framework bind callback. However component framework has a big mutex lock taken for every call to component_add, component_del and bind, unbind callbacks. This can lead to deadlock situation if we are trying to add new/remove component within a bind/unbind callbacks. Which is what was happening with bcm2837 rpi 3. Revert this change till we sort out the mutex issue. Reported-by: Guillaume Tucker Reported-by: Stefan Wahren Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- include/sound/soc.h | 7 ------ sound/soc/soc-core.c | 62 ---------------------------------------------------- 2 files changed, 69 deletions(-) (limited to 'sound/soc') diff --git a/include/sound/soc.h b/include/sound/soc.h index ace474e8649e..41cec42fb456 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -17,7 +17,6 @@ #include #include #include -#include #include #include #include @@ -1091,12 +1090,6 @@ struct snd_soc_card { struct work_struct deferred_resume_work; - /* component framework related */ - bool components_added; - /* set in machine driver to enable/disable auto re-binding */ - bool auto_bind; - struct component_match *match; - /* lists of probed devices belonging to this card */ struct list_head component_dev_list; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 81b27923303d..82eb3868be67 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -279,28 +279,11 @@ static inline void snd_soc_debugfs_exit(void) #endif -static int snd_soc_card_comp_compare(struct device *dev, void *data) -{ - struct snd_soc_component *component; - - lockdep_assert_held(&client_mutex); - list_for_each_entry(component, &component_list, list) { - if (dev == component->dev) { - if (!strcmp(component->name, data)) - return 1; - break; - } - } - - return 0; -} - static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd, struct snd_soc_component *component) { struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_rtdcom_list *new_rtdcom; - char *cname; for_each_rtdcom(rtd, rtdcom) { /* already connected */ @@ -317,13 +300,6 @@ static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd, list_add_tail(&new_rtdcom->list, &rtd->component_list); - if (rtd->card->auto_bind && !rtd->card->components_added) { - cname = devm_kasprintf(rtd->card->dev, GFP_KERNEL, - "%s", component->name); - component_match_add(rtd->card->dev, &rtd->card->match, - snd_soc_card_comp_compare, cname); - } - return 0; } @@ -859,25 +835,6 @@ static bool soc_is_dai_link_bound(struct snd_soc_card *card, return false; } -static int snd_soc_card_comp_bind(struct device *dev) -{ - struct snd_soc_card *card = dev_get_drvdata(dev); - - if (card->instantiated) - return 0; - - return snd_soc_register_card(card); -} - -static void snd_soc_card_comp_unbind(struct device *dev) -{ -} - -static const struct component_master_ops snd_soc_card_comp_ops = { - .bind = snd_soc_card_comp_bind, - .unbind = snd_soc_card_comp_unbind, -}; - static int soc_bind_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -2169,12 +2126,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->instantiated = 1; snd_soc_dapm_sync(&card->dapm); - if (card->auto_bind && !card->components_added) { - component_master_add_with_match(card->dev, - &snd_soc_card_comp_ops, - card->match); - card->components_added = true; - } mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); @@ -2820,9 +2771,6 @@ int snd_soc_unregister_card(struct snd_soc_card *card) dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); } - if (!card->auto_bind && card->components_added) - component_master_del(card->dev, &snd_soc_card_comp_ops); - return 0; } EXPORT_SYMBOL_GPL(snd_soc_unregister_card); @@ -3235,17 +3183,8 @@ int snd_soc_add_component(struct device *dev, snd_soc_component_add(component); - ret = component_add(dev, NULL); - if (ret < 0) { - dev_err(dev, "ASoC: Failed to add Component: %d\n", ret); - goto err_comp; - } - return 0; -err_comp: - soc_remove_component(component); - snd_soc_unregister_dais(component); err_cleanup: snd_soc_component_cleanup(component); err_free: @@ -3293,7 +3232,6 @@ static int __snd_soc_unregister_component(struct device *dev) mutex_unlock(&client_mutex); if (found) { - component_del(dev, NULL); snd_soc_component_cleanup(component); } -- cgit v1.2.3 From 26a6dce8ef99b0199e7682d1c203cf7d0b5fd5b0 Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Thu, 2 Aug 2018 18:21:31 -0500 Subject: ASoC: Intel: bxt: Use refcap device for mono recording The refcap capture device supports mono recording only, this patch adds the channel constraints. Signed-off-by: Yong Zhi Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index be6e4b40bf03..6f052fc8d1e2 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -324,8 +324,22 @@ static const struct snd_pcm_hw_constraint_list constraints_16000 = { .list = rates_16000, }; +static const unsigned int ch_mono[] = { + 1, +}; + +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + static int broxton_refcap_startup(struct snd_pcm_substream *substream) { + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_16000); -- cgit v1.2.3 From 8530ebf1079ccc84ffa32d970cdcae168b2f3684 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Fri, 3 Aug 2018 08:42:11 -0500 Subject: ASoC: smd845: fix memory leak In case memory resources for *card* were allocated, release them before return. Addresses-Coverity-ID: 1472244 ("Resource leak") Fixes: 6b1687bf76ef ("ASoC: qcom: add sdm845 sound card support") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/qcom/sdm845.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index be0cb1122036..c1adb77230eb 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -222,8 +222,10 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) /* Allocate the private data */ data = kzalloc(sizeof(*data), GFP_KERNEL); - if (!data) - return -ENOMEM; + if (!data) { + ret = -ENOMEM; + goto data_alloc_fail; + } card->dev = dev; dev_set_drvdata(dev, card); @@ -248,6 +250,7 @@ register_card_fail: kfree(card->dai_link); parse_dt_fail: kfree(data); +data_alloc_fail: kfree(card); return ret; } -- cgit v1.2.3 From 3b7c88fcc2873eba64f9e8cef34c4af0cba20887 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Fri, 3 Aug 2018 11:28:24 -0500 Subject: ASoC: davinci-i2s: mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1364478 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 807040bb3921..a3206e65e5e5 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -340,6 +340,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, * rate is lowered. */ inv_fs = true; + /* fall through */ case SND_SOC_DAIFMT_DSP_A: dev->mode = MOD_DSP_A; break; -- cgit v1.2.3 From 85c81941d5033fc8a68646823d3157840c9de8b3 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Fri, 3 Aug 2018 11:32:52 -0500 Subject: ASoC: omap-mcpdm: Mark expected switch fall-throughs In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1369526 ("Missing break in switch") Addresses-Coverity-ID: 1369529 ("Missing break in switch") Addresses-Coverity-ID: 1451415 ("Missing break in switch") Addresses-Coverity-ID: 115103 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 0e97360f9890..4c1be36c2207 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -310,15 +310,19 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 4; + /* fall through */ case 4: if (stream == SNDRV_PCM_STREAM_CAPTURE) /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 3; + /* fall through */ case 3: link_mask |= 1 << 2; + /* fall through */ case 2: link_mask |= 1 << 1; + /* fall through */ case 1: link_mask |= 1 << 0; break; -- cgit v1.2.3 From 1a12d5dc7dd1ffd985503f9770b736fb03db2e3f Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Fri, 3 Aug 2018 11:34:30 -0500 Subject: ASoC: core: mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 146568 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 82eb3868be67..9cfe10d8040c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -528,6 +528,7 @@ int snd_soc_suspend(struct device *dev) "ASoC: idle_bias_off CODEC on over suspend\n"); break; } + /* fall through */ case SND_SOC_BIAS_OFF: if (component->driver->suspend) -- cgit v1.2.3 From 16bbeb2b43c3f5d69e1348477e75a24ae6d55d5a Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Fri, 3 Aug 2018 11:29:53 -0500 Subject: ASoC: fsl_esai: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1222121 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 8f43110373b8..c1d1d06783e5 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -249,6 +249,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, break; case ESAI_HCKT_EXTAL: ecr |= ESAI_ECR_ETI; + /* fall through */ case ESAI_HCKR_EXTAL: ecr |= ESAI_ECR_ERI; break; -- cgit v1.2.3 From a773c3b6be185d171e2755ac715e8b1ea099ebbc Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Fri, 3 Aug 2018 11:31:48 -0500 Subject: ASoC: omap-dmic: Mark expected switch fall-throughs In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1468847 ("Missing break in switch") Addresses-Coverity-ID: 1468849 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/omap/omap-dmic.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 51dd7c65096b..fe966272bd0c 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -213,8 +213,10 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream, switch (channels) { case 6: dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE; + /* fall through */ case 4: dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE; + /* fall through */ case 2: dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE; break; -- cgit v1.2.3 From 5019027a566de4986a7f66017cf0d6d794fc155f Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Fri, 3 Aug 2018 11:33:57 -0500 Subject: ASoC: samsung: i2s: Mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1381093 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index f914ed45db7d..d6c62aa13041 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -710,6 +710,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, switch (params_channels(params)) { case 6: val |= MOD_DC2_EN; + /* fall through */ case 4: val |= MOD_DC1_EN; break; -- cgit v1.2.3 From 038541dae968dbef99edca4b22bbb2a7b4afdc83 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Sat, 4 Aug 2018 16:49:02 -0500 Subject: ASoC: max9850: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/max9850.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 74d7f52c7e73..6e6134589588 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -52,9 +52,9 @@ static bool max9850_volatile_register(struct device *dev, unsigned int reg) switch (reg) { case MAX9850_STATUSA: case MAX9850_STATUSB: - return 1; + return true; default: - return 0; + return false; } } -- cgit v1.2.3 From 508e8641f89cc89554adbec3ce59bcd28a4ced4b Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Sat, 4 Aug 2018 16:49:55 -0500 Subject: ASoC: rt5631: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index e52e4670cf65..865f49ac38dd 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -105,9 +105,9 @@ static bool rt5631_volatile_register(struct device *dev, unsigned int reg) case RT5631_INDEX_ADD: case RT5631_INDEX_DATA: case RT5631_EQ_CTRL: - return 1; + return true; default: - return 0; + return false; } } @@ -164,9 +164,9 @@ static bool rt5631_readable_register(struct device *dev, unsigned int reg) case RT5631_VENDOR_ID: case RT5631_VENDOR_ID1: case RT5631_VENDOR_ID2: - return 1; + return true; default: - return 0; + return false; } } -- cgit v1.2.3 From 10754bfc051256ab922b818de3ddd11391862a5f Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Sat, 4 Aug 2018 16:50:27 -0500 Subject: ASoC: tda7419: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/tda7419.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tda7419.c b/sound/soc/codecs/tda7419.c index 225c210ac38f..7f3b79c5a563 100644 --- a/sound/soc/codecs/tda7419.c +++ b/sound/soc/codecs/tda7419.c @@ -142,9 +142,9 @@ struct tda7419_vol_control { static inline bool tda7419_vol_is_stereo(struct tda7419_vol_control *tvc) { if (tvc->reg == tvc->rreg) - return 0; + return false; - return 1; + return true; } static int tda7419_vol_info(struct snd_kcontrol *kcontrol, -- cgit v1.2.3 From 064ee5a370150534e12e5bac4ff81f86528e27ca Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Sat, 4 Aug 2018 16:51:01 -0500 Subject: ASoC: wm8990: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 411b9eee88c2..457bc437ce54 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -40,9 +40,9 @@ static bool wm8990_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8990_RESET: - return 1; + return true; default: - return 0; + return false; } } -- cgit v1.2.3 From c34c4515286f27e7e99d8a0011b4322d1de6c9bc Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Sat, 4 Aug 2018 16:52:01 -0500 Subject: ASoC: cs4270: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 2a7a4168c072..3c266eeb89bf 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -219,7 +219,7 @@ static bool cs4270_reg_is_volatile(struct device *dev, unsigned int reg) { /* Unreadable registers are considered volatile */ if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) - return 1; + return true; return reg == CS4270_CHIPID; } -- cgit v1.2.3 From eb086306bc6b42dc2c538ec2350d44a82d1a835b Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Sat, 4 Aug 2018 16:52:35 -0500 Subject: ASoC: wm8996: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 78a408236cfb..91711f8958c5 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1498,9 +1498,9 @@ static bool wm8996_readable_register(struct device *dev, unsigned int reg) case WM8996_RIGHT_PDM_SPEAKER: case WM8996_PDM_SPEAKER_MUTE_SEQUENCE: case WM8996_PDM_SPEAKER_VOLUME: - return 1; + return true; default: - return 0; + return false; } } @@ -1522,9 +1522,9 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) case WM8996_MIC_DETECT_3: case WM8996_HEADPHONE_DETECT_1: case WM8996_HEADPHONE_DETECT_2: - return 1; + return true; default: - return 0; + return false; } } -- cgit v1.2.3 From 965afd3c1dbaf293bfa9452aaf0a7e53d78d07f3 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Sat, 4 Aug 2018 16:53:10 -0500 Subject: ASoC: da7219: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/da7219.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8bf56d..c0144f2f8174 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -2143,9 +2143,9 @@ static bool da7219_volatile_register(struct device *dev, unsigned int reg) case DA7219_ACCDET_IRQ_EVENT_B: case DA7219_ACCDET_CONFIG_8: case DA7219_SYSTEM_STATUS: - return 1; + return true; default: - return 0; + return false; } } -- cgit v1.2.3 From bc94c8884e5a94914ec950dfc615563a1253e3f3 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Sat, 4 Aug 2018 16:53:38 -0500 Subject: ASoC: twl6040: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index bfd1abd72253..94675da514c8 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -148,7 +148,7 @@ static bool twl6040_can_write_to_chip(struct snd_soc_component *component, case TWL6040_REG_HFRCTL: return priv->dl2_unmuted; default: - return 1; + return true; } } -- cgit v1.2.3 From 1752a35acd8e837463fb7b7d9492b18e2a2ed5e3 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Sat, 4 Aug 2018 16:54:10 -0500 Subject: ASoC: da7213: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/da7213.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 54cb5f24969f..92d006a5283e 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1140,9 +1140,9 @@ static bool da7213_volatile_register(struct device *dev, unsigned int reg) case DA7213_ALC_OFFSET_AUTO_M_R: case DA7213_ALC_OFFSET_AUTO_U_R: case DA7213_ALC_CIC_OP_LVL_DATA: - return 1; + return true; default: - return 0; + return false; } } -- cgit v1.2.3 From 380ae4ec420304aa59adb36776eaf78aef37192c Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Sat, 4 Aug 2018 16:54:54 -0500 Subject: ASoC: wm5100-tables: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100-tables.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index e239f4bf2460..9e987cf07450 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -30,7 +30,7 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) case WM5100_OUTPUT_STATUS_2: case WM5100_INPUT_ENABLES_STATUS: case WM5100_MIC_DETECT_3: - return 1; + return true; default: if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || @@ -41,9 +41,9 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) - return 1; + return true; else - return 0; + return false; } } @@ -798,7 +798,7 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) case WM5100_DSP3_CONTROL_28: case WM5100_DSP3_CONTROL_29: case WM5100_DSP3_CONTROL_30: - return 1; + return true; default: if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || @@ -809,9 +809,9 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) - return 1; + return true; else - return 0; + return false; } } -- cgit v1.2.3 From e1ec62b147c2f2deae66e69ee4f7347dc80123ae Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Sat, 4 Aug 2018 16:55:28 -0500 Subject: ASoC: da9055: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Signed-off-by: Mark Brown --- sound/soc/codecs/da9055.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index afdf90c78884..f6a7bf9560e7 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1041,9 +1041,9 @@ static bool da9055_volatile_register(struct device *dev, case DA9055_HP_R_GAIN_STATUS: case DA9055_LINE_GAIN_STATUS: case DA9055_ALC_CIC_OP_LVL_DATA: - return 1; + return true; default: - return 0; + return false; } } -- cgit v1.2.3 From bee7d3c9f89a1bbe3d5f88ac9d6946de4705731b Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Sat, 4 Aug 2018 16:56:02 -0500 Subject: ASoC: wm8903: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 7b8b6ef2f632..6cb3c153ba19 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -251,10 +251,10 @@ static bool wm8903_volatile_register(struct device *dev, unsigned int reg) case WM8903_DC_SERVO_READBACK_2: case WM8903_DC_SERVO_READBACK_3: case WM8903_DC_SERVO_READBACK_4: - return 1; + return true; default: - return 0; + return false; } } -- cgit v1.2.3 From 8e3684f66e15c354dba5544c2af8ce973ed40cf6 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 6 Aug 2018 11:12:05 +0100 Subject: ASoC: qcom: make common.c as proper module This patch converts common helper functions in to proper module and also fixes below warning. WARNING: sound/soc/qcom/snd-soc-sdm845: 'qcom_snd_parse_of' exported twice. Previous export was in sound/soc/qcom/snd-soc-apq8096.ko Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 5 +++++ sound/soc/qcom/Makefile | 6 ++++-- 2 files changed, 9 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 5f03312ef007..2a4c912d1e48 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -41,6 +41,9 @@ config SND_SOC_APQ8016_SBC APQ8016 SOC-based systems. Say Y if you want to use audio devices on MI2S. +config SND_SOC_QCOM_COMMON + tristate + config SND_SOC_QDSP6_COMMON tristate @@ -86,6 +89,7 @@ config SND_SOC_MSM8996 tristate "SoC Machine driver for MSM8996 and APQ8096 boards" depends on QCOM_APR select SND_SOC_QDSP6 + select SND_SOC_QCOM_COMMON help Support for Qualcomm Technologies LPASS audio block in APQ8096 SoC-based systems. @@ -95,6 +99,7 @@ config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" depends on QCOM_APR select SND_SOC_QDSP6 + select SND_SOC_QCOM_COMMON help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index f0e94d48ba98..41b2c7a23a4d 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -13,13 +13,15 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o # Machine snd-soc-storm-objs := storm.o snd-soc-apq8016-sbc-objs := apq8016_sbc.o -snd-soc-apq8096-objs := apq8096.o common.o -snd-soc-sdm845-objs := sdm845.o common.o +snd-soc-apq8096-objs := apq8096.o +snd-soc-sdm845-objs := sdm845.o +snd-soc-qcom-common-objs := common.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o +obj-$(CONFIG_SND_SOC_QCOM_COMMON) += snd-soc-qcom-common.o #DSP lib obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/ -- cgit v1.2.3 From e9d244b14dd5cf9a78ab256a4463d946e580ca63 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 6 Aug 2018 11:12:06 +0100 Subject: ASoC: apq8096: remove unused header files This patch removes unused header files from the driver. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/apq8096.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 6ee7e66cbaaa..1543e85629f8 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -1,9 +1,7 @@ // SPDX-License-Identifier: GPL-2.0 // Copyright (c) 2018, Linaro Limited -#include #include -#include #include #include #include -- cgit v1.2.3 From 846b2c96808cc7cdf4e0619d00604b3edd15b35a Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 6 Aug 2018 11:12:07 +0100 Subject: ASoC: sdm845: remove unused header files This patch removes unused header files from the driver. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/sdm845.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index c1adb77230eb..2a781d87ee65 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -5,11 +5,9 @@ #include #include -#include #include #include #include -#include #include "common.h" #include "qdsp6/q6afe.h" -- cgit v1.2.3 From d72117d0c89a3f5657ef91d5eef31337964e3cb2 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 6 Aug 2018 11:12:08 +0100 Subject: ASoC: qcom: remove unused header files from common.h This patch removes unused header files from common.h. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/common.h | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/common.h b/sound/soc/qcom/common.h index ad5d2cf27459..f05c05b12bd7 100644 --- a/sound/soc/qcom/common.h +++ b/sound/soc/qcom/common.h @@ -4,7 +4,6 @@ #ifndef __QCOM_SND_COMMON_H__ #define __QCOM_SND_COMMON_H__ -#include #include int qcom_snd_parse_of(struct snd_soc_card *card); -- cgit v1.2.3 From 0961503412e3e13d82f885f9fb3af527edd58e47 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 6 Aug 2018 11:12:09 +0100 Subject: ASoC: qdsp6: q6afe-dai: add SLIM tx AIF_IN dapm Add missing AIF_IN dapm for slim tx ports. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index e988692a3ced..139d24be3fb9 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1117,6 +1117,13 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_RX", "Slimbus5 Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_RX", "Slimbus6 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_0_TX", "Slimbus Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_1_TX", "Slimbus1 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_2_TX", "Slimbus2 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_3_TX", "Slimbus3 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_4_TX", "Slimbus4 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_5_TX", "Slimbus5 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_6_TX", "Slimbus6 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_RX", "Quaternary MI2S Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_MI2S_TX", "Quaternary MI2S Capture", -- cgit v1.2.3 From ad0eaee6195db1db1749dd46b9e6f4466793d178 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Mon, 6 Aug 2018 07:14:51 -0500 Subject: ASoC: wm8994: Fix missing break in switch Add missing break statement in order to prevent the code from falling through to the default case. Addresses-Coverity-ID: 115050 ("Missing break in switch") Reported-by: Valdis Kletnieks Signed-off-by: Gustavo A. R. Silva Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 62f8c5b9ba92..14f1b0c0d286 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2432,7 +2432,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, snd_soc_component_update_bits(component, WM8994_POWER_MANAGEMENT_2, WM8994_OPCLK_ENA, 0); } - /* fall through */ + break; default: return -EINVAL; -- cgit v1.2.3 From bbdb7012b0736cda0b9b00a2949e9207cf2f892f Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Mon, 6 Aug 2018 12:57:14 +0530 Subject: ASoC: AMD: Make ACP->SYSMEM DMA non circular In capture case we don't want ACP to SYSMEM dma to be circular. This is because if an in place DSP filter is applied to captured output then circular DMA can overwrite the filter value with stale data. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 36 +++++++++++++++++++++++++++++++----- 1 file changed, 31 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 94bcf69008df..816abd65a6ed 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -400,7 +400,7 @@ static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, } /* Start a given DMA channel transfer */ -static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) +static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num, bool is_circular) { u32 dma_ctrl; @@ -429,8 +429,11 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) break; } - /* circular for both DMA channel */ - dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + /* enable for ACP to SRAM DMA channel */ + if (is_circular == true) + dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + else + dma_ctrl &= ~ACP_DMA_CNTL_0__Circular_DMA_En_MASK; acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num); } @@ -674,6 +677,7 @@ static int acp_deinit(void __iomem *acp_mmio) /* ACP DMA irq handler routine for playback, capture usecases */ static irqreturn_t dma_irq_handler(int irq, void *arg) { + u16 dscr_idx; u32 intr_flag, ext_intr_status; struct audio_drv_data *irq_data; void __iomem *acp_mmio; @@ -705,6 +709,15 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_14) == + CAPTURE_START_DMA_DESCR_CH15) + dscr_idx = CAPTURE_END_DMA_DESCR_CH14; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH14; + config_acp_dma_channel(acp_mmio, ACP_TO_SYSRAM_CH_NUM, dscr_idx, + 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false); + snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); @@ -712,6 +725,17 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_10) == + CAPTURE_START_DMA_DESCR_CH11) + dscr_idx = CAPTURE_END_DMA_DESCR_CH10; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH10; + config_acp_dma_channel(acp_mmio, + ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + dscr_idx, 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + false); + snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, @@ -1053,9 +1077,11 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_cap_channel_enable(rtd->acp_mmio, CAP_CHANNEL1); } + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + } else { + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + acp_dma_start(rtd->acp_mmio, rtd->ch2, true); } - acp_dma_start(rtd->acp_mmio, rtd->ch1); - acp_dma_start(rtd->acp_mmio, rtd->ch2); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: -- cgit v1.2.3 From 662fb3efe7ee835f0eeba6bc63b81e82a97fc312 Mon Sep 17 00:00:00 2001 From: Mukunda, Vijendar Date: Mon, 6 Aug 2018 12:57:15 +0530 Subject: ASoC: AMD: Modified DMA pointer for capture Give position on ACP->SYSMEM DMA channel for the number of bytes that have been transferred on the basis of current descriptor under service. Signed-off-by: Vijendar Mukunda Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 31 ++++++++++++++++++------------- sound/soc/amd/acp.h | 1 + 2 files changed, 19 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 816abd65a6ed..32f27c5e4d93 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -922,10 +922,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_BLUETOOTH; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH11; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_11; adata->capture_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: @@ -945,10 +942,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_ACP_I2S_1; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_15; adata->capture_i2ssp_stream = substream; } } @@ -1002,6 +996,8 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) u32 buffersize; u32 pos = 0; u64 bytescount = 0; + u16 dscr; + u32 period_bytes; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -1009,11 +1005,20 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) if (!rtd) return -EINVAL; - buffersize = frames_to_bytes(runtime, runtime->buffer_size); - bytescount = acp_get_byte_count(rtd); - - bytescount -= rtd->bytescount; - pos = do_div(bytescount, buffersize); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + period_bytes = frames_to_bytes(runtime, runtime->period_size); + dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); + if (dscr == rtd->dma_dscr_idx_1) + pos = period_bytes; + else + pos = 0; + } else { + buffersize = frames_to_bytes(runtime, runtime->buffer_size); + bytescount = acp_get_byte_count(rtd); + if (bytescount > rtd->bytescount) + bytescount -= rtd->bytescount; + pos = do_div(bytescount, buffersize); + } return bytes_to_frames(runtime, pos); } diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 0a2240bff62e..be3963e8f4fa 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -138,6 +138,7 @@ struct audio_substream_data { u32 sram_bank; u32 byte_cnt_high_reg_offset; u32 byte_cnt_low_reg_offset; + u32 dma_curr_dscr; uint64_t size; u64 bytescount; void __iomem *acp_mmio; -- cgit v1.2.3 From c21c834adb5bc81e7081aa93ac50619c6d060506 Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Mon, 6 Aug 2018 12:57:16 +0530 Subject: ASoC: AMD: Set delay value for the capture case ACP->SYSMEM DMA happens at every I2S->SYSMEM period completion. Thus, there is delay of x frames till I2S->SYSMEM reaches a period length. This delay is communicated to user space. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 32f27c5e4d93..e359938e3d7e 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -922,6 +922,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_BLUETOOTH; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH11; + rtd->byte_cnt_high_reg_offset = + mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH; + rtd->byte_cnt_low_reg_offset = + mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW; rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_11; adata->capture_i2sbt_stream = substream; break; @@ -942,6 +946,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_ACP_I2S_1; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; + rtd->byte_cnt_high_reg_offset = + mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; + rtd->byte_cnt_low_reg_offset = + mmACP_I2S_RECEIVED_BYTE_CNT_LOW; rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_15; adata->capture_i2ssp_stream = substream; } @@ -997,7 +1005,7 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) u32 pos = 0; u64 bytescount = 0; u16 dscr; - u32 period_bytes; + u32 period_bytes, delay; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -1012,6 +1020,11 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) pos = period_bytes; else pos = 0; + bytescount = acp_get_byte_count(rtd); + if (bytescount > rtd->bytescount) + bytescount -= rtd->bytescount; + delay = do_div(bytescount, period_bytes); + runtime->delay = bytes_to_frames(runtime, delay); } else { buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd); -- cgit v1.2.3 From 0b0722e191757ca6851382d78c15277ca9c5c211 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 3 Aug 2018 13:30:03 +0100 Subject: ASoC: compress: make BE and FE order inline with dpcm For some reason order of startup/hw_params/prepare are reversed in dynamic compress usecase when compared to dpcm usecase. This is a issue with platforms like QCOM where it expects the BE to be initialized before FE. Interestingly the compress trigger callback order is inline with dpcm. Am not 100% sure why the compress audio case has been reversed. This patch is making the order inline with dpcm. If the reverse ordering is just co-incendental then this change makes sense and will avoid inventing some new mechanism to cope with ordering. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 96 +++++++++++++++++++++++------------------------- 1 file changed, 46 insertions(+), 50 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b9e1673fea51..409d082e80d1 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -140,6 +140,30 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) stream = SNDRV_PCM_STREAM_CAPTURE; mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime = fe_substream->runtime; + + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) + goto be_err; + else if (ret == 0) + dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n", + fe->dai_link->name, stream ? "capture" : "playback"); + /* calculate valid and active FE <-> BE dpcms */ + dpcm_process_paths(fe, stream, &list, 1); + fe->dpcm[stream].runtime = fe_substream->runtime; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_startup(fe, stream); + if (ret < 0) { + /* clean up all links */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + fe->dpcm[stream].runtime = NULL; + goto out; + } if (cpu_dai->driver->cops && cpu_dai->driver->cops->startup) { ret = cpu_dai->driver->cops->startup(cstream, cpu_dai); @@ -153,7 +177,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) ret = soc_compr_components_open(cstream, &component); if (ret < 0) - goto machine_err; + goto open_err; if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->startup) { ret = fe->dai_link->compr_ops->startup(cstream); @@ -164,31 +188,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) } } - fe->dpcm[stream].runtime = fe_substream->runtime; - - ret = dpcm_path_get(fe, stream, &list); - if (ret < 0) - goto fe_err; - else if (ret == 0) - dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n", - fe->dai_link->name, stream ? "capture" : "playback"); - - /* calculate valid and active FE <-> BE dpcms */ - dpcm_process_paths(fe, stream, &list, 1); - - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; - - ret = dpcm_be_dai_startup(fe, stream); - if (ret < 0) { - /* clean up all links */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) - dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - - dpcm_be_disconnect(fe, stream); - fe->dpcm[stream].runtime = NULL; - goto path_err; - } - dpcm_clear_pending_state(fe, stream); dpcm_path_put(&list); @@ -201,17 +200,14 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) return 0; -path_err: - dpcm_path_put(&list); -fe_err: - if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown) - fe->dai_link->compr_ops->shutdown(cstream); machine_err: soc_compr_components_free(cstream, component); - +open_err: if (cpu_dai->driver->cops && cpu_dai->driver->cops->shutdown) cpu_dai->driver->cops->shutdown(cstream, cpu_dai); out: + dpcm_path_put(&list); +be_err: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; mutex_unlock(&fe->card->mutex); return ret; @@ -551,6 +547,24 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + /* + * Create an empty hw_params for the BE as the machine driver must + * fix this up to match DSP decoder and ASRC configuration. + * I.e. machine driver fixup for compressed BE is mandatory. + */ + memset(&fe->dpcm[fe_substream->stream].hw_params, 0, + sizeof(struct snd_pcm_hw_params)); + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_hw_params(fe, stream); + if (ret < 0) + goto out; + + ret = dpcm_be_dai_prepare(fe, stream); + if (ret < 0) + goto out; + if (cpu_dai->driver->cops && cpu_dai->driver->cops->set_params) { ret = cpu_dai->driver->cops->set_params(cstream, params, cpu_dai); if (ret < 0) @@ -577,24 +591,6 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, goto out; } - /* - * Create an empty hw_params for the BE as the machine driver must - * fix this up to match DSP decoder and ASRC configuration. - * I.e. machine driver fixup for compressed BE is mandatory. - */ - memset(&fe->dpcm[fe_substream->stream].hw_params, 0, - sizeof(struct snd_pcm_hw_params)); - - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; - - ret = dpcm_be_dai_hw_params(fe, stream); - if (ret < 0) - goto out; - - ret = dpcm_be_dai_prepare(fe, stream); - if (ret < 0) - goto out; - dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; -- cgit v1.2.3 From f861e3e28a3016a2064d9f600eaa92a530b732b4 Mon Sep 17 00:00:00 2001 From: Matthias Kaehlcke Date: Tue, 7 Aug 2018 10:19:40 -0700 Subject: ASoC: rt5677: Fix initialization of rt5677_of_match.data The driver expects to find the device id in rt5677_of_match.data, however it is currently assigned to rt5677_of_match.type. Fix this. The problem was found with the help of clang: sound/soc/codecs/rt5677.c:5010:36: warning: expression which evaluates to zero treated as a null pointer constant of type 'const void *' [-Wnon-literal-null-conversion] { .compatible = "realtek,rt5677", RT5677 }, ^~~~~~ Fixes: ddc9e69b9dc2 ("ASoC: rt5677: Hide platform data in the module sources") Signed-off-by: Matthias Kaehlcke Reviewed-by: Guenter Roeck Acked-by: Andy Shevchenko Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 922becfee59b..9b7a1833d331 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5008,7 +5008,7 @@ static const struct regmap_config rt5677_regmap = { }; static const struct of_device_id rt5677_of_match[] = { - { .compatible = "realtek,rt5677", RT5677 }, + { .compatible = "realtek,rt5677", .data = (const void *)RT5677 }, { } }; MODULE_DEVICE_TABLE(of, rt5677_of_match); -- cgit v1.2.3 From b1470d4ce77c2d60661c7d5325d4fb8063e15ff8 Mon Sep 17 00:00:00 2001 From: Ajit Pandey Date: Tue, 7 Aug 2018 18:30:42 +0100 Subject: ASoC: wm_adsp: Correct DSP pointer for preloader control The offset of the DSP core needs to be taken into account for the DSP preloader control get and put. Currently the dsp->preloaded variable will only ever be read/updated on the first DSP, whilst this doesn't affect the operation of the control the readback will be incorrect. Signed-off-by: Ajit Pandey Signed-off-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_adsp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4e7f8525449e..08dc82770273 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2643,7 +2643,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); - struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); + struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct wm_adsp *dsp = &dsps[mc->shift - 1]; ucontrol->value.integer.value[0] = dsp->preloaded; @@ -2655,10 +2658,11 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); - struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); + struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + struct wm_adsp *dsp = &dsps[mc->shift - 1]; char preload[32]; snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift); -- cgit v1.2.3 From 0717edbdfed61b4c1e8291140f78882d3a481042 Mon Sep 17 00:00:00 2001 From: Ryan Lee Date: Tue, 7 Aug 2018 20:06:38 -0700 Subject: ASoC: max98373: Added software reset register to readable registers Signed-off-by: Ryan Lee Signed-off-by: Mark Brown --- sound/soc/codecs/max98373.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a92586106932..92b7125ea169 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -488,6 +488,7 @@ static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv, static bool max98373_readable_register(struct device *dev, unsigned int reg) { switch (reg) { + case MAX98373_R2000_SW_RESET: case MAX98373_R2001_INT_RAW1 ... MAX98373_R200C_INT_EN3: case MAX98373_R2010_IRQ_CTRL: case MAX98373_R2014_THERM_WARN_THRESH -- cgit v1.2.3 From 0a047f07525fecfa8f6fccc5d30afff7e816de8d Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 8 Aug 2018 17:13:38 +0100 Subject: ASoC: wm_adsp: Declare firmware controls from codec driver To allow for more flexibility in naming of DSP-type cores move the creation of the firmware controls to the codec drivers instead of having a hardcoded list in wm_adsp. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/cs47l24.c | 3 +++ sound/soc/codecs/wm2200.c | 10 ++++------ sound/soc/codecs/wm5102.c | 2 ++ sound/soc/codecs/wm5110.c | 5 +++++ sound/soc/codecs/wm_adsp.c | 35 +++++++++-------------------------- sound/soc/codecs/wm_adsp.h | 10 +++++++++- 6 files changed, 32 insertions(+), 33 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 0da52ead91e0..45e50fe3bf25 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -235,6 +235,9 @@ ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP2", 1), +WM_ADSP_FW_CONTROL("DSP3", 2), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 3663b9fd4d65..deff65161504 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1180,6 +1180,9 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, WM2200_SPK1R_MUTE_SHIFT, 1, 1), SOC_ENUM("RxANC Src", wm2200_rxanc_input_sel), + +WM_ADSP_FW_CONTROL("DSP1", 0), +WM_ADSP_FW_CONTROL("DSP2", 1), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); @@ -1553,15 +1556,10 @@ static const struct snd_soc_dapm_route wm2200_dapm_routes[] = { static int wm2200_probe(struct snd_soc_component *component) { struct wm2200_priv *wm2200 = snd_soc_component_get_drvdata(component); - int ret; wm2200->component = component; - ret = snd_soc_add_component_controls(component, wm_adsp_fw_controls, 2); - if (ret != 0) - return ret; - - return ret; + return 0; } static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a01a0c0e01eb..7e817e1877c2 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -985,6 +985,8 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP1", 0), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 00c735c585d9..b0789a03d699 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -927,6 +927,11 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP1", 0), +WM_ADSP_FW_CONTROL("DSP2", 1), +WM_ADSP_FW_CONTROL("DSP3", 2), +WM_ADSP_FW_CONTROL("DSP4", 3), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e39b0e0b04df..fbd0515c49d7 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -684,8 +684,8 @@ static inline void wm_adsp_debugfs_clear(struct wm_adsp *dsp) } #endif -static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -695,9 +695,10 @@ static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, return 0; } +EXPORT_SYMBOL_GPL(wm_adsp_fw_get); -static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -721,8 +722,9 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, return ret; } +EXPORT_SYMBOL_GPL(wm_adsp_fw_put); -static const struct soc_enum wm_adsp_fw_enum[] = { +const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 1, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 2, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), @@ -731,24 +733,7 @@ static const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 5, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 6, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), }; - -const struct snd_kcontrol_new wm_adsp_fw_controls[] = { - SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP5 Firmware", wm_adsp_fw_enum[4], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP6 Firmware", wm_adsp_fw_enum[5], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP7 Firmware", wm_adsp_fw_enum[6], - wm_adsp_fw_get, wm_adsp_fw_put), -}; -EXPORT_SYMBOL_GPL(wm_adsp_fw_controls); +EXPORT_SYMBOL_GPL(wm_adsp_fw_enum); static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, int type) @@ -2884,9 +2869,7 @@ int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *comp dsp->component = component; - return snd_soc_add_component_controls(component, - &wm_adsp_fw_controls[dsp->num - 1], - 1); + return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_component_probe); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index bc6d359f0533..8d58cb9d9bb9 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -121,7 +121,11 @@ struct wm_adsp { .reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_event, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } -extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; +#define WM_ADSP_FW_CONTROL(dspname, num) \ + SOC_ENUM_EXT(dspname " Firmware", wm_adsp_fw_enum[num], \ + wm_adsp_fw_get, wm_adsp_fw_put) + +extern const struct soc_enum wm_adsp_fw_enum[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); @@ -144,6 +148,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream); int wm_adsp_compr_free(struct snd_compr_stream *stream); -- cgit v1.2.3 From 605391d0f4bfdff2f2c6c5477ce0ccf776d8d5c0 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 8 Aug 2018 17:13:39 +0100 Subject: ASoC: wm_adsp: Make DSP name configurable by codec driver Instead of harcoding that a core must always be called "DSPn" add a name member to struct wm_adsp so that the owning codec driver can provide a custom name. This allows for re-use of the wm_adsp driver with parts where the processing cores are named differently. If no name is provided the default DSPn name is used. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 69 ++++++++++++++++++++++++++++++++-------------- sound/soc/codecs/wm_adsp.h | 2 ++ 2 files changed, 50 insertions(+), 21 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 1c12c78dbcce..f61656070225 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -10,6 +10,7 @@ * published by the Free Software Foundation. */ +#include #include #include #include @@ -35,15 +36,15 @@ #include "wm_adsp.h" #define adsp_crit(_dsp, fmt, ...) \ - dev_crit(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_crit(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define adsp_err(_dsp, fmt, ...) \ - dev_err(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_err(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define adsp_warn(_dsp, fmt, ...) \ - dev_warn(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_warn(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define adsp_info(_dsp, fmt, ...) \ - dev_info(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_info(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define adsp_dbg(_dsp, fmt, ...) \ - dev_dbg(_dsp->dev, "DSP%d: " fmt, _dsp->num, ##__VA_ARGS__) + dev_dbg(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) #define ADSP1_CONTROL_1 0x00 #define ADSP1_CONTROL_2 0x02 @@ -608,7 +609,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, struct snd_soc_component *component) { struct dentry *root = NULL; - char *root_name; int i; if (!component->debugfs_root) { @@ -616,13 +616,7 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, goto err; } - root_name = kmalloc(PAGE_SIZE, GFP_KERNEL); - if (!root_name) - goto err; - - snprintf(root_name, PAGE_SIZE, "dsp%d", dsp->num); - root = debugfs_create_dir(root_name, component->debugfs_root); - kfree(root_name); + root = debugfs_create_dir(dsp->name, component->debugfs_root); if (!root) goto err; @@ -1315,12 +1309,12 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, switch (dsp->fw_ver) { case 0: case 1: - snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "DSP%d %s %x", - dsp->num, region_name, alg_region->alg); + snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s %x", + dsp->name, region_name, alg_region->alg); break; default: ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "DSP%d%c %.12s %x", dsp->num, *region_name, + "%s%c %.12s %x", dsp->name, *region_name, wm_adsp_fw_text[dsp->fw], alg_region->alg); /* Truncate the subname from the start if it is too long */ @@ -1648,7 +1642,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) if (file == NULL) return -ENOMEM; - snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.wmfw", dsp->part, dsp->num, + snprintf(file, PAGE_SIZE, "%s-%s-%s.wmfw", dsp->part, dsp->fwf_name, wm_adsp_fw[dsp->fw].file); file[PAGE_SIZE - 1] = '\0'; @@ -2226,7 +2220,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) if (file == NULL) return -ENOMEM; - snprintf(file, PAGE_SIZE, "%s-dsp%d-%s.bin", dsp->part, dsp->num, + snprintf(file, PAGE_SIZE, "%s-%s-%s.bin", dsp->part, dsp->fwf_name, wm_adsp_fw[dsp->fw].file); file[PAGE_SIZE - 1] = '\0'; @@ -2398,8 +2392,38 @@ out: return ret; } +static int wm_adsp_create_name(struct wm_adsp *dsp) +{ + char *p; + + if (!dsp->name) { + dsp->name = devm_kasprintf(dsp->dev, GFP_KERNEL, "DSP%d", + dsp->num); + if (!dsp->name) + return -ENOMEM; + } + + if (!dsp->fwf_name) { + p = devm_kstrdup(dsp->dev, dsp->name, GFP_KERNEL); + if (!p) + return -ENOMEM; + + dsp->fwf_name = p; + for (; *p != 0; ++p) + *p = tolower(*p); + } + + return 0; +} + int wm_adsp1_init(struct wm_adsp *dsp) { + int ret; + + ret = wm_adsp_create_name(dsp); + if (ret) + return ret; + INIT_LIST_HEAD(&dsp->alg_regions); mutex_init(&dsp->pwr_lock); @@ -2672,7 +2696,7 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, struct wm_adsp *dsp = &dsps[mc->shift - 1]; char preload[32]; - snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift); + snprintf(preload, ARRAY_SIZE(preload), "%s Preload", dsp->name); dsp->preloaded = ucontrol->value.integer.value[0]; @@ -2867,8 +2891,7 @@ int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *comp { char preload[32]; - snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", dsp->num); - + snprintf(preload, ARRAY_SIZE(preload), "%s Preload", dsp->name); snd_soc_component_disable_pin(component, preload); wm_adsp2_init_debugfs(dsp, component); @@ -2891,6 +2914,10 @@ int wm_adsp2_init(struct wm_adsp *dsp) { int ret; + ret = wm_adsp_create_name(dsp); + if (ret) + return ret; + switch (dsp->rev) { case 0: /* diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 8d58cb9d9bb9..4b8778b0b06c 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -57,6 +57,8 @@ struct wm_adsp_compr_buf; struct wm_adsp { const char *part; + const char *name; + const char *fwf_name; int rev; int num; int type; -- cgit v1.2.3 From 17c81d2f5a59929c73a2a19fd49fe0b068fda76f Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Thu, 9 Aug 2018 10:48:50 +0100 Subject: ASoC: da7219: Add delays to capture path to remove DC offset noise On some platforms it has been noted that a pop noise can be witnessed when capturing audio, mainly for first time after a headset jack has been inserted. This is due to a DC offset in the Mic PGA and so to avoid this delays are required when powering up the capture path. This commit rectifies the problem by adding delays post Mic PGA and post Mixin PGA. The post Mic PGA delay is determined based on Mic Bias voltage, and is only applied the first time after a headset jack is inserted. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- sound/soc/codecs/da7219-aad.c | 5 +++++ sound/soc/codecs/da7219.c | 44 +++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/da7219.h | 8 ++++++-- 3 files changed, 49 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index a49ab751a036..2c7d5088e6f2 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -59,6 +59,7 @@ static void da7219_aad_btn_det_work(struct work_struct *work) container_of(work, struct da7219_aad_priv, btn_det_work); struct snd_soc_component *component = da7219_aad->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 statusa, micbias_ctrl; bool micbias_up = false; int retries = 0; @@ -86,6 +87,8 @@ static void da7219_aad_btn_det_work(struct work_struct *work) if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES) dev_warn(component->dev, "Mic bias status check timed out"); + da7219->micbias_on_event = true; + /* * Mic bias pulse required to enable mic, must be done before enabling * button detection to prevent erroneous button readings. @@ -439,6 +442,8 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) snd_soc_component_update_bits(component, DA7219_ACCDET_CONFIG_1, DA7219_BUTTON_CONFIG_MASK, 0); + da7219->micbias_on_event = false; + /* Disable mic bias */ snd_soc_dapm_disable_pin(dapm, "Mic Bias"); snd_soc_dapm_sync(dapm); diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index c0144f2f8174..e46e9f4bc994 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -768,6 +768,30 @@ static const struct snd_kcontrol_new da7219_st_out_filtr_mix_controls[] = { * DAPM Events */ +static int da7219_mic_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (da7219->micbias_on_event) { + /* + * Delay only for first capture after bias enabled to + * avoid possible DC offset related noise. + */ + da7219->micbias_on_event = false; + msleep(da7219->mic_pga_delay); + } + break; + default: + break; + } + + return 0; +} + static int da7219_dai_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -937,12 +961,12 @@ static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC"), /* Input PGAs */ - SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL, - DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), - SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL, - DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), + SND_SOC_DAPM_PGA_E("Mic PGA", DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_mic_pga_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("Mixin PGA", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_settling_event, SND_SOC_DAPM_POST_PMU), /* Input Filters */ SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT, @@ -1847,6 +1871,14 @@ static void da7219_handle_pdata(struct snd_soc_component *component) snd_soc_component_write(component, DA7219_MICBIAS_CTRL, micbias_lvl); + /* + * Calculate delay required to compensate for DC offset in + * Mic PGA, based on Mic Bias voltage. + */ + da7219->mic_pga_delay = DA7219_MIC_PGA_BASE_DELAY + + (pdata->micbias_lvl * + DA7219_MIC_PGA_OFFSET_DELAY); + /* Mic */ switch (pdata->mic_amp_in_sel) { case DA7219_MIC_AMP_IN_SEL_DIFF: diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 1b00023e33cd..3a006862f0e7 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -781,8 +781,10 @@ #define DA7219_SYS_STAT_CHECK_DELAY 50 /* Power up/down Delays */ -#define DA7219_SETTLING_DELAY 40 -#define DA7219_MIN_GAIN_DELAY 30 +#define DA7219_SETTLING_DELAY 40 +#define DA7219_MIN_GAIN_DELAY 30 +#define DA7219_MIC_PGA_BASE_DELAY 100 +#define DA7219_MIC_PGA_OFFSET_DELAY 40 enum da7219_clk_src { DA7219_CLKSRC_MCLK = 0, @@ -828,6 +830,8 @@ struct da7219_priv { bool master; bool alc_en; + bool micbias_on_event; + unsigned int mic_pga_delay; u8 gain_ramp_ctrl; }; -- cgit v1.2.3 From f2cf0ef7c0ce141bb38f315c34c56e6ef5667a27 Mon Sep 17 00:00:00 2001 From: Gustavo A. R. Silva Date: Wed, 8 Aug 2018 14:19:33 -0500 Subject: ASoC: adav80x: mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1056531 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adav80x.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index db21ecbe0762..8b9ca7e7a682 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -648,6 +648,7 @@ static int adav80x_set_pll(struct snd_soc_component *component, int pll_id, pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV; break; } + /* fall through */ default: return -EINVAL; } -- cgit v1.2.3