diff options
Diffstat (limited to 'sound/soc/fsl')
30 files changed, 1231 insertions, 286 deletions
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7b1d9970be8b..55ed47c599e2 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -24,6 +24,13 @@ config SND_SOC_FSL_SAI This option is only useful for out-of-tree drivers since in-tree drivers select it automatically. +config SND_SOC_FSL_AUDMIX + tristate "Audio Mixer (AUDMIX) module support" + select REGMAP_MMIO + help + Say Y if you want to add Audio Mixer (AUDMIX) + support for the NXP iMX CPUs. + config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module (SSI) support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n @@ -182,16 +189,17 @@ config SND_MPC52xx_SOC_EFIKA endif # SND_POWERPC_SOC +config SND_SOC_IMX_PCM_FIQ + tristate + default y if SND_SOC_IMX_SSI=y && (SND_SOC_FSL_SSI=m || SND_SOC_FSL_SPDIF=m) && (MXC_TZIC || MXC_AVIC) + select FIQ + if SND_IMX_SOC config SND_SOC_IMX_SSI tristate select SND_SOC_FSL_UTILS -config SND_SOC_IMX_PCM_FIQ - tristate - select FIQ - comment "SoC Audio support for Freescale i.MX boards:" config SND_MXC_SOC_WM1133_EV1 @@ -296,6 +304,15 @@ config SND_SOC_FSL_ASOC_CARD CS4271, CS4272 and SGTL5000. Say Y if you want to add support for Freescale Generic ASoC Sound Card. +config SND_SOC_IMX_AUDMIX + tristate "SoC Audio support for i.MX boards with AUDMIX" + select SND_SOC_FSL_AUDMIX + select SND_SOC_FSL_SAI + help + SoC Audio support for i.MX boards with Audio Mixer + Say Y if you want to add support for SoC audio on an i.MX board with + an Audio Mixer. + endif # SND_IMX_SOC endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 3c0ff315b971..c0dd04422fe9 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -12,6 +12,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-audmix-objs := fsl_audmix.o snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o @@ -22,6 +23,8 @@ snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-micfil-objs := fsl_micfil.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o + +obj-$(CONFIG_SND_SOC_FSL_AUDMIX) += snd-soc-fsl-audmix.o obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o @@ -59,6 +62,7 @@ snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o +snd-soc-imx-audmix-objs := imx-audmix.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o @@ -68,3 +72,4 @@ obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o +obj-$(CONFIG_SND_SOC_IMX_AUDMIX) += snd-soc-imx-audmix.o diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 191426a6d9ad..d648268cb454 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -1,19 +1,13 @@ -/* - * eukrea-tlv320.c -- SoC audio for eukrea_cpuimxXX in I2S mode - * - * Copyright 2010 Eric Bénard, Eukréa Electromatique <eric@eukrea.com> - * - * based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c - * which is Copyright 2009 Simtec Electronics - * and on sound/soc/imx/phycore-ac97.c which is - * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de> - * - * 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+ +// +// eukrea-tlv320.c -- SoC audio for eukrea_cpuimxXX in I2S mode +// +// Copyright 2010 Eric Bénard, Eukréa Electromatique <eric@eukrea.com> +// +// based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c +// which is Copyright 2009 Simtec Electronics +// and on sound/soc/imx/phycore-ac97.c which is +// Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de> #include <linux/errno.h> #include <linux/module.h> @@ -118,13 +112,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "fsl,mux-int-port node missing or invalid.\n"); - return ret; + goto err; } ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); if (ret) { dev_err(&pdev->dev, "fsl,mux-ext-port node missing or invalid.\n"); - return ret; + goto err; } /* diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c new file mode 100644 index 000000000000..3897a54a11fe --- /dev/null +++ b/sound/soc/fsl/fsl_audmix.c @@ -0,0 +1,578 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * NXP AUDMIX ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright 2017 NXP + */ + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/pm_runtime.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "fsl_audmix.h" + +#define SOC_ENUM_SINGLE_S(xreg, xshift, xtexts) \ + SOC_ENUM_SINGLE(xreg, xshift, ARRAY_SIZE(xtexts), xtexts) + +static const char + *tdm_sel[] = { "TDM1", "TDM2", }, + *mode_sel[] = { "Disabled", "TDM1", "TDM2", "Mixed", }, + *width_sel[] = { "16b", "18b", "20b", "24b", "32b", }, + *endis_sel[] = { "Disabled", "Enabled", }, + *updn_sel[] = { "Downward", "Upward", }, + *mask_sel[] = { "Unmask", "Mask", }; + +static const struct soc_enum fsl_audmix_enum[] = { +/* FSL_AUDMIX_CTR enums */ +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MIXCLK_SHIFT, tdm_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_OUTSRC_SHIFT, mode_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_OUTWIDTH_SHIFT, width_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MASKRTDF_SHIFT, mask_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MASKCKDF_SHIFT, mask_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_SYNCMODE_SHIFT, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_SYNCSRC_SHIFT, tdm_sel), +/* FSL_AUDMIX_ATCR0 enums */ +SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR0, 0, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR0, 1, updn_sel), +/* FSL_AUDMIX_ATCR1 enums */ +SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR1, 0, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR1, 1, updn_sel), +}; + +struct fsl_audmix_state { + u8 tdms; + u8 clk; + char msg[64]; +}; + +static const struct fsl_audmix_state prms[4][4] = {{ + /* DIS->DIS, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* DIS->TDM1*/ + { .tdms = 1, .clk = 1, .msg = "DIS->TDM1: TDM1 not started!\n" }, + /* DIS->TDM2*/ + { .tdms = 2, .clk = 2, .msg = "DIS->TDM2: TDM2 not started!\n" }, + /* DIS->MIX */ + { .tdms = 3, .clk = 0, .msg = "DIS->MIX: Please start both TDMs!\n" } +}, { /* TDM1->DIS */ + { .tdms = 1, .clk = 0, .msg = "TDM1->DIS: TDM1 not started!\n" }, + /* TDM1->TDM1, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* TDM1->TDM2 */ + { .tdms = 3, .clk = 2, .msg = "TDM1->TDM2: Please start both TDMs!\n" }, + /* TDM1->MIX */ + { .tdms = 3, .clk = 0, .msg = "TDM1->MIX: Please start both TDMs!\n" } +}, { /* TDM2->DIS */ + { .tdms = 2, .clk = 0, .msg = "TDM2->DIS: TDM2 not started!\n" }, + /* TDM2->TDM1 */ + { .tdms = 3, .clk = 1, .msg = "TDM2->TDM1: Please start both TDMs!\n" }, + /* TDM2->TDM2, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* TDM2->MIX */ + { .tdms = 3, .clk = 0, .msg = "TDM2->MIX: Please start both TDMs!\n" } +}, { /* MIX->DIS */ + { .tdms = 3, .clk = 0, .msg = "MIX->DIS: Please start both TDMs!\n" }, + /* MIX->TDM1 */ + { .tdms = 3, .clk = 1, .msg = "MIX->TDM1: Please start both TDMs!\n" }, + /* MIX->TDM2 */ + { .tdms = 3, .clk = 2, .msg = "MIX->TDM2: Please start both TDMs!\n" }, + /* MIX->MIX, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" } +}, }; + +static int fsl_audmix_state_trans(struct snd_soc_component *comp, + unsigned int *mask, unsigned int *ctr, + const struct fsl_audmix_state prm) +{ + struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp); + /* Enforce all required TDMs are started */ + if ((priv->tdms & prm.tdms) != prm.tdms) { + dev_dbg(comp->dev, "%s", prm.msg); + return -EINVAL; + } + + switch (prm.clk) { + case 1: + case 2: + /* Set mix clock */ + (*mask) |= FSL_AUDMIX_CTR_MIXCLK_MASK; + (*ctr) |= FSL_AUDMIX_CTR_MIXCLK(prm.clk - 1); + break; + default: + break; + } + + return 0; +} + +static int fsl_audmix_put_mix_clk_src(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int reg_val, val, mix_clk; + int ret = 0; + + /* Get current state */ + ret = snd_soc_component_read(comp, FSL_AUDMIX_CTR, ®_val); + if (ret) + return ret; + + mix_clk = ((reg_val & FSL_AUDMIX_CTR_MIXCLK_MASK) + >> FSL_AUDMIX_CTR_MIXCLK_SHIFT); + val = snd_soc_enum_item_to_val(e, item[0]); + + dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val); + + /** + * Ensure the current selected mixer clock is available + * for configuration propagation + */ + if (!(priv->tdms & BIT(mix_clk))) { + dev_err(comp->dev, + "Started TDM%d needed for config propagation!\n", + mix_clk + 1); + return -EINVAL; + } + + if (!(priv->tdms & BIT(val))) { + dev_err(comp->dev, + "The selected clock source has no TDM%d enabled!\n", + val + 1); + return -EINVAL; + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + +static int fsl_audmix_put_out_src(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + u32 out_src, mix_clk; + unsigned int reg_val, val, mask = 0, ctr = 0; + int ret = 0; + + /* Get current state */ + ret = snd_soc_component_read(comp, FSL_AUDMIX_CTR, ®_val); + if (ret) + return ret; + + /* "From" state */ + out_src = ((reg_val & FSL_AUDMIX_CTR_OUTSRC_MASK) + >> FSL_AUDMIX_CTR_OUTSRC_SHIFT); + mix_clk = ((reg_val & FSL_AUDMIX_CTR_MIXCLK_MASK) + >> FSL_AUDMIX_CTR_MIXCLK_SHIFT); + + /* "To" state */ + val = snd_soc_enum_item_to_val(e, item[0]); + + dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val); + + /* Check if state is changing ... */ + if (out_src == val) + return 0; + /** + * Ensure the current selected mixer clock is available + * for configuration propagation + */ + if (!(priv->tdms & BIT(mix_clk))) { + dev_err(comp->dev, + "Started TDM%d needed for config propagation!\n", + mix_clk + 1); + return -EINVAL; + } + + /* Check state transition constraints */ + ret = fsl_audmix_state_trans(comp, &mask, &ctr, prms[out_src][val]); + if (ret) + return ret; + + /* Complete transition to new state */ + mask |= FSL_AUDMIX_CTR_OUTSRC_MASK; + ctr |= FSL_AUDMIX_CTR_OUTSRC(val); + + return snd_soc_component_update_bits(comp, FSL_AUDMIX_CTR, mask, ctr); +} + +static const struct snd_kcontrol_new fsl_audmix_snd_controls[] = { + /* FSL_AUDMIX_CTR controls */ + SOC_ENUM_EXT("Mixing Clock Source", fsl_audmix_enum[0], + snd_soc_get_enum_double, fsl_audmix_put_mix_clk_src), + SOC_ENUM_EXT("Output Source", fsl_audmix_enum[1], + snd_soc_get_enum_double, fsl_audmix_put_out_src), + SOC_ENUM("Output Width", fsl_audmix_enum[2]), + SOC_ENUM("Frame Rate Diff Error", fsl_audmix_enum[3]), + SOC_ENUM("Clock Freq Diff Error", fsl_audmix_enum[4]), + SOC_ENUM("Sync Mode Config", fsl_audmix_enum[5]), + SOC_ENUM("Sync Mode Clk Source", fsl_audmix_enum[6]), + /* TDM1 Attenuation controls */ + SOC_ENUM("TDM1 Attenuation", fsl_audmix_enum[7]), + SOC_ENUM("TDM1 Attenuation Direction", fsl_audmix_enum[8]), + SOC_SINGLE("TDM1 Attenuation Step Divider", FSL_AUDMIX_ATCR0, + 2, 0x00fff, 0), + SOC_SINGLE("TDM1 Attenuation Initial Value", FSL_AUDMIX_ATIVAL0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Up Factor", FSL_AUDMIX_ATSTPUP0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Down Factor", FSL_AUDMIX_ATSTPDN0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Target", FSL_AUDMIX_ATSTPTGT0, + 0, 0x3ffff, 0), + /* TDM2 Attenuation controls */ + SOC_ENUM("TDM2 Attenuation", fsl_audmix_enum[9]), + SOC_ENUM("TDM2 Attenuation Direction", fsl_audmix_enum[10]), + SOC_SINGLE("TDM2 Attenuation Step Divider", FSL_AUDMIX_ATCR1, + 2, 0x00fff, 0), + SOC_SINGLE("TDM2 Attenuation Initial Value", FSL_AUDMIX_ATIVAL1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Up Factor", FSL_AUDMIX_ATSTPUP1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Down Factor", FSL_AUDMIX_ATSTPDN1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Target", FSL_AUDMIX_ATSTPTGT1, + 0, 0x3ffff, 0), +}; + +static int fsl_audmix_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *comp = dai->component; + u32 mask = 0, ctr = 0; + + /* AUDMIX is working in DSP_A format only */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + break; + default: + return -EINVAL; + } + + /* For playback the AUDMIX is slave, and for record is master */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + /* Output data will be written on positive edge of the clock */ + ctr |= FSL_AUDMIX_CTR_OUTCKPOL(0); + break; + case SND_SOC_DAIFMT_NB_NF: + /* Output data will be written on negative edge of the clock */ + ctr |= FSL_AUDMIX_CTR_OUTCKPOL(1); + break; + default: + return -EINVAL; + } + + mask |= FSL_AUDMIX_CTR_OUTCKPOL_MASK; + + return snd_soc_component_update_bits(comp, FSL_AUDMIX_CTR, mask, ctr); +} + +static int fsl_audmix_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsl_audmix *priv = snd_soc_dai_get_drvdata(dai); + + /* Capture stream shall not be handled */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + priv->tdms |= BIT(dai->driver->id); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + priv->tdms &= ~BIT(dai->driver->id); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops fsl_audmix_dai_ops = { + .set_fmt = fsl_audmix_dai_set_fmt, + .trigger = fsl_audmix_dai_trigger, +}; + +static struct snd_soc_dai_driver fsl_audmix_dai[] = { + { + .id = 0, + .name = "audmix-0", + .playback = { + .stream_name = "AUDMIX-Playback-0", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AUDMIX_FORMATS, + }, + .capture = { + .stream_name = "AUDMIX-Capture-0", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AUDMIX_FORMATS, + }, + .ops = &fsl_audmix_dai_ops, + }, + { + .id = 1, + .name = "audmix-1", + .playback = { + .stream_name = "AUDMIX-Playback-1", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AUDMIX_FORMATS, + }, + .capture = { + .stream_name = "AUDMIX-Capture-1", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AUDMIX_FORMATS, + }, + .ops = &fsl_audmix_dai_ops, + }, +}; + +static const struct snd_soc_component_driver fsl_audmix_component = { + .name = "fsl-audmix-dai", + .controls = fsl_audmix_snd_controls, + .num_controls = ARRAY_SIZE(fsl_audmix_snd_controls), +}; + +static bool fsl_audmix_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_AUDMIX_CTR: + case FSL_AUDMIX_STR: + case FSL_AUDMIX_ATCR0: + case FSL_AUDMIX_ATIVAL0: + case FSL_AUDMIX_ATSTPUP0: + case FSL_AUDMIX_ATSTPDN0: + case FSL_AUDMIX_ATSTPTGT0: + case FSL_AUDMIX_ATTNVAL0: + case FSL_AUDMIX_ATSTP0: + case FSL_AUDMIX_ATCR1: + case FSL_AUDMIX_ATIVAL1: + case FSL_AUDMIX_ATSTPUP1: + case FSL_AUDMIX_ATSTPDN1: + case FSL_AUDMIX_ATSTPTGT1: + case FSL_AUDMIX_ATTNVAL1: + case FSL_AUDMIX_ATSTP1: + return true; + default: + return false; + } +} + +static bool fsl_audmix_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_AUDMIX_CTR: + case FSL_AUDMIX_ATCR0: + case FSL_AUDMIX_ATIVAL0: + case FSL_AUDMIX_ATSTPUP0: + case FSL_AUDMIX_ATSTPDN0: + case FSL_AUDMIX_ATSTPTGT0: + case FSL_AUDMIX_ATCR1: + case FSL_AUDMIX_ATIVAL1: + case FSL_AUDMIX_ATSTPUP1: + case FSL_AUDMIX_ATSTPDN1: + case FSL_AUDMIX_ATSTPTGT1: + return true; + default: + return false; + } +} + +static const struct reg_default fsl_audmix_reg[] = { + { FSL_AUDMIX_CTR, 0x00060 }, + { FSL_AUDMIX_STR, 0x00003 }, + { FSL_AUDMIX_ATCR0, 0x00000 }, + { FSL_AUDMIX_ATIVAL0, 0x3FFFF }, + { FSL_AUDMIX_ATSTPUP0, 0x2AAAA }, + { FSL_AUDMIX_ATSTPDN0, 0x30000 }, + { FSL_AUDMIX_ATSTPTGT0, 0x00010 }, + { FSL_AUDMIX_ATTNVAL0, 0x00000 }, + { FSL_AUDMIX_ATSTP0, 0x00000 }, + { FSL_AUDMIX_ATCR1, 0x00000 }, + { FSL_AUDMIX_ATIVAL1, 0x3FFFF }, + { FSL_AUDMIX_ATSTPUP1, 0x2AAAA }, + { FSL_AUDMIX_ATSTPDN1, 0x30000 }, + { FSL_AUDMIX_ATSTPTGT1, 0x00010 }, + { FSL_AUDMIX_ATTNVAL1, 0x00000 }, + { FSL_AUDMIX_ATSTP1, 0x00000 }, +}; + +static const struct regmap_config fsl_audmix_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = FSL_AUDMIX_ATSTP1, + .reg_defaults = fsl_audmix_reg, + .num_reg_defaults = ARRAY_SIZE(fsl_audmix_reg), + .readable_reg = fsl_audmix_readable_reg, + .writeable_reg = fsl_audmix_writeable_reg, + .cache_type = REGCACHE_FLAT, +}; + +static const struct of_device_id fsl_audmix_ids[] = { + { + .compatible = "fsl,imx8qm-audmix", + .data = "imx-audmix", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, fsl_audmix_ids); + +static int fsl_audmix_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct fsl_audmix *priv; + struct resource *res; + const char *mdrv; + const struct of_device_id *of_id; + void __iomem *regs; + int ret; + + of_id = of_match_device(fsl_audmix_ids, dev); + if (!of_id || !of_id->data) + return -EINVAL; + + mdrv = of_id->data; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->regmap = devm_regmap_init_mmio_clk(dev, "ipg", regs, + &fsl_audmix_regmap_config); + if (IS_ERR(priv->regmap)) { + dev_err(dev, "failed to init regmap\n"); + return PTR_ERR(priv->regmap); + } + + priv->ipg_clk = devm_clk_get(dev, "ipg"); + if (IS_ERR(priv->ipg_clk)) { + dev_err(dev, "failed to get ipg clock\n"); + return PTR_ERR(priv->ipg_clk); + } + + platform_set_drvdata(pdev, priv); + pm_runtime_enable(dev); + + ret = devm_snd_soc_register_component(dev, &fsl_audmix_component, + fsl_audmix_dai, + ARRAY_SIZE(fsl_audmix_dai)); + if (ret) { + dev_err(dev, "failed to register ASoC DAI\n"); + return ret; + } + + priv->pdev = platform_device_register_data(dev, mdrv, 0, NULL, 0); + if (IS_ERR(priv->pdev)) { + ret = PTR_ERR(priv->pdev); + dev_err(dev, "failed to register platform %s: %d\n", mdrv, ret); + } + + return ret; +} + +static int fsl_audmix_remove(struct platform_device *pdev) +{ + struct fsl_audmix *priv = dev_get_drvdata(&pdev->dev); + + if (priv->pdev) + platform_device_unregister(priv->pdev); + + return 0; +} + +#ifdef CONFIG_PM +static int fsl_audmix_runtime_resume(struct device *dev) +{ + struct fsl_audmix *priv = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(priv->ipg_clk); + if (ret) { + dev_err(dev, "Failed to enable IPG clock: %d\n", ret); + return ret; + } + + regcache_cache_only(priv->regmap, false); + regcache_mark_dirty(priv->regmap); + + return regcache_sync(priv->regmap); +} + +static int fsl_audmix_runtime_suspend(struct device *dev) +{ + struct fsl_audmix *priv = dev_get_drvdata(dev); + + regcache_cache_only(priv->regmap, true); + + clk_disable_unprepare(priv->ipg_clk); + + return 0; +} +#endif /* CONFIG_PM */ + +static const struct dev_pm_ops fsl_audmix_pm = { + SET_RUNTIME_PM_OPS(fsl_audmix_runtime_suspend, + fsl_audmix_runtime_resume, + NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) +}; + +static struct platform_driver fsl_audmix_driver = { + .probe = fsl_audmix_probe, + .remove = fsl_audmix_remove, + .driver = { + .name = "fsl-audmix", + .of_match_table = fsl_audmix_ids, + .pm = &fsl_audmix_pm, + }, +}; +module_platform_driver(fsl_audmix_driver); + +MODULE_DESCRIPTION("NXP AUDMIX ASoC DAI driver"); +MODULE_AUTHOR("Viorel Suman <viorel.suman@nxp.com>"); +MODULE_ALIAS("platform:fsl-audmix"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_audmix.h b/sound/soc/fsl/fsl_audmix.h new file mode 100644 index 000000000000..7812ffec45c5 --- /dev/null +++ b/sound/soc/fsl/fsl_audmix.h @@ -0,0 +1,102 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * NXP AUDMIX ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright 2017 NXP + */ + +#ifndef __FSL_AUDMIX_H +#define __FSL_AUDMIX_H + +#define FSL_AUDMIX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) +/* AUDMIX Registers */ +#define FSL_AUDMIX_CTR 0x200 /* Control */ +#define FSL_AUDMIX_STR 0x204 /* Status */ + +#define FSL_AUDMIX_ATCR0 0x208 /* Attenuation Control */ +#define FSL_AUDMIX_ATIVAL0 0x20c /* Attenuation Initial Value */ +#define FSL_AUDMIX_ATSTPUP0 0x210 /* Attenuation step up factor */ +#define FSL_AUDMIX_ATSTPDN0 0x214 /* Attenuation step down factor */ +#define FSL_AUDMIX_ATSTPTGT0 0x218 /* Attenuation step target */ +#define FSL_AUDMIX_ATTNVAL0 0x21c /* Attenuation Value */ +#define FSL_AUDMIX_ATSTP0 0x220 /* Attenuation step number */ + +#define FSL_AUDMIX_ATCR1 0x228 /* Attenuation Control */ +#define FSL_AUDMIX_ATIVAL1 0x22c /* Attenuation Initial Value */ +#define FSL_AUDMIX_ATSTPUP1 0x230 /* Attenuation step up factor */ +#define FSL_AUDMIX_ATSTPDN1 0x234 /* Attenuation step down factor */ +#define FSL_AUDMIX_ATSTPTGT1 0x238 /* Attenuation step target */ +#define FSL_AUDMIX_ATTNVAL1 0x23c /* Attenuation Value */ +#define FSL_AUDMIX_ATSTP1 0x240 /* Attenuation step number */ + +/* AUDMIX Control Register */ +#define FSL_AUDMIX_CTR_MIXCLK_SHIFT 0 +#define FSL_AUDMIX_CTR_MIXCLK_MASK BIT(FSL_AUDMIX_CTR_MIXCLK_SHIFT) +#define FSL_AUDMIX_CTR_MIXCLK(i) ((i) << FSL_AUDMIX_CTR_MIXCLK_SHIFT) +#define FSL_AUDMIX_CTR_OUTSRC_SHIFT 1 +#define FSL_AUDMIX_CTR_OUTSRC_MASK (0x3 << FSL_AUDMIX_CTR_OUTSRC_SHIFT) +#define FSL_AUDMIX_CTR_OUTSRC(i) (((i) << FSL_AUDMIX_CTR_OUTSRC_SHIFT)\ + & FSL_AUDMIX_CTR_OUTSRC_MASK) +#define FSL_AUDMIX_CTR_OUTWIDTH_SHIFT 3 +#define FSL_AUDMIX_CTR_OUTWIDTH_MASK (0x7 << FSL_AUDMIX_CTR_OUTWIDTH_SHIFT) +#define FSL_AUDMIX_CTR_OUTWIDTH(i) (((i) << FSL_AUDMIX_CTR_OUTWIDTH_SHIFT)\ + & FSL_AUDMIX_CTR_OUTWIDTH_MASK) +#define FSL_AUDMIX_CTR_OUTCKPOL_SHIFT 6 +#define FSL_AUDMIX_CTR_OUTCKPOL_MASK BIT(FSL_AUDMIX_CTR_OUTCKPOL_SHIFT) +#define FSL_AUDMIX_CTR_OUTCKPOL(i) ((i) << FSL_AUDMIX_CTR_OUTCKPOL_SHIFT) +#define FSL_AUDMIX_CTR_MASKRTDF_SHIFT 7 +#define FSL_AUDMIX_CTR_MASKRTDF_MASK BIT(FSL_AUDMIX_CTR_MASKRTDF_SHIFT) +#define FSL_AUDMIX_CTR_MASKRTDF(i) ((i) << FSL_AUDMIX_CTR_MASKRTDF_SHIFT) +#define FSL_AUDMIX_CTR_MASKCKDF_SHIFT 8 +#define FSL_AUDMIX_CTR_MASKCKDF_MASK BIT(FSL_AUDMIX_CTR_MASKCKDF_SHIFT) +#define FSL_AUDMIX_CTR_MASKCKDF(i) ((i) << FSL_AUDMIX_CTR_MASKCKDF_SHIFT) +#define FSL_AUDMIX_CTR_SYNCMODE_SHIFT 9 +#define FSL_AUDMIX_CTR_SYNCMODE_MASK BIT(FSL_AUDMIX_CTR_SYNCMODE_SHIFT) +#define FSL_AUDMIX_CTR_SYNCMODE(i) ((i) << FSL_AUDMIX_CTR_SYNCMODE_SHIFT) +#define FSL_AUDMIX_CTR_SYNCSRC_SHIFT 10 +#define FSL_AUDMIX_CTR_SYNCSRC_MASK BIT(FSL_AUDMIX_CTR_SYNCSRC_SHIFT) +#define FSL_AUDMIX_CTR_SYNCSRC(i) ((i) << FSL_AUDMIX_CTR_SYNCSRC_SHIFT) + +/* AUDMIX Status Register */ +#define FSL_AUDMIX_STR_RATEDIFF BIT(0) +#define FSL_AUDMIX_STR_CLKDIFF BIT(1) +#define FSL_AUDMIX_STR_MIXSTAT_SHIFT 2 +#define FSL_AUDMIX_STR_MIXSTAT_MASK (0x3 << FSL_AUDMIX_STR_MIXSTAT_SHIFT) +#define FSL_AUDMIX_STR_MIXSTAT(i) (((i) & FSL_AUDMIX_STR_MIXSTAT_MASK) \ + >> FSL_AUDMIX_STR_MIXSTAT_SHIFT) +/* AUDMIX Attenuation Control Register */ +#define FSL_AUDMIX_ATCR_AT_EN BIT(0) +#define FSL_AUDMIX_ATCR_AT_UPDN BIT(1) +#define FSL_AUDMIX_ATCR_ATSTPDIF_SHIFT 2 +#define FSL_AUDMIX_ATCR_ATSTPDFI_MASK \ + (0xfff << FSL_AUDMIX_ATCR_ATSTPDIF_SHIFT) + +/* AUDMIX Attenuation Initial Value Register */ +#define FSL_AUDMIX_ATIVAL_ATINVAL_MASK 0x3FFFF + +/* AUDMIX Attenuation Step Up Factor Register */ +#define FSL_AUDMIX_ATSTPUP_ATSTEPUP_MASK 0x3FFFF + +/* AUDMIX Attenuation Step Down Factor Register */ +#define FSL_AUDMIX_ATSTPDN_ATSTEPDN_MASK 0x3FFFF + +/* AUDMIX Attenuation Step Target Register */ +#define FSL_AUDMIX_ATSTPTGT_ATSTPTG_MASK 0x3FFFF + +/* AUDMIX Attenuation Value Register */ +#define FSL_AUDMIX_ATTNVAL_ATCURVAL_MASK 0x3FFFF + +/* AUDMIX Attenuation Step Number Register */ +#define FSL_AUDMIX_ATSTP_STPCTR_MASK 0x3FFFF + +#define FSL_AUDMIX_MAX_DAIS 2 +struct fsl_audmix { + struct platform_device *pdev; + struct regmap *regmap; + struct clk *ipg_clk; + u8 tdms; +}; + +#endif /* __FSL_AUDMIX_H */ diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 78871de35086..e22508301412 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -1,18 +1,14 @@ -/* - * Freescale DMA ALSA SoC PCM driver - * - * Author: Timur Tabi <timur@freescale.com> - * - * Copyright 2007-2010 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. - * - * This driver implements ASoC support for the Elo DMA controller, which is - * the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms, - * the PCM driver is what handles the DMA buffer. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale DMA ALSA SoC PCM driver +// +// Author: Timur Tabi <timur@freescale.com> +// +// Copyright 2007-2010 Freescale Semiconductor, Inc. +// +// This driver implements ASoC support for the Elo DMA controller, which is +// the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms, +// the PCM driver is what handles the DMA buffer. #include <linux/module.h> #include <linux/init.h> diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h index 78fee97e8036..f19ae765b656 100644 --- a/sound/soc/fsl/fsl_dma.h +++ b/sound/soc/fsl/fsl_dma.h @@ -1,9 +1,6 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* * mpc8610-pcm.h - ALSA PCM interface for the Freescale MPC8610 SoC - * - * 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 _MPC8610_PCM_H diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 3623aa9a6f2e..bad0dfed6b68 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -218,7 +218,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); struct clk *clksrc = esai_priv->extalclk; - bool tx = clk_id <= ESAI_HCKT_EXTAL; + bool tx = (clk_id <= ESAI_HCKT_EXTAL || esai_priv->synchronous); bool in = dir == SND_SOC_CLOCK_IN; u32 ratio, ecr = 0; unsigned long clk_rate; @@ -251,9 +251,9 @@ 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 */ + break; case ESAI_HCKR_EXTAL: - ecr |= ESAI_ECR_ERI; + ecr |= esai_priv->synchronous ? ESAI_ECR_ETI : ESAI_ECR_ERI; break; default: return -EINVAL; @@ -537,10 +537,18 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, bclk = params_rate(params) * slot_width * esai_priv->slots; - ret = fsl_esai_set_bclk(dai, tx, bclk); + ret = fsl_esai_set_bclk(dai, esai_priv->synchronous || tx, bclk); if (ret) return ret; + mask = ESAI_xCR_xSWS_MASK; + val = ESAI_xCR_xSWS(slot_width, width); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + /* Recording in synchronous mode needs to set TCR also */ + if (!tx && esai_priv->synchronous) + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, val); + /* Use Normal mode to support monaural audio */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), ESAI_xCR_xMOD_MASK, params_channels(params) > 1 ? @@ -556,10 +564,9 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); - mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0); - val = ESAI_xCR_xSWS(slot_width, width) | (tx ? ESAI_xCR_PADC : 0); - - regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + if (tx) + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_PADC, ESAI_xCR_PADC); /* Remove ESAI personal reset by configuring ESAI_PCRC and ESAI_PRRC */ regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 40c07e756481..f7f2d29f1bfe 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -151,12 +151,9 @@ static inline int get_clk_div(struct fsl_micfil *micfil, { u32 ctrl2_reg; long mclk_rate; - int osr; int clk_div; regmap_read(micfil->regmap, REG_MICFIL_CTRL2, &ctrl2_reg); - osr = 16 - ((ctrl2_reg & MICFIL_CTRL2_CICOSR_MASK) - >> MICFIL_CTRL2_CICOSR_SHIFT); mclk_rate = clk_get_rate(micfil->mclk); diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index db9e0872f73d..8593269156bd 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -9,6 +9,7 @@ #include <linux/dmaengine.h> #include <linux/module.h> #include <linux/of_address.h> +#include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/slab.h> #include <linux/time.h> @@ -268,12 +269,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_CBS_CFS: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + sai->is_slave_mode = false; break; case SND_SOC_DAIFMT_CBM_CFM: sai->is_slave_mode = true; break; case SND_SOC_DAIFMT_CBS_CFM: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; + sai->is_slave_mode = false; break; case SND_SOC_DAIFMT_CBM_CFS: val_cr4 |= FSL_SAI_CR4_FSD_MSTR; @@ -899,6 +902,8 @@ static int fsl_sai_probe(struct platform_device *pdev) platform_set_drvdata(pdev, sai); + pm_runtime_enable(&pdev->dev); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, &fsl_sai_dai, 1); if (ret) @@ -910,6 +915,13 @@ static int fsl_sai_probe(struct platform_device *pdev) return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); } +static int fsl_sai_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + + return 0; +} + static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, @@ -918,8 +930,8 @@ static const struct of_device_id fsl_sai_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -#ifdef CONFIG_PM_SLEEP -static int fsl_sai_suspend(struct device *dev) +#ifdef CONFIG_PM +static int fsl_sai_runtime_suspend(struct device *dev) { struct fsl_sai *sai = dev_get_drvdata(dev); @@ -929,7 +941,7 @@ static int fsl_sai_suspend(struct device *dev) return 0; } -static int fsl_sai_resume(struct device *dev) +static int fsl_sai_runtime_resume(struct device *dev) { struct fsl_sai *sai = dev_get_drvdata(dev); @@ -941,14 +953,18 @@ static int fsl_sai_resume(struct device *dev) regmap_write(sai->regmap, FSL_SAI_RCSR, 0); return regcache_sync(sai->regmap); } -#endif /* CONFIG_PM_SLEEP */ +#endif /* CONFIG_PM */ static const struct dev_pm_ops fsl_sai_pm_ops = { - SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) + SET_RUNTIME_PM_OPS(fsl_sai_runtime_suspend, + fsl_sai_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) }; static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, + .remove = fsl_sai_remove, .driver = { .name = "fsl-sai", .pm = &fsl_sai_pm_ops, diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 9981668ab590..040d06b89f00 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -71,6 +71,7 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, iprop = of_get_property(dma_np, "cell-index", NULL); if (!iprop) { of_node_put(dma_np); + of_node_put(dma_channel_np); return -EINVAL; } *dma_id = be32_to_cpup(iprop); diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c new file mode 100644 index 000000000000..9aaf3e5b45b9 --- /dev/null +++ b/sound/soc/fsl/imx-audmix.c @@ -0,0 +1,331 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright 2017 NXP + * + * 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 + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/pm_runtime.h> +#include "fsl_sai.h" +#include "fsl_audmix.h" + +struct imx_audmix { + struct platform_device *pdev; + struct snd_soc_card card; + struct platform_device *audmix_pdev; + struct platform_device *out_pdev; + struct clk *cpu_mclk; + int num_dai; + struct snd_soc_dai_link *dai; + int num_dai_conf; + struct snd_soc_codec_conf *dai_conf; + int num_dapm_routes; + struct snd_soc_dapm_route *dapm_routes; +}; + +static const u32 imx_audmix_rates[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 64000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list imx_audmix_rate_constraints = { + .count = ARRAY_SIZE(imx_audmix_rates), + .list = imx_audmix_rates, +}; + +static int imx_audmix_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_audmix *priv = snd_soc_card_get_drvdata(rtd->card); + struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = rtd->card->dev; + unsigned long clk_rate = clk_get_rate(priv->cpu_mclk); + int ret; + + if (clk_rate % 24576000 == 0) { + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &imx_audmix_rate_constraints); + if (ret < 0) + return ret; + } else { + dev_warn(dev, "mclk may be not supported %lu\n", clk_rate); + } + + ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, + 1, 8); + if (ret < 0) + return ret; + + return snd_pcm_hw_constraint_mask64(runtime, SNDRV_PCM_HW_PARAM_FORMAT, + FSL_AUDMIX_FORMATS); +} + +static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF; + u32 channels = params_channels(params); + int ret, dir; + + /* For playback the AUDMIX is slave, and for record is master */ + fmt |= tx ? SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBM_CFM; + dir = tx ? SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN; + + /* set DAI configuration */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (ret) { + dev_err(dev, "failed to set cpu dai fmt: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_MAST1, 0, dir); + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + /* + * Per datasheet, AUDMIX expects 8 slots and 32 bits + * for every slot in TDM mode. + */ + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, BIT(channels) - 1, + BIT(channels) - 1, 8, 32); + if (ret) + dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret); + + return ret; +} + +static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF; + int ret; + + if (!tx) + return 0; + + /* For playback the AUDMIX is slave */ + fmt |= SND_SOC_DAIFMT_CBM_CFM; + + /* set AUDMIX DAI configuration */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (ret) + dev_err(dev, "failed to set AUDMIX DAI fmt: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops imx_audmix_fe_ops = { + .startup = imx_audmix_fe_startup, + .hw_params = imx_audmix_fe_hw_params, +}; + +static struct snd_soc_ops imx_audmix_be_ops = { + .hw_params = imx_audmix_be_hw_params, +}; + +static int imx_audmix_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *audmix_np = NULL, *out_cpu_np = NULL; + struct platform_device *audmix_pdev = NULL; + struct platform_device *cpu_pdev; + struct of_phandle_args args; + struct imx_audmix *priv; + int i, num_dai, ret; + const char *fe_name_pref = "HiFi-AUDMIX-FE-"; + char *be_name, *be_pb, *be_cp, *dai_name, *capture_dai_name; + + if (pdev->dev.parent) { + audmix_np = pdev->dev.parent->of_node; + } else { + dev_err(&pdev->dev, "Missing parent device.\n"); + return -EINVAL; + } + + if (!audmix_np) { + dev_err(&pdev->dev, "Missing DT node for parent device.\n"); + return -EINVAL; + } + + audmix_pdev = of_find_device_by_node(audmix_np); + if (!audmix_pdev) { + dev_err(&pdev->dev, "Missing AUDMIX platform device for %s\n", + np->full_name); + return -EINVAL; + } + put_device(&audmix_pdev->dev); + + num_dai = of_count_phandle_with_args(audmix_np, "dais", NULL); + if (num_dai != FSL_AUDMIX_MAX_DAIS) { + dev_err(&pdev->dev, "Need 2 dais to be provided for %s\n", + audmix_np->full_name); + return -EINVAL; + } + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->num_dai = 2 * num_dai; + priv->dai = devm_kzalloc(&pdev->dev, priv->num_dai * + sizeof(struct snd_soc_dai_link), GFP_KERNEL); + if (!priv->dai) + return -ENOMEM; + + priv->num_dai_conf = num_dai; + priv->dai_conf = devm_kzalloc(&pdev->dev, priv->num_dai_conf * + sizeof(struct snd_soc_codec_conf), + GFP_KERNEL); + if (!priv->dai_conf) + return -ENOMEM; + + priv->num_dapm_routes = 3 * num_dai; + priv->dapm_routes = devm_kzalloc(&pdev->dev, priv->num_dapm_routes * + sizeof(struct snd_soc_dapm_route), + GFP_KERNEL); + if (!priv->dapm_routes) + return -ENOMEM; + + for (i = 0; i < num_dai; i++) { + ret = of_parse_phandle_with_args(audmix_np, "dais", NULL, i, + &args); + if (ret < 0) { + dev_err(&pdev->dev, "of_parse_phandle_with_args failed\n"); + return ret; + } + + cpu_pdev = of_find_device_by_node(args.np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find SAI platform device\n"); + return -EINVAL; + } + put_device(&cpu_pdev->dev); + + dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s%s", + fe_name_pref, args.np->full_name + 1); + + dev_info(pdev->dev.parent, "DAI FE name:%s\n", dai_name); + + if (i == 0) { + out_cpu_np = args.np; + capture_dai_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", + dai_name, "CPU-Capture"); + } + + priv->dai[i].name = dai_name; + priv->dai[i].stream_name = "HiFi-AUDMIX-FE"; + priv->dai[i].codec_dai_name = "snd-soc-dummy-dai"; + priv->dai[i].codec_name = "snd-soc-dummy"; + priv->dai[i].cpu_of_node = args.np; + priv->dai[i].cpu_dai_name = dev_name(&cpu_pdev->dev); + priv->dai[i].platform_of_node = args.np; + priv->dai[i].dynamic = 1; + priv->dai[i].dpcm_playback = 1; + priv->dai[i].dpcm_capture = (i == 0 ? 1 : 0); + priv->dai[i].ignore_pmdown_time = 1; + priv->dai[i].ops = &imx_audmix_fe_ops; + + /* Add AUDMIX Backend */ + be_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "audmix-%d", i); + be_pb = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "AUDMIX-Playback-%d", i); + be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "AUDMIX-Capture-%d", i); + + priv->dai[num_dai + i].name = be_name; + priv->dai[num_dai + i].codec_dai_name = "snd-soc-dummy-dai"; + priv->dai[num_dai + i].codec_name = "snd-soc-dummy"; + priv->dai[num_dai + i].cpu_of_node = audmix_np; + priv->dai[num_dai + i].cpu_dai_name = be_name; + priv->dai[num_dai + i].platform_name = "snd-soc-dummy"; + priv->dai[num_dai + i].no_pcm = 1; + priv->dai[num_dai + i].dpcm_playback = 1; + priv->dai[num_dai + i].dpcm_capture = 1; + priv->dai[num_dai + i].ignore_pmdown_time = 1; + priv->dai[num_dai + i].ops = &imx_audmix_be_ops; + + priv->dai_conf[i].of_node = args.np; + priv->dai_conf[i].name_prefix = dai_name; + + priv->dapm_routes[i].source = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", + dai_name, "CPU-Playback"); + priv->dapm_routes[i].sink = be_pb; + priv->dapm_routes[num_dai + i].source = be_pb; + priv->dapm_routes[num_dai + i].sink = be_cp; + priv->dapm_routes[2 * num_dai + i].source = be_cp; + priv->dapm_routes[2 * num_dai + i].sink = capture_dai_name; + } + + cpu_pdev = of_find_device_by_node(out_cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find SAI platform device\n"); + return -EINVAL; + } + put_device(&cpu_pdev->dev); + + priv->cpu_mclk = devm_clk_get(&cpu_pdev->dev, "mclk1"); + if (IS_ERR(priv->cpu_mclk)) { + ret = PTR_ERR(priv->cpu_mclk); + dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret); + return -EINVAL; + } + + priv->audmix_pdev = audmix_pdev; + priv->out_pdev = cpu_pdev; + + priv->card.dai_link = priv->dai; + priv->card.num_links = priv->num_dai; + priv->card.codec_conf = priv->dai_conf; + priv->card.num_configs = priv->num_dai_conf; + priv->card.dapm_routes = priv->dapm_routes; + priv->card.num_dapm_routes = priv->num_dapm_routes; + priv->card.dev = pdev->dev.parent; + priv->card.owner = THIS_MODULE; + priv->card.name = "imx-audmix"; + + platform_set_drvdata(pdev, &priv->card); + snd_soc_card_set_drvdata(&priv->card, priv); + + ret = devm_snd_soc_register_card(pdev->dev.parent, &priv->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed\n"); + return ret; + } + + return ret; +} + +static struct platform_driver imx_audmix_driver = { + .probe = imx_audmix_probe, + .driver = { + .name = "imx-audmix", + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(imx_audmix_driver); + +MODULE_DESCRIPTION("NXP AUDMIX ASoC machine driver"); +MODULE_AUTHOR("Viorel Suman <viorel.suman@nxp.com>"); +MODULE_ALIAS("platform:imx-audmix"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 99e07b01a2ce..04e59e66711d 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -1,21 +1,11 @@ -/* - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * Copyright 2009 Pengutronix, Sascha Hauer <s.hauer@pengutronix.de> - * - * Initial development of this code was funded by - * Phytec Messtechnik GmbH, http://www.phytec.de - * - * 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. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright 2012 Freescale Semiconductor, Inc. +// Copyright 2012 Linaro Ltd. +// Copyright 2009 Pengutronix, Sascha Hauer <s.hauer@pengutronix.de> +// +// Initial development of this code was funded by +// Phytec Messtechnik GmbH, http://www.phytec.de #include <linux/clk.h> #include <linux/debugfs.h> diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index 9953438086e4..c9d8739b04a9 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.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 <linux/gpio.h> #include <linux/module.h> diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 9d19b808f634..545815a27074 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -1,17 +1,11 @@ -/* - * imx-mc13783.c -- SoC audio for imx based boards with mc13783 codec - * - * Copyright 2012 Philippe Retornaz, <philippe.retornaz@epfl.ch> - * - * Heavly based on phycore-mc13783: - * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de> - * - * 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+ +// +// imx-mc13783.c -- SoC audio for imx based boards with mc13783 codec +// +// Copyright 2012 Philippe Retornaz, <philippe.retornaz@epfl.ch> +// +// Heavly based on phycore-mc13783: +// Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de> #include <linux/module.h> #include <linux/moduleparam.h> diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 0578f3486847..c49aea4fba56 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -1,16 +1,11 @@ -/* - * imx-pcm-fiq.c -- ALSA Soc Audio Layer - * - * Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de> - * - * This code is based on code copyrighted by Freescale, - * Liam Girdwood, Javier Martin and probably others. - * - * 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+ +// imx-pcm-fiq.c -- ALSA Soc Audio Layer +// +// Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de> +// +// This code is based on code copyrighted by Freescale, +// Liam Girdwood, Javier Martin and probably others. + #include <linux/clk.h> #include <linux/delay.h> #include <linux/device.h> diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 133c4470acad..5dd406774d3e 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -1,13 +1,9 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ /* * Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de> * * This code is based on code copyrighted by Freescale, * Liam Girdwood, Javier Martin and probably others. - * - * 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. */ #ifndef _IMX_PCM_H diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 797d66e43d49..4f7f210beb18 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -1,13 +1,6 @@ -/* - * Copyright (C) 2013 Freescale Semiconductor, Inc. - * - * 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 (C) 2013 Freescale Semiconductor, Inc. #include <linux/module.h> #include <linux/of_platform.h> diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 06790615e04e..9038b61317be 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -1,35 +1,28 @@ -/* - * imx-ssi.c -- ALSA Soc Audio Layer - * - * Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de> - * - * This code is based on code copyrighted by Freescale, - * Liam Girdwood, Javier Martin and probably others. - * - * 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. - * - * - * The i.MX SSI core has some nasty limitations in AC97 mode. While most - * sane processor vendors have a FIFO per AC97 slot, the i.MX has only - * one FIFO which combines all valid receive slots. We cannot even select - * which slots we want to receive. The WM9712 with which this driver - * was developed with always sends GPIO status data in slot 12 which - * we receive in our (PCM-) data stream. The only chance we have is to - * manually skip this data in the FIQ handler. With sampling rates different - * from 48000Hz not every frame has valid receive data, so the ratio - * between pcm data and GPIO status data changes. Our FIQ handler is not - * able to handle this, hence this driver only works with 48000Hz sampling - * rate. - * Reading and writing AC97 registers is another challenge. The core - * provides us status bits when the read register is updated with *another* - * value. When we read the same register two times (and the register still - * contains the same value) these status bits are not set. We work - * around this by not polling these bits but only wait a fixed delay. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// imx-ssi.c -- ALSA Soc Audio Layer +// +// Copyright 2009 Sascha Hauer <s.hauer@pengutronix.de> +// +// This code is based on code copyrighted by Freescale, +// Liam Girdwood, Javier Martin and probably others. +// +// The i.MX SSI core has some nasty limitations in AC97 mode. While most +// sane processor vendors have a FIFO per AC97 slot, the i.MX has only +// one FIFO which combines all valid receive slots. We cannot even select +// which slots we want to receive. The WM9712 with which this driver +// was developed with always sends GPIO status data in slot 12 which +// we receive in our (PCM-) data stream. The only chance we have is to +// manually skip this data in the FIQ handler. With sampling rates different +// from 48000Hz not every frame has valid receive data, so the ratio +// between pcm data and GPIO status data changes. Our FIQ handler is not +// able to handle this, hence this driver only works with 48000Hz sampling +// rate. +// Reading and writing AC97 registers is another challenge. The core +// provides us status bits when the read register is updated with *another* +// value. When we read the same register two times (and the register still +// contains the same value) these status bits are not set. We work +// around this by not polling these bits but only wait a fixed delay. #include <linux/clk.h> #include <linux/delay.h> diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index be6562365b6a..19cd0937e740 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -1,8 +1,4 @@ -/* - * 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 */ #ifndef _IMX_SSI_H #define _IMX_SSI_H diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index c1a4544eb16b..ccf9301889fe 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -1,10 +1,10 @@ -/* - * Freescale MPC5200 PSC DMA - * ALSA SoC Platform driver - * - * Copyright (C) 2008 Secret Lab Technologies Ltd. - * Copyright (C) 2009 Jon Smirl, Digispeaker - */ +// SPDX-License-Identifier: GPL-2.0-only +// +// Freescale MPC5200 PSC DMA +// ALSA SoC Platform driver +// +// Copyright (C) 2008 Secret Lab Technologies Ltd. +// Copyright (C) 2009 Jon Smirl, Digispeaker #include <linux/module.h> #include <linux/of_device.h> diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 07ee355ee385..e5b9c04d1565 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -1,13 +1,9 @@ -/* - * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. - * - * Copyright (C) 2009 Jon Smirl, Digispeaker - * Author: Jon Smirl <jonsmirl@gmail.com> - * - * 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 +// +// linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. +// +// Copyright (C) 2009 Jon Smirl, Digispeaker +// Author: Jon Smirl <jonsmirl@gmail.com> #include <linux/module.h> #include <linux/of_device.h> diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index d8232943ccb6..9bc01f374b39 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -1,10 +1,10 @@ -/* - * Freescale MPC5200 PSC in I2S mode - * ALSA SoC Digital Audio Interface (DAI) driver - * - * Copyright (C) 2008 Secret Lab Technologies Ltd. - * Copyright (C) 2009 Jon Smirl, Digispeaker - */ +// SPDX-License-Identifier: GPL-2.0-only +// +// Freescale MPC5200 PSC in I2S mode +// ALSA SoC Digital Audio Interface (DAI) driver +// +// Copyright (C) 2008 Secret Lab Technologies Ltd. +// Copyright (C) 2009 Jon Smirl, Digispeaker #include <linux/module.h> #include <linux/of_device.h> diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index a639b52c16f6..f6261a3eeb0f 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -1,14 +1,10 @@ -/** - * Freescale MPC8610HPCD ALSA SoC Machine driver - * - * Author: Timur Tabi <timur@freescale.com> - * - * Copyright 2007-2010 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 MPC8610HPCD ALSA SoC Machine driver +// +// Author: Timur Tabi <timur@freescale.com> +// +// Copyright 2007-2010 Freescale Semiconductor, Inc. #include <linux/module.h> #include <linux/interrupt.h> diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c index d7ec3d20065c..37a4520aef62 100644 --- a/sound/soc/fsl/mx27vis-aic32x4.c +++ b/sound/soc/fsl/mx27vis-aic32x4.c @@ -1,25 +1,10 @@ -/* - * mx27vis-aic32x4.c - * - * Copyright 2011 Vista Silicon S.L. - * - * Author: Javier Martin <javier.martin@vista-silicon.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., 51 Franklin Street, Fifth Floor, Boston, - * MA 02110-1301, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// mx27vis-aic32x4.c +// +// Copyright 2011 Vista Silicon S.L. +// +// Author: Javier Martin <javier.martin@vista-silicon.com> #include <linux/module.h> #include <linux/moduleparam.h> diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 41c623c55c16..80384f70878d 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -1,14 +1,10 @@ -/** - * Freescale P1022DS ALSA SoC Machine driver - * - * Author: Timur Tabi <timur@freescale.com> - * - * 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 P1022DS ALSA SoC Machine driver +// +// Author: Timur Tabi <timur@freescale.com> +// +// Copyright 2010 Freescale Semiconductor, Inc. #include <linux/module.h> #include <linux/fsl/guts.h> diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c index 4afbdd610bfa..1c32c2d8c6b0 100644 --- a/sound/soc/fsl/p1022_rdk.c +++ b/sound/soc/fsl/p1022_rdk.c @@ -1,21 +1,17 @@ -/** - * Freescale P1022RDK ALSA SoC Machine driver - * - * Author: Timur Tabi <timur@freescale.com> - * - * Copyright 2012 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. - * - * Note: in order for audio to work correctly, the output controls need - * to be enabled, because they control the clock. So for playback, for - * example: - * - * amixer sset 'Left Output Mixer PCM' on - * amixer sset 'Right Output Mixer PCM' on - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale P1022RDK ALSA SoC Machine driver +// +// Author: Timur Tabi <timur@freescale.com> +// +// Copyright 2012 Freescale Semiconductor, Inc. +// +// Note: in order for audio to work correctly, the output controls need +// to be enabled, because they control the clock. So for playback, for +// example: +// +// amixer sset 'Left Output Mixer PCM' on +// amixer sset 'Right Output Mixer PCM' on #include <linux/module.h> #include <linux/fsl/guts.h> diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index e339f36cea95..a7fe4ad25c52 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -1,14 +1,10 @@ -/* - * Phytec pcm030 driver for the PSC of the Freescale MPC52xx - * configured as AC97 interface - * - * Copyright 2008 Jon Smirl, Digispeaker - * Author: Jon Smirl <jonsmirl@gmail.com> - * - * 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 +// +// Phytec pcm030 driver for the PSC of the Freescale MPC52xx +// configured as AC97 interface +// +// Copyright 2008 Jon Smirl, Digispeaker +// Author: Jon Smirl <jonsmirl@gmail.com> #include <linux/init.h> #include <linux/module.h> diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c index 66fb6c4614d2..fe7ba6db7c96 100644 --- a/sound/soc/fsl/phycore-ac97.c +++ b/sound/soc/fsl/phycore-ac97.c @@ -1,14 +1,8 @@ -/* - * phycore-ac97.c -- SoC audio for imx_phycore in AC97 mode - * - * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de> - * - * 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+ +// +// phycore-ac97.c -- SoC audio for imx_phycore in AC97 mode +// +// Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de> #include <linux/module.h> #include <linux/moduleparam.h> diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index 2f80b21b2921..aad24ccbef90 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -1,16 +1,11 @@ -/* - * wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS - * - * Copyright (c) 2010 Wolfson Microelectronics plc - * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> - * - * Based on an earlier driver for the same hardware 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 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+ +// +// wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS +// +// Copyright (c) 2010 Wolfson Microelectronics plc +// Author: Mark Brown <broonie@opensource.wolfsonmicro.com> +// +// Based on an earlier driver for the same hardware by Liam Girdwood. #include <linux/platform_device.h> #include <linux/clk.h> |