From 3a9236e97207f2469254b4098995159b80174d95 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 16 Sep 2019 19:18:51 +0900 Subject: ALSA: dice: fix wrong packet parameter for Alesis iO26 At higher sampling rate (e.g. 192.0 kHz), Alesis iO26 transfers 4 data channels per data block in CIP. Both iO14 and iO26 have the same contents in their configuration ROM. For this reason, ALSA Dice driver attempts to distinguish them according to the value of TX0_AUDIO register at probe callback. Although the way is valid at lower and middle sampling rate, it's lastly invalid at higher sampling rate because because the two models returns the same value for read transaction to the register. In the most cases, users just plug-in the device and ALSA dice driver detects it. In the case, the device runs at lower sampling rate and the driver detects expectedly. For this reason, this commit leaves the way to detect as is. Fixes: 28b208f600a3 ("ALSA: dice: add parameters of stream formats for models produced by Alesis") Cc: # v4.18+ Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20190916101851.30409-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-alesis.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c index 218292bdace6..f5b325263b67 100644 --- a/sound/firewire/dice/dice-alesis.c +++ b/sound/firewire/dice/dice-alesis.c @@ -15,7 +15,7 @@ alesis_io14_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = { static const unsigned int alesis_io26_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = { - {10, 10, 8}, /* Tx0 = Analog + S/PDIF. */ + {10, 10, 4}, /* Tx0 = Analog + S/PDIF. */ {16, 8, 0}, /* Tx1 = ADAT1 + ADAT2. */ }; -- cgit v1.2.3 From 543242211879ebe021b00f7e33eb4e733bcd6c47 Mon Sep 17 00:00:00 2001 From: James McDonnell Date: Mon, 16 Sep 2019 14:53:38 +0000 Subject: ALSA: hda/realtek - Fix alienware headset mic Headset microphone quirk for alienware 15r3. Without this using a headset with mic attached will not work. Signed-off-by: James McDonnell Link: https://lore.kernel.org/r/QB1PR01MB2337D0367C2E3ADB0010134F808C0@QB1PR01MB2337.CANPRD01.PROD.OUTLOOK.COM Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index da1695418731..e27ef434b60d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5817,6 +5817,7 @@ enum { ALC292_FIXUP_DELL_E7X, ALC292_FIXUP_DISABLE_AAMIX, ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, + ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE, ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, @@ -6506,6 +6507,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC292_FIXUP_DISABLE_AAMIX }, + [ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* headset mic w/o jack detect */ + { } + }, + .chained_before = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, [ALC298_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7770,6 +7780,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x17, 0x90170110}, {0x1a, 0x03011020}, {0x21, 0x03211030}), + SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE, + {0x12, 0xb7a60140}, + {0x17, 0x90170110}, + {0x1a, 0x03a11030}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0299, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x12, 0xb7a60130}, -- cgit v1.2.3 From 029d2c0fd61eac74700fb4ffff36fc63bfff7e5e Mon Sep 17 00:00:00 2001 From: Ilya Pshonkin Date: Tue, 17 Sep 2019 10:49:34 +0300 Subject: ALSA: usb-audio: Add Hiby device family to quirks for native DSD support This patch adds quirk VID ID for Hiby portable players family with native DSD playback support. Signed-off-by: Ilya Pshonkin Cc: Link: https://lore.kernel.org/r/20190917074937.157802-1-ilya.pshonkin@netforce.ua Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 25faf2d3c639..5fd4ccce452d 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1658,6 +1658,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x25ce: /* Mytek devices */ case 0x278b: /* Rotel? */ case 0x2ab6: /* T+A devices */ + case 0xc502: /* HiBy devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- cgit v1.2.3 From e75f4940e8ad0dd76527302a10c06b58bf7eb590 Mon Sep 17 00:00:00 2001 From: Mihai Serban Date: Fri, 13 Sep 2019 22:28:05 +0300 Subject: ASoC: fsl_sai: Fix noise when using EDMA EDMA requires the period size to be multiple of maxburst. Otherwise the remaining bytes are not transferred and thus noise is produced. We can handle this issue by adding a constraint on SNDRV_PCM_HW_PARAM_PERIOD_SIZE to be multiple of tx/rx maxburst value. Signed-off-by: Mihai Serban Signed-off-by: Daniel Baluta Acked-by: Nicolin Chen Link: https://lore.kernel.org/r/20190913192807.8423-2-daniel.baluta@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 15 +++++++++++++++ sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 16 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ef0b74693093..b517e4bc1b87 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -628,6 +628,16 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, FSL_SAI_CR3_TRCE_MASK, FSL_SAI_CR3_TRCE); + /* + * EDMA controller needs period size to be a multiple of + * tx/rx maxburst + */ + if (sai->soc_data->use_edma) + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + tx ? sai->dma_params_tx.maxburst : + sai->dma_params_rx.maxburst); + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints); @@ -1026,30 +1036,35 @@ static int fsl_sai_remove(struct platform_device *pdev) static const struct fsl_sai_soc_data fsl_sai_vf610_data = { .use_imx_pcm = false, + .use_edma = false, .fifo_depth = 32, .reg_offset = 0, }; static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { .use_imx_pcm = true, + .use_edma = false, .fifo_depth = 32, .reg_offset = 0, }; static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { .use_imx_pcm = true, + .use_edma = false, .fifo_depth = 16, .reg_offset = 8, }; static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { .use_imx_pcm = true, + .use_edma = false, .fifo_depth = 128, .reg_offset = 8, }; static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .use_imx_pcm = true, + .use_edma = true, .fifo_depth = 64, .reg_offset = 0, }; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index b12cb578f6d0..76b15deea80c 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -157,6 +157,7 @@ struct fsl_sai_soc_data { bool use_imx_pcm; + bool use_edma; unsigned int fifo_depth; unsigned int reg_offset; }; -- cgit v1.2.3 From a0a4bf57a977ed37bcbdfc8027a44485fe086a3d Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 17 Sep 2019 05:03:53 +0800 Subject: ASoC: core: delete component->card_list in soc_remove_component only We add component->card_list in the end of soc_probe_component(). In other words, component->card_list will not be added if there is an error in the soc_probe_component() function. So we can't delete component->card_list in the error handling of soc_probe_component(). Fixes: 22d1423187e5 ("ASoC: soc-core: add soc_cleanup_component()") Signed-off-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190916210353.6318-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 35f48e9c5ead..aff4b4bf4d07 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -978,7 +978,6 @@ static void soc_cleanup_component(struct snd_soc_component *component) /* For framework level robustness */ snd_soc_component_set_jack(component, NULL, NULL); - list_del(&component->card_list); snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); component->card = NULL; @@ -991,7 +990,7 @@ static void soc_remove_component(struct snd_soc_component *component) return; snd_soc_component_remove(component); - + list_del(&component->card_list); soc_cleanup_component(component); } -- cgit v1.2.3 From 0dce49efc70536a8c3b4bb5354a71b727ba31b80 Mon Sep 17 00:00:00 2001 From: Gregory CLEMENT Date: Wed, 18 Sep 2019 12:03:44 +0200 Subject: ASoC: atmel_ssc_dai: Remove wrong spinlock usage A potential bug was reported in the email "[BUG] atmel_ssc_dai: a possible sleep-in-atomic bug in atmel_ssc_shutdown"[1] Indeed in the function atmel_ssc_shutdown() free_irq() was called in a critical section protected by spinlock. However this spinlock is only used in atmel_ssc_shutdown() and atmel_ssc_startup() functions. After further analysis, it occurred that the call to these function are already protected by mutex used on the calling functions. Then we can remove the spinlock which will fix this bug as a side effect. Thanks to this patch the following message disappears: "BUG: sleeping function called from invalid context at kernel/locking/mutex.c:909" [1]: https://www.spinics.net/lists/alsa-devel/msg71286.html Reviewed-by: Alexandre Belloni Signed-off-by: Gregory CLEMENT Link: https://lore.kernel.org/r/20190918100344.23629-1-gregory.clement@bootlin.com Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 12 ++---------- sound/soc/atmel/atmel_ssc_dai.h | 1 - 2 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 48e9eef34c0f..ca603397651c 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -116,19 +116,16 @@ static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { { .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, @@ -317,13 +314,10 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(dai, substream, dma_params); - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) return -EBUSY; - } + ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); return 0; } @@ -355,7 +349,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, dir_mask = 1 << dir; - spin_lock_irq(&ssc_p->lock); ssc_p->dir_mask &= ~dir_mask; if (!ssc_p->dir_mask) { if (ssc_p->initialized) { @@ -369,7 +362,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; ssc_p->forced_divider = 0; } - spin_unlock_irq(&ssc_p->lock); /* Shutdown the SSC clock. */ pr_debug("atmel_ssc_dai: Stopping clock\n"); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index ae764cb541c7..3470b966e449 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -93,7 +93,6 @@ struct atmel_ssc_state { struct atmel_ssc_info { char *name; struct ssc_device *ssc; - spinlock_t lock; /* lock for dir_mask */ unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ unsigned short initialized; /* true if SSC has been initialized */ unsigned short daifmt; -- cgit v1.2.3 From 947ec14c7369a87625f03abaab5b3f4d33ac73ba Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 18 Sep 2019 15:02:38 +0900 Subject: ASoC: rsnd: do error check after rsnd_channel_normalization() SSI need to use rsnd_channel_normalization() for TDM-split mode, thus, channel check need to do after that. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/874l1aw39d.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index f6a7466622ea..fc5d089868df 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -286,6 +286,11 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; + if (rsnd_runtime_is_tdm_split(io)) + chan = rsnd_io_converted_chan(io); + + chan = rsnd_channel_normalization(chan); + if (ssi->usrcnt > 0) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); @@ -300,11 +305,6 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, return 0; } - if (rsnd_runtime_is_tdm_split(io)) - chan = rsnd_io_converted_chan(io); - - chan = rsnd_channel_normalization(chan); - main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx); if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); -- cgit v1.2.3 From d2c63b7dfd06788a466d5ec8a850491f084c5fc2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Sep 2019 09:30:40 +0200 Subject: ALSA: hda - Apply AMD controller workaround for Raven platform It's reported that the garbled sound on HP Envy x360 13z-ag000 (Ryzen Laptop) is fixed by the same workaround applied to other AMD chips. Update the driver_data entry for Raven (1022:15e3) to use the newly introduced preset, AZX_DCAPS_PRESET_AMD_SB. Since it already contains AZX_DCAPS_PM_RUNTIME, we can drop that bit, too. Reported-and-tested-by: Dennis Padiernos Cc: Link: https://lore.kernel.org/r/20190920073040.31764-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 91e71be42fa4..240f4ca76391 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2485,8 +2485,7 @@ static const struct pci_device_id azx_ids[] = { AZX_DCAPS_PM_RUNTIME }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), - .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | - AZX_DCAPS_PM_RUNTIME }, + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, -- cgit v1.2.3 From bd9c10bc663dd2eaac8fe39dad0f18cd21527446 Mon Sep 17 00:00:00 2001 From: Jan-Marek Glogowski Date: Sun, 15 Sep 2019 16:57:28 +0200 Subject: ALSA: hda/realtek - PCI quirk for Medion E4254 The laptop has a combined jack to attach headsets on the right. The BIOS encodes them as two different colored jacks at the front, but otherwise it seems to be configured ok. But any adaption of the pins config on its own doesn't fix the jack detection to work in Linux. Still Windows works correct. This is somehow fixed by chaining ALC256_FIXUP_ASUS_HEADSET_MODE, which seems to register the microphone jack as a headset part and also results in fixing jack sensing, visible in dmesg as: -snd_hda_codec_realtek hdaudioC0D0: Mic=0x19 +snd_hda_codec_realtek hdaudioC0D0: Headset Mic=0x19 [ Actually the essential change is the location of the jack; the driver created "Front Mic Jack" without the matching volume / mute control element due to its jack location, which confused PA. -- tiwai ] Signed-off-by: Jan-Marek Glogowski Cc: Link: https://lore.kernel.org/r/8f4f9b20-0aeb-f8f1-c02f-fd53c09679f1@fbihome.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e27ef434b60d..b000b36ac3c6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5872,6 +5872,7 @@ enum { ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, + ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6937,6 +6938,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x04a11040 }, + { 0x21, 0x04211020 }, + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7200,6 +7211,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ + SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. @@ -7368,6 +7380,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"}, {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, + {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, {} }; #define ALC225_STANDARD_PINS \ -- cgit v1.2.3 From 7a5d9815cc010b055c2a99ccf418c4629365fa43 Mon Sep 17 00:00:00 2001 From: Bard liao Date: Wed, 18 Sep 2019 21:31:31 +0800 Subject: ASoC: core: use list_del_init and move it back to soc_cleanup_component commit a0a4bf57a977 ("ASoC: core: delete component->card_list in soc_remove_component only") was trying to fix a kernel oops when list_del was called twice without re-init the list. Use list_del_init() can solve it, too. Besides, it will be more readable if we cleanup all component related resource at soc_cleanup_component(). Suggested-by: Kuninori Morimoto Signed-off-by: Bard liao Link: https://lore.kernel.org/r/20190918133131.15045-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index aff4b4bf4d07..88978a3036c4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -978,6 +978,7 @@ static void soc_cleanup_component(struct snd_soc_component *component) /* For framework level robustness */ snd_soc_component_set_jack(component, NULL, NULL); + list_del_init(&component->card_list); snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); component->card = NULL; @@ -990,7 +991,7 @@ static void soc_remove_component(struct snd_soc_component *component) return; snd_soc_component_remove(component); - list_del(&component->card_list); + soc_cleanup_component(component); } -- cgit v1.2.3 From 7b2db65b59c30d58c129d3c8b2101feca686155a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 19 Sep 2019 10:16:52 +0300 Subject: ASoC: pcm3168a: The codec does not support S32_LE 24 bits is supported in all modes and 16 bit only when the codec is slave and the DAI is set to RIGHT_J. Remove the unsupported sample format. Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20190919071652.31724-1-peter.ujfalusi@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3168a.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 50ed86d45c26..88b75695fbf7 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -21,8 +21,7 @@ #define PCM3168A_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_3LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S24_LE) #define PCM3168A_FMT_I2S 0x0 #define PCM3168A_FMT_LEFT_J 0x1 -- cgit v1.2.3 From 147162f575152db80000fb3de26358264768ee9f Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Fri, 20 Sep 2019 09:50:18 +0200 Subject: ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies SND_SOC_DM365_VOICE_CODEC is a 'bool' option in a choice statement, meaning it cannot be set to =m, but it selects two other drivers that we may want to be loadable modules after all: WARNING: unmet direct dependencies detected for SND_SOC_CQ0093VC Depends on [m]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] Selected by [y]: - SND_SOC_DM365_VOICE_CODEC [=y] && Selected by [m]: - SND_SOC_ALL_CODECS [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && COMPILE_TEST [=y] Add an intermediate symbol that sets SND_SOC_CQ0093VC and MFD_DAVINCI_VOICECODEC to =m if SND_SOC=m. Signed-off-by: Arnd Bergmann Link: https://lore.kernel.org/r/20190920075046.3210393-1-arnd@arndb.de Signed-off-by: Mark Brown --- sound/soc/ti/Kconfig | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 87a9b9dd4e98..29f61053ab62 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -200,11 +200,18 @@ config SND_SOC_DM365_AIC3X_CODEC config SND_SOC_DM365_VOICE_CODEC bool "Voice Codec - CQ93VC" - select MFD_DAVINCI_VOICECODEC - select SND_SOC_CQ0093VC help Say Y if you want to add support for SoC On-chip voice codec endchoice +config SND_SOC_DM365_VOICE_CODEC_MODULE + def_tristate y + depends on SND_SOC_DM365_VOICE_CODEC && SND_SOC + select MFD_DAVINCI_VOICECODEC + select SND_SOC_CQ0093VC + help + The is an internal symbol needed to ensure that the codec + and MFD driver can be built as loadable modules if necessary. + endmenu -- cgit v1.2.3 From 7b485d175631be676424aedb8cd2f66d0c93da78 Mon Sep 17 00:00:00 2001 From: Shih-Yuan Lee (FourDollars) Date: Fri, 20 Sep 2019 21:40:53 +0800 Subject: ALSA: hda - Add laptop imic fixup for ASUS M9V laptop The same fixup to enable laptop imic is needed for ASUS M9V with AD1986A codec like another HP machine. Signed-off-by: Shih-Yuan Lee (FourDollars) Cc: Link: https://lore.kernel.org/r/20190920134052.GA8035@localhost Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e283966bdbb1..bc9dd8e6fd86 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -357,6 +357,7 @@ static const struct hda_fixup ad1986a_fixups[] = { static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC), + SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9V", AD1986A_FIXUP_LAPTOP_IMIC), SND_PCI_QUIRK(0x1043, 0x1443, "ASUS Z99He", AD1986A_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD), SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK), -- cgit v1.2.3 From f41f900568d9ffd896cc941db7021eb14bd55910 Mon Sep 17 00:00:00 2001 From: Jussi Laako Date: Tue, 24 Sep 2019 10:11:43 +0300 Subject: ALSA: usb-audio: Add DSD support for EVGA NU Audio EVGA NU Audio is actually a USB audio device on a PCIexpress card, with it's own USB controller. It supports both PCM and DSD. Signed-off-by: Jussi Laako Cc: Link: https://lore.kernel.org/r/20190924071143.30911-1-jussi@sonarnerd.net Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5fd4ccce452d..fbfde996fee7 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1658,6 +1658,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x25ce: /* Mytek devices */ case 0x278b: /* Rotel? */ case 0x2ab6: /* T+A devices */ + case 0x3842: /* EVGA */ case 0xc502: /* HiBy devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; -- cgit v1.2.3