From 3f4b57ad07d9237acf1b8cff3f8bf530cacef87a Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Mon, 20 Sep 2021 16:49:39 +0200 Subject: [PATCH 01/15] ASoC: pcm512x: Mend accesses to the I2S_1 and I2S_2 registers Commit 25d27c4f68d2 ("ASoC: pcm512x: Add support for more data formats") breaks the TSE-850 device, which is using a pcm5142 in I2S and CBM_CFS mode (maybe not relevant). Without this fix, the result is: pcm512x 0-004c: Failed to set data format: -16 And after that, no sound. This fix is not 100% correct. The datasheet of at least the pcm5142 states that four bits (0xcc) in the I2S_1 register are "RSV" ("Reserved. Do not access.") and no hint is given as to what the initial values are supposed to be. So, specifying defaults for these bits is wrong. But perhaps better than a broken driver? Fixes: 25d27c4f68d2 ("ASoC: pcm512x: Add support for more data formats") Cc: Liam Girdwood Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Kirill Marinushkin Cc: Peter Ujfalusi Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Peter Rosin Signed-off-by: Peter Ujfalusi Reviewed-by: Peter Ujfalusi Link: https://lore.kernel.org/r/2d221984-7a2e-7006-0f8a-ffb5f64ee885@axentia.se Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4dc844f3c1fc0..60dee41816dc2 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -116,6 +116,8 @@ static const struct reg_default pcm512x_reg_defaults[] = { { PCM512x_FS_SPEED_MODE, 0x00 }, { PCM512x_IDAC_1, 0x01 }, { PCM512x_IDAC_2, 0x00 }, + { PCM512x_I2S_1, 0x02 }, + { PCM512x_I2S_2, 0x00 }, }; static bool pcm512x_readable(struct device *dev, unsigned int reg) From 74b7ee0e7b61838a0a161a84d105aeff0d042646 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 10 Sep 2021 17:18:30 +0800 Subject: [PATCH 02/15] ASoC: fsl_xcvr: Fix channel swap issue with ARC With pause and resume test for ARC, there is occasionally channel swap issue. The reason is that currently driver set the DPATH out of reset first, then start the DMA, the first data got from FIFO may not be the Left channel. Moving DPATH out of reset operation after the dma enablement to fix this issue. Fixes: 28564486866f ("ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1631265510-27384-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_xcvr.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 7ba2fd15132d9..d0556c79fdb15 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -487,8 +487,9 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, return ret; } - /* clear DPATH RESET */ + /* set DPATH RESET */ m_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); + v_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, m_ctl, v_ctl); if (ret < 0) { dev_err(dai->dev, "Error while setting EXT_CTRL: %d\n", ret); @@ -590,10 +591,6 @@ static void fsl_xcvr_shutdown(struct snd_pcm_substream *substream, val |= FSL_XCVR_EXT_CTRL_CMDC_RESET(tx); } - /* set DPATH RESET */ - mask |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - val |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, mask, val); if (ret < 0) { dev_err(dai->dev, "Err setting DPATH RESET: %d\n", ret); @@ -643,6 +640,16 @@ static int fsl_xcvr_trigger(struct snd_pcm_substream *substream, int cmd, dev_err(dai->dev, "Failed to enable DMA: %d\n", ret); return ret; } + + /* clear DPATH RESET */ + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, + FSL_XCVR_EXT_CTRL_DPTH_RESET(tx), + 0); + if (ret < 0) { + dev_err(dai->dev, "Failed to clear DPATH RESET: %d\n", ret); + return ret; + } + break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: From ceef3240f9b7e592dd8d10d619c312c7336117fa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 24 Sep 2021 20:49:56 +0100 Subject: [PATCH 03/15] ASoC: pcm179x: Add missing entries SPI to device ID table Currently autoloading for SPI devices does not use the DT ID table, it uses SPI modalises. Supporting OF modalises is going to be difficult if not impractical, an attempt was made but has been reverted, so ensure that module autoloading works for this driver by adding SPI IDs for parts that only have a compatible listed. Fixes: 96c8395e2166 ("spi: Revert modalias changes") Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20210924194956.46079-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/pcm179x-spi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/pcm179x-spi.c b/sound/soc/codecs/pcm179x-spi.c index 0a542924ec5f9..ebf63ea90a1c4 100644 --- a/sound/soc/codecs/pcm179x-spi.c +++ b/sound/soc/codecs/pcm179x-spi.c @@ -36,6 +36,7 @@ static const struct of_device_id pcm179x_of_match[] = { MODULE_DEVICE_TABLE(of, pcm179x_of_match); static const struct spi_device_id pcm179x_spi_ids[] = { + { "pcm1792a", 0 }, { "pcm179x", 0 }, { }, }; From 0cc3687eadd0971d5d38ff90d14819d88f854960 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 24 Sep 2021 20:48:44 +0100 Subject: [PATCH 04/15] ASoC: cs4341: Add SPI device ID table Currently autoloading for SPI devices does not use the DT ID table, it uses SPI modalises. Supporting OF modalises is going to be difficult if not impractical, an attempt was made but has been reverted, so ensure that module autoloading works for this driver by adding SPI IDs for parts that only have a compatible listed. Fixes: 96c8395e2166 ("spi: Revert modalias changes") Signed-off-by: Mark Brown Cc: patches@opensource.cirrus.com Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/20210924194844.45974-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/cs4341.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c index 7d3e54d8eef36..29d05e32d3417 100644 --- a/sound/soc/codecs/cs4341.c +++ b/sound/soc/codecs/cs4341.c @@ -305,12 +305,19 @@ static int cs4341_spi_probe(struct spi_device *spi) return cs4341_probe(&spi->dev); } +static const struct spi_device_id cs4341_spi_ids[] = { + { "cs4341a" }, + { } +}; +MODULE_DEVICE_TABLE(spi, cs4341_spi_ids); + static struct spi_driver cs4341_spi_driver = { .driver = { .name = "cs4341-spi", .of_match_table = of_match_ptr(cs4341_dt_ids), }, .probe = cs4341_spi_probe, + .id_table = cs4341_spi_ids, }; #endif From 42871e95a3afea8956d8cc567ea725b33a837775 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Wed, 29 Sep 2021 22:15:12 +0200 Subject: [PATCH 05/15] ASoC: nau8824: Fix headphone vs headset, button-press detection no longer working Commit 1d25684e2251 ("ASoC: nau8824: Fix open coded prefix handling") replaced the nau8824_dapm_enable_pin() helper with direct calls to snd_soc_dapm_enable_pin(), but the helper was using snd_soc_dapm_force_enable_pin() and not forcing the MICBIAS + SAR supplies on breaks headphone vs headset and button-press detection. Replace the snd_soc_dapm_enable_pin() calls with snd_soc_dapm_force_enable_pin() to fix this. Cc: stable@vger.kernel.org Fixes: 1d25684e2251 ("ASoC: nau8824: Fix open coded prefix handling") Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20210929201512.460360-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8824.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index db88be48c9980..f946ef65a4c19 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -867,8 +867,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - snd_soc_dapm_enable_pin(dapm, "MICBIAS"); - snd_soc_dapm_enable_pin(dapm, "SAR"); + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); From db0767b8a6e620b99459d2e688c1983c2e5add0d Mon Sep 17 00:00:00 2001 From: Srinivasa Rao Mandadapu Date: Thu, 7 Oct 2021 19:20:19 +0530 Subject: [PATCH 06/15] ASoC: wcd938x: Fix jack detection issue This patch is to fix audio 3.5mm jack detection failure on wcd938x codec based target. Fixes: bcee7ed09b8e (ASoC: codecs: wcd938x: add Multi Button Headset Control support) Signed-off-by: Venkata Prasad Potturu Signed-off-by: Srinivasa Rao Mandadapu Reviewed-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/1633614619-27026-1-git-send-email-srivasam@codeaurora.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index f0daf8defcf1e..52de7d14b1398 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -4144,10 +4144,10 @@ static int wcd938x_codec_set_jack(struct snd_soc_component *comp, { struct wcd938x_priv *wcd = dev_get_drvdata(comp->dev); - if (!jack) + if (jack) return wcd_mbhc_start(wcd->wcd_mbhc, &wcd->mbhc_cfg, jack); - - wcd_mbhc_stop(wcd->wcd_mbhc); + else + wcd_mbhc_stop(wcd->wcd_mbhc); return 0; } From 2577b868a48ef3601116908738efbe570451e605 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Wed, 6 Oct 2021 18:04:25 +0300 Subject: [PATCH 07/15] ASoC: Intel: bytcht_es8316: Get platform data via dev_get_platdata() Access to platform data via dev_get_platdata() getter to make code cleaner. Signed-off-by: Andy Shevchenko Acked-by: Pierre-Louis Bossart Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20211006150428.16434-1-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 055248f104b24..b1f9c9cb33555 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -456,12 +456,12 @@ static const struct dmi_system_id byt_cht_es8316_quirk_table[] = { static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; static const char * const mic_name[] = { "in1", "in2" }; + struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); struct property_entry props[MAX_NO_PROPS] = {}; struct byt_cht_es8316_private *priv; const struct dmi_system_id *dmi_id; - struct device *dev = &pdev->dev; - struct snd_soc_acpi_mach *mach; struct fwnode_handle *fwnode; const char *platform_name; struct acpi_device *adev; @@ -476,7 +476,6 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; - mach = dev->platform_data; /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_cht_es8316_dais); i++) { if (!strcmp(byt_cht_es8316_dais[i].codecs->name, From 6f32c521061b704c0198be3ba9834f5a64ea5605 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Wed, 6 Oct 2021 18:04:26 +0300 Subject: [PATCH 08/15] ASoC: Intel: bytcht_es8316: Use temporary variable for struct device Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20211006150428.16434-2-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index b1f9c9cb33555..efd71e6e42b33 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -493,7 +493,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) put_device(&adev->dev); byt_cht_es8316_dais[dai_index].codecs->name = codec_name; } else { - dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + dev_err(dev, "Error cannot find '%s' dev\n", mach->id); return -ENXIO; } @@ -596,7 +596,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) byt_cht_es8316_card.long_name = long_name; #endif - sof_parent = snd_soc_acpi_sof_parent(&pdev->dev); + sof_parent = snd_soc_acpi_sof_parent(dev); /* set card and driver name */ if (sof_parent) { From 10f4a96543b744c8cc7ef8b0799af21d911dd37d Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Wed, 6 Oct 2021 18:04:27 +0300 Subject: [PATCH 09/15] ASoC: Intel: bytcht_es8316: Switch to use gpiod_get_optional() First of all, replace indexed API by plain one since we have index 0. Second, switch to optional variant and drop duplicated code. Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20211006150428.16434-3-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index efd71e6e42b33..421a04d96d841 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -566,16 +566,12 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios); priv->speaker_en_gpio = - gpiod_get_index(codec_dev, "speaker-enable", 0, - /* see comment in byt_cht_es8316_resume */ - GPIOD_OUT_LOW | GPIOD_FLAGS_BIT_NONEXCLUSIVE); - + gpiod_get_optional(codec_dev, "speaker-enable", + /* see comment in byt_cht_es8316_resume() */ + GPIOD_OUT_LOW | GPIOD_FLAGS_BIT_NONEXCLUSIVE); if (IS_ERR(priv->speaker_en_gpio)) { ret = PTR_ERR(priv->speaker_en_gpio); switch (ret) { - case -ENOENT: - priv->speaker_en_gpio = NULL; - break; default: dev_err(dev, "get speaker GPIO failed: %d\n", ret); fallthrough; From c25d4546ca452b2e8c03bc735e4c65bc6dd751dd Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Wed, 6 Oct 2021 18:04:28 +0300 Subject: [PATCH 10/15] ASoC: Intel: bytcht_es8316: Utilize dev_err_probe() to avoid log saturation dev_err_probe() avoids printing into log when the deferred probe is invoked. This is possible when clock provider is pending to appear. Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20211006150428.16434-4-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 421a04d96d841..4d313d0d0f23e 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -532,11 +532,8 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) /* get the clock */ priv->mclk = devm_clk_get(dev, "pmc_plt_clk_3"); - if (IS_ERR(priv->mclk)) { - ret = PTR_ERR(priv->mclk); - dev_err(dev, "clk_get pmc_plt_clk_3 failed: %d\n", ret); - return ret; - } + if (IS_ERR(priv->mclk)) + return dev_err_probe(dev, PTR_ERR(priv->mclk), "clk_get pmc_plt_clk_3 failed\n"); /* get speaker enable GPIO */ codec_dev = acpi_get_first_physical_node(adev); @@ -570,14 +567,9 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) /* see comment in byt_cht_es8316_resume() */ GPIOD_OUT_LOW | GPIOD_FLAGS_BIT_NONEXCLUSIVE); if (IS_ERR(priv->speaker_en_gpio)) { - ret = PTR_ERR(priv->speaker_en_gpio); - switch (ret) { - default: - dev_err(dev, "get speaker GPIO failed: %d\n", ret); - fallthrough; - case -EPROBE_DEFER: - goto err_put_codec; - } + ret = dev_err_probe(dev, PTR_ERR(priv->speaker_en_gpio), + "get speaker GPIO failed\n"); + goto err_put_codec; } snprintf(components_string, sizeof(components_string), From 5af82c81b2c49cfb1cad84d9eb6eab0e3d1c4842 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Oct 2021 16:17:12 +0200 Subject: [PATCH 11/15] ASoC: DAPM: Fix missing kctl change notifications The put callback of a kcontrol is supposed to return 1 when the value is changed, and this will be notified to user-space. However, some DAPM kcontrols always return 0 (except for errors), hence the user-space misses the update of a control value. This patch corrects the behavior by properly returning 1 when the value gets updated. Reported-and-tested-by: Hans de Goede Cc: Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20211006141712.2439-1-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7b67f1e19ae95..59d07648a7e7f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2561,6 +2561,7 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, const char *pin, int status) { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + int ret = 0; dapm_assert_locked(dapm); @@ -2573,13 +2574,14 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(w, "pin configuration"); dapm_widget_invalidate_input_paths(w); dapm_widget_invalidate_output_paths(w); + ret = 1; } w->connected = status; if (status == 0) w->force = 0; - return 0; + return ret; } /** @@ -3583,14 +3585,15 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; + int ret; if (ucontrol->value.integer.value[0]) - snd_soc_dapm_enable_pin(&card->dapm, pin); + ret = snd_soc_dapm_enable_pin(&card->dapm, pin); else - snd_soc_dapm_disable_pin(&card->dapm, pin); + ret = snd_soc_dapm_disable_pin(&card->dapm, pin); snd_soc_dapm_sync(&card->dapm); - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); @@ -4023,7 +4026,7 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, rtd->params_select = ucontrol->value.enumerated.item[0]; - return 0; + return 1; } static void From 214174d9f56c7f81f4860a26b6b8b961a6b92654 Mon Sep 17 00:00:00 2001 From: Srinivasa Rao Mandadapu Date: Thu, 7 Oct 2021 19:21:15 +0530 Subject: [PATCH 12/15] ASoC: codec: wcd938x: Add irq config support This patch fixes compilation error in wcd98x codec driver. Fixes: 045442228868 ("ASoC: codecs: wcd938x: add audio routing and Kconfig") Signed-off-by: Venkata Prasad Potturu Signed-off-by: Srinivasa Rao Mandadapu Reviewed-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/1633614675-27122-1-git-send-email-srivasam@codeaurora.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233a269d0..216cea04ad704 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1583,6 +1583,7 @@ config SND_SOC_WCD938X_SDW tristate "WCD9380/WCD9385 Codec - SDW" select SND_SOC_WCD938X select SND_SOC_WCD_MBHC + select REGMAP_IRQ depends on SOUNDWIRE select REGMAP_SOUNDWIRE help From c448b7aa3e66042fc0f849d9a0fb90d1af82e948 Mon Sep 17 00:00:00 2001 From: Yang Yingliang Date: Sat, 9 Oct 2021 14:58:40 +0800 Subject: [PATCH 13/15] ASoC: soc-core: fix null-ptr-deref in snd_soc_del_component_unlocked() 'component' is allocated in snd_soc_register_component(), but component->list is not initalized, this may cause snd_soc_del_component_unlocked() deref null ptr in the error handing case. KASAN: null-ptr-deref in range [0x0000000000000000-0x0000000000000007] RIP: 0010:__list_del_entry_valid+0x81/0xf0 Call Trace: snd_soc_del_component_unlocked+0x69/0x1b0 [snd_soc_core] snd_soc_add_component.cold+0x54/0x6c [snd_soc_core] snd_soc_register_component+0x70/0x90 [snd_soc_core] devm_snd_soc_register_component+0x5e/0xd0 [snd_soc_core] tas2552_probe+0x265/0x320 [snd_soc_tas2552] ? tas2552_component_probe+0x1e0/0x1e0 [snd_soc_tas2552] i2c_device_probe+0xa31/0xbe0 Fix by adding INIT_LIST_HEAD() to snd_soc_component_initialize(). Reported-by: Hulk Robot Signed-off-by: Yang Yingliang Link: https://lore.kernel.org/r/20211009065840.3196239-1-yangyingliang@huawei.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c830e96afba24..80ca260595fda 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2599,6 +2599,7 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->dai_list); INIT_LIST_HEAD(&component->dobj_list); INIT_LIST_HEAD(&component->card_list); + INIT_LIST_HEAD(&component->list); mutex_init(&component->io_mutex); component->name = fmt_single_name(dev, &component->id); From aa18457c4af7a9dad1f2b150b11beae1d8ab57aa Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 11 Oct 2021 15:49:03 +0100 Subject: [PATCH 14/15] ASoC: cs42l42: Ensure 0dB full scale volume is used for headsets Ensure the default 0dB playback path is always used. The code that set FULL_SCALE_VOL based on LOAD_DET_RCSTAT was spurious, and resulted in a -6dB attenuation being accidentally inserted into the playback path. Signed-off-by: Stefan Binding Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20211011144903.28915-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index fb1e4c33e27d3..9a463ab54bddc 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -922,7 +922,6 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); unsigned int regval; - u8 fullScaleVol; int ret; if (mute) { @@ -993,20 +992,11 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) cs42l42->stream_use |= 1 << stream; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* Read the headphone load */ - regval = snd_soc_component_read(component, CS42L42_LOAD_DET_RCSTAT); - if (((regval & CS42L42_RLA_STAT_MASK) >> CS42L42_RLA_STAT_SHIFT) == - CS42L42_RLA_STAT_15_OHM) { - fullScaleVol = CS42L42_HP_FULL_SCALE_VOL_MASK; - } else { - fullScaleVol = 0; - } - - /* Un-mute the headphone, set the full scale volume flag */ + /* Un-mute the headphone */ snd_soc_component_update_bits(component, CS42L42_HP_CTL, CS42L42_HP_ANA_AMUTE_MASK | - CS42L42_HP_ANA_BMUTE_MASK | - CS42L42_HP_FULL_SCALE_VOL_MASK, fullScaleVol); + CS42L42_HP_ANA_BMUTE_MASK, + 0); } } From 6b9b546dc00797c74bef491668ce5431ff54e1e2 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Wed, 13 Oct 2021 13:17:04 +0800 Subject: [PATCH 15/15] ASoC: wm8960: Fix clock configuration on slave mode There is a noise issue for 8kHz sample rate on slave mode. Compared with master mode, the difference is the DACDIV setting, after correcting the DACDIV, the noise is gone. There is no noise issue for 48kHz sample rate, because the default value of DACDIV is correct for 48kHz. So wm8960_configure_clocking() should be functional for ADC and DAC function even if it is slave mode. In order to be compatible for old use case, just add condition for checking that sysclk is zero with slave mode. Fixes: 0e50b51aa22f ("ASoC: wm8960: Let wm8960 driver configure its bit clock and frame clock") Signed-off-by: Shengjiu Wang Acked-by: Charles Keepax Link: https://lore.kernel.org/r/1634102224-3922-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9e621a254392c..499604f1e1789 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -742,9 +742,16 @@ static int wm8960_configure_clocking(struct snd_soc_component *component) int i, j, k; int ret; - if (!(iface1 & (1<<6))) { - dev_dbg(component->dev, - "Codec is slave mode, no need to configure clock\n"); + /* + * For Slave mode clocking should still be configured, + * so this if statement should be removed, but some platform + * may not work if the sysclk is not configured, to avoid such + * compatible issue, just add '!wm8960->sysclk' condition in + * this if statement. + */ + if (!(iface1 & (1 << 6)) && !wm8960->sysclk) { + dev_warn(component->dev, + "slave mode, but proceeding with no clock configuration\n"); return 0; }