October 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH next] ALSA: mixer: fix deadlock in snd_mixer_oss_set_volume
by Pavel Skripkin 26 Oct '21

26 Oct '21

In commit 411cef6adfb3 ("ALSA: mixer: oss: Fix racy access to slots") added mutex protection in snd_mixer_oss_set_volume(). Second mutex_lock() in same function looks like typo, fix it. Reported-by: syzbot+ace149a75a9a0a399ac7(a)syzkaller.appspotmail.com Fixes: 411cef6adfb3 ("ALSA: mixer: oss: Fix racy access to slots") Signed-off-by: Pavel Skripkin <paskripkin(a)gmail.com> --- sound/core/oss/mixer_oss.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index d5ddc154a735..9620115cfdc0 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -313,7 +313,7 @@ static int snd_mixer_oss_set_volume(struct snd_mixer_oss_file *fmixer, pslot->volume[1] = right; result = (left & 0xff) | ((right & 0xff) << 8); unlock: - mutex_lock(&mixer->reg_mutex); + mutex_unlock(&mixer->reg_mutex); return result; } -- 2.33.1

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[PATCH] ASoC: qcom: common: Respect status = "disabled" on DAI link nodes
by Stephan Gerhold 25 Oct '21

25 Oct '21

At the moment, the DAI link nodes in the device tree always have to be specified completely in each device tree. However, the available interfaces (e.g. Primary/Secondary/Tertiary/Quaternary MI2S) are common for all devices of a SoC, so the majority of the definitions can be placed in a common device tree include to reduce boilerplate. Make it possible to define such stubs in device tree includes by respecting the "status" property for the DAI link nodes. This is a trivial change that just requires switching to the _available_ OF functions that check the "status" property additionally. This allows defining a stub like: sound_dai_quaternary: dai-link-quaternary { link-name = "Quaternary MI2S"; status = "disabled"; /* Needs extra codec configuration */ cpu { sound-dai = <&q6afedai QUATERNARY_MI2S_RX>; }; platform { sound-dai = <&q6routing>; }; }; where the codec would be filled in by the device-specific device tree. For existing device trees this change does not make any difference. A missing "status" property is treated like status = "okay". Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/qcom/common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 09af00700700..2e1c618f7529 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -44,7 +44,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) return ret; /* Populate links */ - num_links = of_get_child_count(dev->of_node); + num_links = of_get_available_child_count(dev->of_node); /* Allocate the DAI link array */ card->dai_link = devm_kcalloc(dev, num_links, sizeof(*link), GFP_KERNEL); @@ -54,7 +54,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) card->num_links = num_links; link = card->dai_link; - for_each_child_of_node(dev->of_node, np) { + for_each_available_child_of_node(dev->of_node, np) { dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); if (!dlc) { ret = -ENOMEM; -- 2.33.1

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[PATCH v3] ASoC: dt-bindings: lpass: add binding headers for digital codecs
by Srinivasa Rao Mandadapu 25 Oct '21

25 Oct '21

Add header defining for lpass internal digital codecs rx,tx and va dai node id's. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes since v2: -- Rebased to latest sound next and created new patch. Changes since v1: -- Add missing dai node ID's include/dt-bindings/sound/qcom,lpass.h | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/include/dt-bindings/sound/qcom,lpass.h b/include/dt-bindings/sound/qcom,lpass.h index 187af45..a9404c3 100644 --- a/include/dt-bindings/sound/qcom,lpass.h +++ b/include/dt-bindings/sound/qcom,lpass.h @@ -11,9 +11,35 @@ #define LPASS_DP_RX 5 #define LPASS_CDC_DMA_RX0 6 -#define LPASS_CDC_DMA_TX3 7 -#define LPASS_CDC_DMA_VA0 8 -#define LPASS_MAX_PORTS 9 +#define LPASS_CDC_DMA_RX1 7 +#define LPASS_CDC_DMA_RX2 8 +#define LPASS_CDC_DMA_RX3 9 +#define LPASS_CDC_DMA_RX4 10 +#define LPASS_CDC_DMA_RX5 11 +#define LPASS_CDC_DMA_RX6 12 +#define LPASS_CDC_DMA_RX7 13 +#define LPASS_CDC_DMA_RX8 14 +#define LPASS_CDC_DMA_RX9 15 + +#define LPASS_CDC_DMA_TX0 16 +#define LPASS_CDC_DMA_TX1 17 +#define LPASS_CDC_DMA_TX2 18 +#define LPASS_CDC_DMA_TX3 19 +#define LPASS_CDC_DMA_TX4 20 +#define LPASS_CDC_DMA_TX5 21 +#define LPASS_CDC_DMA_TX6 22 +#define LPASS_CDC_DMA_TX7 23 +#define LPASS_CDC_DMA_TX8 24 + +#define LPASS_CDC_DMA_VA_TX0 25 +#define LPASS_CDC_DMA_VA_TX1 26 +#define LPASS_CDC_DMA_VA_TX2 27 +#define LPASS_CDC_DMA_VA_TX3 28 +#define LPASS_CDC_DMA_VA_TX4 29 +#define LPASS_CDC_DMA_VA_TX5 30 +#define LPASS_CDC_DMA_VA_TX6 31 +#define LPASS_CDC_DMA_VA_TX7 32 +#define LPASS_CDC_DMA_VA_TX8 33 #define LPASS_MCLK0 0 -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH] ASoC: topology: Fix stub for snd_soc_tplg_component_remove()
by Mark Brown 25 Oct '21

25 Oct '21

When removing the index argument from snd_soc_topology_component_remove() commit a5b8f71c5477f (ASoC: topology: Remove multistep topology loading) forgot to update the stub for !SND_SOC_TOPOLOGY use, causing build failures for anything that tries to make use of it. Fixes: a5b8f71c5477f (ASoC: topology: Remove multistep topology loading) Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-topology.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 7f33de8ffd95..b4b896f83b94 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -188,8 +188,7 @@ int snd_soc_tplg_widget_bind_event(struct snd_soc_dapm_widget *w, #else -static inline int snd_soc_tplg_component_remove(struct snd_soc_component *comp, - u32 index) +static inline int snd_soc_tplg_component_remove(struct snd_soc_component *comp) { return 0; } -- 2.30.2

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[PATCH 5.14 018/169] ASoC: pcm512x: Mend accesses to the I2S_1 and I2S_2 registers
by Greg Kroah-Hartman 25 Oct '21

25 Oct '21

From: Peter Rosin <peda(a)axentia.se> [ Upstream commit 3f4b57ad07d9237acf1b8cff3f8bf530cacef87a ] 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 <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Kirill Marinushkin <kmarinushkin(a)birdec.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Peter Rosin <peda(a)axentia.se> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/2d221984-7a2e-7006-0f8a-ffb5f64ee885@axentia.se Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 4dc844f3c1fc..60dee41816dc 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) -- 2.33.0

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[PATCH AUTOSEL 5.10 01/13] ASoC: soc-core: fix null-ptr-deref in snd_soc_del_component_unlocked()
by Sasha Levin 25 Oct '21

25 Oct '21

From: Yang Yingliang <yangyingliang(a)huawei.com> [ Upstream commit c448b7aa3e66042fc0f849d9a0fb90d1af82e948 ] '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 <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> Link: https://lore.kernel.org/r/20211009065840.3196239-1-yangyingliang@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 e677422c1058..133296596864 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2454,6 +2454,7 @@ 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); -- 2.33.0

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[PATCH AUTOSEL 5.14 05/18] ALSA: hda/realtek: Fixes HP Spectre x360 15-eb1xxx speakers
by Sasha Levin 25 Oct '21

25 Oct '21

From: Davide Baldo <davide(a)baldo.me> [ Upstream commit d94befbb5ae379f6dfd4fa6d460eacc09fa7b9c3 ] In laptop 'HP Spectre x360 Convertible 15-eb1xxx/8811' both front and rear speakers are silent, this patch fixes that by overriding the pin layout and by initializing the amplifier which needs a GPIO pin to be set to 1 then 0, similar to the existing HP Spectre x360 14 model. In order to have volume control, both front and rear speakers were forced to use the DAC1. This patch also correctly map the mute LED but since there is no microphone on/off switch exposed by the alsa subsystem it never turns on by itself. There are still known audio issues in this laptop: headset microphone doesn't work, the button to mute/unmute microphone is not yet mapped, the LED of the mute/unmute speakers doesn't seems to be exposed via GPIO and never turns on. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213953 Signed-off-by: Davide Baldo <davide(a)baldo.me> Link: https://lore.kernel.org/r/20211015072121.5287-1-davide@baldo.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 46 +++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8e6ff50f0f94..a628b2c050bd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6418,6 +6418,44 @@ static void alc_fixup_no_int_mic(struct hda_codec *codec, } } +/* GPIO1 = amplifier on/off + * GPIO3 = mic mute LED + */ +static void alc285_fixup_hp_spectre_x360_eb1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02 }; + + struct alc_spec *spec = codec->spec; + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170110 }, /* front/high speakers */ + { 0x17, 0x90170130 }, /* back/bass speakers */ + { } + }; + + //enable micmute led + alc_fixup_hp_gpio_led(codec, action, 0x00, 0x04); + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->micmute_led_polarity = 1; + /* needed for amp of back speakers */ + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + snd_hda_apply_pincfgs(codec, pincfgs); + /* share DAC to have unified volume control */ + snd_hda_override_conn_list(codec, 0x14, ARRAY_SIZE(conn), conn); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + break; + case HDA_FIXUP_ACT_INIT: + /* need to toggle GPIO to enable the amp of back speakers */ + alc_update_gpio_data(codec, 0x01, true); + msleep(100); + alc_update_gpio_data(codec, 0x01, false); + break; + } +} + static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6570,6 +6608,7 @@ enum { ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, ALC280_FIXUP_HP_9480M, ALC245_FIXUP_HP_X360_AMP, + ALC285_FIXUP_HP_SPECTRE_X360_EB1, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, @@ -8263,6 +8302,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_spectre_x360, }, + [ALC285_FIXUP_HP_SPECTRE_X360_EB1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_spectre_x360_eb1 + }, [ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_ideapad_s740_coef, @@ -8597,6 +8640,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), + SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), @@ -9018,6 +9063,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, + {.id = ALC285_FIXUP_HP_SPECTRE_X360_EB1, .name = "alc285-hp-spectre-x360-eb1"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, -- 2.33.0

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[PATCH AUTOSEL 5.14 03/18] ASoC: cs42l42: Ensure 0dB full scale volume is used for headsets
by Sasha Levin 25 Oct '21

25 Oct '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> [ Upstream commit aa18457c4af7a9dad1f2b150b11beae1d8ab57aa ] 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 <sbinding(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20211011144903.28915-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 99c022be94a6..8838b9a0de8e 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -901,7 +901,6 @@ 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) { @@ -972,20 +971,11 @@ 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); } } -- 2.33.0

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[PATCH AUTOSEL 5.14 02/18] ASoC: soc-core: fix null-ptr-deref in snd_soc_del_component_unlocked()
by Sasha Levin 25 Oct '21

25 Oct '21

From: Yang Yingliang <yangyingliang(a)huawei.com> [ Upstream commit c448b7aa3e66042fc0f849d9a0fb90d1af82e948 ] '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 <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> Link: https://lore.kernel.org/r/20211009065840.3196239-1-yangyingliang@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 583f2381cfc8..e926985bb2f8 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); -- 2.33.0

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[PATCH] ASoC: doc: update codec example code
by John Keeping 25 Oct '21

25 Oct '21

As the codec API has evolved the documentation has not kept up and still uses old fields that have been removed. Update the examples to represent the current API. Signed-off-by: John Keeping <john(a)metanate.com> --- Documentation/sound/soc/codec.rst | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/soc/codec.rst b/Documentation/sound/soc/codec.rst index 8a9737eb7597..57df149acafc 100644 --- a/Documentation/sound/soc/codec.rst +++ b/Documentation/sound/soc/codec.rst @@ -40,7 +40,7 @@ e.g. .prepare = wm8731_pcm_prepare, .hw_params = wm8731_hw_params, .shutdown = wm8731_shutdown, - .digital_mute = wm8731_mute, + .mute_stream = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, .set_fmt = wm8731_set_dai_fmt, }; @@ -60,7 +60,7 @@ e.g. .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, .ops = &wm8731_dai_ops, - .symmetric_rates = 1, + .symmetric_rate = 1, }; @@ -177,10 +177,10 @@ when the mute is applied or freed. i.e. :: - static int wm8974_mute(struct snd_soc_dai *dai, int mute) + static int wm8974_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; - u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf; + u16 mute_reg = snd_soc_component_read(component, WM8974_DAC) & 0xffbf; if (mute) snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40); -- 2.33.1

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