August 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/2] do exception handling appropriately in imx_es8328_probe()
by Yu Kuai 27 Aug '20

27 Aug '20

Yu Kuai (2): ASoC: fsl: imx-es8328: add missing kfree() call in imx_es8328_probe() ASoC: fsl: imx-es8328: add missing put_device() call in imx_es8328_probe() sound/soc/fsl/imx-es8328.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) -- 2.25.4

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[PATCH v2 0/7] ASoC: merge soc_pcm_open() rollback and soc_pcm_close()
by Kuninori Morimoto 27 Aug '20

27 Aug '20

Hi Mark This is v2 patch-set. soc_pcm_open() does rollback when failed (A), but, it is almost same as soc_pcm_close(). static int soc_pcm_open(xxx) { ... if (ret < 0) goto xxx_err; ... return 0; ^ config_err: | ... | rtd_startup_err: (A) ... | component_err: | ... v return ret; } This kind of duplicated code can be a hotbed of bugs, thus, this patch-set share soc_pcm_close() and rollback. v1 -> v2 - indicate more detail background/logic on git-log Link: https://lore.kernel.org/r/87wo2oku0m.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (7): ASoC: soc-dai: add mark for snd_soc_dai_startup/shutdown() ASoC: soc-link: add mark for snd_soc_link_startup/shutdown() ASoC: soc-component: add mark for soc_pcm_components_open/close() ASoC: soc-component: add mark for snd_soc_pcm_component_pm_runtime_get/put() ASoC: soc-pcm: add soc_pcm_clean() and call it from soc_pcm_open/close() ASoC: soc-pcm: remove unneeded dev_err() for snd_soc_dai_startup() ASoC: soc-pcm: remove unneeded dev_err() for snd_soc_component_module/open() include/sound/soc-component.h | 28 +++++--- include/sound/soc-dai.h | 5 +- include/sound/soc-link.h | 3 +- include/sound/soc.h | 3 + sound/soc/soc-component.c | 73 ++++++++++++++++++++- sound/soc/soc-compress.c | 30 +++------ sound/soc/soc-dai.c | 21 +++++- sound/soc/soc-dapm.c | 4 +- sound/soc/soc-link.c | 21 +++++- sound/soc/soc-pcm.c | 120 ++++++++++++---------------------- 10 files changed, 190 insertions(+), 118 deletions(-) -- 2.25.1

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[PATCH 0/2] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 27 Aug '20

27 Aug '20

Hi Rob, Mark These patches switch to yaml base Documentation for audio-graph-card, and renesas,rsnd which uses audio-graph-card. Kuninori Morimoto (2): ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation ASoC: dt-bindings: renesas,rsnd: switch to yaml base Documentation .../bindings/sound/audio-graph-card.txt | 337 ------------ .../bindings/sound/audio-graph-card.yaml | 135 +++++ .../bindings/sound/renesas,rsnd.txt | 520 ------------------ .../bindings/sound/renesas,rsnd.yaml | 454 +++++++++++++++ .../devicetree/bindings/vendor-prefixes.yaml | 2 +- 5 files changed, 590 insertions(+), 858 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -- 2.25.1

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[PATCH] ASoC: soc-core: move snd_soc_dai_link_set_capabilities() to soc-core.c
by Kuninori Morimoto 27 Aug '20

27 Aug '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 25612477d20b52 ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper") added snd_soc_dai_link_set_capabilities(). But it is using snd_soc_find_dai() (A) which is required client_mutex (B). And client_mutex is soc-core.c local. struct snd_soc_dai *snd_soc_find_dai(xxx) { ... (B) lockdep_assert_held(&client_mutex); ... } void snd_soc_dai_link_set_capabilities(xxx) { ... for_each_pcm_streams(direction) { ... for_each_link_cpus(dai_link, i, cpu) { (A) dai = snd_soc_find_dai(cpu); ... } ... for_each_link_codecs(dai_link, i, codec) { (A) dai = snd_soc_find_dai(codec); ... } } ... } Because of this, we will get WARNING. WARNING: CPU: 2 PID: 53 at sound/soc/soc-core.c:814 snd_soc_find_dai+0xf8/0x100 CPU: 2 PID: 53 Comm: kworker/2:1 Not tainted 5.7.0-rc1+ #328 Hardware name: Renesas H3ULCB Kingfisher board based on r8a77951 (DT) Workqueue: events deferred_probe_work_func pstate: 60000005 (nZCv daif -PAN -UAO) pc : snd_soc_find_dai+0xf8/0x100 lr : snd_soc_find_dai+0xf4/0x100 ... Call trace: snd_soc_find_dai+0xf8/0x100 snd_soc_dai_link_set_capabilities+0xa0/0x16c graph_dai_link_of_dpcm+0x390/0x3c0 graph_for_each_link+0x134/0x200 graph_probe+0x144/0x230 platform_drv_probe+0x5c/0xb0 really_probe+0xe4/0x430 driver_probe_device+0x60/0xf4 Because not only snd_soc_dai_link_set_capabilities() but many drivers are already using snd_soc_find_dai(), 1st idea for this issue was creating snd_soc_find_dai() of _with_mutex / _without_mutex, but it was not enough and/or confusable. soc-dai.c which is implemented snd_soc_dai_link_set_capabilities() currently is mainly for snd_soc_dai related functions, not for snd_soc_dai_link. Thus as 2nd idea, this patch moves snd_soc_dai_link_set_capabilities() from soc-dai.c to soc-core.c, and use mutex_lock(). Fixes: 25612477d20b52 ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 41 +++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-dai.c | 38 -------------------------------------- 2 files changed, 41 insertions(+), 38 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe23e936e2d1..f46979f8d965 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -344,6 +344,47 @@ struct snd_soc_component *snd_soc_lookup_component(struct device *dev, } EXPORT_SYMBOL_GPL(snd_soc_lookup_component); +/* + * snd_soc_dai_link_set_capabilities() - set dai_link properties based on its DAIs + */ +void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link) +{ + struct snd_soc_dai_link_component *cpu; + struct snd_soc_dai_link_component *codec; + struct snd_soc_dai *dai; + bool supported[SNDRV_PCM_STREAM_LAST + 1]; + int direction; + int i; + + mutex_lock(&client_mutex); + + for_each_pcm_streams(direction) { + supported[direction] = true; + + for_each_link_cpus(dai_link, i, cpu) { + dai = snd_soc_find_dai(cpu); + if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { + supported[direction] = false; + break; + } + } + if (!supported[direction]) + continue; + for_each_link_codecs(dai_link, i, codec) { + dai = snd_soc_find_dai(codec); + if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { + supported[direction] = false; + break; + } + } + } + mutex_unlock(&client_mutex); + + dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK]; + dai_link->dpcm_capture = supported[SNDRV_PCM_STREAM_CAPTURE]; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_link_set_capabilities); + struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 693893420bf0..783b7928205e 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -393,44 +393,6 @@ bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir) return stream->channels_min; } -/* - * snd_soc_dai_link_set_capabilities() - set dai_link properties based on its DAIs - */ -void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link) -{ - struct snd_soc_dai_link_component *cpu; - struct snd_soc_dai_link_component *codec; - struct snd_soc_dai *dai; - bool supported[SNDRV_PCM_STREAM_LAST + 1]; - int direction; - int i; - - for_each_pcm_streams(direction) { - supported[direction] = true; - - for_each_link_cpus(dai_link, i, cpu) { - dai = snd_soc_find_dai(cpu); - if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { - supported[direction] = false; - break; - } - } - if (!supported[direction]) - continue; - for_each_link_codecs(dai_link, i, codec) { - dai = snd_soc_find_dai(codec); - if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { - supported[direction] = false; - break; - } - } - } - - dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK]; - dai_link->dpcm_capture = supported[SNDRV_PCM_STREAM_CAPTURE]; -} -EXPORT_SYMBOL_GPL(snd_soc_dai_link_set_capabilities); - void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action) { -- 2.25.1

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[PATCH] ASoC: qcom: common: Fix refcount imbalance on error
by Dinghao Liu 26 Aug '20

26 Aug '20

for_each_child_of_node returns a node pointer np with refcount incremented. So when devm_kzalloc fails, a pairing refcount decrement is needed to keep np's refcount balanced. Fixes: 16395ceee11f8 ("ASoC: qcom: common: Fix NULL pointer in of parser") Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/soc/qcom/common.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 5194d90ddb96..fd69cf8b1f23 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -52,8 +52,10 @@ int qcom_snd_parse_of(struct snd_soc_card *card) for_each_child_of_node(dev->of_node, np) { dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); - if (!dlc) - return -ENOMEM; + if (!dlc) { + ret = -ENOMEM; + goto err; + } link->cpus = &dlc[0]; link->platforms = &dlc[1]; -- 2.17.1

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[PATCH v2 0/5] ASoC: Fix return check for devm_regmap_init_sdw()
by Vinod Koul 26 Aug '20

26 Aug '20

This series attempts to fix return check for devm_regmap_init_sdw() Changes in v2: - Add missing patch for rt711 - Add patch for rt700 Vinod Koul (5): ASoC: max98373: Fix return check for devm_regmap_init_sdw() ASoC: rt1308-sdw: Fix return check for devm_regmap_init_sdw() ASoC: rt711: Fix return check for devm_regmap_init_sdw() ASoC: rt715: Fix return check for devm_regmap_init_sdw() ASoC: rt700: Fix return check for devm_regmap_init_sdw() sound/soc/codecs/max98373-sdw.c | 4 ++-- sound/soc/codecs/rt1308-sdw.c | 4 ++-- sound/soc/codecs/rt700-sdw.c | 4 ++-- sound/soc/codecs/rt711-sdw.c | 4 ++-- sound/soc/codecs/rt715-sdw.c | 4 ++-- 5 files changed, 10 insertions(+), 10 deletions(-) -- 2.26.2

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[PATCH] ALSA: hda/hdmi: always check pin power status in i915 pin fixup
by Kai Vehmanen 26 Aug '20

26 Aug '20

When system is suspended with active audio playback to HDMI/DP, two alternative sequences can happen at resume: a) monitor is detected first and ALSA prepare follows normal stream setup sequence, or b) ALSA prepare is called first, but monitor is not yet detected, so PCM is restarted without a pin, In case of (b), on i915 systems, haswell_verify_D0() is not called at resume and the pin power state may be incorrect. Result is lack of audio after resume with no error reported back to user-space. Fix the problem by always verifying converter and pin state in the i915_pin_cvt_fixup(). BugLink: https://github.com/thesofproject/linux/issues/2388 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3259d713ace9..1e1b13eb7829 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2794,6 +2794,7 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec, hda_nid_t cvt_nid) { if (per_pin) { + haswell_verify_D0(codec, per_pin->cvt_nid, per_pin->pin_nid); snd_hda_set_dev_select(codec, per_pin->pin_nid, per_pin->dev_id); intel_verify_pin_cvt_connect(codec, per_pin); -- 2.27.0

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[PATCH] ASoC: AMD: Clean kernel log from deferred probe error messages
by Akshu Agrawal 26 Aug '20

26 Aug '20

While the driver waits for DAIs to be probed and retries probing, avoid printing error messages. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 406526e79af3..67f80ba51de0 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -471,13 +471,11 @@ static int acp3x_probe(struct platform_device *pdev) } ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", card->name, ret); - return ret; - } - return 0; + return ret; } static const struct acpi_device_id acp3x_audio_acpi_match[] = { -- 2.20.1

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[PATCH v2] ASoC: max9867: shutdown codec when changing filter type
by Pavel Dobias 26 Aug '20

26 Aug '20

Changing filter type without disabling codec results in filter malfunction. Disable codec when changing filter type. Signed-off-by: Pavel Dobias <dobias(a)2n.cz> --- Changes in v2: - don't allow change if component active - avoid codec shutdown in max9867_filter_set if value not changed sound/soc/codecs/max9867.c | 80 +++++++++++++++++++++++++++++++++----- sound/soc/codecs/max9867.h | 4 +- 2 files changed, 73 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index fcb31144d69c..749de1ab1093 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -15,6 +15,13 @@ #include <sound/tlv.h> #include "max9867.h" +struct max9867_priv { + struct regmap *regmap; + const struct snd_pcm_hw_constraint_list *constraints; + unsigned int sysclk, pclk; + bool master, dsp_a; +}; + static const char *const max9867_spmode[] = { "Stereo Diff", "Mono Diff", "Stereo Cap", "Mono Cap", @@ -32,8 +39,68 @@ static const char *const max9867_adc_dac_filter_text[] = { "Butterworth/8-24" }; -static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, - max9867_filter_text); +static int max9867_filter_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); + unsigned int reg; + int ret; + + ret = regmap_read(max9867->regmap, MAX9867_CODECFLTR, &reg); + if (ret) + return -EINVAL; + + if (reg & MAX9867_CODECFLTR_MODE) + ucontrol->value.enumerated.item[0] = 1; + else + ucontrol->value.enumerated.item[0] = 0; + + return 0; +} + +static int max9867_filter_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); + unsigned int reg, mode = ucontrol->value.enumerated.item[0]; + int ret; + + /* don't allow change if component active */ + if (snd_soc_component_active(component)) { + return -EBUSY; + } + + if (mode > 1) + return -EINVAL; + mode = mode ? MAX9867_CODECFLTR_MODE : 0; + + /* read current filter mode */ + ret = regmap_read(max9867->regmap, MAX9867_CODECFLTR, &reg); + if (ret) + return -EINVAL; + + /* check if change is needed */ + if ((reg & MAX9867_CODECFLTR_MODE) == mode) + return 0; + + /* shutdown codec before switching filter mode */ + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_PWRMAN_SHDN, 0); + + /* switch filter mode */ + regmap_update_bits(max9867->regmap, MAX9867_CODECFLTR, + MAX9867_CODECFLTR_MODE, mode); + + /* out of shutdown now */ + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_PWRMAN_SHDN, MAX9867_PWRMAN_SHDN); + + return 0; +} + +static SOC_ENUM_SINGLE_EXT_DECL(max9867_filter, max9867_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_dac_filter, MAX9867_CODECFLTR, 0, max9867_adc_dac_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_adc_filter, MAX9867_CODECFLTR, 4, @@ -76,7 +143,7 @@ static const struct snd_kcontrol_new max9867_snd_controls[] = { SOC_ENUM("Speaker Mode", max9867_spkmode), SOC_SINGLE("Volume Smoothing Switch", MAX9867_MODECONFIG, 6, 1, 0), SOC_SINGLE("Line ZC Switch", MAX9867_MODECONFIG, 5, 1, 0), - SOC_ENUM("DSP Filter", max9867_filter), + SOC_ENUM_EXT("DSP Filter", max9867_filter, max9867_filter_get, max9867_filter_set), SOC_ENUM("ADC Filter", max9867_adc_filter), SOC_ENUM("DAC Filter", max9867_dac_filter), SOC_SINGLE("Mono Playback Switch", MAX9867_IFC1B, 3, 1, 0), @@ -197,13 +264,6 @@ static const struct snd_pcm_hw_constraint_list max9867_constraints_48k = { .count = ARRAY_SIZE(max9867_rates_48k), }; -struct max9867_priv { - struct regmap *regmap; - const struct snd_pcm_hw_constraint_list *constraints; - unsigned int sysclk, pclk; - bool master, dsp_a; -}; - static int max9867_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index 3092c3b99075..b6b880631b13 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -44,7 +44,8 @@ #define MAX9867_IFC1B_PCLK_4 0x05 #define MAX9867_IFC1B_PCLK_8 0x06 #define MAX9867_IFC1B_PCLK_16 0x07 -#define MAX9867_CODECFLTR 0x0a +#define MAX9867_CODECFLTR 0x0a +#define MAX9867_CODECFLTR_MODE (1<<7) #define MAX9867_SIDETONE 0x0b #define MAX9867_DACLEVEL 0x0c #define MAX9867_ADCLEVEL 0x0d @@ -58,6 +59,7 @@ #define MAX9867_MICCONFIG 0x15 #define MAX9867_MODECONFIG 0x16 #define MAX9867_PWRMAN 0x17 +#define MAX9867_PWRMAN_SHDN (1<<7) #define MAX9867_REVISION 0xff #define MAX9867_CACHEREGNUM 10 -- 2.20.1

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[PATCH 1/4] ASoC: max98373: Fix return check for devm_regmap_init_sdw()
by Vinod Koul 26 Aug '20

26 Aug '20

devm_regmap_init_sdw() returns a valid pointer on success or ERR_PTR on failure which should be checked with IS_ERR. Also use PTR_ERR for returning error codes. Reported-by: Takashi Iwai <tiwai(a)suse.de> Fixes: 56a5b7910e96 ("ASoC: codecs: max98373: add SoundWire support") Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- sound/soc/codecs/max98373-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index 5fe724728e84..e4675cfff7b2 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -838,8 +838,8 @@ static int max98373_sdw_probe(struct sdw_slave *slave, /* Regmap Initialization */ regmap = devm_regmap_init_sdw(slave, &max98373_sdw_regmap); - if (!regmap) - return -EINVAL; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); return max98373_init(slave, regmap); } -- 2.26.2

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