July 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: ti: remove sound support for DaVinci DM644x and DM646x
by Lukas Bulwahn 26 Jul '22

26 Jul '22

Commit 7dd33764486d ("ARM: davinci: Delete DM644x board files") and commit b4aed01de486 ("ARM: davinci: Delete DM646x board files") removes the support for DaVinci DM644x and DM646x boards. Hence, remove the sound support for those boards as well. Signed-off-by: Lukas Bulwahn <lukas.bulwahn(a)gmail.com> --- sound/soc/ti/Kconfig | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 40110e9a9e8a..35fddd7c501a 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -180,9 +180,6 @@ config SND_SOC_DAVINCI_EVM select SND_SOC_DAVINCI_ASP if MACH_DAVINCI_DM355_EVM select SND_SOC_DAVINCI_ASP if SND_SOC_DM365_AIC3X_CODEC select SND_SOC_DAVINCI_VCIF if SND_SOC_DM365_VOICE_CODEC - select SND_SOC_DAVINCI_ASP if MACH_DAVINCI_EVM # DM6446 - select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DM6467_EVM - select SND_SOC_SPDIF if MACH_DAVINCI_DM6467_EVM select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DA830_EVM select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DA850_EVM select SND_SOC_TLV320AIC3X @@ -191,8 +188,6 @@ config SND_SOC_DAVINCI_EVM DaVinci EVM platforms: - DM355 - DM365 - - DM6446 - - DM6447 - DM830 - DM850 -- 2.17.1

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Question about DPCM locking
by Kuninori Morimoto 26 Jul '22

26 Jul '22

Hi Takashi May I ask about this patch ? ASoC: soc-pcm: Fix and cleanup DPCM locking b7898396f4bbe160f546d0c5e9fa17cca9a7d153 Q1. It exchanges many function parameters - func(substream, xxx) + func(rtd, substream, xxx) ^^^^ I guess the purpose is it want to call snd_soc_dpcm_mutex_assert_held() inside the func ? But it looks very verbose. we can do like this, but am I misunderstanding ? func(substream, xxx) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); snd_soc_dpcm_mutex_assert_held(rtd); ... } Q2. It added new __soc_pcm_close(). But soc_pcm_close() is using soc_pcm_clean() instead of __soc_pcm_close(). Is this just a mistake ? static int soc_pcm_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); snd_soc_dpcm_mutex_lock(rtd); - soc_pcm_clean(substream, 0); + __soc_pcm_close(substream); snd_soc_dpcm_mutex_unlock(rtd); return 0; } Thank you for your help !! Best regards --- Kuninori Morimoto

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Re: [PATCH v8] ASoc: dt-bindings: tas27xx: revise tas27xx format
by Rob Herring 26 Jul '22

26 Jul '22

On Fri, Jul 22, 2022 at 06:55:12PM +0800, Raphael-Xu wrote: > revise the format Complete sentences. And this is not v8. Each new change is v1. > > Signed-off-by: Raphael-Xu <13691752556(a)139.com> > --- > .../devicetree/bindings/sound/tas27xx.yaml | 16 +++++++++++----- > 1 file changed, 11 insertions(+), 5 deletions(-) > > diff --git a/Documentation/devicetree/bindings/sound/tas27xx.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml > index 66a0df8850ea..573652577462 100644 > --- a/Documentation/devicetree/bindings/sound/tas27xx.yaml > +++ b/Documentation/devicetree/bindings/sound/tas27xx.yaml > @@ -8,7 +8,7 @@ $schema: "http://devicetree.org/meta-schemas/core.yaml#" > title: Texas Instruments TAS2764/TAS2780 Smart PA > > maintainers: > - - Shenghao Ding <shenghao-ding(a)ti.com> > + - Raphael Xu <raphael-xu(a)ti.com> > > description: | > The TAS2764/TAS2780 is a mono, digital input Class-D audio amplifier > @@ -23,17 +23,19 @@ properties: > - ti,tas2780 > > reg: > - maxItems: 1 > description: | > - I2C address of the device can be between 0x38 to 0x45. 0x40-0x45 is no longer valid? > + I2C address of the device can be between 0x38 to 0x3f. > + minimum: 0x38 > + maximum: 0x3f > > reset-gpios: > - maxItems: 1 > description: GPIO used to reset the device. > + maxItems: 1 Unnecessary change. > > shutdown-gpios: > + description: | Don't need '|' if no formatting. > + GPIO used to control the state of the device.Only for TAS2764. Space needed after period. > maxItems: 1 > - description: GPIO used to control the state of the device. > > interrupts: > maxItems: 1 > @@ -41,10 +43,14 @@ properties: > ti,imon-slot-no: > $ref: /schemas/types.yaml#/definitions/uint32 > description: TDM TX current sense time slot. > + minimum: 0 > + maximum: 3 > > ti,vmon-slot-no: > $ref: /schemas/types.yaml#/definitions/uint32 > description: TDM TX voltage sense time slot. > + minimum: 0 > + maximum: 3 > > '#sound-dai-cells': > const: 1 > -- > 2.35.1 > > >

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[PATCH v2 0/5] ALSA: Drop async signal support
by Takashi Iwai 25 Jul '22

25 Jul '22

Hi, this is a revised patch set for dropping fasync support from ALSA core. The async signal itself is very difficult to use properly due to various restrictions (e.g. you cannot perform any I/O in the context), hence it's a feature that has been never used by real applications. OTOH, the real problem is that there have been quite a few syzcaller reports indicating that fasync code path may lead to some potential deadlocks for long time. Dropping the feature is the easiest solution, obviously. The corresponding update for alsa-lib will follow once when we agree with this approach. thanks, Takashi === v1: https://lore.kernel.org/r/20220715102935.4695-1-tiwai@suse.de v1->v2: Fix unused variable warning in patch 2 === Takashi Iwai (5): ALSA: timer: Drop async signal support ALSA: pcm: Drop async signal support ALSA: control: Drop async signal support ALSA: core: Drop async signal support ALSA: doc: Drop stale fasync entry .../kernel-api/writing-an-alsa-driver.rst | 1 - include/sound/control.h | 1 - include/sound/pcm.h | 1 - sound/core/control.c | 11 ----------- sound/core/init.c | 11 +---------- sound/core/pcm_lib.c | 8 +------- sound/core/pcm_native.c | 18 ------------------ sound/core/timer.c | 13 ------------- 8 files changed, 2 insertions(+), 62 deletions(-) -- 2.35.3

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[bug report] ASoC: mediatek: mt8186: support tdm in platform driver
by Dan Carpenter 25 Jul '22

25 Jul '22

Hello Jiaxin Yu, This is a semi-automatic email about new static checker warnings. The patch ae92dcbee8b6: "ASoC: mediatek: mt8186: support tdm in platform driver" from May 23, 2022, leads to the following Smatch complaint: sound/soc/mediatek/mt8186/mt8186-dai-tdm.c:424 mtk_dai_tdm_hw_params() warn: variable dereferenced before check 'tdm_priv' (see line 406) sound/soc/mediatek/mt8186/mt8186-dai-tdm.c 405 struct mtk_afe_tdm_priv *tdm_priv = afe_priv->dai_priv[tdm_id]; 406 unsigned int tdm_mode = tdm_priv->tdm_mode; ^^^^^^^^^^^^^^^^^^^ Lot's of dereferences 407 unsigned int data_mode = tdm_priv->data_mode; 408 unsigned int rate = params_rate(params); 409 unsigned int channels = params_channels(params); 410 snd_pcm_format_t format = params_format(params); 411 unsigned int bit_width = 412 snd_pcm_format_physical_width(format); 413 unsigned int tdm_channels = (data_mode == TDM_DATA_ONE_PIN) ? 414 get_tdm_ch_per_sdata(tdm_mode, channels) : 2; 415 unsigned int lrck_width = 416 get_tdm_lrck_width(format, tdm_mode); 417 unsigned int tdm_con = 0; 418 bool slave_mode = tdm_priv->slave_mode; 419 bool lrck_inv = tdm_priv->lck_invert; 420 bool bck_inv = tdm_priv->bck_invert; 421 unsigned int tran_rate; 422 unsigned int tran_relatch_rate; 423 424 if (!tdm_priv) { ^^^^^^^^^ Checked too late 425 dev_err(afe->dev, "%s(), tdm_priv == NULL", __func__); 426 return -EINVAL; regards, dan carpenter

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[bug report] ASoC: mediatek: mt8186: support adda in platform driver
by Dan Carpenter 25 Jul '22

25 Jul '22

Hello Jiaxin Yu, The patch b65c466220b3: "ASoC: mediatek: mt8186: support adda in platform driver" from May 23, 2022, leads to the following Smatch static checker warning: sound/soc/mediatek/mt8186/mt8186-dai-adda.c:78 get_adda_priv_by_name() warn: strncmp() with weird length: 17 vs 7 sound/soc/mediatek/mt8186/mt8186-dai-adda.c 72 static struct mtk_afe_adda_priv *get_adda_priv_by_name(struct mtk_base_afe *afe, 73 const char *name) 74 { 75 struct mt8186_afe_private *afe_priv = afe->platform_priv; 76 int dai_id; 77 --> 78 if (strncmp(name, "aud_dac_hires_clk", 7) == 0 || 79 strncmp(name, "aud_adc_hires_clk", 7) == 0) 7 is weird. It would be more readable to say it as: if (strncmp(name, "aud_dac", 7) == 0 || strncmp(name, "aud_adc", 7) == 0) 80 dai_id = MT8186_DAI_ADDA; 81 else 82 return NULL; 83 84 return afe_priv->dai_priv[dai_id]; 85 } regards, dan carpenter

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[PATCH] ALSA: hda/realtek: Enable speaker and mute LEDs for HP laptops
by Kai-Heng Feng 25 Jul '22

25 Jul '22

Two more HP laptops that use cs35l41 AMP for speaker and GPIO for mute LEDs. So use the existing quirk to enable them accordingly. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f55bc43bfa9c..cdbee71e83216 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9109,6 +9109,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89c3, "Zbook Studio G9", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8ad1, "HP EliteBook 840 14 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), -- 2.36.1

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[PATCH -next] ALSA: hda: cs35l41: Fix build error unused-function
by Ren Zhijie 25 Jul '22

25 Jul '22

If CONFIG_PM_SLEEP is not set, make ARCH=x86_64 CROSS_COMPILE=x86_64-linux-gnu-, will be failed, like this: sound/pci/hda/cs35l41_hda.c:583:12: error: ‘cs35l41_runtime_resume’ defined but not used [-Werror=unused-function] static int cs35l41_runtime_resume(struct device *dev) ^~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/cs35l41_hda.c:565:12: error: ‘cs35l41_runtime_suspend’ defined but not used [-Werror=unused-function] static int cs35l41_runtime_suspend(struct device *dev) ^~~~~~~~~~~~~~~~~~~~~~~ cc1: all warnings being treated as errors make[3]: *** [sound/pci/hda/cs35l41_hda.o] Error 1 commit 1a3c7bb08826 ("PM: core: Add new *_PM_OPS macros, deprecate old ones"), add new marco RUNTIME_PM_OPS to fix this unused-function problem. Fixes: 1873ebd30cc8 ("ALSA: hda: cs35l41: Support Hibernation during Suspend") Signed-off-by: Ren Zhijie <renzhijie2(a)huawei.com> --- sound/pci/hda/cs35l41_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 28798d5c1cf1..93cf039abb02 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1439,7 +1439,7 @@ void cs35l41_hda_remove(struct device *dev) EXPORT_SYMBOL_NS_GPL(cs35l41_hda_remove, SND_HDA_SCODEC_CS35L41); const struct dev_pm_ops cs35l41_hda_pm_ops = { - SET_RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume, NULL) + RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume, NULL) }; EXPORT_SYMBOL_NS_GPL(cs35l41_hda_pm_ops, SND_HDA_SCODEC_CS35L41); -- 2.17.1

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[PATCH] usb/hiface: fix repeated words in comments
by wangjianli 25 Jul '22

25 Jul '22

Delete the redundant word 'in'. Signed-off-by: wangjianli <wangjianli(a)cdjrlc.com> --- sound/usb/hiface/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 71f17f02f341..cf650fab54d7 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -225,7 +225,7 @@ static int hiface_pcm_stream_start(struct pcm_runtime *rt) } } - /* wait for first out urb to return (sent in in urb handler) */ + /* wait for first out urb to return (sent in urb handler) */ wait_event_timeout(rt->stream_wait_queue, rt->stream_wait_cond, HZ); if (rt->stream_wait_cond) { -- 2.36.1

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[PATCH] usb/6fire: fix repeated words in comments
by wangjianli 25 Jul '22

25 Jul '22

Delete the redundant word 'in'. Signed-off-by: wangjianli <wangjianli(a)cdjrlc.com> --- sound/usb/6fire/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 7168f1c6a37a..32c39d8bd2e5 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -175,7 +175,7 @@ static int usb6fire_pcm_stream_start(struct pcm_runtime *rt) } } - /* wait for first out urb to return (sent in in urb handler) */ + /* wait for first out urb to return (sent in urb handler) */ wait_event_timeout(rt->stream_wait_queue, rt->stream_wait_cond, HZ); if (rt->stream_wait_cond) -- 2.36.1

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