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

[RESEND PATCH] ASoC: wsa881x: mark read_only_wordlength flag
by Srinivas Kandagatla 14 Apr '20

14 Apr '20

WSA881x works in PDM mode so the wordlength is fixed, which also makes the only field "WordLength" in DPN_BlockCtrl1 register a read-only. Writing to this register will throw up errors with Qualcomm Controller. So use ro_blockctrl1_reg flag to mark this field as read-only so that core will not write to this register. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Hi Mark, For some reason this patch was missed during last merge window, Other patch in this series is already in mainline. Without this patch audio is not functional on DB845c and other SDM845 based platforms. Can you please take this for next possible rc. Thanks, srini sound/soc/codecs/wsa881x.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index f2d6f2f81f14..d39d479e2378 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -394,6 +394,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* COMP */ .num = 2, @@ -401,6 +402,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* BOOST */ .num = 3, @@ -408,6 +410,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, }, { /* VISENSE */ .num = 4, @@ -415,6 +418,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, + .read_only_wordlength = true, } }; -- 2.21.0

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[PATCH] ASoC: samsung: s3c24xx-i2s: Fix build after removal of DAI suspend/resume
by Krzysztof Kozlowski 14 Apr '20

14 Apr '20

Commit 450312b640f9 ("ASoC: soc-core: remove DAI suspend/resume") removed the DAI side suspend/resume hooks and switched entirely to component suspend/resume. However the Samsung SoC s3c-i2s-v2 driver was not updated. Move the suspend/resume hooks from s3c-i2s-v2.c to s3c2412-i2s.c while changing dai to component which allows to keep the struct snd_soc_component_driver const. This fixes build errors: sound/soc/samsung/s3c-i2s-v2.c: In function ‘s3c_i2sv2_register_component’: sound/soc/samsung/s3c-i2s-v2.c:730:9: error: ‘struct snd_soc_dai_driver’ has no member named ‘suspend’ dai_drv->suspend = s3c2412_i2s_suspend; Reported-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 450312b640f9 ("ASoC: soc-core: remove DAI suspend/resume") Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Only compile tested. --- sound/soc/samsung/s3c-i2s-v2.c | 57 --------------------------------- sound/soc/samsung/s3c2412-i2s.c | 56 ++++++++++++++++++++++++++++++++ 2 files changed, 56 insertions(+), 57 deletions(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 358887848293..5e95c30fb2ba 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -656,60 +656,6 @@ void s3c_i2sv2_cleanup(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(s3c_i2sv2_cleanup); -#ifdef CONFIG_PM -static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - u32 iismod; - - if (dai->active) { - i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); - i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); - i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); - - /* some basic suspend checks */ - - iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) - pr_warn("%s: RXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) - pr_warn("%s: TXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_IIS_ACTIVE) - pr_warn("%s: IIS active\n", __func__); - } - - return 0; -} - -static int s3c2412_i2s_resume(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - - pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", - dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); - - if (dai->active) { - writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); - writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); - writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); - - writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - ndelay(250); - writel(0x0, i2s->regs + S3C2412_IISFIC); - } - - return 0; -} -#else -#define s3c2412_i2s_suspend NULL -#define s3c2412_i2s_resume NULL -#endif - int s3c_i2sv2_register_component(struct device *dev, int id, const struct snd_soc_component_driver *cmp_drv, struct snd_soc_dai_driver *dai_drv) @@ -727,9 +673,6 @@ int s3c_i2sv2_register_component(struct device *dev, int id, if (!ops->delay) ops->delay = s3c2412_i2s_delay; - dai_drv->suspend = s3c2412_i2s_suspend; - dai_drv->resume = s3c2412_i2s_resume; - return devm_snd_soc_register_component(dev, cmp_drv, dai_drv, 1); } EXPORT_SYMBOL_GPL(s3c_i2sv2_register_component); diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 787a3f6e9f24..b35d828c1cfe 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -117,6 +117,60 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } +#ifdef CONFIG_PM +static int s3c2412_i2s_suspend(struct snd_soc_component *component) +{ + struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component); + u32 iismod; + + if (component->active) { + i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); + i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); + i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); + + /* some basic suspend checks */ + + iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) + pr_warn("%s: RXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) + pr_warn("%s: TXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_IIS_ACTIVE) + pr_warn("%s: IIS active\n", __func__); + } + + return 0; +} + +static int s3c2412_i2s_resume(struct snd_soc_component *component) +{ + struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component); + + pr_info("component_active %d, IISMOD %08x, IISCON %08x\n", + component->active, i2s->suspend_iismod, i2s->suspend_iiscon); + + if (component->active) { + writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); + writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); + writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); + + writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + ndelay(250); + writel(0x0, i2s->regs + S3C2412_IISFIC); + } + + return 0; +} +#else +#define s3c2412_i2s_suspend NULL +#define s3c2412_i2s_resume NULL +#endif + #define S3C2412_I2S_RATES \ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ @@ -146,6 +200,8 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = { static const struct snd_soc_component_driver s3c2412_i2s_component = { .name = "s3c2412-i2s", + .suspend = s3c2412_i2s_suspend, + .resume = s3c2412_i2s_resume, }; static int s3c2412_iis_dev_probe(struct platform_device *pdev) -- 2.17.1

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[RFC v2] ALSA: control: fix a error handling exist in snd_ctl_elem_add
by yangerkun 14 Apr '20

14 Apr '20

CVE-2020-11725 report that 'count = info->owner' may result a SIZE_OVERFLOW. 'info->owner' represent a pid, and actually, we should use info->count. Signed-off-by: yangerkun <yangerkun(a)huawei.com> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) v1->v2: reword the patch head diff --git a/sound/core/control.c b/sound/core/control.c index aa0c0cf182af..c77ca7f39637 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1431,7 +1431,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, return -ENOMEM; /* Check the number of elements for this userspace control. */ - count = info->owner; + count = info->count; if (count == 0) count = 1; -- 2.21.1

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[RFC] ALSA: control: fix a mishandled exits in snd_ctl_elem_add
by yangerkun 14 Apr '20

14 Apr '20

CVE-2020-11725 report that 'count = info->owner' may result a SIZE_OVERFLOW. 'info->owner' represent a pid, and actually, we should use info->count. Signed-off-by: yangerkun <yangerkun(a)huawei.com> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index aa0c0cf182af..c77ca7f39637 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1431,7 +1431,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, return -ENOMEM; /* Check the number of elements for this userspace control. */ - count = info->owner; + count = info->count; if (count == 0) count = 1; -- 2.21.1

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[PATCH v7 0/7] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 14 Apr '20

14 Apr '20

Add new module driver for new ASRC in i.MX8MN, several commits are added for new property fsl,asrc-format Shengjiu Wang (7): ASoC: fsl_asrc: rename asrc_priv to asrc ASoC: dt-bindings: fsl_asrc: Add new property fsl,asrc-format ASoC: fsl-asoc-card: Support new property fsl,asrc-format ASoC: fsl_asrc: Support new property fsl,asrc-format ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers changes in v7 - updated according to Nicoin's comments. - add get_pair_priv_size to replace PAIR_PRIVATE_SIZE changes in v6 - updated according to Nicoin's and Rob's comments. changes in v5 - Add new property fsl,asrc-format, rather than change fsl,asrc-width to fsl,asrc-formt. - code change for above change. changes in v4 - Add several commit for changing DT binding asrc-width to asrc-format - rename asrc_priv to asrc changes in v3 - add new commit "ASoC: fsl_asrc: Change asrc_width to asrc_format" - modify binding doc to yaml format - remove fsl_easrc_dma.c, make fsl_asrc_dma.c useable for easrc. changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,asrc.txt | 4 + .../devicetree/bindings/sound/fsl,easrc.yaml | 101 + sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 24 +- sound/soc/fsl/fsl_asrc.c | 310 +-- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 106 + sound/soc/fsl/fsl_asrc_dma.c | 54 +- sound/soc/fsl/fsl_easrc.c | 2119 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 11 files changed, 3222 insertions(+), 234 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.yaml create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h -- 2.21.0

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[PATCH] ALSA: hda: Allow setting preallocation again for x86
by Takashi Iwai 13 Apr '20

13 Apr '20

The commit c31427d0d21e ("ALSA: hda: No preallocation on x86 platforms") changed CONFIG_SND_HDA_PREALLOC_SIZE setup and its default to zero for x86, as the preallocation should work almost all cases. However, this expectation was too naive; some applications try to allocate as the max buffer size as possible, and it leads to the memory exhaustion. More badly, the commit changed the kconfig no longer adjustable for x86, so you can't fix it statically (although it can be still adjusted via procfs). So, practically seen, it's more recommended to set a reasonable limit for x86, too. This patch follows to that experience, and changes the default to 2048 and allow the kconfig adjustable again. Fixes: c31427d0d21e ("ALSA: hda: No preallocation on x86 platforms") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207223 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/Kconfig | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 4ca6b09056f3..3bc9224d5e4f 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -21,16 +21,17 @@ config SND_HDA_EXT_CORE select SND_HDA_CORE config SND_HDA_PREALLOC_SIZE - int "Pre-allocated buffer size for HD-audio driver" if !SND_DMA_SGBUF + int "Pre-allocated buffer size for HD-audio driver" range 0 32768 - default 0 if SND_DMA_SGBUF + default 2048 if SND_DMA_SGBUF default 64 if !SND_DMA_SGBUF help Specifies the default pre-allocated buffer-size in kB for the HD-audio driver. A larger buffer (e.g. 2048) is preferred for systems using PulseAudio. The default 64 is chosen just for compatibility reasons. - On x86 systems, the default is zero as we need no preallocation. + On x86 systems, the default is 2048 as a reasonable value for + most of modern systems. Note that the pre-allocation size can be changed dynamically via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. -- 2.16.4

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SOF support for Baytrail/CherryTrail
by GitHub pull_request - edited 13 Apr '20

13 Apr '20

alsa-project/alsa-ucm-conf pull request #20 was edited from plbossart: Add symlink and simple test to reuse legacy settings as is. tested on Baytrail/Asus T100TA only with SOF and legacy drivers, all other platforms are just based on same changes. sof-bytcr-rt5640 is here: http://alsa-project.org/db/?f=6f8be0520e2bbda94cb744ce66232174feec03d4 UCM validator seems to have problems with the conditional statement? Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/20 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/20.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Fix/byt
by GitHub pull_request - opened 13 Apr '20

13 Apr '20

alsa-project/alsa-ucm-conf pull request #20 was opened from plbossart: Add symlink and simple test to reuse legacy settings as is. tested on Baytrail/Asus T100TA only with SOF and legacy drivers, all other platforms are just based on same changes. sof-bytcr-rt5640 is here: http://alsa-project.org/db/?f=6f8be0520e2bbda94cb744ce66232174feec03d4 UCM validator seems to have problems with the conditional statement? Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/20 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/20.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 1/5] bytcr-rt5651: Fix dmic check in HiFi-Components.conf
by Hans de Goede 13 Apr '20

13 Apr '20

The kernel sets cfg-mic:dmic not cfg-mic:dmic1 when a digital mic is used. The wrong check for cfg-mic:dmic1 causes there to be no "Mic" device at all, which causes the check for the ConflictingDevices section of the HeadSet (mic) device to fail like e.g. so: [hans@localhost ~]$ alsaucm -c bytcr-rt5651 set _verb HiFi set _enadev Speaker ALSA lib parser.c:1138:(verb_dev_list_add) error: unable to find device 'Mic' ALSA lib parser.c:1433:(parse_verb_file) error: device management error in verb 'HiFi' ALSA lib main.c:962:(snd_use_case_mgr_open) error: failed to import bytcr-rt5651 use case configuration -2 alsaucm: error failed to open sound card bytcr-rt5651: No such file or directory This commit fixes this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/bytcr-rt5651/HiFi-Components.conf | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/ucm2/bytcr-rt5651/HiFi-Components.conf b/ucm2/bytcr-rt5651/HiFi-Components.conf index 53ebcbe..d0a573e 100644 --- a/ucm2/bytcr-rt5651/HiFi-Components.conf +++ b/ucm2/bytcr-rt5651/HiFi-Components.conf @@ -34,11 +34,11 @@ If.headphones { } } -If.dmic1 { +If.dmic { Condition { Type String Haystack "${CardComponents}" - Needle "cfg-mic:dmic1" + Needle "cfg-mic:dmic" } True { <codecs/rt5651/DigitalMic.conf> -- 2.26.0

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[PATCH 0/6] ALSA: hda: Fix/improve no-codec bus
by Takashi Iwai 13 Apr '20

13 Apr '20

Hi, here is a patch set for fixing / improving the error handling of HD-audio bus without codecs, which can be seen on some GPUs, for example. It's been debugged and tested in bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Takashi === Roy Spliet (1): ALSA: hda: Explicitly permit using autosuspend if runtime PM is supported Takashi Iwai (5): ALSA: hda: Don't release card at firmware loading error ALSA: hda: Honor PM disablement in PM freeze and thaw_noirq ops ALSA: hda: Release resources at error in delayed probe ALSA: hda: Keep the controller initialization even if no codecs found ALSA: hda: Skip controller resume if not needed include/sound/hda_codec.h | 5 +++ sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_intel.c | 106 +++++++++++++++++++++++++++------------------- sound/pci/hda/hda_intel.h | 1 + 4 files changed, 69 insertions(+), 45 deletions(-) -- 2.16.4

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