October 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/2] rt5651: Enable platforms with int mic on IN2
by Carlo Caione 18 Oct '17

18 Oct '17

From: Carlo Caione <carlo(a)endlessm.com> While working on enabling a cherry-trail laptop shipping the rt5651 codec I realized that the machine driver needed some fixup. In particular the laptop I'm working on (KIANO SlimNote 14.2) has the internal mic connected to the IN2 port. All the laptop-specific work with the related quirks will follow. Carlo Caione (2): ASoC: intel: byt: Enable quirk logic for custom maps ASoC: intel: byt: Introduce new custom IN2 map sound/soc/intel/boards/bytcr_rt5651.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) -- 2.14.2

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Re: [alsa-devel] linux-next: manual merge of the sound-asoc tree with the drm-misc tree
by Mark Brown 18 Oct '17

18 Oct '17

On Wed, Oct 18, 2017 at 10:57:33AM +0100, Mark Brown wrote: > I fixed it up (see below) and can carry the fix as necessary. This > is now fixed as far as linux-next is concerned, but any non trivial > conflicts should be mentioned to your upstream maintainer when your tree > is submitted for merging. You may also want to consider cooperating > with the maintainer of the conflicting tree to minimise any particularly > complex conflicts. Actually I'm just going to discard the AMD drivers from the ASoC tree because the build produces reams of errors like those below, the changes to move the chip type definitions around weren't fully baked. Please resend both the pull request and the patches with this fixed. Note also that if you're basing something on Linus' tree you should use a tagged release rather than just a random commit. In file included from /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/amdgpu.h:51:0, from /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/amdgpu_connectors.c:31: /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/../include/amd_shared.h:33:6: error: nested redefinition of 'enum amd_asic_type' enum amd_asic_type { ^ /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/../include/amd_shared.h:33:6: error: redeclaration of 'enum amd_asic_type' In file included from /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/../include/amd_shared.h:26:0, from /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/amdgpu.h:51, from /home/broonie/tmpfs/next/drivers/gpu/drm/amd/amdgpu/amdgpu_connectors.c:31: /home/broonie/tmpfs/next/include/drm/amd_asic_type.h:28:6: note: originally defined here enum amd_asic_type { ^

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[alsa-devel] [PATCH 0/4 resend] Add ASoC support for AMD Stoney APUs
by Alex Deucher 18 Oct '17

18 Oct '17

This patch set updates the AMD GPU and Audio CoProcessor (ACP) audio drivers and the designware i2s driver for Stoney (ST). ST is an APU similar to Carrizo (CZ) which already has ACP audio support. The i2s controller and ACP audio DMA engine are part of the GPU and both need updating so I would like to upstream the whole patch set via one tree if possible. I just sent a pull request for the pre-requisite patches which are available here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=linus-4.14-rc4-acp-prereq This is a resend of just the non pre-requisite audio patches at Mark's request. Thanks! Alex v2: - Patch 1 is already applied to the audio tree, just including it for completeness since it's required for this patch set and it's not yet in the drm tree. - New patch to share asic types between gpu and audio drivers - ACPI ID changed for rt5650 machine driver - Integrate feedback on other patches v3: - resend remaining patches that have not been applied yet v4: - Integrate feedback from v3 - Add Mark's acks from v3 - Integrate a machine driver fix from Dylan Reid Akshu Agrawal (1): ASoC: AMD: Add machine driver for cz rt5650 Vijendar Mukunda (3): ASoC: AMD: disabling memory gating in stoney platform ASoC: AMD: DMA driver changes for Stoney Platform ASoC: AMD: Audio buffer related changes for Stoney sound/soc/amd/Kconfig | 7 ++ sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-pcm-dma.c | 249 +++++++++++++++++++++++++++++++++++--------- sound/soc/amd/acp-rt5645.c | 199 +++++++++++++++++++++++++++++++++++ sound/soc/amd/acp.h | 2 + 5 files changed, 408 insertions(+), 51 deletions(-) create mode 100644 sound/soc/amd/acp-rt5645.c -- 2.5.5

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[alsa-devel] [pull v2] amdgpu pre-reqs for stoney ACP audio support
by Alex Deucher 18 Oct '17

18 Oct '17

Hi Dave, Mark, This branch against Linus' 4.14-rc4 contains the 3 pre-requisite patches for Stoney ACP audio. Mark needs these patches to apply the rest of the Stoney ACP patch set on the audio side. I'll resend the stony audio patches shortly, but there are no differences from the last version I send out. Note that depending on when these changes end up in drm-next, there may be a conflict due to f674bd28146045 (drm/amdgpu Moving amdgpu asic types to a separate file). If you end up with a conflict, please take the changes in f674bd28146045, otherwise, you'll end up with the enum defined twice. Please pull. Thanks! The following changes since commit 8a5776a5f49812d29fe4b2d0a2d71675c3facf3f: Linux 4.14-rc4 (2017-10-08 20:53:29 -0700) are available in the git repository at: git://people.freedesktop.org/~agd5f/linux linus-4.14-rc4-acp-prereq for you to fetch changes up to f674bd281460451e8784f3bc917a650e8937a8ed: drm/amdgpu Moving amdgpu asic types to a separate file (2017-10-18 10:56:27 -0400) ---------------------------------------------------------------- Akshu Agrawal (1): drm/amdgpu Moving amdgpu asic types to a separate file Vijendar Mukunda (2): drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data ASoC: AMD: Added asic_type as ACP DMA driver platform data drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 2 ++ drivers/gpu/drm/amd/include/amd_shared.h | 29 ++---------------- include/drm/amd_asic_type.h | 52 ++++++++++++++++++++++++++++++++ sound/soc/amd/acp-pcm-dma.c | 8 ++--- sound/soc/amd/acp.h | 7 +++++ 5 files changed, 65 insertions(+), 33 deletions(-) create mode 100644 include/drm/amd_asic_type.h

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[alsa-devel] [PATCH] ALSA: hda - Add model string for Intel reference board quirk
by Takashi Iwai 18 Oct '17

18 Oct '17

For allowing user to apply the existing quirk on a machine with a different SSID, add a new model string entry, alc700-ref. The quirk itself was introduced in the commit b84e843644f2: "ALSA: hda/realtek - Enable jack detection function for Intel ALC700") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/hd-audio/models.rst | 2 ++ sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 3 insertions(+) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 773d2bfacc6c..1fee5a4f6660 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -82,6 +82,8 @@ tpt460 Lenovo Thinkpad T460/560 setup dual-codecs Lenovo laptops with dual codecs +alc700-ref + Intel reference board with ALC700 codec ALC66x/67x/892 ============== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0ce71111b4e3..1a0163b81f5b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6357,6 +6357,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, {.id = ALC233_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, + {.id = ALC700_FIXUP_INTEL_REFERENCE, .name = "alc700-ref"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.14.2

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[alsa-devel] [PATCH v2 0/9] Baytrail-Cherrytrail updates
by Pierre-Louis Bossart 18 Oct '17

18 Oct '17

As discussed on the alsa-devel mailing list, here is a set of patches to fix 1. previous feedback on bytcht_rt5651 - no new functionality beyond what was already reviewed 2. cosmetic issues (variables in reverse xmas tree order, etc) 3. unnecessary clock tests 4. addition of a helper to remove duplicate code across machine drivers The last two patches were not included in the first pass, they correct an off-by-one issue which has no functional impact and remove the hard-codec compressed dailinks which don't currently work with the mainline code+publicly available firmware. Pierre-Louis Bossart (9): ASoC: Intel: bytcr_rt5651: add MCLK, quirks and cleanups ASoC: Intel: bytcr_rt5640: cosmetic fixes ASoC: Intel: cht_bsw_rt5645: cosmetic fixes ASoC: Intel: bytcht_da7213: cosmetic fixes ASoC: Intel: bytcht_es8316: remove useless code ASoC: Add helper to find codec_dai from dai_name ASoC: Intel: boards: use helper to get codec_dai ASoC: Intel: boards: fix off-by-one dailink id ASoC: Intel: boards: remove hard-coded compressed dailinks include/sound/soc.h | 14 ++ sound/soc/intel/boards/bxt_da7219_max98357a.c | 16 +- sound/soc/intel/boards/bytcht_da7213.c | 17 +- sound/soc/intel/boards/bytcht_es8316.c | 25 +-- sound/soc/intel/boards/bytcht_nocodec.c | 10 +- sound/soc/intel/boards/bytcr_rt5640.c | 105 ++++------- sound/soc/intel/boards/bytcr_rt5651.c | 236 +++++++++++++++++++++--- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 24 +-- sound/soc/intel/boards/cht_bsw_rt5645.c | 103 ++++------- sound/soc/intel/boards/cht_bsw_rt5672.c | 24 +-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 16 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 16 +- 12 files changed, 318 insertions(+), 288 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH 0/6] Baytrail-Cherrytrail updates
by Pierre-Louis Bossart 18 Oct '17

18 Oct '17

As discussed on the alsa-devel mailing list, here is a set of patches to fix 1. previous feedback on bytcht_rt5651 - no new functionality beyond what was already reviewed 2. cosmetic issues (variables in reverse xmas tree order, etc) 3. unnecessary clock tests Since the changes are very small, I grouped them by machine driver instead of generating more patches. let me know if a different split is needed. The last patch addresses Andy's comment that we should not use strncmp(a,b,strlen(b)) when strcmp(a,b) is fine. This change is applied across all machine drivers. Pierre-Louis Bossart (6): ASoC: Intel: bytcr_rt5651: add MCLK support and fix quirks ASoC: Intel: bytcr_rt5651: filter codec name and fix variable declarations ASoC: Intel: bytcr_rt5640: cosmetic fixes ASoC: Intel: cht_bsw_rt5645: cosmetic fixes ASoC: Intel: bytcht_da7213: cosmetic fixes ASoC: Intel: boards: use strcmp with constant literals sound/soc/intel/boards/bxt_da7219_max98357a.c | 3 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 3 +- sound/soc/intel/boards/bytcr_rt5640.c | 79 ++++---- sound/soc/intel/boards/bytcr_rt5651.c | 257 ++++++++++++++++++++++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 3 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 79 ++++---- sound/soc/intel/boards/cht_bsw_rt5672.c | 3 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 3 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 3 +- 10 files changed, 324 insertions(+), 115 deletions(-) -- 2.11.0

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[alsa-devel] Applied "ASoC: Intel: boards: remove hard-coded compressed dailinks" to the asoc tree
by Mark Brown 18 Oct '17

18 Oct '17

The patch ASoC: Intel: boards: remove hard-coded compressed dailinks has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 2193eb9631d272e5a8a7e7534830f555305aa556 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 12 Oct 2017 18:38:05 -0500 Subject: [PATCH] ASoC: Intel: boards: remove hard-coded compressed dailinks The hard-coded compressed dailinks are not supported using publicly-available firmwares, which creates unnecessary user confusion [1]. Even if the firmware was available, the mainline code does not have the required .dynamic=1 and .dpcm_playback=1 fields so probably never worked as is, and last and they conflict with topology-defined streams. Remove them and move on. This can be re-enabled with SOF later in a more flexible manner. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/124868.html Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bytcht_da7213.c | 9 --------- sound/soc/intel/boards/bytcht_es8316.c | 9 --------- sound/soc/intel/boards/bytcht_nocodec.c | 8 -------- sound/soc/intel/boards/bytcr_rt5640.c | 11 +---------- sound/soc/intel/boards/bytcr_rt5651.c | 11 +---------- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 8 -------- sound/soc/intel/boards/cht_bsw_rt5645.c | 8 -------- sound/soc/intel/boards/cht_bsw_rt5672.c | 8 -------- 8 files changed, 2 insertions(+), 70 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index 28a493e5318d..296b50640f46 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -185,14 +185,6 @@ static struct snd_soc_dai_link dailink[] = { .dpcm_playback = 1, .ops = &aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, /* CODEC<->CODEC link */ /* back ends */ { @@ -243,7 +235,6 @@ static int bytcht_da7213_probe(struct platform_device *pdev) card->dev = &pdev->dev; /* fix index of codec dai */ - dai_index = MERR_DPCM_COMPR + 1; for (i = 0; i < ARRAY_SIZE(dailink); i++) { if (!strcmp(dailink[i].codec_name, "i2c-DLGS7213:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 808e825c3b05..6acacc0aeedb 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -194,15 +194,6 @@ static struct snd_soc_dai_link byt_cht_es8316_dais[] = { .ops = &byt_cht_es8316_aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, - /* back ends */ { /* Only SSP2 has been tested here, so BYT-CR platforms that diff --git a/sound/soc/intel/boards/bytcht_nocodec.c b/sound/soc/intel/boards/bytcht_nocodec.c index 511e87601104..b80ec027a0e8 100644 --- a/sound/soc/intel/boards/bytcht_nocodec.c +++ b/sound/soc/intel/boards/bytcht_nocodec.c @@ -133,14 +133,6 @@ static struct snd_soc_dai_link dais[] = { .dpcm_playback = 1, .ops = &aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, /* CODEC<->CODEC link */ /* back ends */ { diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 6c7b71d7fed7..3eeb02036aa1 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -678,14 +678,6 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { .dynamic = 1, .dpcm_playback = 1, .ops = &byt_rt5640_aif1_ops, - }, - [MERR_DPCM_COMPR] = { - .name = "Baytrail Compressed Port", - .stream_name = "Baytrail Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", }, /* back ends */ { @@ -748,7 +740,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) struct sst_acpi_mach *mach; const char *i2c_name = NULL; int ret_val = 0; - int dai_index; + int dai_index = 0; int i; is_bytcr = false; @@ -762,7 +754,6 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(&byt_rt5640_card, priv); /* fix index of codec dai */ - dai_index = MERR_DPCM_COMPR + 1; for (i = 0; i < ARRAY_SIZE(byt_rt5640_dais); i++) { if (!strcmp(byt_rt5640_dais[i].codec_name, "i2c-10EC5640:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index e0caee26fb0a..3076bfc0db5e 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -389,14 +389,6 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { .dpcm_playback = 1, .ops = &byt_rt5651_aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, /* CODEC<->CODEC link */ /* back ends */ { @@ -440,7 +432,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) struct sst_acpi_mach *mach; const char *i2c_name = NULL; int ret_val = 0; - int dai_index; + int dai_index = 0; int i; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); @@ -454,7 +446,6 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(&byt_rt5651_card, priv); /* fix index of codec dai */ - dai_index = MERR_DPCM_COMPR + 1; for (i = 0; i < ARRAY_SIZE(byt_rt5651_dais); i++) { if (!strcmp(byt_rt5651_dais[i].codec_name, "i2c-10EC5651:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index a2194d993615..14dd68edfce4 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -350,14 +350,6 @@ static struct snd_soc_dai_link cht_dailink[] = { .dpcm_playback = 1, .ops = &cht_aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, /* back ends */ { .name = "SSP2-Codec", diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 1901b3b8f4d2..83c923c39393 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -445,14 +445,6 @@ static struct snd_soc_dai_link cht_dailink[] = { .dpcm_playback = 1, .ops = &cht_aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, /* CODEC<->CODEC link */ /* back ends */ { diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 3464cf41c7d1..d135640702e9 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -302,14 +302,6 @@ static struct snd_soc_dai_link cht_dailink[] = { .dpcm_playback = 1, .ops = &cht_aif1_ops, }, - [MERR_DPCM_COMPR] = { - .name = "Compressed Port", - .stream_name = "Compress", - .cpu_dai_name = "compress-cpu-dai", - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", - .platform_name = "sst-mfld-platform", - }, /* Back End DAI links */ { -- 2.14.1

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[alsa-devel] [PATCH] ASoC: wm97xx: fix compilation corner case
by Robert Jarzmik 18 Oct '17

18 Oct '17

When the old AC97 is not used, CONFIG_SND_SOC_AC97_BUS is not defined. As a consequence, in the error path, snd_soc_free_ac97_codec() is not defined and triggers a compilation error. Fix it for wm9705 and wm9712, as wm9713 is correctly written. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/soc/codecs/wm9705.c | 8 ++------ sound/soc/codecs/wm9712.c | 7 ++----- 2 files changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 68c204e3599f..2c09f71fe433 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -321,7 +321,6 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) { struct wm9705_priv *wm9705 = snd_soc_codec_get_drvdata(codec); struct regmap *regmap; - int ret; if (wm9705->mfd_pdata) { wm9705->ac97 = wm9705->mfd_pdata->ac97; @@ -337,8 +336,8 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) regmap = regmap_init_ac97(wm9705->ac97, &wm9705_regmap_config); if (IS_ERR(regmap)) { - ret = PTR_ERR(regmap); - goto err_free_ac97_codec; + snd_soc_free_ac97_codec(wm9705->ac97); + return PTR_ERR(regmap); } #endif } @@ -347,9 +346,6 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) snd_soc_codec_init_regmap(codec, regmap); return 0; -err_free_ac97_codec: - snd_soc_free_ac97_codec(wm9705->ac97); - return ret; } static int wm9705_soc_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 9db5c7783de5..4f6d1a442bc4 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -656,8 +656,8 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) regmap = regmap_init_ac97(wm9712->ac97, &wm9712_regmap_config); if (IS_ERR(regmap)) { - ret = PTR_ERR(regmap); - goto err_free_ac97_codec; + snd_soc_free_ac97_codec(wm9712->ac97); + return PTR_ERR(regmap); } #endif } @@ -668,9 +668,6 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, AC97_VIDEO, 0x3000, 0x3000); return 0; -err_free_ac97_codec: - snd_soc_free_ac97_codec(wm9712->ac97); - return ret; } static int wm9712_soc_remove(struct snd_soc_codec *codec) -- 2.11.0

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[alsa-devel] [PATCH] ASoC: samsung: i2s: disable secondary DAI until it gets fixed
by Marek Szyprowski 18 Oct '17

18 Oct '17

Secondary DAI in Exynos I2S driver is not used by any of the currently supported boards and it causes problems due to some limitations in the ASoC code. Disable it until it gets proper support both by board-specific and ASoC core code. Also disable IDMA support, which relies on secondary DAI presence. This patch fixes following kernel warning: samsung-i2s 3830000.i2s: ASoC: Failed to create component debugfs directory samsung-i2s 3830000.i2s: ASoC: Failed to create component debugfs directory ------------[ cut here ]------------ WARNING: CPU: 3 PID: 82 at fs/proc/generic.c:330 proc_register+0xec/0x10c proc_dir_entry 'sub0/prealloc' already registered Modules linked in: CPU: 3 PID: 82 Comm: kworker/3:1 Not tainted 4.14.0-rc5-next-20171017 #3089 Hardware name: SAMSUNG EXYNOS (Flattened Device Tree) Workqueue: events deferred_probe_work_func [<c0110114>] (unwind_backtrace) from [<c010c900>] (show_stack+0x10/0x14) [<c010c900>] (show_stack) from [<c083e664>] (dump_stack+0x90/0xc8) [<c083e664>] (dump_stack) from [<c011d2b8>] (__warn+0xd4/0x100) [<c011d2b8>] (__warn) from [<c011d384>] (warn_slowpath_fmt+0x38/0x48) [<c011d384>] (warn_slowpath_fmt) from [<c0271268>] (proc_register+0xec/0x10c) [<c0271268>] (proc_register) from [<c027130c>] (proc_create_data+0x84/0xc8) [<c027130c>] (proc_create_data) from [<c061afbc>] (snd_info_register+0x64/0xcc) [<c061afbc>] (snd_info_register) from [<c062a6e0>] (snd_pcm_lib_preallocate_pages1+0x78/0x1a0) [<c062a6e0>] (snd_pcm_lib_preallocate_pages1) from [<c063eef4>] (dmaengine_pcm_new+0xa0/0x1ec) [<c063eef4>] (dmaengine_pcm_new) from [<c062b9f8>] (snd_soc_platform_drv_pcm_new+0x1c/0x28) [<c062b9f8>] (snd_soc_platform_drv_pcm_new) from [<c063d54c>] (soc_new_pcm+0x2f4/0x4f4) [<c063d54c>] (soc_new_pcm) from [<c063107c>] (snd_soc_register_card+0xc4c/0xdc4) [<c063107c>] (snd_soc_register_card) from [<c063db30>] (devm_snd_soc_register_card+0x34/0x70) [<c063db30>] (devm_snd_soc_register_card) from [<c064af60>] (asoc_simple_card_probe+0x230/0x47c) [<c064af60>] (asoc_simple_card_probe) from [<c047f8fc>] (platform_drv_probe+0x50/0xb0) [<c047f8fc>] (platform_drv_probe) from [<c047dee0>] (driver_probe_device+0x2a0/0x46c) [<c047dee0>] (driver_probe_device) from [<c047c0bc>] (bus_for_each_drv+0x44/0x8c) [<c047c0bc>] (bus_for_each_drv) from [<c047db50>] (__device_attach+0xa0/0x134) [<c047db50>] (__device_attach) from [<c047cf7c>] (bus_probe_device+0x88/0x90) [<c047cf7c>] (bus_probe_device) from [<c047d484>] (deferred_probe_work_func+0x3c/0x168) [<c047d484>] (deferred_probe_work_func) from [<c01371f8>] (process_one_work+0x188/0x41c) [<c01371f8>] (process_one_work) from [<c01374b4>] (process_scheduled_works+0x28/0x38) [<c01374b4>] (process_scheduled_works) from [<c01376d4>] (worker_thread+0x210/0x4dc) [<c01376d4>] (worker_thread) from [<c013d9cc>] (kthread+0x128/0x164) [<c013d9cc>] (kthread) from [<c0108848>] (ret_from_fork+0x14/0x2c) ---[ end trace bad8db6ee771d094 ]-- Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- sound/soc/samsung/i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 8d5e1861abb1..233f1c9a4b6c 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1292,6 +1292,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } } } + quirks &= ~(QUIRK_SEC_DAI | QUIRK_SUPPORTS_IDMA); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); pri_dai->addr = devm_ioremap_resource(&pdev->dev, res); -- 2.14.2

2 1

[alsa-devel] [PATCH 1/4] ASoC: rt5659: register power bit of LOUT Amp
by Bard Liao 18 Oct '17

18 Oct '17

There is a power bit for LOUT Amp. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5659.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 71216db..999432e 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2744,7 +2744,8 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5659_hp_event, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_PGA("LOUT Amp", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("LOUT Amp", 1, RT5659_PWR_ANLG_1, RT5659_PWR_LM_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Charge Pump", SND_SOC_NOPM, 0, 0, rt5659_charge_pump_event, SND_SOC_DAPM_PRE_PMU | -- 2.7.4

2 7

[alsa-devel] [PATCH] ALSA: hda: Abort capability probe on invalid capability
by Ughreja, Rakesh A 18 Oct '17

18 Oct '17

From: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> When an invalid capability is discovered, stop traversing the capability link list further. Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> --- sound/hda/hdac_controller.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 978dc18..4fa0cc0 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -314,7 +314,8 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus) break; default: - dev_dbg(bus->dev, "Unknown capability %d\n", cur_cap); + cur_cap = 0; + dev_err(bus->dev, "Unknown capability %d\n", cur_cap); break; } -- 2.7.4

2 3

[alsa-devel] [PATCH] ALSA: hda: Abort capability probe at invalid register read
by Takashi Iwai 18 Oct '17

18 Oct '17

The loop in snd_hdac_bus_parse_capabilities() may go to nirvana when it hits an invalid register value read: BUG: unable to handle kernel paging request at ffffad5dc41f3fff IP: pci_azx_readl+0x5/0x10 [snd_hda_intel] Call Trace: snd_hdac_bus_parse_capabilities+0x3c/0x1f0 [snd_hda_core] azx_probe_continue+0x7d5/0x940 [snd_hda_intel] ..... This happened on a new Intel machine, and we need to check the value and abort the loop accordingly. [Note: the fixes tag below indicates only the commit where this patch can be applied; the original problem was introduced even before that commit] Fixes: 6720b38420a0 ("ALSA: hda - move bus_parse_capabilities to core") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_controller.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 978dc1801b3a..f6d2985b2520 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -284,6 +284,11 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus) dev_dbg(bus->dev, "HDA capability ID: 0x%x\n", (cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF); + if (cur_cap == -1) { + dev_dbg(bus->dev, "Invalid capability reg read\n"); + break; + } + switch ((cur_cap & AZX_CAP_HDR_ID_MASK) >> AZX_CAP_HDR_ID_OFF) { case AZX_ML_CAP_ID: dev_dbg(bus->dev, "Found ML capability\n"); -- 2.14.2

3 8

[alsa-devel] [PATCH v2 0/2] Add snd_card_disconnect_sync() helper
by Takashi Iwai 18 Oct '17

18 Oct '17

Hi, here is a refreshed series containing ALSA core changes Morimoto-san posted. Basically the code is same but it's split, and more comments and an error message are added. I tested lightly with the existing hot-pluggable device and no regression was seen. The commits are found in topic/card-disconnect branch of sound git tree, too. Mark, please pull it to yours when applying the further patches relying on this new helper function. thanks, Takashi === Takashi Iwai (2): ALSA: add snd_card_disconnect_sync() ALSA: pcm: Forcibly stop at disconnect callback include/sound/core.h | 2 ++ sound/core/init.c | 32 ++++++++++++++++++++++++++++++++ sound/core/pcm.c | 4 ++++ 3 files changed, 38 insertions(+) -- 2.14.2

4 22

[alsa-devel] [PATCH] ALSA: seq: Enable 'use' locking in all configurations
by Ben Hutchings 18 Oct '17

18 Oct '17

The 'use' locking macros are no-ops if neither SMP or SND_DEBUG is enabled. This might once have been OK in non-preemptible configurations, but even in that case snd_seq_read() may sleep while relying on a 'use' lock. So always use the proper implementations. Cc: stable(a)vger.kernel.org Signed-off-by: Ben Hutchings <ben.hutchings(a)codethink.co.uk> --- sound/core/seq/seq_lock.c | 4 ---- sound/core/seq/seq_lock.h | 12 ------------ 2 files changed, 16 deletions(-) diff --git a/sound/core/seq/seq_lock.c b/sound/core/seq/seq_lock.c index 0ff7926a5a69..cda64b489e42 100644 --- a/sound/core/seq/seq_lock.c +++ b/sound/core/seq/seq_lock.c @@ -23,8 +23,6 @@ #include <sound/core.h> #include "seq_lock.h" -#if defined(CONFIG_SMP) || defined(CONFIG_SND_DEBUG) - /* wait until all locks are released */ void snd_use_lock_sync_helper(snd_use_lock_t *lockp, const char *file, int line) { @@ -41,5 +39,3 @@ void snd_use_lock_sync_helper(snd_use_lock_t *lockp, const char *file, int line) } } EXPORT_SYMBOL(snd_use_lock_sync_helper); - -#endif diff --git a/sound/core/seq/seq_lock.h b/sound/core/seq/seq_lock.h index 54044bc2c9ef..ac38031c370e 100644 --- a/sound/core/seq/seq_lock.h +++ b/sound/core/seq/seq_lock.h @@ -3,8 +3,6 @@ #include <linux/sched.h> -#if defined(CONFIG_SMP) || defined(CONFIG_SND_DEBUG) - typedef atomic_t snd_use_lock_t; /* initialize lock */ @@ -20,14 +18,4 @@ typedef atomic_t snd_use_lock_t; void snd_use_lock_sync_helper(snd_use_lock_t *lock, const char *file, int line); #define snd_use_lock_sync(lockp) snd_use_lock_sync_helper(lockp, __BASE_FILE__, __LINE__) -#else /* SMP || CONFIG_SND_DEBUG */ - -typedef spinlock_t snd_use_lock_t; /* dummy */ -#define snd_use_lock_init(lockp) /**/ -#define snd_use_lock_use(lockp) /**/ -#define snd_use_lock_free(lockp) /**/ -#define snd_use_lock_sync(lockp) /**/ - -#endif /* SMP || CONFIG_SND_DEBUG */ - #endif /* __SND_SEQ_LOCK_H */ -- 2.15.0.rc0

2 1

[alsa-devel] Applied "drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data" to the asoc tree
by Mark Brown 18 Oct '17

18 Oct '17

The patch drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d4ba4ef2e894c10f2c16497fb0659e5ded502f97 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Date: Mon, 9 Oct 2017 16:35:12 -0400 Subject: [PATCH] drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data asic_type information is passed to ACP DMA Driver as platform data. Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c b/drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c index a52795d9b458..ebca22302ebb 100644 --- a/drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c +++ b/drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c @@ -371,6 +371,8 @@ static int acp_hw_init(void *handle) adev->acp.acp_cell[0].name = "acp_audio_dma"; adev->acp.acp_cell[0].num_resources = 4; adev->acp.acp_cell[0].resources = &adev->acp.acp_res[0]; + adev->acp.acp_cell[0].platform_data = &adev->asic_type; + adev->acp.acp_cell[0].pdata_size = sizeof(adev->asic_type); adev->acp.acp_cell[1].name = "designware-i2s"; adev->acp.acp_cell[1].num_resources = 1; -- 2.14.1

1 0

[alsa-devel] Applied "ASoC: AMD: Added asic_type as ACP DMA driver platform data" to the asoc tree
by Mark Brown 18 Oct '17

18 Oct '17

The patch ASoC: AMD: Added asic_type as ACP DMA driver platform data has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 01a6b1634d50dba65133a89fe19f182173bb8e8f Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Date: Mon, 9 Oct 2017 16:36:08 -0400 Subject: [PATCH] ASoC: AMD: Added asic_type as ACP DMA driver platform data asic_type information is passed to ACP DMA Driver as platform data. Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 8 ++------ sound/soc/amd/acp.h | 7 +++++++ 2 files changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 08b1399d1da2..dcbf9973884d 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -73,12 +73,6 @@ static const struct snd_pcm_hardware acp_pcm_hardware_capture = { .periods_max = CAPTURE_MAX_NUM_PERIODS, }; -struct audio_drv_data { - struct snd_pcm_substream *play_stream; - struct snd_pcm_substream *capture_stream; - void __iomem *acp_mmio; -}; - static u32 acp_reg_read(void __iomem *acp_mmio, u32 reg) { return readl(acp_mmio + (reg * 4)); @@ -916,6 +910,7 @@ static int acp_audio_probe(struct platform_device *pdev) int status; struct audio_drv_data *audio_drv_data; struct resource *res; + const u32 *pdata = pdev->dev.platform_data; audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), GFP_KERNEL); @@ -932,6 +927,7 @@ static int acp_audio_probe(struct platform_device *pdev) audio_drv_data->play_stream = NULL; audio_drv_data->capture_stream = NULL; + audio_drv_data->asic_type = *pdata; res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); if (!res) { diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 330832ef4e5e..28cf9140f49c 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -84,6 +84,13 @@ struct audio_substream_data { void __iomem *acp_mmio; }; +struct audio_drv_data { + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; + void __iomem *acp_mmio; + u32 asic_type; +}; + enum { ACP_TILE_P1 = 0, ACP_TILE_P2, -- 2.14.1

1 0

[alsa-devel] [PATCH 0/9 v2] Add ASoC support for AMD Stoney APUs
by Alex Deucher 18 Oct '17

18 Oct '17

This patch set updates the AMD GPU and Audio CoProcessor (ACP) audio drivers and the designware i2s driver for Stoney (ST). ST is an APU similar to Carrizo (CZ) which already has ACP audio support. The i2s controller and ACP audio DMA engine are part of the GPU and both need updating so I would like to upstream the whole patch set via one tree if possible. The current code is based on drm-next, but I'm happy to rebase on whatever tree this ends up going through if there are any problems applying. The entire patch set can be viewed here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=stoney_acp Thanks! Alex v2: - Patch 1 is already applied to the audio tree, just including it for completeness since it's required for this patch set and it's not yet in the drm tree. - New patch to share asic types between gpu and audio drivers - ACPI ID changed for rt5650 machine driver - Integrate feedback on other patches Akshu Agrawal (2): drm/amdgpu Moving amdgpu asic types to a separate file ASoC: AMD: Add machine driver for cz rt5650 Vijendar Mukunda (7): ASoC: dwc: Added a quirk DW_I2S_QUIRK_16BIT_IDX_OVERRIDE to dwc driver drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data drm/amd/amdgpu: Added a quirk for Stoney platform ASoC: AMD: disabling memory gating in stoney platform ASoC: AMD: DMA driver changes for Stoney Platform ASoC: AMD: Audio buffer related changes for Stoney drm/amd/amdgpu: Disabling Power Gating for Stoney platform drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 74 ++++++---- drivers/gpu/drm/amd/include/amd_shared.h | 29 +--- include/drm/amd_asic_type.h | 52 +++++++ include/sound/designware_i2s.h | 1 + sound/soc/amd/Kconfig | 7 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-pcm-dma.c | 243 ++++++++++++++++++++++++------- sound/soc/amd/acp-rt5645.c | 210 ++++++++++++++++++++++++++ sound/soc/amd/acp.h | 9 ++ sound/soc/dwc/dwc-i2s.c | 6 + 10 files changed, 525 insertions(+), 108 deletions(-) create mode 100644 include/drm/amd_asic_type.h create mode 100644 sound/soc/amd/acp-rt5645.c -- 2.5.5

2 14

[alsa-devel] [PATCH 0/6 v4] Add ASoC support for AMD Stoney APUs
by Alex Deucher 18 Oct '17

18 Oct '17

This patch set updates the AMD GPU and Audio CoProcessor (ACP) audio drivers and the designware i2s driver for Stoney (ST). ST is an APU similar to Carrizo (CZ) which already has ACP audio support. The i2s controller and ACP audio DMA engine are part of the GPU and both need updating so I would like to upstream the whole patch set via one tree if possible. The entire patch set can be viewed here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=stoney_acp3 I just need acks from audio on the remaining patches that touch audio (1, 5, 6), and then I can send a pull request. Thanks! Alex v2: - Patch 1 is already applied to the audio tree, just including it for completeness since it's required for this patch set and it's not yet in the drm tree. - New patch to share asic types between gpu and audio drivers - ACPI ID changed for rt5650 machine driver - Integrate feedback on other patches v3: - resend remaining patches that have not been applied yet v4: - Integrate feedback from v3 - Add Mark's acks from v3 - Integrate a machine driver fix from Dylan Reid Akshu Agrawal (2): drm/amdgpu Moving amdgpu asic types to a separate file ASoC: AMD: Add machine driver for cz rt5650 Vijendar Mukunda (4): drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data ASoC: AMD: disabling memory gating in stoney platform ASoC: AMD: DMA driver changes for Stoney Platform ASoC: AMD: Audio buffer related changes for Stoney drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 2 + drivers/gpu/drm/amd/include/amd_shared.h | 29 +--- include/drm/amd_asic_type.h | 52 +++++++ sound/soc/amd/Kconfig | 7 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-pcm-dma.c | 253 ++++++++++++++++++++++++------- sound/soc/amd/acp-rt5645.c | 199 ++++++++++++++++++++++++ sound/soc/amd/acp.h | 9 ++ 8 files changed, 471 insertions(+), 82 deletions(-) create mode 100644 include/drm/amd_asic_type.h create mode 100644 sound/soc/amd/acp-rt5645.c -- 2.5.5

3 14

[alsa-devel] [PATCH] ALSA: emu10k1: remove redundant assignment to tmp
by Colin King 17 Oct '17

17 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> The assignment to variable tmp is redundant as the value is never read and a new value is assigned to tmp in the following for-loop, so remove the assignment. Cleans up clang warning: Value stored to 'tmp' is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/emu10k1/emuproc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index cf05229b569b..bde0d1954f56 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -280,7 +280,6 @@ static void snd_emu10k1_proc_rates_read(struct snd_info_entry *entry, struct snd_emu10k1 *emu = entry->private_data; unsigned int val, tmp, n; val = snd_emu10k1_ptr20_read(emu, CAPTURE_RATE_STATUS, 0); - tmp = (val >> 16) & 0x8; for (n = 0; n < 4; n++) { tmp = val >> (16 + (n*4)); if (tmp & 0x8) snd_iprintf(buffer, "Channel %d: Rate=%d\n", n, samplerate[tmp & 0x7]); -- 2.14.1

2 1

Re: [alsa-devel] stable-rc build: 9 warnings 0 failures (stable-rc/v3.18.75-20-g4ffe554)
by Arnd Bergmann 17 Oct '17

17 Oct '17

On Tue, Oct 17, 2017 at 12:22 PM, Olof's autobuilder <build(a)lixom.net> wrote: > Warnings: > > 9 sound/usb/mixer.c:2161:27: warning: passing argument 1 of 'snd_usb_mixer_disconnect' from incompatible pointer type [-Wincompatible-pointer-types] This broke after backporting 124751d5e63c ("ALSA: usb-audio: Kill stray URB at exiting") into linux-3.18. The interface was changed in commit a6cece9d8199 ("ALSA: usb-audio: Pass direct struct pointer instead of list_head"), which was merged in linux-3.19. I don't immediately see how this can be fixed. Backporting a6cece9d8199 would probably work. Arnd

3 2

[alsa-devel] [PATCH] ALSA: au88x0: remove redundant assignment of variable i
by Colin King 17 Oct '17

17 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> The assignment to variable i is redundant as it is never read because it is updated a little later on. Remove this assignment. Cleans up clang warning: Value stored to 'i' is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/au88x0/au88x0_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index c308a4f70550..4083c8b01619 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2628,7 +2628,7 @@ static void vortex_spdif_init(vortex_t * vortex, int spdif_sr, int spdif_mode) else edi = 0x1ffff; } else { - i = edi = 0x800; + edi = 0x800; } /* this_04 and this_08 are the CASp4Src's (samplerate converters) */ vortex_src_setupchannel(vortex, this_04, edi, 0, 1, -- 2.14.1

2 1

[alsa-devel] [PATCH] ASoC: tpa61xx: add back chip name to the controls
by Lucas Stach 17 Oct '17

17 Oct '17

Commit cb7e62256e99( ASoC: tpa6130a2: Register component) removed the chip name from the controls, which causes issues as the TPA61xx is not a standalone codec, but a headphone amplifier, which may be connected to the line out of a codec which already exposes a "Headphone Playback Volume" control. In that case we end up with duplicate controls, causing the the probe of the soundcard to be aborted. Signed-off-by: Lucas Stach <l.stach(a)pengutronix.de> --- sound/soc/codecs/tpa6130a2.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 2e014c80d113..abca7eedb6be 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -136,7 +136,7 @@ static const DECLARE_TLV_DB_RANGE(tpa6130_tlv, ); static const struct snd_kcontrol_new tpa6130a2_controls[] = { - SOC_SINGLE_TLV("Headphone Playback Volume", + SOC_SINGLE_TLV("TPA6130A2 Headphone Playback Volume", TPA6130A2_REG_VOL_MUTE, 0, 0x3f, 0, tpa6130_tlv), }; @@ -148,7 +148,7 @@ static const DECLARE_TLV_DB_RANGE(tpa6140_tlv, ); static const struct snd_kcontrol_new tpa6140a2_controls[] = { - SOC_SINGLE_TLV("Headphone Playback Volume", + SOC_SINGLE_TLV("TPA6140A2 Headphone Playback Volume", TPA6130A2_REG_VOL_MUTE, 1, 0x1f, 0, tpa6140_tlv), }; -- 2.11.0

2 2

Re: [alsa-devel] [PATCH] ASoC: sgtl5000: add support for a fixed sysclk frequency
by Fabio Estevam 17 Oct '17

17 Oct '17

[Sorry for the top-posting. I did not see your message in the mailing list] Hi Jakob, arch/arm/boot/dts/imx51-babbage.dts has SYS_MCLK generated by an external fixed oscillator and it works fine. Do you really need this new property? ________________________________ From: Jakob Unterwurzacher <jakob.unterwurzacher(a)theobroma-systems.com> Sent: Friday, October 6, 2017 5:34:44 AM To: linux-kernel(a)vger.kernel.org Cc: Jakob Unterwurzacher; Klaus Goger; Liam Girdwood; Mark Brown; Rob Herring; Mark Rutland; Jaroslav Kysela; Takashi Iwai; Richard Leitner; Fabio Estevam; Bhumika Goyal; alsa-devel(a)alsa-project.org; devicetree(a)vger.kernel.org Subject: [PATCH] ASoC: sgtl5000: add support for a fixed sysclk frequency The system clock, also called SYS_MCLK in the SGTL5000 datasheet, is usually generated by the I2S master with a frequency proportional to the sample rate. However, this is not always the case. Compute modules following the Qseven specification, for example, do not even have a pin for I2S SYS_MCLK on their board-to-board connector. Only I2S_WS, I2S_RST# I2S_CLK, I2S_SDI and I2S_SDO are specified. Instead, SYS_MCLK is usually generated by an external fixed oscillator. Add support for this scenario, enabled by the new "sysclk-fixed" devicetree property. Tested on a "Puma" RK3399-Q7 compute module with a "Haikou" baseboard that has a fixed 24.576 MHz oscillator connected to the codec's SYS_MCLK. Signed-off-by: Jakob Unterwurzacher <jakob.unterwurzacher(a)theobroma-systems.com> Cc: Klaus Goger <klaus.goger(a)theobroma-systems.com> --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 5 +++++ sound/soc/codecs/sgtl5000.c | 12 ++++++++++++ 2 files changed, 17 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 7a73a9d62015..0d0dd58b2d4a 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -35,6 +35,11 @@ VDDIO 1.8V 2.5V 3.3V 2 = 3.33 mA 5.74 mA 8.03 mA 3 = 4.99 mA 8.61 mA 12.05 mA +- sysclk-fixed : If this property exists, the sysclk is considered to be + generated by a fixed-frequency oscillator. The frequency value is read + from the clock defined in the "clocks" property. set_sysclk requests + are ignored. + Example: codec: sgtl5000@0a { diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f2bb4feba3b6..be3aea89371c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -133,6 +133,7 @@ struct sgtl5000_priv { u8 micbias_resistor; u8 micbias_voltage; u8 lrclk_strength; + u8 sysclk_fixed; /* sysclk rate cannot be changed */ }; /* @@ -613,6 +614,10 @@ static int sgtl5000_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case SGTL5000_SYSCLK: + if (sgtl5000->sysclk_fixed) { + dev_dbg(codec->dev, "sysclk-fixed: ignoring set_sysclk request\n"); + return 0; + } sgtl5000->sysclk = freq; break; default: @@ -1444,6 +1449,13 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, } else { sgtl5000->micbias_voltage = 0; } + if (of_property_read_bool(np, "sysclk-fixed")) { + sgtl5000->sysclk = clk_get_rate(sgtl5000->mclk); + sgtl5000->sysclk_fixed = 1; + dev_dbg(&client->dev, + "sysclk-fixed: setting sysclk to fixed value %d\n", + sgtl5000->sysclk); + } } sgtl5000->lrclk_strength = I2S_LRCLK_STRENGTH_LOW; -- 2.11.0

3 6

[alsa-devel] [PATCH] ALSA: usb-audio: Convert timers to use timer_setup()
by Kees Cook 17 Oct '17

17 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Clemens Ladisch <clemens(a)ladisch.de> Cc: Andrej Krutak <dev(a)andree.sk> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: Dan Carpenter <dan.carpenter(a)oracle.com> Cc: "Hans P. Möller Ebner" <hmoller(a)uc.cl> Cc: Bhumika Goyal <bhumirks(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/usb/line6/driver.c | 4 ++-- sound/usb/line6/driver.h | 3 +-- sound/usb/line6/pod.c | 11 +++++------ sound/usb/line6/podhd.c | 10 +++++----- sound/usb/line6/toneport.c | 7 +++---- sound/usb/line6/variax.c | 32 ++++++++++++++++---------------- sound/usb/midi.c | 7 +++---- 7 files changed, 35 insertions(+), 39 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 8d5a454842f4..4f9613e5fc9e 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -199,9 +199,9 @@ static int line6_send_raw_message_async_part(struct message *msg, Setup and start timer. */ void line6_start_timer(struct timer_list *timer, unsigned long msecs, - void (*function)(unsigned long), unsigned long data) + void (*function)(struct timer_list *t)) { - setup_timer(timer, function, data); + timer->function = (TIMER_FUNC_TYPE)function; mod_timer(timer, jiffies + msecs_to_jiffies(msecs)); } EXPORT_SYMBOL_GPL(line6_start_timer); diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index dc97895547be..61425597eb61 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -198,8 +198,7 @@ extern int line6_send_sysex_message(struct usb_line6 *line6, extern ssize_t line6_set_raw(struct device *dev, struct device_attribute *attr, const char *buf, size_t count); extern void line6_start_timer(struct timer_list *timer, unsigned long msecs, - void (*function)(unsigned long), - unsigned long data); + void (*function)(struct timer_list *t)); extern int line6_version_request_async(struct usb_line6 *line6); extern int line6_write_data(struct usb_line6 *line6, unsigned address, void *data, unsigned datalen); diff --git a/sound/usb/line6/pod.c b/sound/usb/line6/pod.c index 358224cc5638..020c81818951 100644 --- a/sound/usb/line6/pod.c +++ b/sound/usb/line6/pod.c @@ -174,7 +174,7 @@ static const char pod_version_header[] = { }; /* forward declarations: */ -static void pod_startup2(unsigned long data); +static void pod_startup2(struct timer_list *t); static void pod_startup3(struct usb_line6_pod *pod); static char *pod_alloc_sysex_buffer(struct usb_line6_pod *pod, int code, @@ -286,13 +286,12 @@ static void pod_startup1(struct usb_line6_pod *pod) CHECK_STARTUP_PROGRESS(pod->startup_progress, POD_STARTUP_INIT); /* delay startup procedure: */ - line6_start_timer(&pod->startup_timer, POD_STARTUP_DELAY, pod_startup2, - (unsigned long)pod); + line6_start_timer(&pod->startup_timer, POD_STARTUP_DELAY, pod_startup2); } -static void pod_startup2(unsigned long data) +static void pod_startup2(struct timer_list *t) { - struct usb_line6_pod *pod = (struct usb_line6_pod *)data; + struct usb_line6_pod *pod = from_timer(pod, t, startup_timer); struct usb_line6 *line6 = &pod->line6; CHECK_STARTUP_PROGRESS(pod->startup_progress, POD_STARTUP_VERSIONREQ); @@ -413,7 +412,7 @@ static int pod_init(struct usb_line6 *line6, line6->process_message = line6_pod_process_message; line6->disconnect = line6_pod_disconnect; - init_timer(&pod->startup_timer); + timer_setup(&pod->startup_timer, NULL, 0); INIT_WORK(&pod->startup_work, pod_startup4); /* create sysfs entries: */ diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index 46fa8c046cb5..36ed9c85c0eb 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -158,7 +158,7 @@ static struct line6_pcm_properties podx3_pcm_properties = { }; static struct usb_driver podhd_driver; -static void podhd_startup_start_workqueue(unsigned long data); +static void podhd_startup_start_workqueue(struct timer_list *t); static void podhd_startup_workqueue(struct work_struct *work); static int podhd_startup_finalize(struct usb_line6_podhd *pod); @@ -208,12 +208,12 @@ static void podhd_startup(struct usb_line6_podhd *pod) /* delay startup procedure: */ line6_start_timer(&pod->startup_timer, PODHD_STARTUP_DELAY, - podhd_startup_start_workqueue, (unsigned long)pod); + podhd_startup_start_workqueue); } -static void podhd_startup_start_workqueue(unsigned long data) +static void podhd_startup_start_workqueue(struct timer_list *t) { - struct usb_line6_podhd *pod = (struct usb_line6_podhd *)data; + struct usb_line6_podhd *pod = from_timer(pod, t, startup_timer); CHECK_STARTUP_PROGRESS(pod->startup_progress, PODHD_STARTUP_SCHEDULE_WORKQUEUE); @@ -319,7 +319,7 @@ static int podhd_init(struct usb_line6 *line6, line6->disconnect = podhd_disconnect; - init_timer(&pod->startup_timer); + timer_setup(&pod->startup_timer, NULL, 0); INIT_WORK(&pod->startup_work, podhd_startup_workqueue); if (pod->line6.properties->capabilities & LINE6_CAP_CONTROL) { diff --git a/sound/usb/line6/toneport.c b/sound/usb/line6/toneport.c index ba7975c0d03d..750467fb95db 100644 --- a/sound/usb/line6/toneport.c +++ b/sound/usb/line6/toneport.c @@ -241,9 +241,9 @@ static int snd_toneport_source_put(struct snd_kcontrol *kcontrol, return 1; } -static void toneport_start_pcm(unsigned long arg) +static void toneport_start_pcm(struct timer_list *t) { - struct usb_line6_toneport *toneport = (struct usb_line6_toneport *)arg; + struct usb_line6_toneport *toneport = from_timer(toneport, t, timer); struct usb_line6 *line6 = &toneport->line6; line6_pcm_acquire(line6->line6pcm, LINE6_STREAM_MONITOR, true); @@ -415,8 +415,7 @@ static int toneport_init(struct usb_line6 *line6, struct usb_line6_toneport *toneport = (struct usb_line6_toneport *) line6; toneport->type = id->driver_info; - setup_timer(&toneport->timer, toneport_start_pcm, - (unsigned long)toneport); + timer_setup(&toneport->timer, toneport_start_pcm, 0); line6->disconnect = line6_toneport_disconnect; diff --git a/sound/usb/line6/variax.c b/sound/usb/line6/variax.c index 0c4512d0382e..e8c852b2ce35 100644 --- a/sound/usb/line6/variax.c +++ b/sound/usb/line6/variax.c @@ -82,9 +82,9 @@ static const char variax_activate[] = { }; /* forward declarations: */ -static void variax_startup2(unsigned long data); -static void variax_startup4(unsigned long data); -static void variax_startup5(unsigned long data); +static void variax_startup2(struct timer_list *t); +static void variax_startup4(struct timer_list *t); +static void variax_startup5(struct timer_list *t); static void variax_activate_async(struct usb_line6_variax *variax, int a) { @@ -106,12 +106,12 @@ static void variax_startup1(struct usb_line6_variax *variax) /* delay startup procedure: */ line6_start_timer(&variax->startup_timer1, VARIAX_STARTUP_DELAY1, - variax_startup2, (unsigned long)variax); + variax_startup2); } -static void variax_startup2(unsigned long data) +static void variax_startup2(struct timer_list *t) { - struct usb_line6_variax *variax = (struct usb_line6_variax *)data; + struct usb_line6_variax *variax = from_timer(variax, t, startup_timer1); struct usb_line6 *line6 = &variax->line6; /* schedule another startup procedure until startup is complete: */ @@ -120,7 +120,7 @@ static void variax_startup2(unsigned long data) variax->startup_progress = VARIAX_STARTUP_VERSIONREQ; line6_start_timer(&variax->startup_timer1, VARIAX_STARTUP_DELAY1, - variax_startup2, (unsigned long)variax); + variax_startup2); /* request firmware version: */ line6_version_request_async(line6); @@ -132,12 +132,12 @@ static void variax_startup3(struct usb_line6_variax *variax) /* delay startup procedure: */ line6_start_timer(&variax->startup_timer2, VARIAX_STARTUP_DELAY3, - variax_startup4, (unsigned long)variax); + variax_startup4); } -static void variax_startup4(unsigned long data) +static void variax_startup4(struct timer_list *t) { - struct usb_line6_variax *variax = (struct usb_line6_variax *)data; + struct usb_line6_variax *variax = from_timer(variax, t, startup_timer2); CHECK_STARTUP_PROGRESS(variax->startup_progress, VARIAX_STARTUP_ACTIVATE); @@ -145,12 +145,12 @@ static void variax_startup4(unsigned long data) /* activate device: */ variax_activate_async(variax, 1); line6_start_timer(&variax->startup_timer2, VARIAX_STARTUP_DELAY4, - variax_startup5, (unsigned long)variax); + variax_startup5); } -static void variax_startup5(unsigned long data) +static void variax_startup5(struct timer_list *t) { - struct usb_line6_variax *variax = (struct usb_line6_variax *)data; + struct usb_line6_variax *variax = from_timer(variax, t, startup_timer2); CHECK_STARTUP_PROGRESS(variax->startup_progress, VARIAX_STARTUP_WORKQUEUE); @@ -190,7 +190,7 @@ static void line6_variax_process_message(struct usb_line6 *line6) } else if (memcmp(buf + 1, variax_init_done + 1, sizeof(variax_init_done) - 1) == 0) { /* notify of complete initialization: */ - variax_startup4((unsigned long)variax); + variax_startup4(&variax->startup_timer2); } break; } @@ -222,8 +222,8 @@ static int variax_init(struct usb_line6 *line6, line6->process_message = line6_variax_process_message; line6->disconnect = line6_variax_disconnect; - init_timer(&variax->startup_timer1); - init_timer(&variax->startup_timer2); + timer_setup(&variax->startup_timer1, NULL, 0); + timer_setup(&variax->startup_timer2, NULL, 0); INIT_WORK(&variax->startup_work, variax_startup6); /* initialize USB buffers: */ diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 7ab25de5ca0a..2c1aaa3292bf 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -352,9 +352,9 @@ static void snd_usbmidi_out_tasklet(unsigned long data) } /* called after transfers had been interrupted due to some USB error */ -static void snd_usbmidi_error_timer(unsigned long data) +static void snd_usbmidi_error_timer(struct timer_list *t) { - struct snd_usb_midi *umidi = (struct snd_usb_midi *)data; + struct snd_usb_midi *umidi = from_timer(umidi, t, error_timer); unsigned int i, j; spin_lock(&umidi->disc_lock); @@ -2369,8 +2369,7 @@ int __snd_usbmidi_create(struct snd_card *card, usb_id = USB_ID(le16_to_cpu(umidi->dev->descriptor.idVendor), le16_to_cpu(umidi->dev->descriptor.idProduct)); umidi->usb_id = usb_id; - setup_timer(&umidi->error_timer, snd_usbmidi_error_timer, - (unsigned long)umidi); + timer_setup(&umidi->error_timer, snd_usbmidi_error_timer, 0); /* detect the endpoint(s) to use */ memset(endpoints, 0, sizeof(endpoints)); -- 2.7.4 -- Kees Cook Pixel Security

2 1

[alsa-devel] [PATCH] ASoC: cht-bsw-rt5645: Also enable/disable the clock on Cherry Trail
by Hans de Goede 16 Oct '17

16 Oct '17

Just like on Bay Trail the codec mclk is also controlled by the common clk framework as pmc_plt_clk_3 on Cherry Trail and the cht-bsw-rt5645 needs to control it so that it has the right frequency and gets en/disabled as appropriate. This fixes the sound crackling (like a bonfire) when the clock was not set up properly before Linux boots and turns the clock off when not in use saving power. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=196351 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 5bcde01d15e6..d553e2b67c92 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -682,14 +682,12 @@ static int snd_cht_mc_probe(struct platform_device *pdev) cht_rt5645_cpu_dai_name; } - if (is_valleyview()) { - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); - if (IS_ERR(drv->mclk)) { - dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); - return PTR_ERR(drv->mclk); - } + drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (IS_ERR(drv->mclk)) { + dev_err(&pdev->dev, + "Failed to get MCLK from pmc_plt_clk_3: %ld\n", + PTR_ERR(drv->mclk)); + return PTR_ERR(drv->mclk); } snd_soc_card_set_drvdata(card, drv); -- 2.14.2

2 1

[alsa-devel] [PATCH] ALSA: 6fire: remove unused variable card
by Colin King 16 Oct '17

16 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> The pointer card is being assigned a value but it is never used. Remove this redundant variable. Cleans up clang warning: Value stored to 'card' is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/6fire/chip.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c index c7641cb50616..17d5e3ee6d73 100644 --- a/sound/usb/6fire/chip.c +++ b/sound/usb/6fire/chip.c @@ -174,11 +174,9 @@ static int usb6fire_chip_probe(struct usb_interface *intf, static void usb6fire_chip_disconnect(struct usb_interface *intf) { struct sfire_chip *chip; - struct snd_card *card; chip = usb_get_intfdata(intf); if (chip) { /* if !chip, fw upload has been performed */ - card = chip->card; chip->intf_count--; if (!chip->intf_count) { mutex_lock(&register_mutex); -- 2.14.1

2 1

[alsa-devel] [PATCH] asihpi: clean up a couple of build warnings
by Colin King 16 Oct '17

16 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> Variable dpcm is never used and hence can be removed. Variable runtime is being assigned but is never read, so the assignment is redundant and can be removed. Cleans up two clang warnings: Value stored to 'dpcm' is never read Value stored to 'runtime' is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/asihpi/asihpi.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 614322171c79..720361455c60 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -573,10 +573,8 @@ static void snd_card_asihpi_pcm_int_start(struct snd_pcm_substream *substream) static void snd_card_asihpi_pcm_int_stop(struct snd_pcm_substream *substream) { - struct snd_card_asihpi_pcm *dpcm; struct snd_card_asihpi *card; - dpcm = (struct snd_card_asihpi_pcm *)substream->runtime->private_data; card = snd_pcm_substream_chip(substream); hpi_handle_error(hpi_adapter_set_property(card->hpi->adapter->index, @@ -863,7 +861,6 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) snd_pcm_group_for_each_entry(s, substream) { struct snd_card_asihpi_pcm *ds = s->runtime->private_data; - runtime = s->runtime; /* don't link Cap and Play */ if (substream->stream != s->stream) -- 2.14.1

2 1

[alsa-devel] [PATCH] ALSA: emux: remove unused redundant variable p2
by Colin King 16 Oct '17

16 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> The variable p2 is being assigned but never used, it is redundant and can be safely removed. Cleans up clang warning: Value stored to 'p2' is never read. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/synth/emux/emux_oss.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index de19e108974a..764ff4bc2089 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -430,7 +430,6 @@ gusspec_control(struct snd_emux *emu, struct snd_emux_port *port, int cmd, { int voice; unsigned short p1; - short p2; int plong; struct snd_midi_channel *chan; @@ -445,7 +444,6 @@ gusspec_control(struct snd_emux *emu, struct snd_emux_port *port, int cmd, chan = &port->chset.channels[voice]; p1 = *(unsigned short *) &event[4]; - p2 = *(short *) &event[6]; plong = *(int*) &event[4]; switch (cmd) { -- 2.14.1

2 1

[alsa-devel] [PATCH] ALSA: ens137x: remove redundant variable result
by Colin King 16 Oct '17

16 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> Variable result is being assigned a value from a calculation however the variable is never read, so this redundant variable can be removed. Cleans up clang warning: Value stored to 'result' is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/ens1370.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index d4cd6451fdca..39f79a6b5283 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -732,7 +732,7 @@ static void snd_es1371_codec_wait(struct snd_ac97 *ac97) static void snd_es1371_adc_rate(struct ensoniq * ensoniq, unsigned int rate) { - unsigned int n, truncm, freq, result; + unsigned int n, truncm, freq; mutex_lock(&ensoniq->src_mutex); n = rate / 3000; @@ -740,7 +740,6 @@ static void snd_es1371_adc_rate(struct ensoniq * ensoniq, unsigned int rate) n--; truncm = (21 * n - 1) | 1; freq = ((48000UL << 15) / rate) * n; - result = (48000UL << 15) / (freq / n); if (rate >= 24000) { if (truncm > 239) truncm = 239; -- 2.14.1

2 1

[alsa-devel] [PATCH] ALSA: pcm: remove redundant variable runtime
by Colin King 16 Oct '17

16 Oct '17

From: Colin Ian King <colin.king(a)canonical.com> An earlier commit removed the access to variable runtime and we are now left with unused variable that is redundant, so remove it. Cleans up the clang warning: Value stored to 'runtime' is never read Fixes: e11f0f90a626 ("ALSA: pcm: remove SNDRV_PCM_IOCTL1_INFO internal command") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/core/pcm_native.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 2fec2feac387..a4d92e46c459 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -195,7 +195,6 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irqrestore); int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info) { - struct snd_pcm_runtime *runtime; struct snd_pcm *pcm = substream->pcm; struct snd_pcm_str *pstr = substream->pstr; @@ -211,7 +210,6 @@ int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info) info->subdevices_count = pstr->substream_count; info->subdevices_avail = pstr->substream_count - pstr->substream_opened; strlcpy(info->subname, substream->name, sizeof(info->subname)); - runtime = substream->runtime; return 0; } -- 2.14.1

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add native DSD support for Pro-Ject Pre Box S2 Digital
by Jussi Laako 16 Oct '17

16 Oct '17

Add native DSD support quirk for Pro-Ject Pre Box S2 Digital USB id 2772:0230. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 9aeb05f..5a040e3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1348,6 +1348,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */ case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */ case USB_ID(0x22d9, 0x0416): /* OPPO HA-1 */ + case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.7.4

2 1

[alsa-devel] 4.13 regression: get_kctl_0dB_offset doesn't handle all possible callbacks
by PaX Team 16 Oct '17

16 Oct '17

hello devs, upstream commit 99b5c5bb9a5435a5ae5d46445ac0f2bf6aa5ee52 removed the use of set_fs in get_kctl_0dB_offset under the assumption that the only runtime value of kctl->tlv.c was snd_hda_mixer_amp_tlv. alas, recently, the KERNEXEC and UDEREF features in PaX reported a violation of this assumption as it turns out that _snd_ctl_add_slave sets this callback to slave_tlv_cmd instead. it seems to me that there're several other possible values for this callback so it's not clear why it was assumed that get_kctl_0dB_offset would only be called with only one of them in practice, but nevertheless, it happens here so clearly something is wrong. the backtrace that shows how things go wrong in this case: PAX: suspicious general protection fault: 0000 [#1] PREEMPT SMP Modules linked in: CPU: 4 PID: 31 Comm: kworker/4:0 Not tainted 4.13.5-i386-pax #17 Workqueue: events azx_probe_work task: eb61c880 task.stack: eb62a000 EIP: 0060:[<009bbd2d>] init_slave_0dB+0x2d/0xa0 EFLAGS: 00210202 CPU: 4 EAX: 00991fd0 EBX: e9883a80 ECX: e9883a80 EDX: eb62bd74 ESI: eb62bd74 EDI: e9098800 EBP: eb62bd08 ESP: eb62bcec DS: 0068 ES: 0068 FS: 00d8 GS: 0068 SS: 0068 CR0: 80050033 CR2: 00000000 CR3: 03b2d100 CR4: 000406f0 shadow CR4: 000406f0 Call Trace: [<009bb469>] map_slaves+0xb9/0xe0 [<009bce0e>] __snd_hda_add_vmaster+0xde/0x110 [<009c82f0>] snd_hda_gen_build_controls+0x1a0/0x1b0 [<009d0a4d>] cx_auto_build_controls+0xd/0x70 [<009bdf76>] snd_hda_codec_build_controls+0x186/0x1d0 [<009b92ad>] hda_codec_driver_probe+0x6d/0xf0 [<006a3be9>] driver_probe_device+0x289/0x420 [<006a3ed6>] __device_attach_driver+0x76/0x100 [<006a1edf>] bus_for_each_drv+0x3f/0x70 [<006a3813>] __device_attach+0xa3/0x110 [<006a3f9d>] device_initial_probe+0xd/0x10 [<006a2d6f>] bus_probe_device+0x6f/0x80 [<006a100f>] device_add+0x2cf/0x590 [<009d8d8c>] snd_hdac_device_register+0xc/0x40 [<009b90b4>] snd_hda_codec_configure+0x34/0x140 [<009c2875>] azx_codec_configure+0x25/0x50 [<009d8081>] azx_probe_continue+0x621/0x9e0 [<009d84bd>] azx_probe_work+0xd/0x10 [<0006fff2>] process_one_work+0x122/0x2a0 [<000701a9>] worker_thread+0x39/0x390 [<00074df2>] kthread+0xe2/0x110 [<00cee619>] ret_from_fork+0x19/0x30 Code: e5 57 89 c7 56 89 d6 53 89 cb 83 ec 10 8b 41 6c a9 00 00 00 10 74 09 81 79 58 90 bc 9b 00 74 4e a8 10 74 38 8b 43 58 85 c0 74 31 <83> 38 01 75 2c 8b 48 0c 81 e1 ff ff fe ff 74 21 8b 16 39 d1 74 EIP: [<009bbd2d>] init_slave_0dB+0x2d/0xa0 SS:ESP: 0068:eb62bcec what KERNEXEC on i386 does is that it executes kernel code in its own 0-based code segment hence the 'low' code addresses. due to the current logic that checks for SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK in get_kctl_0dB_offset, this callback address is instead treated as a data pointer (as apparently SNDRV_CTL_ELEM_ACCESS_TLV_READ is also set) and since it's not a valid kernel address, it causes a GPF under the UDEREF PaX feature (without UDEREF it'd have been an oops since such low addresses are not mapped for kernel threads). on vanilla kernels all this is a silent read of kernel code bytes that are then interpreted as the tlv[] array content, which is probably not what you want either. as for fixing this, first the above mentioned assumption should be re-evaluated. if it's considered correct then there is some logic bug in my case (i can help debug it if you tell me what to do) otherwise the current pattern of if (SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK && callback == snd_hda_mixer_amp_tlv) { call wrapped function underneath snd_hda_mixer_amp_tlv } else if (SNDRV_CTL_ELEM_ACCESS_TLV_READ) { treat unrecognized callback address as data ptr } should be changed to if (SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK( { if (callback == snd_hda_mixer_amp_tlv) { call wrapped function underneath snd_hda_mixer_amp_tlv } else if (callback == others) { handle others, WARN/BUG/etc } } else if (SNDRV_CTL_ELEM_ACCESS_TLV_READ) { no longer treat unrecognized callback address as data ptr } and all other callbacks with userland access should be refactored the same way as snd_hda_mixer_amp_tlv was. i'd also suggest that you at least do the above suggested if/else pattern change in order to prevent the misuse of unexpected callbacks in the future. cheers, PaX Team

3 2

[alsa-devel] [PATCH 0/2] sound: oss: make the function argument and structure const
by Bhumika Goyal 16 Oct '17

16 Oct '17

Make the function argument of function sound_install_audiodrv const. After this change, make the structures audio_driver const as they are only passed to the const argument (made const in Patch[1/2]) of the function sound_install_audiodrv . Bhumika Goyal (2): sound: oss: make the function argument as const sound: oss: make audio_driver structure const sound/oss/dev_table.c | 2 +- sound/oss/dev_table.h | 2 +- sound/oss/pas2_pcm.c | 2 +- sound/oss/waveartist.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) -- 1.9.1

2 3

[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: Decrease loglevel for topology loading
by Chintan Patel 16 Oct '17

16 Oct '17

In case of failure in loading customize topology firmware, dfw_sst.bin gets loaded. However, current log provides this message as error even after successfully falling back to default topology "dfw_sst.bin". Hence to convey proper message, changing log level and message. Signed-off-by: Chintan Patel <chintan.m.patel(a)intel.com> --- Changes in v2: Addressing review comments from Takashi Iwai by changing loglevel as info sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 22f768ca3c73..12f383fd7c39 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3404,7 +3404,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_err(bus->dev, "tplg fw %s load failed with %d\n", + dev_info(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", skl->tplg_name, ret); ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { -- 1.9.1

2 1

[alsa-devel] Surface 3 audio broken between 4.11 and 4.13 (rt5645)
by Bastien Nocera 15 Oct '17

15 Oct '17

Hey, As I was trying to test PulseAudio HDMI LPE audio support fixes, I realised that the internal sound card of my Surface 3 wasn't working anymore, using the same user-space configuration. Looks like detection of the device regressed between 2 versions. Does this look familiar to anyone, or should I start bisecting? (which will take me absolutely ages, as the CPU is anemic on that machine) Working: $ dmesg | grep rt5 [ 13.707825] rt5640 i2c-10EC5640:00: Device with ID register 0x6308 is not rt5640/39 [ 15.010548] cht-bsw-rt5645 cht-bsw-rt5645: snd-soc-dummy-dai <-> media-cpu-dai mapping ok [ 15.011132] cht-bsw-rt5645 cht-bsw-rt5645: snd-soc-dummy-dai <-> deepbuffer-cpu-dai mapping ok [ 15.012756] cht-bsw-rt5645 cht-bsw-rt5645: rt5645-aif1 <-> ssp2-port mapping ok [ 15.100152] input: chtrt5645 Headset as /devices/pci0000:00/808622A8:00/cht-bsw-rt5645/sound/card0/input27 $ uname -r 4.11.0-0.rc0.git4.21.surface3.fc26.x86_64 Not working: $ journalctl -b -2 | grep rt5 Oct 13 20:10:19 surface kernel: rt5640 i2c-10EC5640:00: Device with ID register 0x6308 is not rt5640/39 Oct 13 20:10:21 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC1_MAP enabled Oct 13 20:10:21 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC enabled Oct 13 20:10:21 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk MCLK_EN enabled Oct 13 20:10:21 surface kernel: bytcr_rt5640 bytcr_rt5640: ASoC: CODEC DAI rt5640-aif1 not registered Oct 13 20:10:21 surface kernel: bytcr_rt5640 bytcr_rt5640: devm_snd_soc_register_card failed -517 Oct 13 20:10:22 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC1_MAP enabled Oct 13 20:10:22 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC enabled Oct 13 20:10:22 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk MCLK_EN enabled Oct 13 20:10:22 surface kernel: bytcr_rt5640 bytcr_rt5640: ASoC: CODEC DAI rt5640-aif1 not registered Oct 13 20:10:22 surface kernel: bytcr_rt5640 bytcr_rt5640: devm_snd_soc_register_card failed -517 Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC1_MAP enabled Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC enabled Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk MCLK_EN enabled Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: ASoC: CODEC DAI rt5640-aif1 not registered Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: devm_snd_soc_register_card failed -517 Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC1_MAP enabled Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk DMIC enabled Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: quirk MCLK_EN enabled Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: ASoC: CODEC DAI rt5640-aif1 not registered Oct 13 20:10:23 surface kernel: bytcr_rt5640 bytcr_rt5640: devm_snd_soc_register_card failed -517 $ journalctl -b -2 | head -n3 | tail -n1 Oct 13 22:10:10 surface kernel: Linux version 4.13.0-0.rc2.git0.26.surface3.fc27.x86_64 (hadess@classic) (gcc version 7.1.1 20170526 (Red Hat 7.1.1-2) (GCC)) #1 SMP Tue Jul 25 18:47:03 CEST 2017 Cheers

2 2

[alsa-devel] [PATCH 0/3] ASoC: Intel: Skylake: Add support for loading FE dais from topology
by Guneshwor Singh 13 Oct '17

13 Oct '17

This series adds support for loading FE dais using topology. A flag is introduced to decide whether to register FE dais defined in the platform driver or use from topology. By default, it is set to register the FE dais defined in the platform driver. While at it, fix a missing sentinel too. Guneshwor Singh (3): ASoC: Intel: Skylake: Add flag to check to register FE dais from topology ASoC: Intel: Skylake: Add dai load ops for dais from topology ASoC: Intel: Skylake: Fix missing sentinel in sst_acpi_mach sound/soc/intel/skylake/skl-pcm.c | 46 +++++++++++++++++++++++++++++++--- sound/soc/intel/skylake/skl-topology.c | 1 + sound/soc/intel/skylake/skl-topology.h | 3 +++ sound/soc/intel/skylake/skl.c | 10 +++++++- sound/soc/intel/skylake/skl.h | 4 ++- 5 files changed, 58 insertions(+), 6 deletions(-) -- 2.14.2

3 7

[alsa-devel] [PATCH] ASoC: rsnd: more clear ADG clock debug info
by Kuninori Morimoto 13 Oct '17

13 Oct '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ADG inputs clock from CLK{A,B,C,I} and outputs clock from CLKOUT{0,1,2,3} which is selected by BRG{A,B}. Now, ADG is assuming BRGA is for 44100Hz related clocks, BRGB is for 48000Hz related clocks. Clock related debug is very difficult/confusable. This patch cleanups clock related debug info. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 59 ++++++++++++++++++++++++++++++++++--------------- 1 file changed, 41 insertions(+), 18 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 19213077..8ddb087 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -57,6 +57,13 @@ struct rsnd_adg { i++) #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg) +static const char * const clk_name[] = { + [CLKA] = "clk_a", + [CLKB] = "clk_b", + [CLKC] = "clk_c", + [CLKI] = "clk_i", +}; + static u32 rsnd_adg_calculate_rbgx(unsigned long div) { int i, ratio; @@ -279,6 +286,7 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) struct rsnd_priv *priv = rsnd_mod_to_priv(ssi_mod); struct rsnd_adg *adg = rsnd_priv_to_adg(priv); struct rsnd_mod *adg_mod = rsnd_mod_get(adg); + struct device *dev = rsnd_priv_to_dev(priv); int id = rsnd_mod_id(ssi_mod); int shift = (id % 4) * 8; u32 mask = 0xFF << shift; @@ -305,12 +313,13 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) rsnd_mod_bset(adg_mod, AUDIO_CLK_SEL2, mask, val); break; } + + dev_dbg(dev, "AUDIO_CLK_SEL is 0x%x\n", val); } int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate) { struct rsnd_adg *adg = rsnd_priv_to_adg(priv); - struct device *dev = rsnd_priv_to_dev(priv); struct clk *clk; int i; int sel_table[] = { @@ -320,8 +329,6 @@ int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate) [CLKI] = 0x0, }; - dev_dbg(dev, "request clock = %d\n", rate); - /* * find suitable clock from * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI. @@ -377,9 +384,10 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *ssi_mod, unsigned int rate) rsnd_mod_write(adg_mod, BRRA, adg->rbga); rsnd_mod_write(adg_mod, BRRB, adg->rbgb); - dev_dbg(dev, "ADG: %s[%d] selects 0x%x for %d\n", - rsnd_mod_name(ssi_mod), rsnd_mod_id(ssi_mod), - data, rate); + dev_dbg(dev, "CLKOUT is based on BRG%c (= %dHz)\n", + (ckr) ? 'B' : 'A', + (ckr) ? adg->rbgb_rate_for_48khz : + adg->rbga_rate_for_441khz); return 0; } @@ -408,21 +416,12 @@ static void rsnd_adg_get_clkin(struct rsnd_priv *priv, { struct device *dev = rsnd_priv_to_dev(priv); struct clk *clk; - static const char * const clk_name[] = { - [CLKA] = "clk_a", - [CLKB] = "clk_b", - [CLKC] = "clk_c", - [CLKI] = "clk_i", - }; int i; for (i = 0; i < CLKMAX; i++) { clk = devm_clk_get(dev, clk_name[i]); adg->clk[i] = IS_ERR(clk) ? NULL : clk; } - - for_each_rsnd_clk(clk, adg, i) - dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk)); } static void rsnd_adg_get_clkout(struct rsnd_priv *priv, @@ -571,12 +570,35 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, adg->ckr = ckr; adg->rbga = rbga; adg->rbgb = rbgb; +} + +#ifdef DEBUG +static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct clk *clk; + int i; + + for_each_rsnd_clk(clk, adg, i) + dev_dbg(dev, "%s : %p : %ld\n", + clk_name[i], clk, clk_get_rate(clk)); - for_each_rsnd_clkout(clk, adg, i) - dev_dbg(dev, "clkout %d : %p : %ld\n", i, clk, clk_get_rate(clk)); dev_dbg(dev, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n", - ckr, rbga, rbgb); + adg->ckr, adg->rbga, adg->rbgb); + dev_dbg(dev, "BRGA (for 44100 base) = %d\n", adg->rbga_rate_for_441khz); + dev_dbg(dev, "BRGB (for 48000 base) = %d\n", adg->rbgb_rate_for_48khz); + + /* + * Actual CLKOUT will be exchanged in rsnd_adg_ssi_clk_try_start() + * by BRGCKR::BRGCKR_31 + */ + for_each_rsnd_clkout(clk, adg, i) + dev_dbg(dev, "clkout %d : %p : %ld\n", i, + clk, clk_get_rate(clk)); } +#else +#define rsnd_adg_clk_dbg_info(priv, adg) +#endif int rsnd_adg_probe(struct rsnd_priv *priv) { @@ -595,6 +617,7 @@ int rsnd_adg_probe(struct rsnd_priv *priv) rsnd_adg_get_clkin(priv, adg); rsnd_adg_get_clkout(priv, adg); + rsnd_adg_clk_dbg_info(priv, adg); priv->adg = adg; -- 1.9.1

2 1

[alsa-devel] No sound on samsung galaxy book 12 (HDA Intel ALC298)
by Nathanael inkson 13 Oct '17

13 Oct '17

Hi, I have just installed ubuntu 16.04.3 on a samsung galaxy book 12. I have no sound from either speakers or headphones. the intel driver seems to be loaded . I have tried all the tricks of unmuting sound in the alsa mixers to no avail. aplay -l gives: card 0: PCH [HDA Intel PCH], device 0: ALC298 Analog [ALC298 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 Can anyone help? The output of the alsa script is upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.64 !!################################ !!Script ran on: Fri Oct 13 13:41:05 UTC 2017 !!Linux Distribution !!------------------ Ubuntu 16.04.3 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 16.04.3 LTS" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 16.04.3 LTS" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL="http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" UBUNTU_CODENAME=xenial !!DMI Information !!--------------- Manufacturer: SAMSUNG ELECTRONICS CO., LTD. Product Name: Galaxy Book 12 Product Version: P02HAC Firmware Version: P02HAC.004.170626.WY.1442 Board Vendor: SAMSUNG ELECTRONICS CO., LTD. Board Name: SM-W720NZKBBTU !!ACPI Device Status Information !!--------------- /sys/bus/acpi/devices/ACPI0003:00/status 31 /sys/bus/acpi/devices/ACPI000C:00/status 15 /sys/bus/acpi/devices/INT33A1:00/status 15 /sys/bus/acpi/devices/INT33D2:00/status 11 /sys/bus/acpi/devices/INT33D3:00/status 15 /sys/bus/acpi/devices/INT33D4:00/status 15 /sys/bus/acpi/devices/INT33D6:00/status 15 /sys/bus/acpi/devices/INT3400:00/status 15 /sys/bus/acpi/devices/INT3403:01/status 15 /sys/bus/acpi/devices/INT3403:02/status 15 /sys/bus/acpi/devices/INT3406:00/status 15 /sys/bus/acpi/devices/INT3446:00/status 15 /sys/bus/acpi/devices/INT3447:00/status 15 /sys/bus/acpi/devices/INT344B:00/status 15 /sys/bus/acpi/devices/INT3472:01/status 15 /sys/bus/acpi/devices/INT3472:02/status 15 /sys/bus/acpi/devices/INT347F:00/status 15 /sys/bus/acpi/devices/INT3F0D:00/status 15 /sys/bus/acpi/devices/MSF0001:00/status 15 /sys/bus/acpi/devices/MSFT0101:00/status 15 /sys/bus/acpi/devices/PNP0103:00/status 15 /sys/bus/acpi/devices/PNP0C02:03/status 3 /sys/bus/acpi/devices/PNP0C02:05/status 3 /sys/bus/acpi/devices/PNP0C0A:00/status 31 /sys/bus/acpi/devices/PNP0C0F:00/status 9 /sys/bus/acpi/devices/PNP0C0F:01/status 9 /sys/bus/acpi/devices/PNP0C0F:02/status 9 /sys/bus/acpi/devices/PNP0C0F:03/status 9 /sys/bus/acpi/devices/PNP0C0F:04/status 9 /sys/bus/acpi/devices/PNP0C0F:05/status 9 /sys/bus/acpi/devices/PNP0C0F:06/status 9 /sys/bus/acpi/devices/PNP0C0F:07/status 9 /sys/bus/acpi/devices/SAM0101:00/status 15 /sys/bus/acpi/devices/SAM0201:00/status 15 /sys/bus/acpi/devices/SAM0203:01/status 15 /sys/bus/acpi/devices/SAM0204:00/status 15 /sys/bus/acpi/devices/SAM0603:00/status 15 /sys/bus/acpi/devices/SAM0701:00/status 15 /sys/bus/acpi/devices/SONY258A:00/status 15 /sys/bus/acpi/devices/SSE4752:00/status 15 /sys/bus/acpi/devices/SYNP1234:00/status 15 /sys/bus/acpi/devices/WCOM0027:00/status 15 /sys/bus/acpi/devices/device:10/status 15 /sys/bus/acpi/devices/device:11/status 15 /sys/bus/acpi/devices/device:4f/status 15 /sys/bus/acpi/devices/device:50/status 15 /sys/bus/acpi/devices/device:51/status 15 /sys/bus/acpi/devices/device:52/status 15 /sys/bus/acpi/devices/device:53/status 15 /sys/bus/acpi/devices/device:54/status 15 /sys/bus/acpi/devices/device:55/status 15 /sys/bus/acpi/devices/device:56/status 15 /sys/bus/acpi/devices/device:5b/status 15 /sys/bus/acpi/devices/device:5c/status 15 /sys/bus/acpi/devices/device:5d/status 15 /sys/bus/acpi/devices/device:5e/status 15 /sys/bus/acpi/devices/device:5f/status 15 /sys/bus/acpi/devices/device:60/status 15 /sys/bus/acpi/devices/device:61/status 15 !!Kernel Information !!------------------ Kernel release: 4.10.0-37-generic Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k4.10.0-37-generic Library version: 1.1.0 Utilities version: 1.1.0 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xdf638000 irq 280 !!PCI Soundcards installed in the system !!-------------------------------------- 00:05.0 Multimedia controller: Intel Corporation Sky Lake Imaging Unit (rev 01) 00:14.3 Multimedia controller: Intel Corporation Device 9d32 (rev 01) 00:1f.3 Multimedia audio controller: Intel Corporation Device 9d71 (rev 21) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1f.3 0401: 8086:9d71 (rev 21) Subsystem: 144d:c14f !!Modprobe options (Sound related) !!-------------------------------- snd_pcsp: index=-2 snd_usb_audio: index=-2 snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_usb_audio: index=-2 snd_usb_caiaq: index=-2 snd_usb_ua101: index=-2 snd_usb_us122l: index=-2 snd_usb_usx2y: index=-2 snd_cmipci: mpu_port=0x330 fm_port=0x388 snd_pcsp: index=-2 snd_usb_audio: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : -1 !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC298 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0298 Subsystem Id: 0x144dc14f Revision Id: 0x100103 No Modem Function Group found Default PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 D3cold CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=8, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Master Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC298 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x01, mute=0 Amp-Out vals: [0x73 0x73] Converter: stream=1, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x01, mute=0 Amp-Out vals: [0x7f 0x7f] Converter: stream=0, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x40]: 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x43, nsteps=0x7f, stepsize=0x01, mute=1 Amp-In vals: [0xc3 0xc3] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC298 Analog", type="Audio", device=0 Amp-In caps: ofs=0x43, nsteps=0x7f, stepsize=0x01, mute=1 Amp-In vals: [0x6e 0x6e] Converter: stream=1, channel=0 SDI-Select: 0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100511: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x40]: 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x25 Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x9d 0x9d] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 4 0x18 0x19 0x1a 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] Connection: 1 0x03 Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x43, nsteps=0x7f, stepsize=0x01, mute=1 Amp-In vals: [0xc3 0xc3] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x24 Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x90a601c0: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0xc, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x13 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x14 [Pin Complex] wcaps 0x40050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00010010: OUT EAPD EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 2 0x0c* 0x0d Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Master Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 3 0x0c* 0x0d 0x06 Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x04a11020: [Jack] Mic at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001373c: IN OUT HP EAPD Detect Vref caps: HIZ 50 GRD 80 100 EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 2 0x0c* 0x0d Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400400: Mono Pincap 0x00000020: IN Pin Default 0x40420105: [N/A] SPDIF Out at Ext N/A Conn = 1/4, Color = Unknown DefAssociation = 0x0, Sequence = 0x5 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x1e [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=150 Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000001c: OUT HP Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 2 0x0c* 0x0d Node 0x22 [Audio Selector] wcaps 0x300101: Stereo Connection: 6 0x18 0x19 0x1a 0x1d 0x0b 0x12* Node 0x23 [Audio Selector] wcaps 0x300101: Stereo Connection: 6 0x18 0x19 0x1a* 0x1d 0x0b 0x13 Node 0x24 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x12* 0x13 Node 0x25 [Audio Selector] wcaps 0x300101: Stereo Connection: 1 0x1f Codec: Intel Kabylake HDMI Address: 2 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x8086280b Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1a]: 16 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1a]: 16 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1a]: 16 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 2 Oct 13 14:39 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 8 Oct 13 14:39 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 9 Oct 13 14:39 /dev/snd/hwC0D2 crw-rw----+ 1 root audio 116, 4 Oct 13 14:39 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 3 Oct 13 14:39 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 5 Oct 13 14:39 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 6 Oct 13 14:39 /dev/snd/pcmC0D7p crw-rw----+ 1 root audio 116, 7 Oct 13 14:39 /dev/snd/pcmC0D8p crw-rw----+ 1 root audio 116, 1 Oct 13 14:39 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Oct 13 14:39 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Oct 13 14:39 . drwxr-xr-x 3 root root 260 Oct 13 14:39 .. lrwxrwxrwx 1 root root 12 Oct 13 14:39 pci-0000:00:1f.3 -> ../controlC0 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC298 Analog [ALC298 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC298 Analog [ALC298 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [PCH] Card hw:0 'PCH'/'HDA Intel PCH at 0xdf638000 irq 280' Mixer name : 'Realtek ALC298' Components : 'HDA:10ec0298,144dc14f,00100103 HDA:8086280b,80860101,00100000' Controls : 36 Simple ctrls : 10 Simple mixer control 'Master',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 115 [91%] [-6.00dB] [on] Front Right: Playback 115 [91%] [-6.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 29 [94%] [9.00dB] [off] Front Right: Playback 29 [94%] [9.00dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 110 [87%] [21.50dB] [on] Front Right: Capture 110 [87%] [21.50dB] [on] Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Loopback Mixing',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' !!Alsactl output !!-------------- --startcollapse-- state.PCH { control.1 { iface MIXER name 'Master Playback Volume' value.0 115 value.1 115 comment { access 'read write' type INTEGER count 2 range '0 - 127' dbmin -6350 dbmax 0 dbvalue.0 -600 dbvalue.1 -600 } } control.2 { iface MIXER name 'Master Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Loopback Mixing' value Disabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.4 { iface MIXER name 'Mic Playback Volume' value.0 29 value.1 29 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 900 dbvalue.1 900 } } control.5 { iface MIXER name 'Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.6 { iface MIXER name 'Capture Volume' value.0 110 value.1 110 comment { access 'read write' type INTEGER count 2 range '0 - 127' dbmin -3350 dbmax 3000 dbvalue.0 2150 dbvalue.1 2150 } } control.7 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.8 { iface MIXER name 'Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.9 { iface MIXER name 'Internal Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.10 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.11 { iface CARD name 'Internal Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.12 { iface CARD name 'Speaker Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.13 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.14 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.15 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.16 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.17 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.18 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.19 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.20 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.21 { iface CARD name 'HDMI/DP,pcm=7 Jack' value false comment { access read type BOOLEAN count 1 } } control.22 { iface MIXER name 'IEC958 Playback Con Mask' index 1 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.23 { iface MIXER name 'IEC958 Playback Pro Mask' index 1 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.24 { iface MIXER name 'IEC958 Playback Default' index 1 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.25 { iface MIXER name 'IEC958 Playback Switch' index 1 value true comment { access 'read write' type BOOLEAN count 1 } } control.26 { iface PCM device 7 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.27 { iface CARD name 'HDMI/DP,pcm=8 Jack' value false comment { access read type BOOLEAN count 1 } } control.28 { iface MIXER name 'IEC958 Playback Con Mask' index 2 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.29 { iface MIXER name 'IEC958 Playback Pro Mask' index 2 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.30 { iface MIXER name 'IEC958 Playback Default' index 2 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.31 { iface MIXER name 'IEC958 Playback Switch' index 2 value true comment { access 'read write' type BOOLEAN count 1 } } control.32 { iface PCM device 8 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.33 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.34 { iface PCM device 7 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.35 { iface PCM device 8 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.36 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } } --endcollapse-- !!All Loaded Modules !!------------------ Module ccm rfcomm arc4 bnep snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic binfmt_misc nls_iso8859_1 joydev 8250_dw i2c_designware_platform i2c_designware_core snd_soc_skl snd_soc_skl_ipc snd_soc_sst_ipc snd_soc_sst_dsp snd_hda_ext_core snd_soc_sst_match snd_soc_core snd_compress ac97_bus snd_pcm_dmaengine snd_hda_intel snd_hda_codec snd_hda_core snd_hwdep snd_pcm snd_seq_midi snd_seq_midi_event snd_rawmidi intel_rapl x86_pkg_temp_thermal intel_powerclamp coretemp kvm_intel kvm irqbypass crct10dif_pclmul crc32_pclmul ghash_clmulni_intel pcbc snd_seq snd_seq_device snd_timer ath10k_pci hid_multitouch ath10k_core snd ath aesni_intel mac80211 aes_x86_64 crypto_simd glue_helper cryptd cfg80211 soundcore input_leds serio_raw btusb btrtl btbcm btintel shpchp bluetooth idma64 mei_me virt_dma mei processor_thermal_device intel_pch_thermal intel_soc_dts_iosf intel_lpss_pci intel_vbtn int3403_thermal soc_button_array int340x_thermal_zone mac_hid intel_lpss_acpi intel_lpss int3406_thermal tpm_crb dptf_power int3400_thermal acpi_thermal_rel intel_hid sparse_keymap acpi_pad parport_pc ppdev lp parport autofs4 mmc_block hid_generic usbhid i915 i2c_algo_bit drm_kms_helper syscopyarea sysfillrect sysimgblt fb_sys_fops sdhci_pci drm sdhci ahci libahci video i2c_hid hid wmi pinctrl_sunrisepoint pinctrl_intel fjes !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x12 0x90a601c0 0x13 0x411111f0 0x14 0x411111f0 0x17 0x90170110 0x18 0x04a11020 0x19 0x411111f0 0x1a 0x411111f0 0x1d 0x40420105 0x1e 0x411111f0 0x1f 0x411111f0 0x21 0x411111f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: /sys/class/sound/hwC0D2/init_pin_configs: 0x05 0x18560010 0x06 0x18560010 0x07 0x18560010 /sys/class/sound/hwC0D2/driver_pin_configs: /sys/class/sound/hwC0D2/user_pin_configs: /sys/class/sound/hwC0D2/init_verbs: /sys/class/sound/hwC0D2/hints: !!ALSA/HDA dmesg !!-------------- [ 2.431375] intel_rapl: Found RAPL domain dram [ 2.462483] snd_hda_intel 0000:00:1f.3: enabling device (0000 -> 0002) [ 2.570682] input: SYNP1234:00 06CB:5810 as /devices/pci0000:00/INT3446:00/i2c_designware.0/i2c-6/i2c-SYNP1234:00/0018:06CB:5810.0004/input/input12 -- [ 2.855095] hid-generic 0018:2D1F:000B.0005: input,hidraw4: I2C HID v1.00 Mouse [WCOM0027:00 2D1F:000B] on i2c-WCOM0027:00 [ 3.588848] snd_hda_intel 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) [ 3.613596] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC298: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 3.613598] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 3.613599] snd_hda_codec_realtek hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 3.613600] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 3.613600] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 3.613601] snd_hda_codec_realtek hdaudioC0D0: Mic=0x18 [ 3.613602] snd_hda_codec_realtek hdaudioC0D0: Internal Mic=0x12 [ 3.626328] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input18 [ 3.626383] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input19 [ 3.626430] input: HDA Intel PCH HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input20 [ 3.626479] input: HDA Intel PCH HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input21 [ 3.749827] audit: type=1400 audit(1507901955.077:2): apparmor="STATUS" operation="profile_load" profile="unconfined" name="/sbin/dhclient" pid=737 comm="apparmor_parser"

1 0

[alsa-devel] [PATCH] ALSA: hda - silence uninitialized variable warning in activate_amp_in()
by Dan Carpenter 13 Oct '17

13 Oct '17

If snd_hda_get_conn_list() fails then "conn" isn't initialized. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 28e265a88383..5cc65093d941 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -795,6 +795,8 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, hda_nid_t nid = path->path[i]; nums = snd_hda_get_conn_list(codec, nid, &conn); + if (nums < 0) + return; type = get_wcaps_type(get_wcaps(codec, nid)); if (type == AC_WID_PIN || (type == AC_WID_AUD_IN && codec->single_adc_amp)) {

2 1

[alsa-devel] [PATCH v2] ALSA: hda/ca0132 - use ARRAY_SIZE
by Jérémy Lefaure 13 Oct '17

13 Oct '17

Using the ARRAY_SIZE macro improves the readability of the code. Found with Coccinelle with the following semantic patch: @r depends on (org || report)@ type T; T[] E; position p; @@ ( (sizeof(E)@p /sizeof(*E)) | (sizeof(E)@p /sizeof(E[...])) | (sizeof(E)@p /sizeof(T)) ) Signed-off-by: Jérémy Lefaure <jeremy.lefaure(a)lse.epita.fr> --- Changes since v1: - keep sound/oss untouched sound/pci/hda/patch_ca0132.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 3e73d5c6ccfc..768ea8651993 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -27,6 +27,7 @@ #include <linux/mutex.h> #include <linux/module.h> #include <linux/firmware.h> +#include <linux/kernel.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" @@ -3605,8 +3606,7 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, static int ca0132_voicefx_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - unsigned int items = sizeof(ca0132_voicefx_presets) - / sizeof(struct ct_voicefx_preset); + unsigned int items = ARRAY_SIZE(ca0132_voicefx_presets); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; @@ -3635,10 +3635,8 @@ static int ca0132_voicefx_put(struct snd_kcontrol *kcontrol, struct ca0132_spec *spec = codec->spec; int i, err = 0; int sel = ucontrol->value.enumerated.item[0]; - unsigned int items = sizeof(ca0132_voicefx_presets) - / sizeof(struct ct_voicefx_preset); - if (sel >= items) + if (sel >= ARRAY_SIZE(ca0132_voicefx_presets)) return 0; codec_dbg(codec, "ca0132_voicefx_put: sel=%d, preset=%s\n", -- 2.14.2

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: add support for CX2072X CODEC
by simon.ho.cnxt＠gmail.com 13 Oct '17

13 Oct '17

From: Simon Ho <simon.ho(a)conexant.com> Hi Takashi, Mark I corrected the driver based on your input as below. Please help review the code. Mark: - Corrected the coding style. - EQ/DRC settings can be configured from userspace via mixer controls. - Wrongly cast void pointers. - Print I2C error code. - Use memcmp() to check if the new EQ value is different. - Removed unused TDM code. - Removed noisy "out of memory" message. - Corrected ACPI and I2C match table. - Removed unnessary CONFIG_ACPI define. Takashi: - Fixed printf int argument. - Fixed unexpected error code at no MCLK fallback. Thanks, Simon Simon Ho (2): ASoC: cx2072x: Add DT bingings documentation for CX2072X CODEC ASoC: add support for Conexant CX2072X CODEC .../devicetree/bindings/sound/cx2072x.txt | 36 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2072x.c | 2253 ++++++++++++++++++++ sound/soc/codecs/cx2072x.h | 319 +++ 5 files changed, 2615 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cx2072x.txt create mode 100644 sound/soc/codecs/cx2072x.c create mode 100644 sound/soc/codecs/cx2072x.h -- 2.7.4

4 10

[alsa-devel] [PATCH] ASoC: max98925: Return proper error if revision mismatch
by Axel Lin 13 Oct '17

13 Oct '17

Return proper error instead of 0 if the revision does not match. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98925.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 327eaa25c9bd..921f95fc396d 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -579,7 +579,7 @@ static int max98925_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(max98925->regmap); dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); - goto err_out; + return ret; } if (!of_property_read_u32(i2c->dev.of_node, "vmon-slot-no", &value)) { @@ -596,16 +596,20 @@ static int max98925_i2c_probe(struct i2c_client *i2c, } max98925->i_slot = value; } - ret = regmap_read(max98925->regmap, - MAX98925_REV_VERSION, &reg); - if ((ret < 0) || - ((reg != MAX98925_VERSION) && - (reg != MAX98925_VERSION1))) { - dev_err(&i2c->dev, - "device initialization error (%d 0x%02X)\n", + + ret = regmap_read(max98925->regmap, MAX98925_REV_VERSION, &reg); + if (ret < 0) { + dev_err(&i2c->dev, "Read revision failed\n"); + return ret; + } + + if ((reg != MAX98925_VERSION) && (reg != MAX98925_VERSION1)) { + ret = -ENODEV; + dev_err(&i2c->dev, "Invalid revision (%d 0x%02X)\n", ret, reg); - goto err_out; + return ret; } + dev_info(&i2c->dev, "device version 0x%02X\n", reg); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98925, @@ -613,7 +617,6 @@ static int max98925_i2c_probe(struct i2c_client *i2c, if (ret < 0) dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); -err_out: return ret; } -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: cht_bsw_max98090_ti: Fix NULL pointer dereference while accessing jack
by Guenter Roeck 12 Oct '17

12 Oct '17

From: Guenter Roeck <groeck(a)chromium.org> Commit f2ed6b07645e ("ASoC: Make aux_dev more like a generic component") caused a regression on this driver, since now a kernel oops is seen when the driver is loaded, or more specifically when ts3a227e_enable_jack_detect() is called. That commit changed the probing of aux_devs before checking new DAI links, so cht_max98090_headset_init is now called before cht_codec_init. The kernel crashes due a NULL pointer dereference in cht_max98090_headset_init since there is a call that tries to access the jack pointer which has not been allocated yet. This patch moves the new jack object creation from cht_codec_init to cht_max98090_headset_init, making sure the jack is created before is accessed. Also see upstream commit 5f22449344d9 ("ASoC: rockchip-max98090: Fix NULL pointer dereference while accessing to jack.") for a similar fix in the rockchip-max98090 driver. Fixes: f2ed6b07645e ("ASoC: Make aux_dev more like a generic component") Cc: Mengdong Lin <mengdong.lin(a)linux.intel.com> Signed-off-by: Guenter Roeck <groeck(a)chromium.org> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 55 ++++++++++++++-------------- 1 file changed, 27 insertions(+), 28 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 20755ecc7f9e..c1518db7bb61 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -111,37 +111,12 @@ static struct notifier_block cht_jack_nb = { static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { - int ret; - int jack_type; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); - struct snd_soc_jack *jack = &ctx->jack; - - /** - * TI supports 4 butons headset detection - * KEY_MEDIA - * KEY_VOICECOMMAND - * KEY_VOLUMEUP - * KEY_VOLUMEDOWN - */ - if (ctx->ts3a227e_present) - jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | - SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3; - else - jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE; - - ret = snd_soc_card_jack_new(runtime->card, "Headset Jack", - jack_type, jack, NULL, 0); - - if (ret) { - dev_err(runtime->dev, "Headset Jack creation failed %d\n", ret); - return ret; - } if (ctx->ts3a227e_present) - snd_soc_jack_notifier_register(jack, &cht_jack_nb); + snd_soc_jack_notifier_register(&ctx->jack, &cht_jack_nb); - return ret; + return 0; } static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, @@ -188,8 +163,32 @@ static int cht_max98090_headset_init(struct snd_soc_component *component) { struct snd_soc_card *card = component->card; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_jack *jack = &ctx->jack; + int jack_type; + int ret; + + /* + * TI supports 4 butons headset detection + * KEY_MEDIA + * KEY_VOICECOMMAND + * KEY_VOLUMEUP + * KEY_VOLUMEDOWN + */ + if (ctx->ts3a227e_present) + jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3; + else + jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE; + + ret = snd_soc_card_jack_new(card, "Headset Jack", jack_type, jack, + NULL, 0); + if (ret) { + dev_err(card->dev, "Headset Jack creation failed %d\n", ret); + return ret; + } - return ts3a227e_enable_jack_detect(component, &ctx->jack); + return ts3a227e_enable_jack_detect(component, jack); } static const struct snd_soc_ops cht_aif1_ops = { -- 2.7.4

2 1

[alsa-devel] regression introduced by commit 70d435ba1cd6 ("ASoC: imx-pcm-dma: simplify pcm_config")
by Lothar Waßmann 12 Oct '17

12 Oct '17

Hi, the commit 70d435ba1cd6 ("ASoC: imx-pcm-dma: simplify pcm_config") introduced a regression on the Ka-Ro TX series modules (and possibly other hardware as well). The problem is, that this commit restricts the supported sound formats to S24_LE only rather than S16_LE, S24_LE, S20_3LE without the patch. Therefore aplay fails with the following error messages, when trying to play a sound file with S16_LE format: |ALSA lib alsa-lib-1.1.2/src/pcm/pcm_params.c:2162:(snd1_pcm_hw_refine_slave) Slave PCM not usable |aplay: set_params:1204: Broken configuration for this PCM: no configurations available This is with the simple-sound-card driver as well as the imx-sgtl5000 driver. Can someone tell me on what hardware and with what software has this patch been tested? Lothar Waßmann

3 3

[alsa-devel] [PATCH] ASoC: rsnd: add rsnd_kctrl_xxx() macro
by Kuninori Morimoto 12 Oct '17

12 Oct '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current CTU/MIX/DVC are directly using rsnd_kctrl_cfg_m/s to control val etc, but it is difficult to read/understand. And there was no uniformity in access method. This patch adds new rsnd_kctrl_xxx() and implements uniformed access method. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ctu.c | 80 ++++++++++++++++++++++++------------------------ sound/soc/sh/rcar/dvc.c | 19 ++++++------ sound/soc/sh/rcar/mix.c | 10 +++--- sound/soc/sh/rcar/rsnd.h | 4 +++ 4 files changed, 59 insertions(+), 54 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index ad07ba8..d201d55 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -133,7 +133,7 @@ static void rsnd_ctu_value_init(struct rsnd_dai_stream *io, int i; for (i = 0; i < RSND_MAX_CHANNELS; i++) { - u32 val = ctu->pass.val[i]; + u32 val = rsnd_kctrl_valm(ctu->pass, i); cpmdr |= val << (28 - (i * 4)); @@ -150,44 +150,44 @@ static void rsnd_ctu_value_init(struct rsnd_dai_stream *io, rsnd_mod_write(mod, CTU_SCMDR, scmdr); if (scmdr > 0) { - rsnd_mod_write(mod, CTU_SV00R, ctu->sv0.val[0]); - rsnd_mod_write(mod, CTU_SV01R, ctu->sv0.val[1]); - rsnd_mod_write(mod, CTU_SV02R, ctu->sv0.val[2]); - rsnd_mod_write(mod, CTU_SV03R, ctu->sv0.val[3]); - rsnd_mod_write(mod, CTU_SV04R, ctu->sv0.val[4]); - rsnd_mod_write(mod, CTU_SV05R, ctu->sv0.val[5]); - rsnd_mod_write(mod, CTU_SV06R, ctu->sv0.val[6]); - rsnd_mod_write(mod, CTU_SV07R, ctu->sv0.val[7]); + rsnd_mod_write(mod, CTU_SV00R, rsnd_kctrl_valm(ctu->sv0, 0)); + rsnd_mod_write(mod, CTU_SV01R, rsnd_kctrl_valm(ctu->sv0, 1)); + rsnd_mod_write(mod, CTU_SV02R, rsnd_kctrl_valm(ctu->sv0, 2)); + rsnd_mod_write(mod, CTU_SV03R, rsnd_kctrl_valm(ctu->sv0, 3)); + rsnd_mod_write(mod, CTU_SV04R, rsnd_kctrl_valm(ctu->sv0, 4)); + rsnd_mod_write(mod, CTU_SV05R, rsnd_kctrl_valm(ctu->sv0, 5)); + rsnd_mod_write(mod, CTU_SV06R, rsnd_kctrl_valm(ctu->sv0, 6)); + rsnd_mod_write(mod, CTU_SV07R, rsnd_kctrl_valm(ctu->sv0, 7)); } if (scmdr > 1) { - rsnd_mod_write(mod, CTU_SV10R, ctu->sv1.val[0]); - rsnd_mod_write(mod, CTU_SV11R, ctu->sv1.val[1]); - rsnd_mod_write(mod, CTU_SV12R, ctu->sv1.val[2]); - rsnd_mod_write(mod, CTU_SV13R, ctu->sv1.val[3]); - rsnd_mod_write(mod, CTU_SV14R, ctu->sv1.val[4]); - rsnd_mod_write(mod, CTU_SV15R, ctu->sv1.val[5]); - rsnd_mod_write(mod, CTU_SV16R, ctu->sv1.val[6]); - rsnd_mod_write(mod, CTU_SV17R, ctu->sv1.val[7]); + rsnd_mod_write(mod, CTU_SV10R, rsnd_kctrl_valm(ctu->sv1, 0)); + rsnd_mod_write(mod, CTU_SV11R, rsnd_kctrl_valm(ctu->sv1, 1)); + rsnd_mod_write(mod, CTU_SV12R, rsnd_kctrl_valm(ctu->sv1, 2)); + rsnd_mod_write(mod, CTU_SV13R, rsnd_kctrl_valm(ctu->sv1, 3)); + rsnd_mod_write(mod, CTU_SV14R, rsnd_kctrl_valm(ctu->sv1, 4)); + rsnd_mod_write(mod, CTU_SV15R, rsnd_kctrl_valm(ctu->sv1, 5)); + rsnd_mod_write(mod, CTU_SV16R, rsnd_kctrl_valm(ctu->sv1, 6)); + rsnd_mod_write(mod, CTU_SV17R, rsnd_kctrl_valm(ctu->sv1, 7)); } if (scmdr > 2) { - rsnd_mod_write(mod, CTU_SV20R, ctu->sv2.val[0]); - rsnd_mod_write(mod, CTU_SV21R, ctu->sv2.val[1]); - rsnd_mod_write(mod, CTU_SV22R, ctu->sv2.val[2]); - rsnd_mod_write(mod, CTU_SV23R, ctu->sv2.val[3]); - rsnd_mod_write(mod, CTU_SV24R, ctu->sv2.val[4]); - rsnd_mod_write(mod, CTU_SV25R, ctu->sv2.val[5]); - rsnd_mod_write(mod, CTU_SV26R, ctu->sv2.val[6]); - rsnd_mod_write(mod, CTU_SV27R, ctu->sv2.val[7]); + rsnd_mod_write(mod, CTU_SV20R, rsnd_kctrl_valm(ctu->sv2, 0)); + rsnd_mod_write(mod, CTU_SV21R, rsnd_kctrl_valm(ctu->sv2, 1)); + rsnd_mod_write(mod, CTU_SV22R, rsnd_kctrl_valm(ctu->sv2, 2)); + rsnd_mod_write(mod, CTU_SV23R, rsnd_kctrl_valm(ctu->sv2, 3)); + rsnd_mod_write(mod, CTU_SV24R, rsnd_kctrl_valm(ctu->sv2, 4)); + rsnd_mod_write(mod, CTU_SV25R, rsnd_kctrl_valm(ctu->sv2, 5)); + rsnd_mod_write(mod, CTU_SV26R, rsnd_kctrl_valm(ctu->sv2, 6)); + rsnd_mod_write(mod, CTU_SV27R, rsnd_kctrl_valm(ctu->sv2, 7)); } if (scmdr > 3) { - rsnd_mod_write(mod, CTU_SV30R, ctu->sv3.val[0]); - rsnd_mod_write(mod, CTU_SV31R, ctu->sv3.val[1]); - rsnd_mod_write(mod, CTU_SV32R, ctu->sv3.val[2]); - rsnd_mod_write(mod, CTU_SV33R, ctu->sv3.val[3]); - rsnd_mod_write(mod, CTU_SV34R, ctu->sv3.val[4]); - rsnd_mod_write(mod, CTU_SV35R, ctu->sv3.val[5]); - rsnd_mod_write(mod, CTU_SV36R, ctu->sv3.val[6]); - rsnd_mod_write(mod, CTU_SV37R, ctu->sv3.val[7]); + rsnd_mod_write(mod, CTU_SV30R, rsnd_kctrl_valm(ctu->sv3, 0)); + rsnd_mod_write(mod, CTU_SV31R, rsnd_kctrl_valm(ctu->sv3, 1)); + rsnd_mod_write(mod, CTU_SV32R, rsnd_kctrl_valm(ctu->sv3, 2)); + rsnd_mod_write(mod, CTU_SV33R, rsnd_kctrl_valm(ctu->sv3, 3)); + rsnd_mod_write(mod, CTU_SV34R, rsnd_kctrl_valm(ctu->sv3, 4)); + rsnd_mod_write(mod, CTU_SV35R, rsnd_kctrl_valm(ctu->sv3, 5)); + rsnd_mod_write(mod, CTU_SV36R, rsnd_kctrl_valm(ctu->sv3, 6)); + rsnd_mod_write(mod, CTU_SV37R, rsnd_kctrl_valm(ctu->sv3, 7)); } rsnd_mod_write(mod, CTU_CTUIR, 0); @@ -199,17 +199,17 @@ static void rsnd_ctu_value_reset(struct rsnd_dai_stream *io, struct rsnd_ctu *ctu = rsnd_mod_to_ctu(mod); int i; - if (!ctu->reset.val) + if (!rsnd_kctrl_vals(ctu->reset)) return; for (i = 0; i < RSND_MAX_CHANNELS; i++) { - ctu->pass.val[i] = 0; - ctu->sv0.val[i] = 0; - ctu->sv1.val[i] = 0; - ctu->sv2.val[i] = 0; - ctu->sv3.val[i] = 0; + rsnd_kctrl_valm(ctu->pass, i) = 0; + rsnd_kctrl_valm(ctu->sv0, i) = 0; + rsnd_kctrl_valm(ctu->sv1, i) = 0; + rsnd_kctrl_valm(ctu->sv2, i) = 0; + rsnd_kctrl_valm(ctu->sv3, i) = 0; } - ctu->reset.val = 0; + rsnd_kctrl_vals(ctu->reset) = 0; } static int rsnd_ctu_init(struct rsnd_mod *mod, diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 4ef318a..dbe54f0 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -73,8 +73,9 @@ static void rsnd_dvc_halt(struct rsnd_mod *mod) rsnd_mod_write(mod, DVC_SWRSR, 0); } -#define rsnd_dvc_get_vrpdr(dvc) (dvc->rup.val << 8 | dvc->rdown.val) -#define rsnd_dvc_get_vrdbr(dvc) (0x3ff - (dvc->volume.val[0] >> 13)) +#define rsnd_dvc_get_vrpdr(dvc) (rsnd_kctrl_vals(dvc->rup) << 8 | \ + rsnd_kctrl_vals(dvc->rdown)) +#define rsnd_dvc_get_vrdbr(dvc) (0x3ff - (rsnd_kctrl_valm(dvc->volume, 0) >> 13)) static void rsnd_dvc_volume_parameter(struct rsnd_dai_stream *io, struct rsnd_mod *mod) @@ -84,12 +85,12 @@ static void rsnd_dvc_volume_parameter(struct rsnd_dai_stream *io, int i; /* Enable Ramp */ - if (dvc->ren.val) + if (rsnd_kctrl_vals(dvc->ren)) for (i = 0; i < RSND_MAX_CHANNELS; i++) - val[i] = dvc->volume.cfg.max; + val[i] = rsnd_kctrl_max(dvc->volume); else for (i = 0; i < RSND_MAX_CHANNELS; i++) - val[i] = dvc->volume.val[i]; + val[i] = rsnd_kctrl_valm(dvc->volume, i); /* Enable Digital Volume */ rsnd_mod_write(mod, DVC_VOL0R, val[0]); @@ -119,7 +120,7 @@ static void rsnd_dvc_volume_init(struct rsnd_dai_stream *io, dvucr |= 0x101; /* Enable Ramp */ - if (dvc->ren.val) { + if (rsnd_kctrl_vals(dvc->ren)) { dvucr |= 0x10; /* @@ -161,10 +162,10 @@ static void rsnd_dvc_volume_update(struct rsnd_dai_stream *io, u32 vrdbr = 0; int i; - for (i = 0; i < dvc->mute.cfg.size; i++) - zcmcr |= (!!dvc->mute.cfg.val[i]) << i; + for (i = 0; i < rsnd_kctrl_size(dvc->mute); i++) + zcmcr |= (!!rsnd_kctrl_valm(dvc->mute, i)) << i; - if (dvc->ren.val) { + if (rsnd_kctrl_vals(dvc->ren)) { vrpdr = rsnd_dvc_get_vrpdr(dvc); vrdbr = rsnd_dvc_get_vrdbr(dvc); } diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 912cfec..7998380 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -84,7 +84,7 @@ static void rsnd_mix_halt(struct rsnd_mod *mod) #define rsnd_mix_get_vol(mix, X) \ rsnd_flags_has(mix, HAS_VOL##X) ? \ - (VOL_MAX - mix->volume##X.cfg.val[0]) : 0 + (VOL_MAX - rsnd_kctrl_vals(mix->volume##X)) : 0 static void rsnd_mix_volume_parameter(struct rsnd_dai_stream *io, struct rsnd_mod *mod) { @@ -116,9 +116,9 @@ static void rsnd_mix_volume_init(struct rsnd_dai_stream *io, rsnd_mod_write(mod, MIX_ADINR, rsnd_runtime_channel_after_ctu(io)); /* volume step */ - rsnd_mod_write(mod, MIX_MIXMR, mix->ren.cfg.val[0]); - rsnd_mod_write(mod, MIX_MVPDR, mix->rup.cfg.val[0] << 8 | - mix->rdw.cfg.val[0]); + rsnd_mod_write(mod, MIX_MIXMR, rsnd_kctrl_vals(mix->ren)); + rsnd_mod_write(mod, MIX_MVPDR, rsnd_kctrl_vals(mix->rup) << 8 | + rsnd_kctrl_vals(mix->rdw)); /* common volume parameter */ rsnd_mix_volume_parameter(io, mod); @@ -217,7 +217,7 @@ static int rsnd_mix_pcm_new(struct rsnd_mod *mod, volume, VOL_MAX); if (ret < 0) return ret; - volume->cfg.val[0] = VOL_MAX; + rsnd_kctrl_vals(*volume) = VOL_MAX; if (rsnd_flags_has(mix, ONCE_KCTRL_INITIALIZED)) return ret; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 0d9f9c3..2a224fa 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -631,6 +631,10 @@ struct rsnd_kctrl_cfg_s { struct rsnd_kctrl_cfg cfg; u32 val; }; +#define rsnd_kctrl_size(x) ((x).cfg.size) +#define rsnd_kctrl_max(x) ((x).cfg.max) +#define rsnd_kctrl_valm(x, i) ((x).val[i]) /* = (x).cfg.val[i] */ +#define rsnd_kctrl_vals(x) ((x).val) /* = (x).cfg.val[0] */ int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io); int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io); -- 1.9.1

2 1

[alsa-devel] Pulseaudio say me to report a issue.
by Diniz Bortolotto 12 Oct '17

12 Oct '17

Hi buddies! My name is Diniz and I'm new here in alsa-devel list. I just entered in the list because pulseaudio reported a error in my syslog. I'm running a Slackware 64bits current machine and receive the messages below: pulseaudio[1938]: [alsa-sink-ALC887-VD Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. pulseaudio[1938]: [alsa-sink-ALC887-VD Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver "snd_hda_intel". Please report this issue to the ALSA developers. pulseaudio[1938]: [alsa-sink-ALC887-VD Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. I don't know if that is a big deal. Sorry if it is not. I only wanna help. Sorry my poor english. :-) Best Regards, Diniz Bortolotto

1 0

[alsa-devel] Click sound when playing "silence" buffer
by Владимир Мартьянов 11 Oct '17

11 Oct '17

Hello! I'm completely new to ALSA and I tried to do a simply buzzer using it. During the debug I noticed a strange behaviour: I used SND_PCM_FORMAT_S8 format, tried to play the buffer filled with zero bytes and I heard click sound when my program starts and end playing the buffer. When I filled the buffer with 0x80 bytes I didn't hear any clicks. The closer the bytes value to 0x80, the quieter the click is. I tried snd_pcm_format_set_silence() instead of memset() and got clicks too. Why I got them? Does it mean what when I use some formats, the values in buffer for zero output voltage are not 0x00? My source code is in attachment, my sound card is Intel HDA.

1 0

[alsa-devel] [PATCH] ALSA: seq: Fix use-after-free at creating a port
by Takashi Iwai 11 Oct '17

11 Oct '17

There is a potential race window opened at creating and deleting a port via ioctl, as spotted by fuzzing. snd_seq_create_port() creates a port object and returns its pointer, but it doesn't take the refcount, thus it can be deleted immediately by another thread. Meanwhile, snd_seq_ioctl_create_port() still calls the function snd_seq_system_client_ev_port_start() with the created port object that is being deleted, and this triggers use-after-free like: BUG: KASAN: use-after-free in snd_seq_ioctl_create_port+0x504/0x630 [snd_seq] at addr ffff8801f2241cb1 ============================================================================= BUG kmalloc-512 (Tainted: G B ): kasan: bad access detected ----------------------------------------------------------------------------- INFO: Allocated in snd_seq_create_port+0x94/0x9b0 [snd_seq] age=1 cpu=3 pid=4511 ___slab_alloc+0x425/0x460 __slab_alloc+0x20/0x40 kmem_cache_alloc_trace+0x150/0x190 snd_seq_create_port+0x94/0x9b0 [snd_seq] snd_seq_ioctl_create_port+0xd1/0x630 [snd_seq] snd_seq_do_ioctl+0x11c/0x190 [snd_seq] snd_seq_ioctl+0x40/0x80 [snd_seq] do_vfs_ioctl+0x54b/0xda0 SyS_ioctl+0x79/0x90 entry_SYSCALL_64_fastpath+0x16/0x75 INFO: Freed in port_delete+0x136/0x1a0 [snd_seq] age=1 cpu=2 pid=4717 __slab_free+0x204/0x310 kfree+0x15f/0x180 port_delete+0x136/0x1a0 [snd_seq] snd_seq_delete_port+0x235/0x350 [snd_seq] snd_seq_ioctl_delete_port+0xc8/0x180 [snd_seq] snd_seq_do_ioctl+0x11c/0x190 [snd_seq] snd_seq_ioctl+0x40/0x80 [snd_seq] do_vfs_ioctl+0x54b/0xda0 SyS_ioctl+0x79/0x90 entry_SYSCALL_64_fastpath+0x16/0x75 Call Trace: [<ffffffff81b03781>] dump_stack+0x63/0x82 [<ffffffff81531b3b>] print_trailer+0xfb/0x160 [<ffffffff81536db4>] object_err+0x34/0x40 [<ffffffff815392d3>] kasan_report.part.2+0x223/0x520 [<ffffffffa07aadf4>] ? snd_seq_ioctl_create_port+0x504/0x630 [snd_seq] [<ffffffff815395fe>] __asan_report_load1_noabort+0x2e/0x30 [<ffffffffa07aadf4>] snd_seq_ioctl_create_port+0x504/0x630 [snd_seq] [<ffffffffa07aa8f0>] ? snd_seq_ioctl_delete_port+0x180/0x180 [snd_seq] [<ffffffff8136be50>] ? taskstats_exit+0xbc0/0xbc0 [<ffffffffa07abc5c>] snd_seq_do_ioctl+0x11c/0x190 [snd_seq] [<ffffffffa07abd10>] snd_seq_ioctl+0x40/0x80 [snd_seq] [<ffffffff8136d433>] ? acct_account_cputime+0x63/0x80 [<ffffffff815b515b>] do_vfs_ioctl+0x54b/0xda0 ..... We may fix this in a few different ways, and in this patch, it's fixed simply by taking the refcount properly at snd_seq_create_port() and letting the caller unref the object after use. Also, there is another potential use-after-free by sprintf() call in snd_seq_create_port(), and this is moved inside the lock. This fix covers CVE-2017-15265. Reported-and-tested-by: Michael23 Yu <ycqzsy(a)gmail.com> Suggested-by: Linus Torvalds <torvalds(a)linux-foundation.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_clientmgr.c | 6 +++++- sound/core/seq/seq_ports.c | 7 +++++-- 2 files changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index ea2d0ae85bd3..6c9cba2166d9 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1259,6 +1259,7 @@ static int snd_seq_ioctl_create_port(struct snd_seq_client *client, void *arg) struct snd_seq_port_info *info = arg; struct snd_seq_client_port *port; struct snd_seq_port_callback *callback; + int port_idx; /* it is not allowed to create the port for an another client */ if (info->addr.client != client->number) @@ -1269,7 +1270,9 @@ static int snd_seq_ioctl_create_port(struct snd_seq_client *client, void *arg) return -ENOMEM; if (client->type == USER_CLIENT && info->kernel) { - snd_seq_delete_port(client, port->addr.port); + port_idx = port->addr.port; + snd_seq_port_unlock(port); + snd_seq_delete_port(client, port_idx); return -EINVAL; } if (client->type == KERNEL_CLIENT) { @@ -1290,6 +1293,7 @@ static int snd_seq_ioctl_create_port(struct snd_seq_client *client, void *arg) snd_seq_set_port_info(port, info); snd_seq_system_client_ev_port_start(port->addr.client, port->addr.port); + snd_seq_port_unlock(port); return 0; } diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 0a7020c82bfc..d21ece9f8d73 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -122,7 +122,9 @@ static void port_subs_info_init(struct snd_seq_port_subs_info *grp) } -/* create a port, port number is returned (-1 on failure) */ +/* create a port, port number is returned (-1 on failure); + * the caller needs to unref the port via snd_seq_port_unlock() appropriately + */ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, int port) { @@ -151,6 +153,7 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, snd_use_lock_init(&new_port->use_lock); port_subs_info_init(&new_port->c_src); port_subs_info_init(&new_port->c_dest); + snd_use_lock_use(&new_port->use_lock); num = port >= 0 ? port : 0; mutex_lock(&client->ports_mutex); @@ -165,9 +168,9 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, list_add_tail(&new_port->list, &p->list); client->num_ports++; new_port->addr.port = num; /* store the port number in the port */ + sprintf(new_port->name, "port-%d", num); write_unlock_irqrestore(&client->ports_lock, flags); mutex_unlock(&client->ports_mutex); - sprintf(new_port->name, "port-%d", num); return new_port; } -- 2.14.2

2 2

[alsa-devel] PCM2902: click/pop sound when it should be silent.
by Theodor van Nahl 11 Oct '17

11 Oct '17

Hello, I do have a Behringer U-PHORIA UMC22 USB soundcard which uses the PCM2902 soundchip. I am using that device with pulseaudio on top and the trouble occurs when I do not use the device. Problem ======= In various intervals, but only when I don't play audio, I do hear a pop or click sound comparable to the sound a power amp does when connecting or disconnecting the power. My Analysis =========== I did watch the usb port using wireshark and the following frame has been the last frame before that sound: Frame 3470: 64 bytes on wire (512 bits), 64 bytes captured (512 bits) on interface 0 USB URB [Source: host] [Destination: 2.4.0] URB id: 0xffff9522ee93f9c0 URB type: URB_SUBMIT ('S') URB transfer type: URB_CONTROL (0x02) Endpoint: 0x00, Direction: OUT Device: 4 URB bus id: 2 Device setup request: relevant (0) Data: present (0) URB sec: 1507602131 URB usec: 34054 URB status: Operation now in progress (-EINPROGRESS) (-115) URB length [bytes]: 0 Data length [bytes]: 0 [Response in: 3471] Interval: 0 Start frame: 0 Copy of Transfer Flags: 0x00000000 Number of ISO descriptors: 0 URB setup bmRequestType: 0x01 0... .... = Direction: Host-to-device .00. .... = Type: Standard (0x0) ...0 0001 = Recipient: Interface (0x01) bRequest: SET INTERFACE (11) bAlternateSetting: 0 wInterface: 1 wLength: 0 From what I understand from the PCM2902 Manual [0] on page 22 is that the streaming interface gets deactivated (zero bandwith setting) by this frame. If I read the documentation of the PCM2902 correctly that step is not nesessary, the chip suspends itself when no data comes in (p. 26). Reading *linux/sound/usb/endpoint.c* I am convinced that the frame above and its counter part are send by this code and send everytime a first endpoint (pulseaudio, aplay, …) connects and the last endpoint disconnected. Due to that I do believe that this is a bug in the driver used for the pcm2902 chip. References ========== That bug is not unheard of as you can see in [1] and [2] but the workarounds provided in those links I cannot confirm to work. I hope that my analysis is as conclusive for you as they sound to me. From this point I am unsure where to head next and was hoping for help here. So my followup questions are: * What do you believe where and "why there" this should be fixed? * Why is the frame above beeing send to the device? The soundcard does seem to have power after that frame. * Isn't it possible to never set the pcm2902 into "zero bandwith mode"? Greetings, Theodor [0]: http://www.ti.com/product/pcm2902 [1]: https://ubuntuforums.org/showthread.php?t=2292143&p=13345100#post13345100 [2]: https://www.reddit.com/r/linux/comments/3idb1h/clicky_noise_before_and_afte…

1 0

[alsa-devel] [PATCH v2 0/9] sound: Add sanity checks for invalid EPs
by Takashi Iwai 11 Oct '17

11 Oct '17

Hi, this is a revised patch set to cover the codes that may submit URBs containing invalid EPs without validation, which result in the kernel warning from the USB core. The first patch adds a new helper for simplifying the EP check, and the rest applies it at needed places. The original issues were spotted by syzkaller, and I put a few others for covering more similar cases. The only updated code in v2 is about the first patch to add usb_urb_ep_type_check() helper (in addition to tested-by tag from Andrey). Typos were fixed and it's called also from usb_submit_urb() as Greg suggested, too. USB devs: does this look OK now? If yes, and if I get acks, I can take the patches to sound tree. Or I don't mind if you take all to usb tree, too. Or maybe better, I'll prepare an immutable branch based on 4.4.14-rc (rc4 for now) in case you want to pull into both trees. Let me know your wish. thanks, Takashi === Takashi Iwai (9): usb: core: Add a helper function to check the validity of EP type in URB ALSA: bcd2000: Add a sanity check for invalid EPs ALSA: caiaq: Add a sanity check for invalid EPs ALSA: line6: Add a sanity check for invalid EPs ALSA: usb-audio: Add sanity checks for invalid EPs ALSA: usx2y: Add sanity checks for invalid EPs ALSA: hiface: Add sanity checks for invalid EPs ALSA: caiaq: Add yet more sanity checks for invalid EPs ALSA: line6: Add yet more sanity checks for invalid EPs drivers/usb/core/urb.c | 30 ++++++++++++++++++++++++++---- include/linux/usb.h | 2 ++ sound/usb/bcd2000/bcd2000.c | 7 +++++++ sound/usb/caiaq/device.c | 7 +++++++ sound/usb/caiaq/input.c | 9 +++++++++ sound/usb/hiface/pcm.c | 9 +++++++-- sound/usb/line6/driver.c | 30 ++++++++++++++++++++++-------- sound/usb/line6/midi.c | 17 +++++++++++------ sound/usb/midi.c | 38 ++++++++++++++++++++++++++++++-------- sound/usb/usx2y/usbusx2y.c | 8 ++++++++ sound/usb/usx2y/usbusx2yaudio.c | 3 +++ 11 files changed, 132 insertions(+), 28 deletions(-) -- 2.14.2

3 26

[alsa-devel] [PATCH 0/3] ALSA: add snd_card_disconnect_sync()
by Kuninori Morimoto 11 Oct '17

11 Oct '17

Hi Takashi-san, Mark Current ALSA (SoC) might be kernel panic if user unbinds sound driver during playback. > aplay xxx.wav & > echo xxxx > /sys/bus/platform/drivers/xxxx/unbind Main issue is we *can't* skip unbind with return -Exxx, because unbind operation doesn't checks return value from each driver. Thus, we *must* stop driver immediately when unbind. 1) is for ALSA 2) is for ALSA SoC 3) is for Renesas sound driver. I think almost all driver want to have 3) type patch on .remove but this patch-set doesn't care at this point. Please let me know if they should have it. Takashi Iwai (1): 1) ALSA: add snd_card_disconnect_sync() Kuninori Morimoto (2): 2) ASoC: add snd_soc_card_disconnect_sync() 3) ASoC: rsnd: call snd_soc_card_disconnect_sync() when remove include/sound/core.h | 2 ++ include/sound/soc.h | 2 ++ sound/core/init.c | 18 ++++++++++++++++++ sound/core/pcm.c | 4 ++++ sound/soc/sh/rcar/core.c | 2 ++ sound/soc/soc-core.c | 10 ++++++++++ 6 files changed, 38 insertions(+) -- 1.9.1 Best regards --- Kuninori Morimoto

3 8

[alsa-devel] [PATCH] ASoC: rsnd: stop all working stream when .remove
by Kuninori Morimoto 11 Oct '17

11 Oct '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Driver should stop all working stream when .remove timing. Current Renesas sound driver is assuming that all stream was stopped when .remove but it was wrong. This patch stops all working stream when .remove, otherwise kernel will get damage for example in below case. Special thanks to Truong, Hiep > cd /sys/bus/platform/drivers/rcar_sound > aplay xxx.wav & > echo ec500000.sound > unbind Reported-by: Hiep Cao Minh <cm-hiep(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 68 ++++++++++++++++++++++++++++++++++-------------- 1 file changed, 48 insertions(+), 20 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1071332..1bf472f 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -197,12 +197,6 @@ void rsnd_mod_interrupt(struct rsnd_mod *mod, } } -int rsnd_io_is_working(struct rsnd_dai_stream *io) -{ - /* see rsnd_dai_stream_init/quit() */ - return !!io->substream; -} - int rsnd_runtime_channel_original(struct rsnd_dai_stream *io) { struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); @@ -610,20 +604,24 @@ struct rsnd_dai_stream *rsnd_rdai_to_io(struct rsnd_dai *rdai, return &rdai->capture; } -static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) +int rsnd_io_is_working(struct rsnd_dai_stream *io) +{ + /* see rsnd_dai_stream_init/quit() */ + return !!io->substream; +} + +#define rsnd_io_start(priv, io, sub) rsnd_io_operation(priv, io, sub) +#define rsnd_io_stop(priv, io) rsnd_io_operation(priv, io, NULL) +static int rsnd_io_operation(struct rsnd_priv *priv, + struct rsnd_dai_stream *io, + struct snd_pcm_substream *substream) { - struct rsnd_priv *priv = rsnd_dai_to_priv(dai); - struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); int ret; unsigned long flags; spin_lock_irqsave(&priv->lock, flags); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: + if (substream) { rsnd_dai_stream_init(io, substream); ret = rsnd_dai_call(init, io, priv); @@ -638,9 +636,7 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, if (ret < 0) goto dai_trigger_end; - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: + } else { ret = rsnd_dai_call(irq, io, priv, 0); ret |= rsnd_dai_call(stop, io, priv); @@ -648,9 +644,6 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, ret |= rsnd_dai_call(quit, io, priv); rsnd_dai_stream_quit(io); - break; - default: - ret = -EINVAL; } dai_trigger_end: @@ -659,6 +652,30 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct rsnd_priv *priv = rsnd_dai_to_priv(dai); + struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + ret = rsnd_io_start(priv, io, substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + ret = rsnd_io_stop(priv, io); + break; + default: + ret = -EINVAL; + } + + return ret; +} + static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); @@ -1477,6 +1494,17 @@ static int rsnd_remove(struct platform_device *pdev) }; int ret = 0, i; + /* + * Stop all working io + */ + for_each_rsnd_dai(rdai, priv, i) { + if (rsnd_io_is_working(&rdai->playback)) + rsnd_io_stop(priv, &rdai->playback); + + if (rsnd_io_is_working(&rdai->capture)) + rsnd_io_stop(priv, &rdai->capture); + } + pm_runtime_disable(&pdev->dev); for_each_rsnd_dai(rdai, priv, i) { -- 1.9.1

3 21

[alsa-devel] [PATCH 0/3] ALSA: add snd_card_disconnect_sync()
by Kuninori Morimoto 11 Oct '17

11 Oct '17

Hi Takashi-san, Mark Current ALSA (SoC) might be kernel panic if user unbinds sound driver during playback. > aplay xxx.wav & > echo xxxx > /sys/bus/platform/drivers/xxxx/unbind Main issue is we *can't* skip unbind with return -Exxx, because unbind operation doesn't checks return value from each driver. Thus, we *must* stop driver immediately when unbind. 1) is for ALSA 2) is for ALSA SoC 3) is for Renesas sound driver. I think almost all driver want to have 3) type patch on .remove but this patch-set doesn't care at this point. Please let me know if they should have it. Takashi Iwai (1): 1) ALSA: add snd_card_disconnect_sync() Kuninori Morimoto (2): 2) ASoC: add snd_soc_card_disconnect_sync() 3) ASoC: rsnd: call snd_soc_card_disconnect_sync() when remove include/sound/core.h | 2 ++ include/sound/soc.h | 2 ++ sound/core/init.c | 18 ++++++++++++++++++ sound/core/pcm.c | 4 ++++ sound/soc/sh/rcar/core.c | 2 ++ sound/soc/soc-core.c | 10 ++++++++++ 6 files changed, 38 insertions(+) -- 1.9.1 Best regards --- Kuninori Morimoto

1 0

[alsa-devel] [PATCH 0/3] ALSA: add snd_card_disconnect_sync()
by Kuninori Morimoto 11 Oct '17

11 Oct '17

Hi Takashi-san, Mark Current ALSA (SoC) might be kernel panic if user unbinds sound driver during playback. > aplay xxx.wav & > echo xxxx > /sys/bus/platform/drivers/xxxx/unbind Main issue is we *can't* skip unbind with return -Exxx, because unbind operation doesn't checks return value from each driver. Thus, we *must* stop driver immediately when unbind. 1) is for ALSA 2) is for ALSA SoC 3) is for Renesas sound driver. I think almost all driver want to have 3) type patch on .remove but this patch-set doesn't care at this point. Please let me know if they should have it. Takashi Iwai (1): 1) ALSA: add snd_card_disconnect_sync() Kuninori Morimoto (2): 2) ASoC: add snd_soc_card_disconnect_sync() 3) ASoC: rsnd: call snd_soc_card_disconnect_sync() when remove include/sound/core.h | 2 ++ include/sound/soc.h | 2 ++ sound/core/init.c | 18 ++++++++++++++++++ sound/core/pcm.c | 4 ++++ sound/soc/sh/rcar/core.c | 2 ++ sound/soc/soc-core.c | 10 ++++++++++ 6 files changed, 38 insertions(+) -- 1.9.1 Best regards --- Kuninori Morimoto

1 0

[alsa-devel] [PATCH 0/9] sound: Add sanity checks for invalid EPs
by Takashi Iwai 10 Oct '17

10 Oct '17

Hi, this is a patchset to cover the codes that may submit URBs containing invalid EPs without validation, which result in the kernel warning from the USB core. The first patch adds a new helper for simplifying the EP check, and the rest applies it at needed places. USB devs: could you check the first patch? I tried usb_find_common_endpoints() and friends at first, but it made the code more complex in the end, because we're dealing with the fixed endpoints and the validation is required for them. The original issues were spotted by syzkaller, and I put a few others for covering more similar cases. thanks, Takashi === Takashi Iwai (9): usb: core: Add a helper function to check the validity of EP type in URB ALSA: bcd2000: Add a sanity check for invalid EPs ALSA: caiaq: Add a sanity check for invalid EPs ALSA: line6: Add a sanity check for invalid EPs ALSA: usb-audio: Add sanity checks for invalid EPs ALSA: usx2y: Add sanity checks for invalid EPs ALSA: hiface: Add sanity checks for invalid EPs ALSA: caiaq: Add yet more sanity checks for invalid EPs ALSA: line6: Add yet more sanity checks for invalid EPs drivers/usb/core/urb.c | 28 +++++++++++++++++++++++++--- include/linux/usb.h | 2 ++ sound/usb/bcd2000/bcd2000.c | 7 +++++++ sound/usb/caiaq/device.c | 7 +++++++ sound/usb/caiaq/input.c | 9 +++++++++ sound/usb/hiface/pcm.c | 9 +++++++-- sound/usb/line6/driver.c | 30 ++++++++++++++++++++++-------- sound/usb/line6/midi.c | 17 +++++++++++------ sound/usb/midi.c | 38 ++++++++++++++++++++++++++++++-------- sound/usb/usx2y/usbusx2y.c | 8 ++++++++ sound/usb/usx2y/usbusx2yaudio.c | 3 +++ 11 files changed, 131 insertions(+), 27 deletions(-) -- 2.14.2

3 14

[alsa-devel] [PATCH 6/8] Documentation: fix sound related doc refs
by Tom Saeger 10 Oct '17

10 Oct '17

Make sound doc refs valid. Signed-off-by: Tom Saeger <tom.saeger(a)oracle.com> --- Documentation/sound/hd-audio/notes.rst | 2 +- Documentation/sound/kernel-api/writing-an-alsa-driver.rst | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst index f59c3cdbfaf4..9f7347830ba4 100644 --- a/Documentation/sound/hd-audio/notes.rst +++ b/Documentation/sound/hd-audio/notes.rst @@ -192,7 +192,7 @@ preset model instead of PCI (and codec-) SSID look-up. What ``model`` option values are available depends on the codec chip. Check your codec chip from the codec proc file (see "Codec Proc-File" section below). It will show the vendor/product name of your codec -chip. Then, see Documentation/sound/HD-Audio-Models.rst file, +chip. Then, see Documentation/sound/hd-audio/models.rst file, the section of HD-audio driver. You can find a list of codecs and ``model`` options belonging to each codec. For example, for Realtek ALC262 codec chip, pass ``model=ultra`` for devices that are compatible diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 58ffa3f5bda7..a0b268466cb1 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -2498,7 +2498,7 @@ Mic boost Mic-boost switch is set as “Mic Boost” or “Mic Boost (6dB)”. More precise information can be found in -``Documentation/sound/alsa/ControlNames.txt``. +``Documentation/sound/designs/control-names.rst``. Access Flags ------------ -- 2.14.2

2 1

[alsa-devel] [PATCH 3/8] Documentation: fix input related doc refs
by Tom Saeger 10 Oct '17

10 Oct '17

Make `input` document refs valid including: - joystick - joystick-parport Signed-off-by: Tom Saeger <tom.saeger(a)oracle.com> --- Documentation/admin-guide/kernel-parameters.txt | 10 +++++----- Documentation/hid/hiddev.txt | 2 +- Documentation/input/devices/xpad.rst | 3 ++- Documentation/sound/cards/joystick.rst | 2 +- 4 files changed, 9 insertions(+), 8 deletions(-) diff --git a/Documentation/admin-guide/kernel-parameters.txt b/Documentation/admin-guide/kernel-parameters.txt index e857bbbc8575..411b41127eee 100644 --- a/Documentation/admin-guide/kernel-parameters.txt +++ b/Documentation/admin-guide/kernel-parameters.txt @@ -314,7 +314,7 @@ amijoy.map= [HW,JOY] Amiga joystick support Map of devices attached to JOY0DAT and JOY1DAT Format: <a>,<b> - See also Documentation/input/joystick.txt + See also Documentation/input/joydev/joystick.rst analog.map= [HW,JOY] Analog joystick and gamepad support Specifies type or capabilities of an analog joystick @@ -724,7 +724,7 @@ db9.dev[2|3]= [HW,JOY] Multisystem joystick support via parallel port (one device per port) Format: <port#>,<type> - See also Documentation/input/joystick-parport.txt + See also Documentation/input/devices/joystick-parport.rst ddebug_query= [KNL,DYNAMIC_DEBUG] Enable debug messages at early boot time. See @@ -1220,7 +1220,7 @@ [HW,JOY] Multisystem joystick and NES/SNES/PSX pad support via parallel port (up to 5 devices per port) Format: <port#>,<pad1>,<pad2>,<pad3>,<pad4>,<pad5> - See also Documentation/input/joystick-parport.txt + See also Documentation/input/devices/joystick-parport.rst gamma= [HW,DRM] @@ -1766,7 +1766,7 @@ ivrs_acpihid[00:14.5]=AMD0020:0 js= [HW,JOY] Analog joystick - See Documentation/input/joystick.txt. + See Documentation/input/joydev/joystick.rst. nokaslr [KNL] When CONFIG_RANDOMIZE_BASE is set, this disables @@ -4199,7 +4199,7 @@ TurboGraFX parallel port interface Format: <port#>,<js1>,<js2>,<js3>,<js4>,<js5>,<js6>,<js7> - See also Documentation/input/joystick-parport.txt + See also Documentation/input/devices/joystick-parport.rst udbg-immortal [PPC] When debugging early kernel crashes that happen after console_init() and before a proper diff --git a/Documentation/hid/hiddev.txt b/Documentation/hid/hiddev.txt index 6e8c9f1d2f22..638448707aa2 100644 --- a/Documentation/hid/hiddev.txt +++ b/Documentation/hid/hiddev.txt @@ -12,7 +12,7 @@ To support these disparate requirements, the Linux USB system provides HID events to two separate interfaces: * the input subsystem, which converts HID events into normal input device interfaces (such as keyboard, mouse and joystick) and a -normalised event interface - see Documentation/input/input.txt +normalised event interface - see Documentation/input/input.rst * the hiddev interface, which provides fairly raw HID events The data flow for a HID event produced by a device is something like diff --git a/Documentation/input/devices/xpad.rst b/Documentation/input/devices/xpad.rst index 5a709ab77c8d..b8bd65962dd8 100644 --- a/Documentation/input/devices/xpad.rst +++ b/Documentation/input/devices/xpad.rst @@ -230,4 +230,5 @@ Historic Edits 2005-03-19 - Dominic Cerquetti <binary1230(a)yahoo.com> - added stuff for dance pads, new d-pad->axes mappings -Later changes may be viewed with 'git log Documentation/input/xpad.txt' +Later changes may be viewed with +'git log --follow Documentation/input/devices/xpad.rst' diff --git a/Documentation/sound/cards/joystick.rst b/Documentation/sound/cards/joystick.rst index a6e468c81d02..488946fc1079 100644 --- a/Documentation/sound/cards/joystick.rst +++ b/Documentation/sound/cards/joystick.rst @@ -11,7 +11,7 @@ General First of all, you need to enable GAMEPORT support on Linux kernel for using a joystick with the ALSA driver. For the details of gameport -support, refer to Documentation/input/joystick.txt. +support, refer to Documentation/input/joydev/joystick.rst. The joystick support of ALSA drivers is different between ISA and PCI cards. In the case of ISA (PnP) cards, it's usually handled by the -- 2.14.2

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[alsa-devel] [RESEND PATCH] ASoC: codecs: msm8916-wcd-digital: fix RX2 MIX1 and RX3 MIX1
by Jean-François Têtu 10 Oct '17

10 Oct '17

The kcontrol for the third input (rxN_mix1_inp3) of both RX2 and RX3 mixers are not using the correct control register. This simple patch fixes this. Signed-off-by: Jean-François Têtu <jean-francois.tetu(a)savoirfairelinux.com> --- sound/soc/codecs/msm8916-wcd-digital.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 66df8f810f0d..a10a724eb448 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -238,7 +238,7 @@ static const struct soc_enum rx_mix2_inp1_chain_enum = SOC_ENUM_SINGLE( static const struct soc_enum rx2_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B1_CTL, 0, 6, rx_mix1_text), SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B1_CTL, 3, 6, rx_mix1_text), - SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B1_CTL, 0, 6, rx_mix1_text), + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B2_CTL, 0, 6, rx_mix1_text), }; /* RX2 MIX2 */ @@ -249,7 +249,7 @@ static const struct soc_enum rx2_mix2_inp1_chain_enum = SOC_ENUM_SINGLE( static const struct soc_enum rx3_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B1_CTL, 0, 6, rx_mix1_text), SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B1_CTL, 3, 6, rx_mix1_text), - SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B1_CTL, 0, 6, rx_mix1_text), + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B2_CTL, 0, 6, rx_mix1_text), }; /* DEC */ -- 2.14.2

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[alsa-devel] usb/sound: use-after-free in snd_usb_mixer_interrupt
by Andrey Konovalov 10 Oct '17

10 Oct '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit 8a5776a5f49812d29fe4b2d0a2d71675c3facf3f (4.14-rc4). gadgetfs: bound to dummy_udc driver usb 1-1: new full-speed USB device number 2 using dummy_hcd gadgetfs: connected gadgetfs: disconnected gadgetfs: connected usb 1-1: config 1 has an invalid interface number: 1 but max is 0 usb 1-1: config 1 has 2 interfaces, different from the descriptor's value: 1 usb 1-1: config 1 interface 0 altsetting 0 endpoint 0x81 has an invalid bInterval 0, changing to 10 usb 1-1: too many endpoints for config 1 interface 1 altsetting 174: 131, using maximum allowed: 30 usb 1-1: config 1 interface 1 altsetting 174 has 0 endpoint descriptors, different from the interface de scriptor's value: 131 usb 1-1: config 1 interface 1 has no altsetting 0 usb 1-1: New USB device found, idVendor=0dba, idProduct=1000 usb 1-1: New USB device strings: Mfr=0, Product=0, SerialNumber=252 usb 1-1: SerialNumber: a gadgetfs: configuration #1 hub 1-1:1.0: USB hub found hub 1-1:1.0: config failed, can't read hub descriptor (err -22) snd-usb-audio: probe of 1-1:1.0 failed with error -22 ================================================================== BUG: KASAN: use-after-free in snd_usb_mixer_interrupt+0x604/0x6f0 Read of size 8 at addr ffff8800636f8900 by task swapper/1/0 CPU: 1 PID: 0 Comm: swapper/1 Not tainted 4.14.0-rc4-43418-g43a3f84d2109-dirty #391 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Call Trace: <IRQ> __dump_stack lib/dump_stack.c:16 dump_stack+0x292/0x395 lib/dump_stack.c:52 print_address_description+0x78/0x280 mm/kasan/report.c:252 kasan_report_error mm/kasan/report.c:351 kasan_report+0x23d/0x350 mm/kasan/report.c:409 __asan_report_load8_noabort+0x19/0x20 mm/kasan/report.c:430 snd_usb_mixer_interrupt+0x604/0x6f0 sound/usb/mixer.c:2490 __usb_hcd_giveback_urb+0x2e0/0x650 drivers/usb/core/hcd.c:1779 usb_hcd_giveback_urb+0x337/0x420 drivers/usb/core/hcd.c:1845 dummy_timer+0xba9/0x39f0 drivers/usb/gadget/udc/dummy_hcd.c:1965 call_timer_fn+0x2a2/0x940 kernel/time/timer.c:1281 expire_timers kernel/time/timer.c:1320 __run_timers+0x87f/0xd40 kernel/time/timer.c:1620 run_timer_softirq+0x83/0x140 kernel/time/timer.c:1646 __do_softirq+0x2ee/0xc0f kernel/softirq.c:284 invoke_softirq kernel/softirq.c:364 irq_exit+0x171/0x1a0 kernel/softirq.c:405 exiting_irq ./arch/x86/include/asm/apic.h:638 smp_apic_timer_interrupt+0x2b9/0x8d0 arch/x86/kernel/apic/apic.c:1048 apic_timer_interrupt+0x9d/0xb0 </IRQ> RIP: 0010:native_safe_halt+0x6/0x10 ./arch/x86/include/asm/irqflags.h:53 RSP: 0018:ffff88006bac7b90 EFLAGS: 00000282 ORIG_RAX: ffffffffffffff10 RAX: dffffc0000000020 RBX: 1ffff1000d758f76 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000001 RDI: ffff88006baa3a64 RBP: ffff88006bac7b90 R08: ffffffff813d5301 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 1ffff1000d758f82 R13: ffff88006bac7cd0 R14: ffffffff8881ecc8 R15: 0000000000000000 arch_safe_halt ./arch/x86/include/asm/paravirt.h:93 default_idle+0x127/0x690 arch/x86/kernel/process.c:341 arch_cpu_idle+0xf/0x20 arch/x86/kernel/process.c:332 default_idle_call+0x3b/0x60 kernel/sched/idle.c:98 cpuidle_idle_call kernel/sched/idle.c:156 do_idle+0x33a/0x410 kernel/sched/idle.c:246 cpu_startup_entry+0x1d/0x20 kernel/sched/idle.c:351 start_secondary+0x3de/0x500 arch/x86/kernel/smpboot.c:278 secondary_startup_64+0xa5/0xa5 arch/x86/kernel/head_64.S:235 Allocated by task 1484: save_stack_trace+0x1b/0x20 arch/x86/kernel/stacktrace.c:59 save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 kasan_kmalloc+0xad/0xe0 mm/kasan/kasan.c:551 kmem_cache_alloc_trace+0x11e/0x2d0 mm/slub.c:2772 kmalloc ./include/linux/slab.h:493 kzalloc ./include/linux/slab.h:666 snd_usb_create_mixer+0x145/0x1010 sound/usb/mixer.c:2540 create_standard_mixer_quirk+0x58/0x80 sound/usb/quirks.c:516 snd_usb_create_quirk+0x92/0x100 sound/usb/quirks.c:560 create_composite_quirk+0x1c4/0x3e0 sound/usb/quirks.c:59 snd_usb_create_quirk+0x92/0x100 sound/usb/quirks.c:560 usb_audio_probe+0x1040/0x2c10 sound/usb/card.c:618 usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_set_configuration+0x104e/0x1870 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_new_device+0x7b8/0x1020 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431 Freed by task 1484: save_stack_trace+0x1b/0x20 arch/x86/kernel/stacktrace.c:59 save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 kasan_slab_free+0x72/0xc0 mm/kasan/kasan.c:524 slab_free_hook mm/slub.c:1390 slab_free_freelist_hook mm/slub.c:1412 slab_free mm/slub.c:2988 kfree+0xf6/0x2f0 mm/slub.c:3919 snd_usb_mixer_free+0x11a/0x160 sound/usb/mixer.c:2244 snd_usb_mixer_dev_free+0x36/0x50 sound/usb/mixer.c:2250 __snd_device_free+0x1ff/0x380 sound/core/device.c:91 snd_device_free_all+0x8f/0xe0 sound/core/device.c:244 snd_card_do_free sound/core/init.c:461 release_card_device+0x47/0x170 sound/core/init.c:181 device_release+0x13f/0x210 drivers/base/core.c:814 kobject_cleanup lib/kobject.c:648 kobject_release lib/kobject.c:677 kref_put ./include/linux/kref.h:70 kobject_put+0x145/0x240 lib/kobject.c:694 put_device+0x25/0x30 drivers/base/core.c:1931 snd_card_free_when_closed sound/core/init.c:489 snd_card_free+0x156/0x220 sound/core/init.c:514 usb_audio_probe+0x1e36/0x2c10 sound/usb/card.c:648 usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_set_configuration+0x104e/0x1870 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_new_device+0x7b8/0x1020 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431 The buggy address belongs to the object at ffff8800636f8900 which belongs to the cache kmalloc-192 of size 192 The buggy address is located 0 bytes inside of 192-byte region [ffff8800636f8900, ffff8800636f89c0) The buggy address belongs to the page: page:ffffea00018dbe00 count:1 mapcount:0 mapping: (null) index:0x0 flags: 0x100000000000100(slab) raw: 0100000000000100 0000000000000000 0000000000000000 0000000180100010 raw: ffffea00018dbc00 0000000400000004 ffff88006c003380 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8800636f8800: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ffff8800636f8880: 00 00 00 00 00 00 00 00 fc fc fc fc fc fc fc fc >ffff8800636f8900: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8800636f8980: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff8800636f8a00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ==================================================================

2 3

[alsa-devel] [PATCH] ALSA: seq: Add sanity check for user-space pointer delivery
by Takashi Iwai 10 Oct '17

10 Oct '17

The sequencer event may contain a user-space pointer with its SNDRV_SEQ_EXT_USRPTR bit, and we assure that its delivery is limited with non-atomic mode. Otherwise the copy_from_user() may hit the fault and cause a problem. Although the core code doesn't set such a flag (only set at snd_seq_write()), any wild driver may set it mistakenly and lead to an unexpected crash. This patch adds a sanity check of such events at the delivery core code to filter out the invalid invocation in the atomic mode. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_clientmgr.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 6c9cba2166d9..84759d9c54bf 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -802,6 +802,10 @@ static int snd_seq_deliver_event(struct snd_seq_client *client, struct snd_seq_e return -EMLINK; } + if (snd_seq_ev_is_variable(event) && + snd_BUG_ON(atomic && (event->data.ext.len & SNDRV_SEQ_EXT_USRPTR))) + return -EINVAL; + if (event->queue == SNDRV_SEQ_ADDRESS_SUBSCRIBERS || event->dest.client == SNDRV_SEQ_ADDRESS_SUBSCRIBERS) result = deliver_to_subscribers(client, event, atomic, hop); -- 2.14.2

1 0

[alsa-devel] [PATCH 1/2] ASoC: dapm: add initialization for w_param_text pointer
by Arnd Bergmann 10 Oct '17

10 Oct '17

We now allocate the array conditionally, but we always pass the pointer to the new snd_soc_dapm_free_kcontrol() function, which introduces a warning for the case that it is not initialized: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': sound/soc/soc-dapm.c:3940:2: error: 'w_param_text' may be used uninitialized in this function [-Werror=maybe-uninitialized] As snd_soc_dapm_free_kcontrol() is global, it doesn't get inlined and gcc fails to notice that we don't actually access the array in that case, so the code is actually safe. Adding an initialization for the array pointer shuts up the warning. Fixes: 19ad683abc85 ("ASoC: dapm: Avoid creating kcontrol for params") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ceec185c871f..19babdae1a06 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3905,6 +3905,8 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, ret = -ENOMEM; goto param_fail; } + } else { + w_param_text = NULL; } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); -- 2.9.0

2 3

[alsa-devel] [PATCH 0/2 v2] ASoC: soc-core: add missing component functions
by Kuninori Morimoto 10 Oct '17

10 Oct '17

Hi Mark These are v2 of one of "replace to component" patch set. These add missing functions which exist in "platforms" but not exist in "component". Kuninori Morimoto (3): ASoC: soc-core: add component lookup functions ASoC: soc-core: add snd_soc_add_component() ASoC: soc-core: remove unnecessary message from snd_soc_register_component() include/sound/soc.h | 7 +++++++ sound/soc/soc-core.c | 58 ++++++++++++++++++++++++++++++++++++++++++---------- 2 files changed, 54 insertions(+), 11 deletions(-) -- 1.9.1

2 4

[alsa-devel] [bug report] ASoC: Intel: Refactor common IPC/mailbox code into generic APIs
by Dan Carpenter 10 Oct '17

10 Oct '17

Hello Jin Yao, The patch a33c1ec5cf82: "ASoC: Intel: Refactor common IPC/mailbox code into generic APIs" from Apr 7, 2015, leads to the following static checker warning: sound/soc/intel/common/sst-ipc.c:242 sst_ipc_reply_find_msg() error: uninitialized symbol 'mask'. sound/soc/intel/common/sst-ipc.c 226 struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc, 227 u64 header) 228 { 229 struct ipc_message *msg; 230 u64 mask; 231 232 if (ipc->ops.reply_msg_match != NULL) 233 header = ipc->ops.reply_msg_match(header, &mask); What should "mask" be initialized to if ops.reply_msg_match is NULL? 234 235 if (list_empty(&ipc->rx_list)) { 236 dev_err(ipc->dev, "error: rx list empty but received 0x%llx\n", 237 header); 238 return NULL; 239 } 240 241 list_for_each_entry(msg, &ipc->rx_list, list) { 242 if ((msg->header & mask) == header) ^^^^^^^^^^^^^^^^^^ Used uninitialized here. 243 return msg; 244 } 245 246 return NULL; 247 } regards, dan carpenter

1 0

[alsa-devel] [pull] amdgpu pre-reqs for stoney ACP audio support
by Alex Deucher 10 Oct '17

10 Oct '17

Hi Dave, Mark, This branch against Linus' master contains the 3 pre-requisite patches for Stoney ACP audio. Mark needs these patches to apply the rest of the Stoney ACP patch set on the audio side. Please pull. Thanks! The following changes since commit 68ebe3cbe786a954481d6ded0dc83f7d551640bf: Merge tag 'nfs-for-4.14-3' of git://git.linux-nfs.org/projects/trondmy/linux-nfs (2017-10-09 10:55:37 -0700) are available in the git repository at: git://people.freedesktop.org/~agd5f/linux linus-acp-prereq for you to fetch changes up to 1e4448648333a8fb7830a7a4b76c86e160a79b1e: drm/amdgpu Moving amdgpu asic types to a separate file (2017-10-09 17:54:24 -0400) ---------------------------------------------------------------- Akshu Agrawal (1): drm/amdgpu Moving amdgpu asic types to a separate file Vijendar Mukunda (2): drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data ASoC: AMD: Added asic_type as ACP DMA driver platform data drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 2 ++ drivers/gpu/drm/amd/include/amd_shared.h | 29 ++---------------- include/drm/amd_asic_type.h | 52 ++++++++++++++++++++++++++++++++ sound/soc/amd/acp-pcm-dma.c | 8 ++--- sound/soc/amd/acp.h | 7 +++++ 5 files changed, 65 insertions(+), 33 deletions(-) create mode 100644 include/drm/amd_asic_type.h

1 0

[alsa-devel] [PATCH v2] ALSA: seq: resize buffer for overflow
by Mark Salyzyn 09 Oct '17

09 Oct '17

Can not replicate, issue discovered in fuzzing. Stack trace below. No functional or performance testing done regarding the fix. Trap at (reformatted): snd_seq_oss_readq_puts(struct seq_oss_readq *q, int dev, unsigned char *data, int len) { union evrec rec; int result; memset(&rec, 0, sizeof(rec)); rec.c[0] = SEQ_MIDIPUTC; rec.c[2] = dev; while (len-- > 0) { rec.c[1] = *data++; // data is RBX HERE 'data' pointer just passed a page boundary, so the buffer supplied was short. Caller must have been handed an ev->type equal to SNDDRV_SEQ_EVENT_SYSEX, which resulted in handing off ev->data.ext.ptr[ev->data.ext.len] buffer. Intuited that the source of the event and buffer was referenced in snd_midi_event_encode_byte() passing a larger length than the allocated buffer. BUG: unable to handle kernel paging request at ffffc900008ab000 IP: [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 PGD 1da091067 PUD 1da092067 Oops: 0000 [#1] PREEMPT SMP KASAN Dumping ftrace buffer: (ftrace buffer empty) Modules linked in: CPU: 1 PID: 3264 Comm: XXXXXXXX Hardware name: XXXXXXXXXX task: ffff8801cdd9e000 task.stack: ffff8801ce648000 RIP: 0010:[<ffffffff82e2fc05>] [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 RSP: 0018:ffff8801ce64f1c0 EFLAGS: 00010246 RAX: 0000000000000000 RBX: ffffc900008ab000 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000001 RDI: ffffffff858e5780 RBP: ffff8801ce64f260 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 1ffff10039cc9df2 R12: 000000003fffffa4 R13: dffffc0000000000 R14: ffff8801ce64f238 R15: ffffc900008ab001 FS: 00007fe3d3d9e700(0000) GS:ffff8801db300000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: ffffc900008ab000 CR3: 00000001d19b7000 CR4: 00000000001406e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Stack: 1ffff10039cc9e3b ffff8801ce64f1f8 ffff8801d0b9aa00 0000000041b58ab3 ffffffff841daf3c ffffffff82e2fb30 0000000000000286 0000000000000005 ffffffff838aa5d5 ffffffff861962c0 dffffc0000000000 ffff8801ce64f260 Call Trace: [<ffffffff82e2ebbe>] send_midi_event sound/core/seq/oss/seq_oss_midi.c:616 [inline] [<ffffffff82e2ebbe>] snd_seq_oss_midi_input+0x8ce/0xa70 sound/core/seq/oss/seq_oss_midi.c:535 [<ffffffff82e2758d>] snd_seq_oss_event_input+0x15d/0x220 sound/core/seq/oss/seq_oss_event.c:439 [<ffffffff82e0b650>] snd_seq_deliver_single_event.constprop.11+0x310/0x7c0 sound/core/seq/seq_clientmgr.c:621 [<ffffffff82e0be16>] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:676 [inline] [<ffffffff82e0be16>] snd_seq_deliver_event+0x316/0x740 sound/core/seq/seq_clientmgr.c:807 [<ffffffff82e0cf9e>] snd_seq_kernel_client_dispatch+0x11e/0x150 sound/core/seq/seq_clientmgr.c:2314 [<ffffffff82e31775>] dummy_input+0x235/0x320 sound/core/seq/seq_dummy.c:104 [<ffffffff82e0b650>] snd_seq_deliver_single_event.constprop.11+0x310/0x7c0 sound/core/seq/seq_clientmgr.c:621 [<ffffffff82e0bc2d>] snd_seq_deliver_event+0x12d/0x740 sound/core/seq/seq_clientmgr.c:818 [<ffffffff82e0d0ed>] snd_seq_dispatch_event+0x11d/0x520 sound/core/seq/seq_clientmgr.c:892 [<ffffffff82e1117e>] snd_seq_check_queue.part.3+0x38e/0x510 sound/core/seq/seq_queue.c:285 [<ffffffff82e11d8d>] snd_seq_check_queue sound/core/seq/seq_queue.c:357 [inline] [<ffffffff82e11d8d>] snd_seq_enqueue_event+0x32d/0x3d0 sound/core/seq/seq_queue.c:363 [<ffffffff82e0c444>] snd_seq_client_enqueue_event+0x204/0x3e0 sound/core/seq/seq_clientmgr.c:951 [<ffffffff82e0cc55>] kernel_client_enqueue.part.10+0xb5/0xd0 sound/core/seq/seq_clientmgr.c:2251 [<ffffffff82e0cd3f>] kernel_client_enqueue sound/core/seq/seq_clientmgr.c:2241 [inline] [<ffffffff82e0cd3f>] snd_seq_kernel_client_enqueue_blocking+0xcf/0x110 sound/core/seq/seq_clientmgr.c:2279 [<ffffffff82e27f68>] insert_queue sound/core/seq/oss/seq_oss_rw.c:189 [inline] [<ffffffff82e27f68>] snd_seq_oss_write+0x538/0x850 sound/core/seq/oss/seq_oss_rw.c:148 [<ffffffff82e1fab4>] odev_write+0x64/0x90 sound/core/seq/oss/seq_oss.c:177 [<ffffffff8156ab43>] __vfs_write+0x103/0x680 fs/read_write.c:510 [<ffffffff8156ec70>] vfs_write+0x170/0x4e0 fs/read_write.c:560 [<ffffffff81572669>] SYSC_write fs/read_write.c:607 [inline] [<ffffffff81572669>] SyS_write+0xd9/0x1b0 fs/read_write.c:599 [<ffffffff838aad85>] entry_SYSCALL_64_fastpath+0x23/0xc6 Code: 4d 9a eb 4e e8 2d aa 53 fe 4c 8d 7b 01 48 89 d8 48 89 d9 48 c1 e8 03 83 e1 07 42 0f b6 04 28 38 c8 7f 08 84 c0 0f 85 80 00 00 00 <41> 0f b6 47 ff 41 83 ec 01 48 8b b5 68 ff ff ff 48 8b bd 70 ff RIP [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 RSP <ffff8801ce64f1c0> CR2: ffffc900008ab000 Signed-off-by: Mark Salyzyn <salyzyn(a)android.com> v2: removed nested locks in snd_midi_event_resize_buffer --- sound/core/seq/seq_midi_event.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index 90bbbdbeba03..ed3206ef0cd4 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -192,8 +192,7 @@ EXPORT_SYMBOL(snd_midi_event_no_status); /* * resize buffer */ -#if 0 -int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) +static int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) { unsigned char *new_buf, *old_buf; unsigned long flags; @@ -203,16 +202,13 @@ int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) new_buf = kmalloc(bufsize, GFP_KERNEL); if (new_buf == NULL) return -ENOMEM; - spin_lock_irqsave(&dev->lock, flags); old_buf = dev->buf; dev->buf = new_buf; dev->bufsize = bufsize; reset_encode(dev); - spin_unlock_irqrestore(&dev->lock, flags); kfree(old_buf); return 0; } -#endif /* 0 */ /* * read bytes and encode to sequencer event if finished @@ -297,6 +293,8 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, } else if (dev->type == ST_SYSEX) { if (c == MIDI_CMD_COMMON_SYSEX_END || dev->read >= dev->bufsize) { + if (dev->read > dev->bufsize) + snd_midi_event_resize_buffer(dev, dev->read); ev->flags &= ~SNDRV_SEQ_EVENT_LENGTH_MASK; ev->flags |= SNDRV_SEQ_EVENT_LENGTH_VARIABLE; ev->type = SNDRV_SEQ_EVENT_SYSEX; -- 2.14.2.920.gcf0c67979c-goog

2 2

[alsa-devel] usb/sound/line6: trying to register non-static key in podhd_disconnect
by Andrey Konovalov 09 Oct '17

09 Oct '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit e19b205be43d11bff638cad4487008c48d21c103 (4.14-rc2). INFO: trying to register non-static key. the code is fine but needs lockdep annotation. turning off the locking correctness validator. CPU: 1 PID: 1845 Comm: kworker/1:2 Not tainted 4.14.0-rc2-42613-g1488251d1a98 #238 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:16 dump_stack+0x292/0x395 lib/dump_stack.c:52 register_lock_class+0x6c4/0x1a00 kernel/locking/lockdep.c:769 __lock_acquire+0x27e/0x4550 kernel/locking/lockdep.c:3385 lock_acquire+0x259/0x620 kernel/locking/lockdep.c:4002 del_timer_sync+0x12c/0x280 kernel/time/timer.c:1237 podhd_disconnect+0x8c/0x160 sound/usb/line6/podhd.c:299 line6_probe+0x844/0x1310 sound/usb/line6/driver.c:783 podhd_probe+0x64/0x70 sound/usb/line6/podhd.c:474 usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_set_configuration+0x104e/0x1870 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_new_device+0x7b8/0x1020 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431

2 7

[alsa-devel] [PATCH] ASoC: Intel: cht_bsw_max98090: remove useless code, align with ChromeOS driver
by Pierre-Louis Bossart 09 Oct '17

09 Oct '17

Remove dead code which was missed in previous changes. This is not visible with git diff but there is a test+return on the same condition just above. Also reuse local variable instead of fetching the jack information twice. Tested on Acer R11 (cyan) Fixes: 3bbda5a38601 ('ASoC: Intel: cht_bsw_max98090_ti: Fix jack initialization') Cc: Thierry Escande <thierry.escande(a)collabora.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 74c047a4b05c..119da5503fee 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -219,10 +219,6 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } - - if (ctx->ts3a227e_present) - snd_soc_jack_notifier_register(jack, &cht_jack_nb); - ret = snd_soc_jack_add_gpiods(runtime->card->dev->parent, jack, ARRAY_SIZE(hs_jack_gpios), hs_jack_gpios); @@ -323,7 +319,7 @@ static int cht_max98090_headset_init(struct snd_soc_component *component) return ret; } - return ts3a227e_enable_jack_detect(component, &ctx->jack); + return ts3a227e_enable_jack_detect(component, jack); } static const struct snd_soc_ops cht_aif1_ops = { -- 2.11.0

3 2

[alsa-devel] [BUG] sound/core/seq: A possible sleep-in-atomic bug in snd_virmidi_dev_receive_event
by Jia-Ju Bai 09 Oct '17

09 Oct '17

According to seq_virmidi.c, the driver may sleep under a read spinlock. The function call path is: snd_virmidi_dev_receive_event (acquire the spinlock) snd_seq_dump_var_event copy_from_user --> may sleep This bug is found by my static analysis tool and my code review. Thanks, Jia-Ju Bai

2 1

[alsa-devel] Applied "ASoC: cygnus: Remove unnecessary active_slots check" to the asoc tree
by Mark Brown 09 Oct '17

09 Oct '17

The patch ASoC: cygnus: Remove unnecessary active_slots check has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4c75968a1bbfb3f190e2e624c83929451e4730ac Mon Sep 17 00:00:00 2001 From: Christos Gkekas <chris.gekas(a)gmail.com> Date: Sun, 8 Oct 2017 19:20:30 +0100 Subject: [PATCH] ASoC: cygnus: Remove unnecessary active_slots check Variable active_slots is unsigned so checking whether it is less than zero is not necessary. Signed-off-by: Christos Gkekas <chris.gekas(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/bcm/cygnus-ssp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index cd8aef8ed8a5..abafadc0b534 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -986,7 +986,7 @@ static int cygnus_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, active_slots = hweight32(tx_mask); - if ((active_slots < 0) || (active_slots > 16)) + if (active_slots > 16) return -EINVAL; /* Slot value must be even */ -- 2.14.1

1 0

[alsa-devel] [PATCH] ASoC: img-parallel-out: Check pm_runtime_get_sync return code
by Ed Blake 09 Oct '17

09 Oct '17

Check the the pm_runtime_get_sync return code and return if an error. Signed-off-by: Ed Blake <ed.blake(a)sondrel.com> --- sound/soc/img/img-parallel-out.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c index 2fc8a6372206..acc005217be0 100644 --- a/sound/soc/img/img-parallel-out.c +++ b/sound/soc/img/img-parallel-out.c @@ -153,6 +153,7 @@ static int img_prl_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct img_prl_out *prl = snd_soc_dai_get_drvdata(dai); u32 reg, control_set = 0; + int ret; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -164,7 +165,10 @@ static int img_prl_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - pm_runtime_get_sync(prl->dev); + ret = pm_runtime_get_sync(prl->dev); + if (ret < 0) + return ret; + reg = img_prl_out_readl(prl, IMG_PRL_OUT_CTL); reg = (reg & ~IMG_PRL_OUT_CTL_EDGE_MASK) | control_set; img_prl_out_writel(prl, reg, IMG_PRL_OUT_CTL); -- 2.11.0

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: img-i2s-in: Add power management
by Ed Blake 09 Oct '17

09 Oct '17

This patch set adds power managment to the img-i2s-in driver. The first patch adds system PM suspend / resume handling and the second patch adds runtime PM. Changes in v2: - Changed from ASoC specific pm callbacks to device ones - Errors returned from pm_runtime_get_sync are now handled Ed Blake (2): ASoC: img-i2s-in: Add suspend / resume handling ASoC: img-i2s-in: Add runtime PM sound/soc/img/img-i2s-in.c | 130 ++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 118 insertions(+), 12 deletions(-) -- 2.11.0

2 4

[alsa-devel] [PATCH v2 0/2] ASoC: img-i2s-out: Add power management
by Ed Blake 09 Oct '17

09 Oct '17

This patch set adds power managment to the img-i2s-out driver. The first patch adds system PM suspend / resume handling and the second patch adds control of the sys clock to runtime PM. Changes in v2: - Changed from ASoC specific pm callbacks to device ones - Errors returned from pm_runtime_get_sync are now handled Ed Blake (2): ASoC: img-i2s-out: Add suspend / resume handling ASoC: img-i2s-out: Add control of sys clock to runtime PM sound/soc/img/img-i2s-out.c | 108 ++++++++++++++++++++++++++++++++++---------- 1 file changed, 85 insertions(+), 23 deletions(-) -- 2.11.0

2 4

[alsa-devel] [PATCH v2 0/2] ASoC: img-spdif-in: Add power management
by Ed Blake 09 Oct '17

09 Oct '17

This patch set adds power managment to the img-spdif-in driver. The first patch adds system PM suspend / resume handling and the second patch adds runtime PM. Changes in v2: - Changed from ASoC specific pm callbacks to device ones Ed Blake (2): ASoC: img-spdif-in: Add suspend / resume handling ASoC: img-spdif-in: Add runtime PM sound/soc/img/img-spdif-in.c | 110 ++++++++++++++++++++++++++++++++++++++----- 1 file changed, 99 insertions(+), 11 deletions(-) -- 2.11.0

2 4

[alsa-devel] [PATCH v2 0/2] dma: img-spdif-out: Add power management
by Ed Blake 09 Oct '17

09 Oct '17

This patch set adds power managment to the img-spdif-out driver. The first patch adds system PM suspend / resume handling and the second patch adds control of the sys clock to runtime PM. Changes in v2: - Changed from ASoC specific pm callbacks to device ones Ed Blake (2): ASoC: img-spdif-out: Add suspend / resume handling ASoC: img-spdif-out: Add control of sys clock to runtime PM sound/soc/img/img-spdif-out.c | 73 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 60 insertions(+), 13 deletions(-) -- 2.11.0

2 3

[alsa-devel] [PATCH 0/3] dma: img-spdif-out: Add power management
by Ed Blake 09 Oct '17

09 Oct '17

This patch set adds power managment to the img-spdif-out driver. The first patch renames existing 'suspend' and 'resume' functions to 'runtime_suspend' and 'runtime_resume', which is what they actually are, the second patch adds system PM suspend / resume handling and the third patch adds control of the sys clock to runtime PM. Ed Blake (3): ASoC: img-spdif-out: Rename suspend / resume funcs ASoC: img-spdif-out: Add suspend / resume handling ASoC: img-spdif-out: Add control of sys clock to runtime PM sound/soc/img/img-spdif-out.c | 89 +++++++++++++++++++++++++++++++++---------- 1 file changed, 68 insertions(+), 21 deletions(-) -- 1.9.1

2 7

[alsa-devel] [PATCH] sound/soc: fix typos in Kconfig (trivial)
by Nik Nyby 09 Oct '17

09 Oct '17

This fixes a few typos in the snd_designware description. Signed-off-by: Nik Nyby <nikolas(a)gnu.org> --- sound/soc/dwc/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index c6fd95fa5ca6..aa0c6ec4d93c 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -4,8 +4,8 @@ config SND_DESIGNWARE_I2S select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for I2S driver for - Synopsys desigwnware I2S device. The device supports upto - maximum of 8 channels each for play and record. + Synopsys designware I2S device. The device supports up to + a maximum of 8 channels each for play and record. config SND_DESIGNWARE_PCM bool "PCM PIO extension for I2S driver" -- 2.14.2

2 2

[alsa-devel] [PATCH v3] ASoC: dapm: Avoid creating kcontrol for params
by yesanishhere＠gmail.com 09 Oct '17

09 Oct '17

From: anish kumar <yesanishhere(a)gmail.com> Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- Changes in this version: - Fixed the coding guideline comment from charles. Changes in v2: - Review comments from charles to move common function of error handling in seperate function. Changes in v1: - Included the review comments from charles regarding releasing memory in error path. sound/soc/soc-dapm.c | 140 +++++++++++++++++++++++++++++++-------------------- 1 file changed, 86 insertions(+), 54 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d55cac6..dbf3838 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3778,18 +3778,27 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, return 0; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +void +snd_soc_dapm_free_kcontrol(struct snd_soc_card *card, + unsigned long *private_value, + int num_params, + const char **w_param_text) +{ + int count; + + devm_kfree(card->dev, (void *)*private_value); + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); +} + +static struct snd_kcontrol_new * +snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card, + char *link_name, + const struct snd_soc_pcm_stream *params, + int num_params, const char **w_param_text, + unsigned long *private_value) { - struct snd_soc_dapm_widget template; - struct snd_soc_dapm_widget *w; - char *link_name; - int ret, count; - unsigned long private_value; - const char **w_param_text; struct soc_enum w_param_enum[] = { SOC_ENUM_SINGLE(0, 0, 0, NULL), }; @@ -3798,19 +3807,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, snd_soc_dapm_dai_link_get, snd_soc_dapm_dai_link_put), }; + struct snd_kcontrol_new *kcontrol_news; const struct snd_soc_pcm_stream *config = params; - - w_param_text = devm_kcalloc(card->dev, num_params, - sizeof(char *), GFP_KERNEL); - if (!w_param_text) - return -ENOMEM; - - link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", - source->name, sink->name); - if (!link_name) { - ret = -ENOMEM; - goto outfree_w_param; - } + int count; for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { @@ -3827,47 +3826,86 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, strlen(config->stream_name) + 1, GFP_KERNEL); } - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + if (!w_param_text[count]) + goto outfree_w_param; config++; } + w_param_enum[0].items = num_params; w_param_enum[0].texts = w_param_text; - memset(&template, 0, sizeof(template)); - template.reg = SND_SOC_NOPM; - template.id = snd_soc_dapm_dai_link; - template.name = link_name; - template.event = snd_soc_dai_link_event; - template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD; - template.num_kcontrols = 1; - /* duplicate w_param_enum on heap so that memory persists */ - private_value = + *private_value = (unsigned long) devm_kmemdup(card->dev, (void *)(kcontrol_dai_link[0].private_value), sizeof(struct soc_enum), GFP_KERNEL); - if (!private_value) { + if (!*private_value) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_link_name; + goto outfree_w_param; } - kcontrol_dai_link[0].private_value = private_value; + kcontrol_dai_link[0].private_value = *private_value; /* duplicate kcontrol_dai_link on heap so that memory persists */ - template.kcontrol_news = - devm_kmemdup(card->dev, &kcontrol_dai_link[0], + kcontrol_news = devm_kmemdup(card->dev, &kcontrol_dai_link[0], sizeof(struct snd_kcontrol_new), GFP_KERNEL); - if (!template.kcontrol_news) { + if (!kcontrol_news) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_private_value; + goto outfree_w_param; } + return kcontrol_news; +outfree_w_param: + snd_soc_dapm_free_kcontrol(card, private_value, num_params, w_param_text); + return NULL; +} + +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + const char **w_param_text; + unsigned long private_value; + char *link_name; + int ret; + + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); + if (!link_name) + return -ENOMEM; + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + template.kcontrol_news = NULL; + + /* allocate memory for control, only in case of multiple configs */ + if (num_params > 1) { + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) { + ret = -ENOMEM; + goto param_fail; + } + + template.num_kcontrols = 1; + template.kcontrol_news = + snd_soc_dapm_alloc_kcontrol(card, + link_name, params, num_params, + w_param_text, &private_value); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto param_fail; + } + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); @@ -3899,15 +3937,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); -outfree_private_value: - devm_kfree(card->dev, (void *)private_value); -outfree_link_name: + snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); +param_fail: devm_kfree(card->dev, link_name); - for (count = 0 ; count < num_params; count++) - devm_kfree(card->dev, (void *)w_param_text[count]); -outfree_w_param: - devm_kfree(card->dev, w_param_text); - return ret; } -- 2.5.4 (Apple Git-61)

4 5

[alsa-devel] [PATCH] sound/pci/riptide or drivers/base/firmware: Fix a possible sleep-in-atomic bug
by Jia-Ju Bai 09 Oct '17

09 Oct '17

The riptide driver may sleep under a spinlock, and the function call path is: snd_riptide_prepare (acquire the spinlock) setsampleformat sendcmd riptide_reset try_to_load_firmware request_firmware _request_firmware (drivers/base/firmware_class.c) _request_firmware_prepare kzalloc(GFP_KERNEL) --> may sleep To fix it, GFP_KERNEL is replaced with GFP_ATOMIC in _request_firmware_prepare. This bug is found by my static analysis tool and my code review. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)163.com> --- drivers/base/firmware_class.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/base/firmware_class.c b/drivers/base/firmware_class.c index 4b57cf5..d89f676 100644 --- a/drivers/base/firmware_class.c +++ b/drivers/base/firmware_class.c @@ -1138,7 +1138,7 @@ static inline void kill_pending_fw_fallback_reqs(bool only_kill_custom) { } struct firmware_buf *buf; int ret; - *firmware_p = firmware = kzalloc(sizeof(*firmware), GFP_KERNEL); + *firmware_p = firmware = kzalloc(sizeof(*firmware), GFP_ATOMIC); if (!firmware) { dev_err(device, "%s: kmalloc(struct firmware) failed\n", __func__); -- 1.7.9.5

2 2

[alsa-devel] [PATCH] opl3: Fix a possible sleep-in-atomic bug in snd_opl3_find_patch
by Jia-Ju Bai 09 Oct '17

09 Oct '17

The driver may sleep under a spinlock, and the function call path is: snd_opl3_note_on (acquire the spinlock) snd_opl3_find_patch kzalloc(GFP_KERNEL) --> may sleep To fix it, GFP_KERNEL is replaced with GFP_ATOMIC. This bug is found by my static analysis tool and my code review. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)163.com> --- sound/drivers/opl3/opl3_synth.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index ddcc1a3..2e1cb2b 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -325,7 +325,7 @@ struct fm_patch *snd_opl3_find_patch(struct snd_opl3 *opl3, int prog, int bank, if (!create_patch) return NULL; - patch = kzalloc(sizeof(*patch), GFP_KERNEL); + patch = kzalloc(sizeof(*patch), GFP_ATOMIC); if (!patch) return NULL; patch->prog = prog; -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: cygnus: Remove unnecessary active_slots check
by Christos Gkekas 08 Oct '17

08 Oct '17

Variable active_slots is unsigned so checking whether it is less than zero is not necessary. Signed-off-by: Christos Gkekas <chris.gekas(a)gmail.com> --- sound/soc/bcm/cygnus-ssp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index 15c438f..cb46128 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -998,7 +998,7 @@ static int cygnus_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, active_slots = hweight32(tx_mask); - if ((active_slots < 0) || (active_slots > 16)) + if (active_slots > 16) return -EINVAL; /* Slot value must be even */ -- 2.7.4

1 0

[alsa-devel] [BUG] haswell: A possible sleep-in-atomic bug in hsw_irq_thread
by Jia-Ju Bai 08 Oct '17

08 Oct '17

According to sst-haswell-ipc.c, the driver may sleep under a spinlock, and the function call path is: hsw_irq_thread (acquire the spinlock) hsw_process_notification hsw_log_message mutex_lock --> may sleep This bug is found by my static analysis tool and my code review. Thanks, Jia-Ju Bai

1 0

[alsa-devel] [PATCH] ALSA: seq: resize buffer for overflow
by Mark Salyzyn 06 Oct '17

06 Oct '17

Can not replicate, issue discovered in fuzzing. Stack trace below. No functional or performance testing done regarding the fix. Trap at (reformatted): snd_seq_oss_readq_puts(struct seq_oss_readq *q, int dev, unsigned char *data, int len) { union evrec rec; int result; memset(&rec, 0, sizeof(rec)); rec.c[0] = SEQ_MIDIPUTC; rec.c[2] = dev; while (len-- > 0) { rec.c[1] = *data++; // data is RBX HERE 'data' pointer just passed a page boundary, so the buffer supplied was short. Caller must have been handed an ev->type equal to SNDDRV_SEQ_EVENT_SYSEX, which resulted in handing off ev->data.ext.ptr[ev->data.ext.len] buffer. Intuited that the source of the event and buffer was referenced in snd_midi_event_encode_byte() passing a larger length than the allocated buffer. BUG: unable to handle kernel paging request at ffffc900008ab000 IP: [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 PGD 1da091067 PUD 1da092067 Oops: 0000 [#1] PREEMPT SMP KASAN Dumping ftrace buffer: (ftrace buffer empty) Modules linked in: CPU: 1 PID: 3264 Comm: XXXXXXXX Hardware name: XXXXXXXXXX task: ffff8801cdd9e000 task.stack: ffff8801ce648000 RIP: 0010:[<ffffffff82e2fc05>] [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 RSP: 0018:ffff8801ce64f1c0 EFLAGS: 00010246 RAX: 0000000000000000 RBX: ffffc900008ab000 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000001 RDI: ffffffff858e5780 RBP: ffff8801ce64f260 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 1ffff10039cc9df2 R12: 000000003fffffa4 R13: dffffc0000000000 R14: ffff8801ce64f238 R15: ffffc900008ab001 FS: 00007fe3d3d9e700(0000) GS:ffff8801db300000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: ffffc900008ab000 CR3: 00000001d19b7000 CR4: 00000000001406e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Stack: 1ffff10039cc9e3b ffff8801ce64f1f8 ffff8801d0b9aa00 0000000041b58ab3 ffffffff841daf3c ffffffff82e2fb30 0000000000000286 0000000000000005 ffffffff838aa5d5 ffffffff861962c0 dffffc0000000000 ffff8801ce64f260 Call Trace: [<ffffffff82e2ebbe>] send_midi_event sound/core/seq/oss/seq_oss_midi.c:616 [inline] [<ffffffff82e2ebbe>] snd_seq_oss_midi_input+0x8ce/0xa70 sound/core/seq/oss/seq_oss_midi.c:535 [<ffffffff82e2758d>] snd_seq_oss_event_input+0x15d/0x220 sound/core/seq/oss/seq_oss_event.c:439 [<ffffffff82e0b650>] snd_seq_deliver_single_event.constprop.11+0x310/0x7c0 sound/core/seq/seq_clientmgr.c:621 [<ffffffff82e0be16>] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:676 [inline] [<ffffffff82e0be16>] snd_seq_deliver_event+0x316/0x740 sound/core/seq/seq_clientmgr.c:807 [<ffffffff82e0cf9e>] snd_seq_kernel_client_dispatch+0x11e/0x150 sound/core/seq/seq_clientmgr.c:2314 [<ffffffff82e31775>] dummy_input+0x235/0x320 sound/core/seq/seq_dummy.c:104 [<ffffffff82e0b650>] snd_seq_deliver_single_event.constprop.11+0x310/0x7c0 sound/core/seq/seq_clientmgr.c:621 [<ffffffff82e0bc2d>] snd_seq_deliver_event+0x12d/0x740 sound/core/seq/seq_clientmgr.c:818 [<ffffffff82e0d0ed>] snd_seq_dispatch_event+0x11d/0x520 sound/core/seq/seq_clientmgr.c:892 [<ffffffff82e1117e>] snd_seq_check_queue.part.3+0x38e/0x510 sound/core/seq/seq_queue.c:285 [<ffffffff82e11d8d>] snd_seq_check_queue sound/core/seq/seq_queue.c:357 [inline] [<ffffffff82e11d8d>] snd_seq_enqueue_event+0x32d/0x3d0 sound/core/seq/seq_queue.c:363 [<ffffffff82e0c444>] snd_seq_client_enqueue_event+0x204/0x3e0 sound/core/seq/seq_clientmgr.c:951 [<ffffffff82e0cc55>] kernel_client_enqueue.part.10+0xb5/0xd0 sound/core/seq/seq_clientmgr.c:2251 [<ffffffff82e0cd3f>] kernel_client_enqueue sound/core/seq/seq_clientmgr.c:2241 [inline] [<ffffffff82e0cd3f>] snd_seq_kernel_client_enqueue_blocking+0xcf/0x110 sound/core/seq/seq_clientmgr.c:2279 [<ffffffff82e27f68>] insert_queue sound/core/seq/oss/seq_oss_rw.c:189 [inline] [<ffffffff82e27f68>] snd_seq_oss_write+0x538/0x850 sound/core/seq/oss/seq_oss_rw.c:148 [<ffffffff82e1fab4>] odev_write+0x64/0x90 sound/core/seq/oss/seq_oss.c:177 [<ffffffff8156ab43>] __vfs_write+0x103/0x680 fs/read_write.c:510 [<ffffffff8156ec70>] vfs_write+0x170/0x4e0 fs/read_write.c:560 [<ffffffff81572669>] SYSC_write fs/read_write.c:607 [inline] [<ffffffff81572669>] SyS_write+0xd9/0x1b0 fs/read_write.c:599 [<ffffffff838aad85>] entry_SYSCALL_64_fastpath+0x23/0xc6 Code: 4d 9a eb 4e e8 2d aa 53 fe 4c 8d 7b 01 48 89 d8 48 89 d9 48 c1 e8 03 83 e1 07 42 0f b6 04 28 38 c8 7f 08 84 c0 0f 85 80 00 00 00 <41> 0f b6 47 ff 41 83 ec 01 48 8b b5 68 ff ff ff 48 8b bd 70 ff RIP [<ffffffff82e2fc05>] snd_seq_oss_readq_puts+0xd5/0x170 sound/core/seq/oss/seq_oss_readq.c:112 RSP <ffff8801ce64f1c0> CR2: ffffc900008ab000 Signed-off-by: Mark Salyzyn <salyzyn(a)android.com> --- sound/core/seq/seq_midi_event.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c index 90bbbdbeba03..cbc501fd47c2 100644 --- a/sound/core/seq/seq_midi_event.c +++ b/sound/core/seq/seq_midi_event.c @@ -192,8 +192,7 @@ EXPORT_SYMBOL(snd_midi_event_no_status); /* * resize buffer */ -#if 0 -int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) +static int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) { unsigned char *new_buf, *old_buf; unsigned long flags; @@ -212,7 +211,6 @@ int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize) kfree(old_buf); return 0; } -#endif /* 0 */ /* * read bytes and encode to sequencer event if finished @@ -297,6 +295,8 @@ int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, } else if (dev->type == ST_SYSEX) { if (c == MIDI_CMD_COMMON_SYSEX_END || dev->read >= dev->bufsize) { + if (dev->read > dev->bufsize) + snd_midi_event_resize_buffer(dev, dev->read); ev->flags &= ~SNDRV_SEQ_EVENT_LENGTH_MASK; ev->flags |= SNDRV_SEQ_EVENT_LENGTH_VARIABLE; ev->type = SNDRV_SEQ_EVENT_SYSEX; -- 2.14.2.920.gcf0c67979c-goog

1 0

Re: [alsa-devel] [PATCH] alsactl: Move systemd unit start-up from basic.target to sound.target
by Takashi Iwai 06 Oct '17

06 Oct '17

On Tue, 26 Sep 2017 20:36:12 +0200, Chris Mayo wrote: > > Ensures soundcard is ready before restoring state. > > sound.target added to systemd in v18: > https://cgit.freedesktop.org/systemd/systemd/commit/?id=88dfa2938af > > Simplify dependencies: > - After=alsa-state.service is not needed because both units test for > @daemonswitch@ with opposite outcomes. > > - After=sysinit.target is automatically added by systemd. > > First proposed by Tom Yan. > > Signed-off-by: Chris Mayo <aklhfex(a)gmail.com> > --- > > I need this because my Sound Blaster Titanium is not ready by the time > basic.target is reached. > > It was proposed last year by Tom Yan but looks like it didn't get in > because requested config checks and documentation didn't materialise. > https://www.spinics.net/lists/alsa-devel/msg45537.html > > I've added those (PKG_CHECK_MODULES outputs a message at configure time). > > I've left ConditionPathExistsGlob=/dev/snd/control* in > alsa-restore.service because although it would normally be started by the > udev rule and sound.target there's no guarantee that is how it was started. Thanks, this looks like a sensitive change. Applied now. Takashi

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Add sample rate quirk for Plantronics P610
by Kai-Heng Feng 06 Oct '17

06 Oct '17

Like other Plantronics devices, P610 does not support sample rate reading. Apply sample rate quirk to it. BugLink: https://bugs.launchpad.net/bugs/1719853 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b8cb57aeec77..9ddaae3784f5 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1138,6 +1138,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x047F, 0xC022): /* Plantronics C310 */ + case USB_ID(0x047F, 0xC02F): /* Plantronics P610 */ case USB_ID(0x047F, 0xC036): /* Plantronics C520-M */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ -- 2.14.1

2 3

[alsa-devel] [asoc:topic/dapm 4/4] sound//soc/soc-dapm.c:3940:2: warning: 'w_param_text' may be used uninitialized in this function
by kbuild test robot 05 Oct '17

05 Oct '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/dapm head: 19ad683abc8534cd755403381c73c0ddce7b9f6d commit: 19ad683abc8534cd755403381c73c0ddce7b9f6d [4/4] ASoC: dapm: Avoid creating kcontrol for params config: i386-randconfig-x079-201740 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 19ad683abc8534cd755403381c73c0ddce7b9f6d # save the attached .config to linux build tree make ARCH=i386 Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): sound//soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': >> sound//soc/soc-dapm.c:3940:2: warning: 'w_param_text' may be used uninitialized in this function [-Wmaybe-uninitialized] snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vim +/w_param_text +3940 sound//soc/soc-dapm.c 3862 3863 int snd_soc_dapm_new_pcm(struct snd_soc_card *card, 3864 const struct snd_soc_pcm_stream *params, 3865 unsigned int num_params, 3866 struct snd_soc_dapm_widget *source, 3867 struct snd_soc_dapm_widget *sink) 3868 { 3869 struct snd_soc_dapm_widget template; 3870 struct snd_soc_dapm_widget *w; 3871 const char **w_param_text; 3872 unsigned long private_value; 3873 char *link_name; 3874 int ret; 3875 3876 link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", 3877 source->name, sink->name); 3878 if (!link_name) 3879 return -ENOMEM; 3880 3881 memset(&template, 0, sizeof(template)); 3882 template.reg = SND_SOC_NOPM; 3883 template.id = snd_soc_dapm_dai_link; 3884 template.name = link_name; 3885 template.event = snd_soc_dai_link_event; 3886 template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | 3887 SND_SOC_DAPM_PRE_PMD; 3888 template.kcontrol_news = NULL; 3889 3890 /* allocate memory for control, only in case of multiple configs */ 3891 if (num_params > 1) { 3892 w_param_text = devm_kcalloc(card->dev, num_params, 3893 sizeof(char *), GFP_KERNEL); 3894 if (!w_param_text) { 3895 ret = -ENOMEM; 3896 goto param_fail; 3897 } 3898 3899 template.num_kcontrols = 1; 3900 template.kcontrol_news = 3901 snd_soc_dapm_alloc_kcontrol(card, 3902 link_name, params, num_params, 3903 w_param_text, &private_value); 3904 if (!template.kcontrol_news) { 3905 ret = -ENOMEM; 3906 goto param_fail; 3907 } 3908 } 3909 dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); 3910 3911 w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); 3912 if (IS_ERR(w)) { 3913 ret = PTR_ERR(w); 3914 /* Do not nag about probe deferrals */ 3915 if (ret != -EPROBE_DEFER) 3916 dev_err(card->dev, 3917 "ASoC: Failed to create %s widget (%d)\n", 3918 link_name, ret); 3919 goto outfree_kcontrol_news; 3920 } 3921 if (!w) { 3922 dev_err(card->dev, "ASoC: Failed to create %s widget\n", 3923 link_name); 3924 ret = -ENOMEM; 3925 goto outfree_kcontrol_news; 3926 } 3927 3928 w->params = params; 3929 w->num_params = num_params; 3930 3931 ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); 3932 if (ret) 3933 goto outfree_w; 3934 return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); 3935 3936 outfree_w: 3937 devm_kfree(card->dev, w); 3938 outfree_kcontrol_news: 3939 devm_kfree(card->dev, (void *)template.kcontrol_news); > 3940 snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); 3941 param_fail: 3942 devm_kfree(card->dev, link_name); 3943 return ret; 3944 } 3945 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 00/18] use ARRAY_SIZE macro
by Jérémy Lefaure 05 Oct '17

05 Oct '17

Hi everyone, Using ARRAY_SIZE improves the code readability. I used coccinelle (I made a change to the array_size.cocci file [1]) to find several places where ARRAY_SIZE could be used instead of other macros or sizeof division. I tried to divide the changes into a patch per subsystem (excepted for staging). If one of the patch should be split into several patches, let me know. In order to reduce the size of the To: and Cc: lines, each patch of the series is sent only to the maintainers and lists concerned by the patch. This cover letter is sent to every list concerned by this series. This series is based on linux-next next-20170929. Each patch has been tested by building the relevant files with W=1. This series contains the following patches: [PATCH 01/18] sound: use ARRAY_SIZE [PATCH 02/18] tracing/filter: use ARRAY_SIZE [PATCH 03/18] media: use ARRAY_SIZE [PATCH 04/18] IB/mlx5: Use ARRAY_SIZE [PATCH 05/18] net: use ARRAY_SIZE [PATCH 06/18] drm: use ARRAY_SIZE [PATCH 07/18] scsi: bfa: use ARRAY_SIZE [PATCH 08/18] ecryptfs: use ARRAY_SIZE [PATCH 09/18] nfsd: use ARRAY_SIZE [PATCH 10/18] orangefs: use ARRAY_SIZE [PATCH 11/18] dm space map metadata: use ARRAY_SIZE [PATCH 12/18] x86: use ARRAY_SIZE [PATCH 13/18] tpm: use ARRAY_SIZE [PATCH 14/18] ipmi: use ARRAY_SIZE [PATCH 15/18] acpi: use ARRAY_SIZE [PATCH 16/18] media: staging: atomisp: use ARRAY_SIZE [PATCH 17/18] staging: rtl8723bs: use ARRAY_SIZE [PATCH 18/18] staging: rtlwifi: use ARRAY_SIZE [1]: https://lkml.org/lkml/2017/9/13/689

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[alsa-devel] [PATCH] ALSA: sh: aica: Convert timers to use timer_setup()
by Kees Cook 05 Oct '17

05 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. This requires adding a pointer to hold the timer's target substream, as there won't be a way to pass this in the future. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Bhumika Goyal <bhumirks(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Kees Cook <keescook(a)chromium.org> --- This requires commit 686fef928bba ("timer: Prepare to change timer callback argument type") in v4.14-rc3, but should be otherwise stand-alone. --- sound/sh/aica.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index fdc680ae8aa0..2b26311405a4 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -299,14 +299,14 @@ static void run_spu_dma(struct work_struct *work) } } -static void aica_period_elapsed(unsigned long timer_var) +static void aica_period_elapsed(struct timer_list *t) { + struct snd_card_aica *dreamcastcard = from_timer(dreamcastcard, + t, timer); + struct snd_pcm_substream *substream = dreamcastcard->timer_substream; /*timer function - so cannot sleep */ int play_period; struct snd_pcm_runtime *runtime; - struct snd_pcm_substream *substream; - struct snd_card_aica *dreamcastcard; - substream = (struct snd_pcm_substream *) timer_var; runtime = substream->runtime; dreamcastcard = substream->pcm->private_data; /* Have we played out an additional period? */ @@ -336,12 +336,12 @@ static void spu_begin_dma(struct snd_pcm_substream *substream) /*get the queue to do the work */ schedule_work(&(dreamcastcard->spu_dma_work)); /* Timer may already be running */ - if (unlikely(dreamcastcard->timer.data)) { + if (unlikely(dreamcastcard->timer_substream)) { mod_timer(&dreamcastcard->timer, jiffies + 4); return; } - setup_timer(&dreamcastcard->timer, aica_period_elapsed, - (unsigned long) substream); + timer_setup(&dreamcastcard->timer, aica_period_elapsed, 0); + dreamcastcard->timer_substream = substream; mod_timer(&dreamcastcard->timer, jiffies + 4); } @@ -379,7 +379,7 @@ static int snd_aicapcm_pcm_close(struct snd_pcm_substream { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; flush_work(&(dreamcastcard->spu_dma_work)); - if (dreamcastcard->timer.data) + if (dreamcastcard->timer_substream) del_timer(&dreamcastcard->timer); kfree(dreamcastcard->channel); spu_disable(); @@ -600,7 +600,7 @@ static int snd_aica_probe(struct platform_device *devptr) { int err; struct snd_card_aica *dreamcastcard; - dreamcastcard = kmalloc(sizeof(struct snd_card_aica), GFP_KERNEL); + dreamcastcard = kzalloc(sizeof(struct snd_card_aica), GFP_KERNEL); if (unlikely(!dreamcastcard)) return -ENOMEM; err = snd_card_new(&devptr->dev, index, SND_AICA_DRIVER, @@ -619,8 +619,6 @@ static int snd_aica_probe(struct platform_device *devptr) err = snd_aicapcmchip(dreamcastcard, 0); if (unlikely(err < 0)) goto freedreamcast; - dreamcastcard->timer.data = 0; - dreamcastcard->channel = NULL; /* Add basic controls */ err = add_aicamixer_controls(dreamcastcard); if (unlikely(err < 0)) -- 2.7.4 -- Kees Cook Pixel Security

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[alsa-devel] [PATCH 0/2] ASoC: img-i2s-in: Add power management
by Ed Blake 05 Oct '17

05 Oct '17

This patch set adds power managment to the img-i2s-in driver. The first patch adds system PM suspend / resume handling and the second patch adds runtime PM. Ed Blake (2): ASoC: img-i2s-in: Add suspend / resume handling ASoC: img-i2s-in: Add runtime PM sound/soc/img/img-i2s-in.c | 126 ++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 114 insertions(+), 12 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] conf: HdmiLpeAudio: remove the "front" pcm definition
by Tanu Kaskinen 05 Oct '17

05 Oct '17

PulseAudio assumes that the "front" pcm device always refers to an analog device, not HDMI. While that assumption is not really valid, the reality is that without that assumption PulseAudio can't know whether "front" and "hdmi" refer to a different or the same device. The HDMI LPE driver doesn't allow audio streaming while the HDMI cable is unplugged, so PulseAudio has to know when it's plugged in and when it's not. If both "front" and "hdmi" devices exist, PulseAudio will notice that HDMI is unplugged, but it doesn't know that "front" refers to the same device, and PulseAudio will try to use the "front" device with bad consequences. The kernel driver's refusal to stream any audio makes PulseAudio enter an infinite loop and then the kernel kills PulseAudio, because it consumes too much CPU time in a realtime thread. While the looping in PulseAudio could probably be fixed, that wouldn't change the fact that PulseAudio thinks that there is an analog device. I believe it's best to avoid having the same device as both "front" and "hdmi" in alsa-lib. I removed also the surround configuration includes. I don't think they had any effect anyway, so I wonder why they were there in the first place. BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=100488 --- src/conf/cards/HdmiLpeAudio.conf | 24 ------------------------ 1 file changed, 24 deletions(-) diff --git a/src/conf/cards/HdmiLpeAudio.conf b/src/conf/cards/HdmiLpeAudio.conf index 9fa30da0..a1e493da 100644 --- a/src/conf/cards/HdmiLpeAudio.conf +++ b/src/conf/cards/HdmiLpeAudio.conf @@ -2,30 +2,6 @@ # Configuration for the Intel HDMI/DP LPE audio # -<confdir:pcm/front.conf> - -HdmiLpeAudio.pcm.front.0 { - @args [ CARD ] - @args.CARD { - type string - } - type softvol - slave.pcm { - type hw - card $CARD - } - control { - name "PCM Playback Volume" - card $CARD - } -} - -<confdir:pcm/surround40.conf> -<confdir:pcm/surround21.conf> -<confdir:pcm/surround41.conf> -<confdir:pcm/surround50.conf> -<confdir:pcm/surround51.conf> - <confdir:pcm/hdmi.conf> HdmiLpeAudio.pcm.hdmi.0 { -- 2.14.2

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[alsa-devel] SEGFault when optininal snd_ctl_ext_callback::read_event() not set
by Wischer, Timo (ADITG/ESB) 05 Oct '17

05 Oct '17

Hi all, snd_ctl_ext_callback::read_event() callback is mentioned as optional (see http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=include/control_extern…) but there is no NULL check and the NULL pointer will be called if the read_event function callback pointer is not set. (see http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/control/control_ex…) I think a default function has to be provided which will be called when the callback is not set or the read_event() callback should not be marked as optional. What is your opinion? Best regards Timo Wischer Advanced Driver Information Technology GmbH Engineering Software Base (ADITG/ESB) Robert-Bosch-Str. 200 31139 Hildesheim Germany Tel. +49 5121 49 6938 Fax +49 5121 49 6999 twischer(a)de.adit-jv.com ADIT is a joint venture company of Robert Bosch GmbH/Robert Bosch Car Multimedia GmbH and DENSO Corporation Sitz: Hildesheim, Registergericht: Amtsgericht Hildesheim HRB 3438 Geschäftsführung: Wilhelm Grabow, Ken Yaguchi

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[alsa-devel] notifying userspace of hwparams change
by Tim Harvey 05 Oct '17

05 Oct '17

Greetings, I've got an HDMI receiver chip I'm working on a driver for which provides the audio signal from the HDMI source. An HDMI source can be a variety of sample-rates, channels, widths, and formats. If I understand correctly I need to register the device as supporting all the possible rates/channels/formats then constrain these to what is actually available at the time in the startup callback with snd_pcm_hw_contstraint_minmax and snd_pcm_hw_constraint_mask. How do I notify userspace when the chip tells me that the HDMI audio source rate/channels/format has changed? I know on the Video side the driver sends a V4L2_EVENT_SOURCE_CHANGE event to userspace when input format/resolution has changed so that userspace can stop streaming, call VIDIOC_QUERY_DV_TIMINGS and then VIDIOC_S_DV_TIMINGS with new timings. Regards, Tim

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[alsa-devel] [PATCH] ALSA: core: Add new ACCESS_TLV flag for coefficient TLV controls
by Charles Keepax 05 Oct '17

05 Oct '17

ASoC can use the TLV mechanism to pass coefficient data for controls that exceed the normal 512 byte limit. However, there is no mechanism to inform user-space the control expects data through the TLV mechanism. Currently amixer uses a combination of the absense of non-TLV read/write permissions and the type being BYTES. This is not ideal as it is not a clear signal and prevents controls that might support both access mechanisms. Add a specific flag SNDRV_CTL_ELEM_ACCESS_TLV_COEFF to let user-space know when it is dealing with a TLV control that is being used to pass actual coefficient data. Then add this flag to the controls generated by the current users of these types of controls. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Hi, This builds on work done by Sakamoto-san [1] and the changes mentioned to amixer in the commit message are found in [2]. I think something like this should probably be added, what I am less clear on is where we go from here? Sakamoto-san's original patches also added a tag type SNDRV_CTL_TVLT_COEFF, to keep the data matching the normal format found in TLV controls. The trouble is neither of the two users really follow this convention at the moment. The ADSP code just takes the data as is with no additional headers and the Skylake code does have a tag and a length as part of the data passed through the TLV however the tag part looks like it is used for some internal purpose, so I am far from clear it would support a fixed value to identify the data without some changes as well. Changing the ADSP code is probably acceptable since we have had very few users in the field, and since this work has been done improvements in the firmware tool chains mean most firmwares are created with much smaller dedicated purpose controls rather than the large block ones that were initially the reason for using these controls on our side. Finally, since [3] tinymix has been updated to add some TLV information into control writes/reads which will break the ADSP code once it trickles down to users. So I would be keen to discuss this at this years conference and decide how we are going to handle these controls? As Mark says [4] maybe we should see if there is a better approach we should all migrate across to? Thanks, Charles [1] https://patchwork.kernel.org/patch/9304649/ [2] https://patchwork.kernel.org/patch/8147001/ [3] https://github.com/tinyalsa/tinyalsa/commit/3f813e47784674a3909fe1277bc10b7… [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-September/125743.… include/sound/soc.h | 3 ++- include/uapi/sound/asound.h | 1 + sound/core/control.c | 1 + sound/soc/codecs/wm_adsp.c | 3 ++- sound/soc/intel/skylake/skl-topology.c | 1 + sound/soc/soc-ops.c | 3 +++ 6 files changed, 10 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d776cdee30d7..d82297a95caf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -323,7 +323,8 @@ #define SND_SOC_BYTES_TLV(xname, xcount, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK | \ + SNDRV_CTL_ELEM_ACCESS_TLV_COEFF, \ .tlv.c = (snd_soc_bytes_tlv_callback), \ .info = snd_soc_bytes_info_ext, \ .private_value = (unsigned long)&(struct soc_bytes_ext) \ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1949923a40bf..4bf0b2f835e9 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -853,6 +853,7 @@ typedef int __bitwise snd_ctl_elem_iface_t; #define SNDRV_CTL_ELEM_ACCESS_TLV_WRITE (1<<5) /* TLV write is possible */ #define SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE (SNDRV_CTL_ELEM_ACCESS_TLV_READ|SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) #define SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND (1<<6) /* TLV command is possible */ +#define SNDRV_CTL_ELEM_ACCESS_TLV_COEFF (1<<7) /* Coefficient data is accepted */ #define SNDRV_CTL_ELEM_ACCESS_INACTIVE (1<<8) /* control does actually nothing, but may be updated */ #define SNDRV_CTL_ELEM_ACCESS_LOCK (1<<9) /* write lock */ #define SNDRV_CTL_ELEM_ACCESS_OWNER (1<<10) /* write lock owner */ diff --git a/sound/core/control.c b/sound/core/control.c index 56b3e2d49c82..e7a74d48633b 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -266,6 +266,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, SNDRV_CTL_ELEM_ACCESS_INACTIVE | SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND | + SNDRV_CTL_ELEM_ACCESS_TLV_COEFF | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK); err = snd_ctl_new(&kctl, count, access, NULL); diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 65c059b5ffd7..1d7fe0d8a18a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1156,7 +1156,8 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) wr = SNDRV_CTL_ELEM_ACCESS_TLV_WRITE; vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; - out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK | + SNDRV_CTL_ELEM_ACCESS_TLV_COEFF; } else { rd = SNDRV_CTL_ELEM_ACCESS_READ; wr = SNDRV_CTL_ELEM_ACCESS_WRITE; diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 27bcb62568fb..01b01c6a2bf3 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2870,6 +2870,7 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, tplg_bc = container_of(hdr, struct snd_soc_tplg_bytes_control, hdr); if (kctl->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + kctl->access |= SNDRV_CTL_ELEM_ACCESS_TLV_COEFF; sb = (struct soc_bytes_ext *)kctl->private_value; if (tplg_bc->priv.size) return skl_init_algo_data( diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 500f98c730b9..5ac2afe12b5d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -776,6 +776,9 @@ int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, unsigned int count = size < params->max ? size : params->max; int ret = -ENXIO; + if (!(kcontrol->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_COEFF)) + return -ENXIO; + switch (op_flag) { case SNDRV_CTL_TLV_OP_READ: if (params->get) -- 2.11.0

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[alsa-devel] [PATCH] sound/pci/asihpi: Convert timers to use timer_setup()
by Kees Cook 05 Oct '17

05 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Kees Cook <keescook(a)chromium.org> --- This requires commit 686fef928bba ("timer: Prepare to change timer callback argument type") in v4.14-rc3, but should be otherwise stand-alone. --- sound/pci/asihpi/asihpi.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 70d023a85bf5..614322171c79 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -749,9 +749,9 @@ static inline unsigned int modulo_min(unsigned int a, unsigned int b, /** Timer function, equivalent to interrupt service routine for cards */ -static void snd_card_asihpi_timer_function(unsigned long data) +static void snd_card_asihpi_timer_function(struct timer_list *t) { - struct snd_card_asihpi_pcm *dpcm = (struct snd_card_asihpi_pcm *)data; + struct snd_card_asihpi_pcm *dpcm = from_timer(dpcm, t, timer); struct snd_pcm_substream *substream = dpcm->substream; struct snd_card_asihpi *card = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime; @@ -948,7 +948,7 @@ static void snd_card_asihpi_int_task(unsigned long data) asihpi = (struct snd_card_asihpi *)a->snd_card->private_data; if (asihpi->llmode_streampriv) snd_card_asihpi_timer_function( - (unsigned long)asihpi->llmode_streampriv); + &asihpi->llmode_streampriv->timer); } static void snd_card_asihpi_isr(struct hpi_adapter *a) @@ -1059,8 +1059,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) If internal and other stream playing, can't switch */ - setup_timer(&dpcm->timer, snd_card_asihpi_timer_function, - (unsigned long) dpcm); + timer_setup(&dpcm->timer, snd_card_asihpi_timer_function, 0); dpcm->substream = substream; runtime->private_data = dpcm; runtime->private_free = snd_card_asihpi_runtime_free; @@ -1240,8 +1239,7 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) if (err) return -EIO; - setup_timer(&dpcm->timer, snd_card_asihpi_timer_function, - (unsigned long) dpcm); + timer_setup(&dpcm->timer, snd_card_asihpi_timer_function, 0); dpcm->substream = substream; runtime->private_data = dpcm; runtime->private_free = snd_card_asihpi_runtime_free; -- 2.7.4 -- Kees Cook Pixel Security

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[alsa-devel] [PATCH] sound/core: Convert timers to use timer_setup()
by Kees Cook 05 Oct '17

05 Oct '17

In preparation for unconditionally passing the struct timer_list pointer to all timer callbacks, switch to using the new timer_setup() and from_timer() to pass the timer pointer explicitly. This adds a pointer back to struct snd_timer. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Ingo Molnar <mingo(a)kernel.org> Cc: Dan Carpenter <dan.carpenter(a)oracle.com> Cc: alsa-devel(a)alsa-project.org Cc: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Kees Cook <keescook(a)chromium.org> --- This requires commit 686fef928bba ("timer: Prepare to change timer callback argument type") in v4.14-rc3, but should be otherwise stand-alone. --- sound/core/timer.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 6cdd04a45962..09acaf2b2e57 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1028,15 +1028,17 @@ EXPORT_SYMBOL(snd_timer_global_register); struct snd_timer_system_private { struct timer_list tlist; + struct snd_timer *snd_timer; unsigned long last_expires; unsigned long last_jiffies; unsigned long correction; }; -static void snd_timer_s_function(unsigned long data) +static void snd_timer_s_function(struct timer_list *t) { - struct snd_timer *timer = (struct snd_timer *)data; - struct snd_timer_system_private *priv = timer->private_data; + struct snd_timer_system_private *priv = from_timer(priv, t, + tlist); + struct snd_timer *timer = priv->snd_timer; unsigned long jiff = jiffies; if (time_after(jiff, priv->last_expires)) priv->correction += (long)jiff - (long)priv->last_expires; @@ -1118,7 +1120,8 @@ static int snd_timer_register_system(void) snd_timer_free(timer); return -ENOMEM; } - setup_timer(&priv->tlist, snd_timer_s_function, (unsigned long) timer); + priv->snd_timer = timer; + timer_setup(&priv->tlist, snd_timer_s_function, 0); timer->private_data = priv; timer->private_free = snd_timer_free_system; return snd_timer_global_register(timer); -- 2.7.4 -- Kees Cook Pixel Security

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[alsa-devel] [PATCH] ASoC: fsi: Use of_device_get_match_data() helper
by Geert Uytterhoeven 04 Oct '17

04 Oct '17

Use the of_device_get_match_data() helper instead of open coding. Note that when used with DT, there's always a valid match. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/sh/fsi.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 6d3c7706d93fbc10..c3aaf4788557c4e1 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1932,14 +1932,9 @@ static int fsi_probe(struct platform_device *pdev) core = NULL; if (np) { - const struct of_device_id *of_id; - - of_id = of_match_device(fsi_of_match, &pdev->dev); - if (of_id) { - core = of_id->data; - fsi_of_parse("fsia", np, &info.port_a, &pdev->dev); - fsi_of_parse("fsib", np, &info.port_b, &pdev->dev); - } + core = of_device_get_match_data(&pdev->dev); + fsi_of_parse("fsia", np, &info.port_a, &pdev->dev); + fsi_of_parse("fsib", np, &info.port_b, &pdev->dev); } else { const struct platform_device_id *id_entry = pdev->id_entry; if (id_entry) -- 2.7.4

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[alsa-devel] usb/sound/bcd2000: warning in bcd2000_init_device
by Andrey Konovalov 04 Oct '17

04 Oct '17

Hi! I've got the following report while fuzzing the kernel with syzkaller. On commit e19b205be43d11bff638cad4487008c48d21c103 (4.14-rc2). It seems that there's no check of the endpoint type. usb 1-1: BOGUS urb xfer, pipe 1 != type 3 ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1846 at drivers/usb/core/urb.c:449 usb_submit_urb+0xf8a/0x11d0 Modules linked in: CPU: 0 PID: 1846 Comm: kworker/0:2 Not tainted 4.14.0-rc2-42613-g1488251d1a98 #238 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Workqueue: usb_hub_wq hub_event task: ffff880064296300 task.stack: ffff8800643b0000 RIP: 0010:usb_submit_urb+0xf8a/0x11d0 drivers/usb/core/urb.c:448 RSP: 0018:ffff8800643b6140 EFLAGS: 00010286 RAX: 0000000000000029 RBX: ffff880063842400 RCX: 0000000000000000 RDX: 0000000000000029 RSI: ffffffff85a58800 RDI: ffffed000c876c1a RBP: ffff8800643b6240 R08: 1ffff1000c876ac0 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 1ffff1000c876c2f R13: 0000000000000003 R14: 0000000000000001 R15: ffff88006325c768 FS: 0000000000000000(0000) GS:ffff88006c800000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fa2a51c0000 CR3: 000000006acc8000 CR4: 00000000000006f0 Call Trace: bcd2000_init_device sound/usb/bcd2000/bcd2000.c:289 bcd2000_init_midi sound/usb/bcd2000/bcd2000.c:345 bcd2000_probe+0xe64/0x19e0 sound/usb/bcd2000/bcd2000.c:406 usb_probe_interface+0x35d/0x8e0 drivers/usb/core/driver.c:361 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_set_configuration+0x104e/0x1870 drivers/usb/core/message.c:1932 generic_probe+0x73/0xe0 drivers/usb/core/generic.c:174 usb_probe_device+0xaf/0xe0 drivers/usb/core/driver.c:266 really_probe drivers/base/dd.c:413 driver_probe_device+0x610/0xa00 drivers/base/dd.c:557 __device_attach_driver+0x230/0x290 drivers/base/dd.c:653 bus_for_each_drv+0x161/0x210 drivers/base/bus.c:463 __device_attach+0x26e/0x3d0 drivers/base/dd.c:710 device_initial_probe+0x1f/0x30 drivers/base/dd.c:757 bus_probe_device+0x1eb/0x290 drivers/base/bus.c:523 device_add+0xd0b/0x1660 drivers/base/core.c:1835 usb_new_device+0x7b8/0x1020 drivers/usb/core/hub.c:2457 hub_port_connect drivers/usb/core/hub.c:4903 hub_port_connect_change drivers/usb/core/hub.c:5009 port_event drivers/usb/core/hub.c:5115 hub_event+0x194d/0x3740 drivers/usb/core/hub.c:5195 process_one_work+0xc7f/0x1db0 kernel/workqueue.c:2119 worker_thread+0x221/0x1850 kernel/workqueue.c:2253 kthread+0x3a1/0x470 kernel/kthread.c:231 ret_from_fork+0x2a/0x40 arch/x86/entry/entry_64.S:431 Code: 48 8b 85 30 ff ff ff 48 8d b8 98 00 00 00 e8 6e 91 8b ff 45 89 e8 44 89 f1 4c 89 fa 48 89 c6 48 c7 c7 c0 5a c8 85 e8 10 50 dd fd <0f> ff e9 9b f7 ff ff e8 ba cf 26 fe e9 80 f7 ff ff e8 a0 a5 f4 ---[ end trace bad127706d5fe2d6 ]---

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[alsa-devel] [PATCH v2] ASoC: codecs: msm8916-wcd-analog: configure micbias in mbhc setup
by Damien Riegel 04 Oct '17

04 Oct '17

The very first time a headset is plugged in, detection is unreliable because bias hasn't been configured yet, it's done once a mechanical insertion interrupt has been triggered, so following insertions (and thus detections) are not affected. To fix the very first detection, the bias must also be configured in the function that setup the MBHC. Move pm8916_wcd_setup_mbhc after pm8916_mbhc_configure_bias to avoid a forward declaration. Signed-off-by: Damien Riegel <damien.riegel(a)savoirfairelinux.com> --- Changes in v2: - squash the two patches in one patch only sound/soc/codecs/msm8916-wcd-analog.c | 94 +++++++++++++++++++---------------- 1 file changed, 50 insertions(+), 44 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index f562f2d86907..71494e9dbdcb 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -443,50 +443,6 @@ static int pm8916_wcd_analog_enable_micbias_int1(struct wcd->micbias1_cap_mode); } -static void pm8916_wcd_setup_mbhc(struct pm8916_wcd_analog_priv *wcd) -{ - struct snd_soc_codec *codec = wcd->codec; - u32 plug_type = 0; - u32 int_en_mask; - - snd_soc_write(codec, CDC_A_MBHC_DET_CTL_1, - CDC_A_MBHC_DET_CTL_L_DET_EN | - CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_INSERTION | - CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_AUTO | - CDC_A_MBHC_DET_CTL_MBHC_BIAS_EN); - - if (wcd->hphl_jack_type_normally_open) - plug_type |= CDC_A_HPHL_PLUG_TYPE_NO; - - if (wcd->gnd_jack_type_normally_open) - plug_type |= CDC_A_GND_PLUG_TYPE_NO; - - snd_soc_write(codec, CDC_A_MBHC_DET_CTL_2, - CDC_A_MBHC_DET_CTL_HS_L_DET_PULL_UP_CTRL_I_3P0 | - CDC_A_MBHC_DET_CTL_HS_L_DET_COMPA_CTRL_V0P9_VDD | - plug_type | - CDC_A_MBHC_DET_CTL_HPHL_100K_TO_GND_EN); - - - snd_soc_write(codec, CDC_A_MBHC_DBNC_TIMER, - CDC_A_MBHC_DBNC_TIMER_INSREM_DBNC_T_256_MS | - CDC_A_MBHC_DBNC_TIMER_BTN_DBNC_T_16MS); - - /* enable MBHC clock */ - snd_soc_update_bits(codec, CDC_D_CDC_DIG_CLK_CTL, - DIG_CLK_CTL_D_MBHC_CLK_EN_MASK, - DIG_CLK_CTL_D_MBHC_CLK_EN); - - int_en_mask = MBHC_SWITCH_INT; - if (wcd->mbhc_btn_enabled) - int_en_mask |= MBHC_BUTTON_PRESS_DET | MBHC_BUTTON_RELEASE_DET; - - snd_soc_update_bits(codec, CDC_D_INT_EN_CLR, int_en_mask, 0); - snd_soc_update_bits(codec, CDC_D_INT_EN_SET, int_en_mask, int_en_mask); - wcd->mbhc_btn0_released = false; - wcd->detect_accessory_type = true; -} - static int pm8916_mbhc_configure_bias(struct pm8916_wcd_analog_priv *priv, bool micbias2_enabled) { @@ -534,6 +490,56 @@ static int pm8916_mbhc_configure_bias(struct pm8916_wcd_analog_priv *priv, return 0; } +static void pm8916_wcd_setup_mbhc(struct pm8916_wcd_analog_priv *wcd) +{ + struct snd_soc_codec *codec = wcd->codec; + bool micbias_enabled = false; + u32 plug_type = 0; + u32 int_en_mask; + + snd_soc_write(codec, CDC_A_MBHC_DET_CTL_1, + CDC_A_MBHC_DET_CTL_L_DET_EN | + CDC_A_MBHC_DET_CTL_MECH_DET_TYPE_INSERTION | + CDC_A_MBHC_DET_CTL_MIC_CLAMP_CTL_AUTO | + CDC_A_MBHC_DET_CTL_MBHC_BIAS_EN); + + if (wcd->hphl_jack_type_normally_open) + plug_type |= CDC_A_HPHL_PLUG_TYPE_NO; + + if (wcd->gnd_jack_type_normally_open) + plug_type |= CDC_A_GND_PLUG_TYPE_NO; + + snd_soc_write(codec, CDC_A_MBHC_DET_CTL_2, + CDC_A_MBHC_DET_CTL_HS_L_DET_PULL_UP_CTRL_I_3P0 | + CDC_A_MBHC_DET_CTL_HS_L_DET_COMPA_CTRL_V0P9_VDD | + plug_type | + CDC_A_MBHC_DET_CTL_HPHL_100K_TO_GND_EN); + + + snd_soc_write(codec, CDC_A_MBHC_DBNC_TIMER, + CDC_A_MBHC_DBNC_TIMER_INSREM_DBNC_T_256_MS | + CDC_A_MBHC_DBNC_TIMER_BTN_DBNC_T_16MS); + + /* enable MBHC clock */ + snd_soc_update_bits(codec, CDC_D_CDC_DIG_CLK_CTL, + DIG_CLK_CTL_D_MBHC_CLK_EN_MASK, + DIG_CLK_CTL_D_MBHC_CLK_EN); + + if (snd_soc_read(codec, CDC_A_MICB_2_EN) & CDC_A_MICB_2_EN_ENABLE) + micbias_enabled = true; + + pm8916_mbhc_configure_bias(wcd, micbias_enabled); + + int_en_mask = MBHC_SWITCH_INT; + if (wcd->mbhc_btn_enabled) + int_en_mask |= MBHC_BUTTON_PRESS_DET | MBHC_BUTTON_RELEASE_DET; + + snd_soc_update_bits(codec, CDC_D_INT_EN_CLR, int_en_mask, 0); + snd_soc_update_bits(codec, CDC_D_INT_EN_SET, int_en_mask, int_en_mask); + wcd->mbhc_btn0_released = false; + wcd->detect_accessory_type = true; +} + static int pm8916_wcd_analog_enable_micbias_int2(struct snd_soc_dapm_widget *w, struct snd_kcontrol -- 2.14.1

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[alsa-devel] [PATCH] ASoC: rockchip: Allocate enough memory so we don't overflow routes
by Douglas Anderson 04 Oct '17

04 Oct '17

In the recent commit d9f9c167edae ("ASoC: rockchip: Init dapm routes dynamically") we improperly allocated memory for the card->dapm_routes causing us to overflow the allocation on every boot. Oops. Let's allocate the correct amount of memory. We'll also add a check to make sure that we don't overrun memory even if we encounter some sort of weird device tree. Fixes: d9f9c167edae ("ASoC: rockchip: Init dapm routes dynamically") Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- sound/soc/rockchip/rk3399_gru_sound.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 30eed83e8a13..d64fbbd50544 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -494,13 +494,17 @@ static int rockchip_sound_of_parse_dais(struct device *dev, struct snd_soc_dai_link *dai; struct snd_soc_dapm_route *routes; int i, index; + int num_routes; card->dai_link = devm_kzalloc(dev, sizeof(rockchip_dais), GFP_KERNEL); if (!card->dai_link) return -ENOMEM; - routes = devm_kzalloc(dev, sizeof(rockchip_routes), + num_routes = 0; + for (i = 0; i < ARRAY_SIZE(rockchip_routes); i++) + num_routes += rockchip_routes[i].num_routes; + routes = devm_kzalloc(dev, num_routes * sizeof(*routes), GFP_KERNEL); if (!routes) return -ENOMEM; @@ -538,6 +542,12 @@ static int rockchip_sound_of_parse_dais(struct device *dev, dai->platform_of_node = np_cpu; dai->cpu_of_node = np_cpu; + if (card->num_dapm_routes + rockchip_routes[index].num_routes > + num_routes) { + dev_err(dev, "Too many routes\n"); + return -EINVAL; + } + memcpy(routes + card->num_dapm_routes, rockchip_routes[index].routes, rockchip_routes[index].num_routes * sizeof(*routes)); -- 2.14.2.822.g60be5d43e6-goog

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[alsa-devel] [PATCH 0/3] ASoC: img-i2s-out: Add power management
by Ed Blake 04 Oct '17

04 Oct '17

This patch set adds power managment to the img-i2s-out driver. The first patch renames existing 'suspend' and 'resume' functions to 'runtime_suspend' and 'runtime_resume', which is what they actually are, the second patch adds system PM suspend / resume handling and the third patch adds control of the sys clock to runtime PM. Ed Blake (3): ASoC: img-i2s-out: Rename suspend / resume funcs ASoC: img-i2s-out: Add suspend / resume handling ASoC: img-i2s-out: Add control of sys clock to runtime PM sound/soc/img/img-i2s-out.c | 116 +++++++++++++++++++++++++++++++++----------- 1 file changed, 87 insertions(+), 29 deletions(-) -- 1.9.1

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