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

[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

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[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…

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[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

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[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

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