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

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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

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[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

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[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

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[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

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