September 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/5] ALSA: soc: fix sparse warnings
by Takashi Sakamoto 28 Sep '16

28 Sep '16

Hi, This patchset is to fix some sparse warnings in ALSA SoC part. Below items are going to be fixed: * fix a condition statement with a logical operation * purge old way to initialize array * missing 'static' qualifier for file local symbols Although I got many warnings from Intel SST drivers due to confusions of address spaces (__iomem and __kernel), I still left them as they are, because they require more codes to fix. I hope Intel developers will do additional work for them... Takashi Sakamoto (5): ALSA: soc-da7219: fix inappropriate condition statement ALSA: soc-sst-bxt-rt298: fix obsoleted initializers for array ALSA: soc-sst-bxt-da7219_max98357a: fix obsoleted initializers for array ALSA: soc-rt5616: add static qualifier for file local symbols ALSA: soc-tpa6130a2: add static qualifier for file local symbols sound/soc/codecs/da7219.c | 2 +- sound/soc/codecs/rt5616.c | 4 ++-- sound/soc/codecs/tpa6130a2.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 12 ++++++------ sound/soc/intel/boards/bxt_rt298.c | 14 +++++++------- 5 files changed, 17 insertions(+), 17 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: trivial: system spelling fix
by Marek Vasut 28 Sep '16

28 Sep '16

Do s/ststem/system/ . Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index b95ae49..d023959 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -13,4 +13,4 @@ config SND_SIMPLE_SCU_CARD select SND_SIMPLE_CARD_UTILS help This option enables generic simple SCU sound card support. - It supports DPCM of multi CPU single Codec ststem. + It supports DPCM of multi CPU single Codec system. -- 2.9.3

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[alsa-devel] [PATCH v2 0/3] ASoC: max9867/rt5616/wm8960: remove usage of obsoleted TLV-related macro
by Takashi Sakamoto 28 Sep '16

28 Sep '16

This patchset update my previous one: [alsa-devel] [PATCH] ASOC: max9867/rt5616/wm8960: remove usage of obsoleted TLV-related macro Changes v2: - Split to several patches for each modules Takashi Sakamoto (3): ASoC: max9867: remove usage of obsoleted TLV-related macro ASoC: rt5616: remove usage of obsoleted TLV-related macro ASoC: wm8960: remove usage of obsoleted TLV-related macro sound/soc/codecs/max9867.c | 5 ++--- sound/soc/codecs/rt5616.c | 5 ++--- sound/soc/codecs/wm8960.c | 5 ++--- 3 files changed, 6 insertions(+), 9 deletions(-) -- 2.7.4

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[alsa-devel] [RFC PATCH 0/3] support DP MST audio
by libin.yang＠linux.intel.com 28 Sep '16

28 Sep '16

From: Libin Yang <libin.yang(a)linux.intel.com> This patchset starts to support DP MST audio. This patchset is based on drm-intel-nightly on git://anongit.freedesktop.org/drm-intel Libin Yang (3): ALSA: hda - codec add DP MST support for connection list ALSA: hda - add DP mst verb support ALSA: hda - add DP mst audio support sound/pci/hda/hda_codec.c | 129 ++++++++++++++++++----- sound/pci/hda/hda_codec.h | 18 ++-- sound/pci/hda/hda_generic.c | 18 ++-- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/patch_analog.c | 18 ++-- sound/pci/hda/patch_hdmi.c | 228 +++++++++++++++++++++++++++++++---------- sound/pci/hda/patch_realtek.c | 22 ++-- sound/pci/hda/patch_sigmatel.c | 4 +- sound/pci/hda/patch_via.c | 10 +- 9 files changed, 328 insertions(+), 121 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: tlv320aic31xx: do not declare support for mono DAI
by Nikita Yushchenko 27 Sep '16

27 Sep '16

This hardware supports only 2-channel DAI, even mono ADC digital output has two channels with the same data. Having min_channels=1 results in broken playback of mono files in setups where CPU DAI supports mono. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- sound/soc/codecs/tlv320aic31xx.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 725173b12725..be1a64bfd320 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1213,7 +1213,7 @@ static struct snd_soc_dai_driver dac31xx_dai_driver[] = { .name = "tlv32dac31xx-hifi", .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC31XX_RATES, .formats = AIC31XX_FORMATS, @@ -1228,14 +1228,14 @@ static struct snd_soc_dai_driver aic31xx_dai_driver[] = { .name = "tlv320aic31xx-hifi", .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC31XX_RATES, .formats = AIC31XX_FORMATS, }, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC31XX_RATES, .formats = AIC31XX_FORMATS, -- 2.1.4

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[alsa-devel] [PATCH 0/3] ASOC: wm8991/stac9766: fix wrong usage of TLV-related macro
by Takashi Sakamoto 27 Sep '16

27 Sep '16

Hi, This patchset is for Linux 4.8-rc8. In current implementation of drivers for wm899a and stac9766, DECLARE_TLV_DB_LINEAR() is used to represent information about attenuation and gain of volumes. According to their datasheet, the attenuation and volumes are first-degree equation with dB coefficiency. It's not linear and usage of the macro is wrong[1]. This patchset fixes the bug. As of 2016, these CODECs seem to EOL. So this patchset is just for maintenance purpose to remove misuse of TLV-related macros from tree. [1] Re: [RFC] TLV entry for linear volume http://www.spinics.net/lists/alsa-devel/msg03485.html Takashi Sakamoto (3): ASOC: wm8991: fix wrong usage of DECLARE_TLV_DB_LINEAR() ASOC: wm8991: remove unused variable ASOC: stac9766: fix wrong usage of DECLARE_TLV_DB_LINEAR() sound/soc/codecs/stac9766.c | 8 ++++---- sound/soc/codecs/wm8991.c | 28 +++++++++++++++++++--------- 2 files changed, 23 insertions(+), 13 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 1/2] ASoC: arizona: Add debug prints for output power up/down times
by Charles Keepax 27 Sep '16

27 Sep '16

When debugging it is useful to check the total power up/down delay that is executed as part of the coalesced output delay. This patch adds some debug prints for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 780cf4f..eaf1dbb 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -891,6 +891,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, case ARIZONA_OUT3R_ENA_SHIFT: priv->out_up_pending--; if (!priv->out_up_pending) { + dev_dbg(codec->dev, "Power up delay: %d\n", + priv->out_up_delay); msleep(priv->out_up_delay); priv->out_up_delay = 0; } @@ -925,6 +927,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, case ARIZONA_OUT3R_ENA_SHIFT: priv->out_down_pending--; if (!priv->out_down_pending) { + dev_dbg(codec->dev, "Power down delay: %d\n", + priv->out_down_delay); msleep(priv->out_down_delay); priv->out_down_delay = 0; } -- 2.1.4

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[alsa-devel] Applied "ASoC: sst-bxt-da7219_max98357a: fix obsoleted initializers for array" to the asoc tree
by Mark Brown 27 Sep '16

27 Sep '16

The patch ASoC: sst-bxt-da7219_max98357a: fix obsoleted initializers for array 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 43d443dc257c00ef3a3f940b6abfb7537c4fcbe8 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Date: Tue, 27 Sep 2016 07:45:09 +0900 Subject: [PATCH] ASoC: sst-bxt-da7219_max98357a: fix obsoleted initializers for array Sparse reports below warnings. bxt_da7219_max98357a.c:250:9: warning: obsolete array initializer, use C99 syntax bxt_da7219_max98357a.c:275:9: warning: obsolete array initializer, use C99 syntax bxt_da7219_max98357a.c:290:9: warning: obsolete array initializer, use C99 syntax bxt_da7219_max98357a.c:304:9: warning: obsolete array initializer, use C99 syntax bxt_da7219_max98357a.c:317:9: warning: obsolete array initializer, use C99 syntax There's no need to use obsoleted way. This commit fixes it. Fixes: 723bad3fef8b (ASoC: Intel: Add Broxton-P Dialog Maxim machine driver) Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Acked-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index bb4533c305c7..6532b8f0ab2f 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -302,7 +302,7 @@ static struct snd_soc_ops broxton_refcap_ops = { /* broxton digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link broxton_dais[] = { /* Front End DAI links */ - [BXT_DPCM_AUDIO_PB] + [BXT_DPCM_AUDIO_PB] = { .name = "Bxt Audio Port", .stream_name = "Audio", @@ -318,7 +318,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .dpcm_playback = 1, .ops = &broxton_da7219_fe_ops, }, - [BXT_DPCM_AUDIO_CP] + [BXT_DPCM_AUDIO_CP] = { .name = "Bxt Audio Capture Port", .stream_name = "Audio Record", @@ -333,7 +333,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .dpcm_capture = 1, .ops = &broxton_da7219_fe_ops, }, - [BXT_DPCM_AUDIO_REF_CP] + [BXT_DPCM_AUDIO_REF_CP] = { .name = "Bxt Audio Reference cap", .stream_name = "Refcap", @@ -362,7 +362,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .dynamic = 1, .ops = &broxton_dmic_ops, }, - [BXT_DPCM_AUDIO_HDMI1_PB] + [BXT_DPCM_AUDIO_HDMI1_PB] = { .name = "Bxt HDMI Port1", .stream_name = "Hdmi1", @@ -375,7 +375,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .nonatomic = 1, .dynamic = 1, }, - [BXT_DPCM_AUDIO_HDMI2_PB] + [BXT_DPCM_AUDIO_HDMI2_PB] = { .name = "Bxt HDMI Port2", .stream_name = "Hdmi2", @@ -388,7 +388,7 @@ static struct snd_soc_dai_link broxton_dais[] = { .nonatomic = 1, .dynamic = 1, }, - [BXT_DPCM_AUDIO_HDMI3_PB] + [BXT_DPCM_AUDIO_HDMI3_PB] = { .name = "Bxt HDMI Port3", .stream_name = "Hdmi3", -- 2.9.3

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[alsa-devel] [PATCH] ASOC: max9867/rt5616/wm8960: remove usage of obsoleted TLV-related macro
by Takashi Sakamoto 27 Sep '16

27 Sep '16

TLV_DB_RANGE_HEAD macro was obsoleted by commit bf1d1c9b6179 ("ALSA: tlv: add DECLARE_TLV_DB_RANGE()"). This commit removes usage of the macro, with the obsoleting macro renamed to SNDRV_CTL_TLVD_DECLARE_DB_RANGE(). Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: patches(a)opensource.wolfsonmicro.com Cc: anish kumar <yesanishhere(a)gmail.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/soc/codecs/max9867.c | 5 ++--- sound/soc/codecs/rt5616.c | 5 ++--- sound/soc/codecs/wm8960.c | 5 ++--- 3 files changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 2a22fdd..1aa838f 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -38,11 +38,10 @@ static DECLARE_TLV_DB_SCALE(max9860_capture_tlv, -600, 200, 0); static DECLARE_TLV_DB_SCALE(max9860_mic_tlv, 2000, 100, 1); static DECLARE_TLV_DB_SCALE(max9860_adc_left_tlv, -1200, 100, 1); static DECLARE_TLV_DB_SCALE(max9860_adc_right_tlv, -1200, 100, 1); -static const unsigned int max98088_micboost_tlv[] = { - TLV_DB_RANGE_HEAD(2), +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max98088_micboost_tlv, 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), -}; +); static const struct snd_kcontrol_new max9867_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", MAX9867_LEFTVOL, diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index f527b5b..851704f 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -294,8 +294,7 @@ static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ -static unsigned int bst_tlv[] = { - TLV_DB_RANGE_HEAD(7), +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(bst_tlv, 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), @@ -303,7 +302,7 @@ static unsigned int bst_tlv[] = { 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), -}; +); static const struct snd_kcontrol_new rt5616_snd_controls[] = { /* Headphone Output Volume */ diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index d7f444f..3d8c756 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -226,11 +226,10 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); static const DECLARE_TLV_DB_SCALE(lineinboost_tlv, -1500, 300, 1); -static const unsigned int micboost_tlv[] = { - TLV_DB_RANGE_HEAD(2), +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(micboost_tlv, 0, 1, TLV_DB_SCALE_ITEM(0, 1300, 0), 2, 3, TLV_DB_SCALE_ITEM(2000, 900, 0), -}; +); static const struct snd_kcontrol_new wm8960_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL, -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda - Add the top speaker pin config for HP Spectre x360
by Takashi Iwai 27 Sep '16

27 Sep '16

HP Spectre x360 with CX20724 codec has two speaker outputs while the BIOS sets up only the bottom one (NID 0x17) and disables the top one (NID 0x1d). This patch adds a fixup simply defining the proper pincfg for NID 0x1d so that the top speaker works as is. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=169071 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 56fefbd85782..ed62748a6d55 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -261,6 +261,7 @@ enum { CXT_FIXUP_HP_530, CXT_FIXUP_CAP_MIX_AMP_5047, CXT_FIXUP_MUTE_LED_EAPD, + CXT_FIXUP_HP_SPECTRE, }; /* for hda_fixup_thinkpad_acpi() */ @@ -765,6 +766,14 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_mute_led_eapd, }, + [CXT_FIXUP_HP_SPECTRE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* enable NID 0x1d for the speaker on top */ + { 0x1d, 0x91170111 }, + { } + } + }, }; static const struct snd_pci_quirk cxt5045_fixups[] = { @@ -814,6 +823,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC), + SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), -- 2.10.0

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