April 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RESEND RFC PATCH 0/3] ASoC: Mediatek: Add support for MT8173 SOC
by Koro Chen 04 May '15

04 May '15

Refactor and resend this series because of the 60KB limitation. This adds basic support for the Mediatek AFE unit for MT8173 SoC. This patch is based on Linux 4.0-rc1, Sascha's clk patch [1], and his SCPSYS power patch [2]. The AFE unit comprises several memory interfaces that communicate with CPU, a multi input multi output digital audio interconnect, and several external interfaces (e.g. I2S). The AFE unit is a DMA master, so no external DMA engine is needed. Each external interface is presented as a DAI to ASoC. A memory interface must be connected via the interconnect to an external interface. The connection pathes are configured through the device tree. [1] http://lists.infradead.org/pipermail/linux-mediatek/2015-March/000079.html [2] http://lists.infradead.org/pipermail/linux-mediatek/2015-March/000132.html Koro Chen (3): ASoC: mediatek: Add binding support for AFE driver ASoC: mediatek: Add AFE connection control ASoC: mediatek: Add AFE platform driver .../devicetree/bindings/sound/mtk-afe-pcm.txt | 105 ++ include/dt-bindings/sound/mtk-afe.h | 36 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mediatek/Kconfig | 8 + sound/soc/mediatek/Makefile | 2 + sound/soc/mediatek/mtk-afe-common.h | 105 ++ sound/soc/mediatek/mtk-afe-connection.c | 416 ++++++ sound/soc/mediatek/mtk-afe-connection.h | 30 + sound/soc/mediatek/mtk-afe-pcm.c | 1497 ++++++++++++++++++++ 10 files changed, 2201 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt create mode 100644 include/dt-bindings/sound/mtk-afe.h create mode 100644 sound/soc/mediatek/Kconfig create mode 100644 sound/soc/mediatek/Makefile create mode 100644 sound/soc/mediatek/mtk-afe-common.h create mode 100644 sound/soc/mediatek/mtk-afe-connection.c create mode 100644 sound/soc/mediatek/mtk-afe-connection.h create mode 100644 sound/soc/mediatek/mtk-afe-pcm.c --

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Re: [alsa-devel] [PATCH] ALSA: hda - Add dock support for ThinkPad X250 (17aa:2226)
by Takashi Iwai 02 May '15

02 May '15

At Sat, 11 Apr 2015 09:31:35 +0200, Yves-Alexis Perez wrote: > > This model uses the same dock port as the previous generation. > > Signed-off-by: Yves-Alexis Perez <corsac(a)debian.org> Thanks, applied with Cc to stable. But, at the next time, please put maintainers to Cc. Takashi > --- > sound/pci/hda/patch_realtek.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 7438213..2087728 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -5030,6 +5030,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), > SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), > SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > + SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), > SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), > SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), > SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > -- > 2.1.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

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[alsa-devel] [PATCH RFT] ASoC: mc13783: Fix wrong mask value used in mc13xxx_reg_rmw() calls
by Axel Lin 01 May '15

01 May '15

mc13xxx_reg_rmw() won't change any bit if passing 0 to the mask field. Pass AUDIO_SSI_SEL instead of 0 for the mask field to set AUDIO_SSI_SEL bit. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi, I don't have this h/w for test, so please help review and test this patch. Thanks. sound/soc/codecs/mc13783.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 2ffb9a0..3d44fc5 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -623,14 +623,14 @@ static int mc13783_probe(struct snd_soc_codec *codec) AUDIO_SSI_SEL, 0); else mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC, - 0, AUDIO_SSI_SEL); + AUDIO_SSI_SEL, AUDIO_SSI_SEL); if (priv->dac_ssi_port == MC13783_SSI1_PORT) mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, AUDIO_SSI_SEL, 0); else mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, - 0, AUDIO_SSI_SEL); + AUDIO_SSI_SEL, AUDIO_SSI_SEL); return 0; } -- 2.1.0

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[alsa-devel] [PATCH 1/3] ALSA: hda: add hdac_adsp_enable module flag
by Vinod Koul 01 May '15

01 May '15

Some Intel HDA controllers sport a DSP. These systems can also be enabled with ASoC HDA driver as well. So add a flag in hda-core to enable/disable aDSP This flag for now is false, and should be true once the ASoC based systems mature. The integrators/OS vendors can configure this flag based on system preference. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/hdaudio.h | 11 +++++++++++ sound/hda/hda_bus_type.c | 4 ++++ 2 files changed, 15 insertions(+) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b97c59eab7ab..015bec1079f9 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -12,6 +12,17 @@ #include <sound/memalloc.h> #include <sound/hda_verbs.h> +/* + * hdac_adsp_enable: exported HD-A aDSP enable configuration. + * + * Some Intel HDA controllers sport a DSP, for these platform we can bypass + * aDSP and use as regular HDA controller or enable aDSP and use aDSP + * along with I2S codecs etc. + * hdac_adsp_enable would enable the aDSP based HDA controller if the + * platform supports it + */ +extern bool hdac_adsp_enable; + /* codec node id */ typedef u16 hda_nid_t; diff --git a/sound/hda/hda_bus_type.c b/sound/hda/hda_bus_type.c index 519914a12e8a..80e0570ffbf4 100644 --- a/sound/hda/hda_bus_type.c +++ b/sound/hda/hda_bus_type.c @@ -10,6 +10,10 @@ MODULE_DESCRIPTION("HD-audio bus"); MODULE_LICENSE("GPL"); +bool hdac_adsp_enable = false; +module_param(hdac_adsp_enable, bool, 0444); +MODULE_PARM_DESC(hdac_adsp_enable, "Enable aDSP on Intel HDA based systems"); + static int hda_bus_match(struct device *dev, struct device_driver *drv) { struct hdac_device *hdev = dev_to_hdac_dev(dev); -- 1.9.1

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[alsa-devel] [PATCH v2 1/5] ASoC: Correct typo in SOC_VALUE_ENUM_SINGLE macro
by Charles Keepax 01 May '15

01 May '15

xnitmes is clearly intended to be xnitems, but all other macros just refer to this as xitems, so change it to that. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Use first entry of values array as off value for the mux Thanks, Charles include/sound/soc.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a73d855..45cfd69 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -190,8 +190,8 @@ #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} -#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ - SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) +#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xitems, xtexts, xvalues) \ + SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xitems, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts) #define SOC_ENUM(xname, xenum) \ -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: fix simple_return.cocci warnings
by kbuild test robot 30 Apr '15

30 Apr '15

sound/soc/codecs/adau1977.c:496:5-8: WARNING: end returns can be simpified Simplify a trivial if-return sequence. Possibly combine with a preceding function call. Generated by: scripts/coccinelle/misc/simple_return.cocci CC: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- adau1977.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -493,10 +493,7 @@ static int adau1977_set_bias_level(struc break; } - if (ret) - return ret; - - return 0; + return ret; } static int adau1977_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,

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[alsa-devel] [PATCH] ASoC: fix simple_return.cocci warnings
by kbuild test robot 30 Apr '15

30 Apr '15

sound/soc/codecs/ssm2518.c:521:5-8: WARNING: end returns can be simpified Simplify a trivial if-return sequence. Possibly combine with a preceding function call. Generated by: scripts/coccinelle/misc/simple_return.cocci CC: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- ssm2518.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -518,10 +518,7 @@ static int ssm2518_set_bias_level(struct break; } - if (ret) - return ret; - - return 0; + return ret; } static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Correct pm status check in suspend callback
by Peter Ujfalusi 30 Apr '15

30 Apr '15

pm_runtime_enabled() will only tell if the pm runtime has been enabled for the device, which is done at probe time but will not tell the actual power state of the device. pm_runtime_active() provides this information. This patch fixes a kernel crash when doing suspend when McASP is not active. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Mark, can you schedule this for the next -rc? Thank you, Peter sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index ddf7e97d8f0b..1ab3fcdc6bba 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1247,7 +1247,7 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) u32 reg; int i; - context->pm_state = pm_runtime_enabled(mcasp->dev); + context->pm_state = pm_runtime_active(mcasp->dev); if (!context->pm_state) pm_runtime_get_sync(mcasp->dev); -- 2.3.5

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[alsa-devel] [PATCH v2 0/3] Calculate BCLK using TDM slots and remove channels rule
by Jyri Sarha 30 Apr '15

30 Apr '15

The first and the last patch are excatly the same as in the first series, but in the middle thee is a new patch to constraint out unsupported channel counts in multiple serializers case. - The first patch is a bug fix and also fixes Dan Carpenter's concern about stack usage in davinci_mcasp_hw_rule_channels() - The second is also a bug fix, even thou the bug has existed in the dirver for a very long time. It is just that nobody has used multiple serializers before. - The last patch is a simple optimization All the patches have been tested are internally reviewed. Jyri Sarha (3): ASoC: davinci-mcasp: Calculate BCLK using TDM slots and remove channels rule ASoC: davinci-mcasp: Channel count constraints for multi-serializer case ASoC: davinci-macsp: Optimize implicit BLCK sample-rate rule sound/soc/davinci/davinci-mcasp.c | 167 ++++++++++++++++++++------------------ 1 file changed, 90 insertions(+), 77 deletions(-) -- 1.9.1

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[alsa-devel] ASoC fixes for v4.1
by Mark Brown 30 Apr '15

30 Apr '15

The following changes since commit b787f68c36d49bb1d9236f403813641efa74a031: Linux 4.1-rc1 (2015-04-26 17:59:10 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.1-rc1 for you to fetch changes up to d839c98f98826f0c13e46d5a4cecb46dd357b50e: Merge remote-tracking branches 'asoc/fix/rt5677', 'asoc/fix/samsung' and 'asoc/fix/tfa9879' into asoc-linus (2015-04-29 13:37:31 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.1 A few fixes for v4.1, none earth shattering and mostly driver related except for one change to fix !PM builds for Intel platforms which is done by adding stubs in the core so other platforms don't run into the same issue. ---------------------------------------------------------------- Axel Lin (1): ASoC: rt5645: Fix mask for setting RT5645_DMIC_2_DP_GPIO12 bit Bard Liao (2): ASoC: rt5677: add register patch for PLL ASoC: rt5677: fixed wrong DMIC ref clock Charles Keepax (1): ASoC: dapm: Enable autodisable on SOC_DAPM_SINGLE_TLV_AUTODISABLE Fabio Estevam (1): ASoC: fsl_ssi: Fix platform_get_irq() error handling Fang, Yang A (1): ASoC: rt5645: Add ACPI match ID Jie Yang (1): ASoC: add static inline funcs to fix a compiling issue Jin Yao (1): ASoC: Intel: Remove invalid kfree of devm allocated data Kuninori Morimoto (1): ASoC: rsnd: set dmaen->chan = NULL when error case Mark Brown (5): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/email', 'asoc/fix/fsl-ssi', 'asoc/fix/pm', 'asoc/fix/qcom' and 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5677', 'asoc/fix/samsung' and 'asoc/fix/tfa9879' into asoc-linus Rajeev Kumar (1): ASoC: Update email-id of Rajeev Kumar Takashi Iwai (1): ASoC: qcom: Return an error for invalid PCM trigger command Vinod Koul (1): ASoC: Intel: fix the makefile for atom code Wei Yongjun (3): ASoC: tfa9879: Fix return value check in tfa9879_i2c_probe() ASoC: samsung: s3c24xx-i2s: Fix return value check in s3c24xx_iis_dev_probe() ASoC: Intel: sst_byt: remove kfree for memory allocated with devm_kzalloc include/sound/designware_i2s.h | 2 +- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 12 ++++++++++++ include/sound/spear_dma.h | 2 +- sound/soc/codecs/rt5645.c | 13 ++++++++++++- sound/soc/codecs/rt5677.c | 5 ++++- sound/soc/codecs/tfa9879.c | 4 ++-- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 1 - sound/soc/intel/haswell/sst-haswell-ipc.c | 1 - sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- sound/soc/sh/rcar/dma.c | 1 + 14 files changed, 39 insertions(+), 14 deletions(-)

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