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

[alsa-devel] [bug report] ASoC: dapm: Implement and instantiate DAI widgets
by Dan Carpenter 20 Jan '17

20 Jan '17

Hello Mark Brown, The patch 888df395ebc5: "ASoC: dapm: Implement and instantiate DAI widgets" from Feb 16, 2012, leads to the following static checker warning: sound/soc/soc-dapm.c:3944 snd_soc_dapm_new_dai_widgets() error: 'w' dereferencing possible ERR_PTR() sound/soc/soc-dapm.c 3929 if (dai->driver->playback.stream_name) { 3930 template.id = snd_soc_dapm_dai_in; 3931 template.name = dai->driver->playback.stream_name; 3932 template.sname = dai->driver->playback.stream_name; 3933 3934 dev_dbg(dai->dev, "ASoC: adding %s widget\n", 3935 template.name); 3936 3937 w = snd_soc_dapm_new_control_unlocked(dapm, &template); This can either return NULL or ERR_PTR(-EPROBE_DEFER). 3938 if (!w) { 3939 dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", 3940 dai->driver->playback.stream_name); 3941 return -ENOMEM; 3942 } 3943 3944 w->priv = dai; ^^^^^^^^^^^^^ 3945 dai->playback_widget = w; 3946 } regards, dan carpenter

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[alsa-devel] [PATCH v2 0/2] ASoC: sun4i-spdif: Add support for the H3 SoC
by codekipper＠gmail.com 20 Jan '17

20 Jan '17

From: Marcus Cooper <codekipper(a)gmail.com> The H3 SoC uses the same SPDIF block as found in earlier SoCs, but the transmit fifo is at a different address. This has been tested on the Beelink X2. BR, CK --- Changes since v1: - Now a patch set with bindings and driver changes. --- Marcus Cooper (2): ASoC: sunxi: Add bindings for sun8i to SPDIF ASoC: sun4i-spdif: Add support for the H3 SoC Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt | 1 + sound/soc/sunxi/sun4i-spdif.c | 11 +++++++++++ 2 files changed, 12 insertions(+) -- 2.11.0

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[alsa-devel] [PATCH V3 0/5] Add support for Legacy HDMI audio drivers
by Jerome Anand 20 Jan '17

20 Jan '17

Legacy (CherryTrail/ Baytrail) HDMI audio drivers added Legacy hdmi audio-Gfx interaction/ interfacing is updated to use irq chip framework This patch series has only been tested on hardware with a single HDMI connector/pipe and additional work may be needed for newer machines with 2 connectors Jerome Anand (5): drm/i915: setup bridge for HDMI LPE audio driver drm/i915: Add support for audio driver notifications ALSA: add Intel HDMI LPE audio driver for BYT/CHT-T ALSA: x86: hdmi: Add audio support for BYT and CHT ALSA: x86: hdmi: continue playback even when display resolution changes Documentation/gpu/i915.rst | 9 + drivers/gpu/drm/i915/Makefile | 3 + drivers/gpu/drm/i915/i915_drv.c | 8 +- drivers/gpu/drm/i915/i915_drv.h | 17 + drivers/gpu/drm/i915/i915_irq.c | 16 + drivers/gpu/drm/i915/i915_reg.h | 3 + drivers/gpu/drm/i915/intel_audio.c | 8 + drivers/gpu/drm/i915/intel_hdmi.c | 1 + drivers/gpu/drm/i915/intel_lpe_audio.c | 389 +++++++ include/drm/intel_lpe_audio.h | 46 + sound/Kconfig | 2 + sound/Makefile | 2 +- sound/x86/Kconfig | 16 + sound/x86/Makefile | 8 + sound/x86/intel_hdmi_audio.c | 1932 ++++++++++++++++++++++++++++++++ sound/x86/intel_hdmi_audio.h | 201 ++++ sound/x86/intel_hdmi_audio_if.c | 551 +++++++++ sound/x86/intel_hdmi_lpe_audio.c | 617 ++++++++++ sound/x86/intel_hdmi_lpe_audio.h | 685 +++++++++++ 19 files changed, 4511 insertions(+), 3 deletions(-) create mode 100644 drivers/gpu/drm/i915/intel_lpe_audio.c create mode 100644 include/drm/intel_lpe_audio.h create mode 100644 sound/x86/Kconfig create mode 100644 sound/x86/Makefile create mode 100644 sound/x86/intel_hdmi_audio.c create mode 100644 sound/x86/intel_hdmi_audio.h create mode 100644 sound/x86/intel_hdmi_audio_if.c create mode 100644 sound/x86/intel_hdmi_lpe_audio.c create mode 100644 sound/x86/intel_hdmi_lpe_audio.h -- 2.9.3 base-commit: eb5c556a1e5eff388e48e4b7cf61e95783019e83

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[alsa-devel] ASoC: fsl_sai: Has anybody attempted to run the i.MX6UL in TDM mode with channels > 2?
by Caleb Crome 19 Jan '17

19 Jan '17

Hi all, Now that the MX6 fsl_ssi seems to be getting stable for TDM with channels > 2 (thanks all!), the MX6 UL and MX7 have only the SAI port. Does anybody have experience with running the SAI with more than 2 channels? Any thoughts or comments would be welcome! Thanks, -Caleb

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[alsa-devel] [PATCH] ASoC: arizona: Propagate errors from arizona_spk_init
by Charles Keepax 19 Jan '17

19 Jan '17

arizona_spk_init uses snd_soc_dapm_new_control which since commit 37e1df8c95e2 ("ASoC: dapm: handle probe deferrals") will occasionally request a probe deferral. Which means we should propagate the error out of our driver from it. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs47l24.c | 5 ++++- sound/soc/codecs/wm5102.c | 5 ++++- sound/soc/codecs/wm5110.c | 5 ++++- sound/soc/codecs/wm8997.c | 6 +++++- sound/soc/codecs/wm8998.c | 6 +++++- 5 files changed, 22 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 0dd721e..47e6fdd 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1124,7 +1124,10 @@ static int cs47l24_codec_probe(struct snd_soc_codec *codec) priv->core.arizona->dapm = dapm; - arizona_init_spk(codec); + ret = arizona_init_spk(codec); + if (ret < 0) + return ret; + arizona_init_gpio(codec); arizona_init_mono(codec); arizona_init_notifiers(codec); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 9edd239..1fe358e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1946,7 +1946,10 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) if (ret) goto err_adsp2_codec_probe; - arizona_init_spk(codec); + ret = arizona_init_spk(codec); + if (ret < 0) + return ret; + arizona_init_gpio(codec); arizona_init_notifiers(codec); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 01b5b6c..1bc9421 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2284,7 +2284,10 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) priv->core.arizona->dapm = dapm; - arizona_init_spk(codec); + ret = arizona_init_spk(codec); + if (ret < 0) + return ret; + arizona_init_gpio(codec); arizona_init_mono(codec); arizona_init_notifiers(codec); diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index ee0c863..49401a8 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1062,8 +1062,12 @@ static int wm8997_codec_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct snd_soc_component *component = snd_soc_dapm_to_component(dapm); struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = arizona_init_spk(codec); + if (ret < 0) + return ret; - arizona_init_spk(codec); arizona_init_notifiers(codec); snd_soc_component_disable_pin(component, "HAPTICS"); diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 3694f59..44f4471 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1321,10 +1321,14 @@ static int wm8998_codec_probe(struct snd_soc_codec *codec) struct wm8998_priv *priv = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct snd_soc_component *component = snd_soc_dapm_to_component(dapm); + int ret; priv->core.arizona->dapm = dapm; - arizona_init_spk(codec); + ret = arizona_init_spk(codec); + if (ret < 0) + return ret; + arizona_init_gpio(codec); arizona_init_notifiers(codec); -- 2.1.4

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[alsa-devel] [PATCH 00/12] ASoC: Intel: Skylake: Driver updates
by jeeja.kp＠intel.com 19 Jan '17

19 Jan '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series includes driver updates for the handling scenarios during System S0/S3 transitions when audio stream is stopped during system. o move DMA configuration in DSP Gateway Module widget event handler to ensure DMA is configured after resume. o Enable SSP MLCK using Supply widget. This will ensure the MCLK is enabled when the widget is power on. o Check DMA decouple register state before coupling/decoupling the DMA. o In case of HDMI pass-through pipe, reset only the FE Pipe when stream in in XRUN state. o Resume the stream from HW renderer position. o HDMI Codec pin and converter state need to be restored when stream is resumed. Jeeja KP (12): ASoC: Intel: Skylake: Add helper function to setup host/link dma ASoC: Intel: Skylake: Configure DMA in PRE_PMD handler of Mixer ASoC: Intel: Skylake: Removed unused skl_get_format() ALSA: hda: check stream decoupled register state ASoC: Intel: Skylake: Add set_tristate DAI ops to enable SSP MCLK ASoC: Intel: Skylake: Remove unused SSP BE prepare DAI ops ASoC: Intel: Skylake: Add supply widget as non DSP widget ASoC: Intel: Skylake: Add supply widget in skl_nau_max machine ASoC: Intel: Skylake: Add supply widget in bxt_da_max machine ASoC: Intel: Skylake: Don't reset pass-through pipe in BE prepare ASoC: Intel: Skylake: set the resume point to LPIB ASoC: hdac_hdmi: Enable pin and converter in prepare sound/hda/ext/hdac_ext_stream.c | 15 +- sound/soc/codecs/hdac_hdmi.c | 28 ++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 34 ++++ sound/soc/intel/boards/skl_nau88l25_max98357a.c | 34 ++++ sound/soc/intel/skylake/skl-pcm.c | 210 +++++++++++++----------- sound/soc/intel/skylake/skl-topology.c | 27 ++- sound/soc/intel/skylake/skl-topology.h | 6 + 7 files changed, 240 insertions(+), 114 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH v4 0/9] ASoC: Intel: Skylake: Driver updates
by jeeja.kp＠intel.com 19 Jan '17

19 Jan '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series includes driver updates for the handling scenarios during System S0/S3 transitions when audio stream is stopped during system. o move DMA configuration in DSP Gateway Module widget event handler to ensure DMA is configured after resume. o Enable SSP MLCK using Supply widget. This will ensure the MCLK is enabled when the widget is power on. o Check DMA decouple register state before coupling/decoupling the DMA. o In case of HDMI pass-through pipe, reset only the FE Pipe when stream in in XRUN state. o Resume the stream from HW renderer position. o HDMI Codec pin and converter state need to be restored when stream is resumed. Changes in v2: - Addressed Mark's comment on patch 1, split this into 2 as this had additional changes other than configuring the DMA. - Addressed Takashi's comment on the usage of snd_hdac_updatel() Changes in v3: - Addressed Takashi's comment on superfluous shift used in reading the value of PPCTL register and also cleaned the condition for updating the reg value. Changes in v4: - Acked by Takashi for patch "ALSA: hda: check stream decoupled register state". - Removed compilation warning in patch "ASoC: hdac_hdmi: Enable pin and converter in prepare". - Remove 4 patches which are already applied. Jeeja KP (9): ALSA: hda: check stream decoupled register state ASoC: Intel: Skylake: Add set_tristate DAI ops to enable SSP MCLK ASoC: Intel: Skylake: Remove unused SSP BE prepare DAI ops ASoC: Intel: Skylake: Add supply widget as non DSP widget ASoC: Intel: Skylake: Add supply widget in skl_nau_max machine ASoC: Intel: Skylake: Add supply widget in bxt_da_max machine ASoC: Intel: Skylake: Don't reset pass-through pipe in BE prepare ASoC: Intel: Skylake: set the resume point to LPIB ASoC: hdac_hdmi: Enable pin and converter in prepare sound/hda/ext/hdac_ext_stream.c | 15 ++++--- sound/soc/codecs/hdac_hdmi.c | 37 ++++++++---------- sound/soc/intel/boards/bxt_da7219_max98357a.c | 34 ++++++++++++++++ sound/soc/intel/boards/skl_nau88l25_max98357a.c | 34 ++++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 52 +++++++++++++++---------- sound/soc/intel/skylake/skl-topology.c | 6 ++- 6 files changed, 130 insertions(+), 48 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH] ASoC: Intel: boards: Remove ignore_suspend for WoV streams
by Yong Zhi 19 Jan '17

19 Jan '17

When Ref capture is used during S0IX, only the DSP pipelines are needed, thus remove the ignore_suspend for WoV streams so that DMA can be suspended, but keep them for WoV endpoints. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 1b4330c..02439ac 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -357,7 +357,6 @@ static int broxton_refcap_startup(struct snd_pcm_substream *substream) .platform_name = "0000:00:0e.0", .init = NULL, .dpcm_capture = 1, - .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, .ops = &broxton_refcap_ops, -- 1.9.1

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[alsa-devel] DMA circular buffers - little help wanted
by Petr Kulhavy 19 Jan '17

19 Jan '17

Hi, I'm trying to write a sound card driver and need a bit of help understanding the logic behind the DMA circular buffers. I can understand there is a buffer of runtime->buffer_size frames, with two pointers: runtime->control->appl_ptr and runtime->status->hw_ptr pointing into it (they are actually offsets from the buffer start). One pointer is the producer, the other consumer. So far a standard circular buffer as I know it. Please correct me if I'm wrong. But what is runtime->boundary good for? It seems the arithmetics is not module buffer_size but modulo boundary, which is pretty confusing me. According to the code and comments some values (like buffer "pointers", available space, free, space, etc.) are supposed to be within 0 ... boundary - 1 and some within 0 ... buffer_size - 1. Not clear to me what is what and why. Could someone please help me to understand the logic behind? The other thing I would like to understand, how is the data positioned in time in terms of synchronisation? I mean where in the buffer, relative to the hw_ptr, is the frame that is supposed to play "now", for an arbitrary "now". Thanks Petr

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[alsa-devel] [PATCH RFC] ASoC: rt5663: use msleep() for uncritical delay
by Nicholas Mc Guire 19 Jan '17

19 Jan '17

The delay here does not seem to be critical with respect to longer delays than 10ms as this delay is to ensure that the write took effect before the next soc_update_bits/write call only, thus a high resolution timer makes little sense here - msleep() should do. Fixes: commit df7c52168ee1 ("ASoC: add rt5663 codec driver") Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Problem was found by cocinelle script. msleep() of 10ms can take up to 20ms on HZ=100 systems but even this does not seem critical here as this is not a very frequent operation (that is not called in a loop) In any case the delta of only 5 us seems unnecessarily small given that jitter on almost all systems would be expected to be larger than 5us and the microsecond precision really is not needed here. This does throw a checkpatch warning with: WARNING: msleep < 20ms can sleep for up to 20ms;" but that seems ok in this case, presumably there are very few systems running audio subsystem that have HZ=100 set. Patch was compile tested with: x86_64_defconfig + CONFIG_COMPILE_TEST=y SND_SOC=m + SND_SOC_ALL_CODECS=m (implies SND_SOC_RT5663=m) Patch is aginast 4.10-rc3 (localversion-next is next-20170111) sound/soc/codecs/rt5663.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index ed94830..2ecc600 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -2764,7 +2764,7 @@ static int rt5663_set_bias_level(struct snd_soc_codec *codec, RT5663_PWR_FV1_MASK | RT5663_PWR_FV2_MASK | RT5663_PWR_MB_MASK, RT5663_PWR_VREF1 | RT5663_PWR_VREF2 | RT5663_PWR_MB); - usleep_range(10000, 10005); + msleep(10); if (rt5663->codec_ver == CODEC_VER_1) { snd_soc_update_bits(codec, RT5663_SIG_CLK_DET, RT5663_EN_ANA_CLK_DET_MASK | -- 2.1.4

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