August 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 0/4] wm8904: adapt driver for use with audio-graph-card
by Michał Mirosław 29 Aug '19

29 Aug '19

This series allows to use WM8904 codec as audio-graph-card component. It starts with rework of FLL handling in the codec's driver, and as an example includes (untested) rework for codec with similar FLL: WM8994. Series based on tiwai/sound/for-next tree. You can also pull from: https://rere.qmqm.pl/git/linux branch: wm8904 (branch includes two fixes already sent to alsa-devel, but not merged yet). Michał Mirosław (4): ASoC: wm_fll: extract common code for Wolfson FLLs ASoC: wm8904: use common FLL code ASoC: wm8904: automatically choose clock source [RFT] ASoC: wm8994: use common FLL code sound/soc/atmel/atmel_wm8904.c | 11 +- sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8904.c | 516 +++++++++++--------------------- sound/soc/codecs/wm8904.h | 5 - sound/soc/codecs/wm8994.c | 281 +++++------------- sound/soc/codecs/wm8994.h | 4 +- sound/soc/codecs/wm_fll.c | 518 +++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_fll.h | 60 ++++ 9 files changed, 849 insertions(+), 557 deletions(-) create mode 100644 sound/soc/codecs/wm_fll.c create mode 100644 sound/soc/codecs/wm_fll.h -- 2.20.1

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[alsa-devel] [PATCH] ASoC: cs42xx8: Force suspend/resume during system suspend/resume
by Shengjiu Wang 29 Aug '19

29 Aug '19

Use force_suspend/resume to make sure clocks are disabled/enabled accordingly during system suspend/resume. Signed-off-by: Dong Aisheng <aisheng.dong(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/cs42xx8.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 5b049fcdba20..94b1adb088fd 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -684,6 +684,8 @@ static int cs42xx8_runtime_suspend(struct device *dev) #endif const struct dev_pm_ops cs42xx8_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) SET_RUNTIME_PM_OPS(cs42xx8_runtime_suspend, cs42xx8_runtime_resume, NULL) }; EXPORT_SYMBOL_GPL(cs42xx8_pm); -- 2.21.0

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[alsa-devel] [PATCH v2 00/25] ASoC: cleanup patches for soc-core
by Kuninori Morimoto 29 Aug '19

29 Aug '19

Hi Mark These are v2 of soc-core cleanup patches. I removed 3 patches which are reviewd from Ranjani / Pierre-Louis. Kuninori Morimoto (25): ASoC: soc-core: use device_register() ASoC: soc-core: set component->debugfs_root NULL ASoC: soc-core: add comment for for_each_xxx ASoC: soc-core: check return value of snd_soc_add_dai_link() ASoC: soc-core: don't use for_each_card_links_safe() at snd_soc_find_dai_link() ASoC: soc-core: reuse rtdcom at snd_soc_rtdcom_add() ASoC: soc-core: tidyup for snd_soc_dapm_new_controls() ASoC: soc-core: tidyup for snd_soc_add_component_controls() ASoC: soc-core: tidyup for snd_soc_dapm_add_routes() ASoC: soc-core: tidyup for snd_soc_add_card_controls() ASoC: soc-core: call snd_soc_dapm_debugfs_init() at soc_init_card_debugfs() ASoC: soc-core: remove unneeded list_empty() check for snd_soc_try_rebind_card() ASoC: soc-core: tidyup for card->deferred_resume_work ASoC: soc-core: define soc_dpcm_debugfs_add() for non CONFIG_DEBUG_FS ASoC: soc-core: dai_link check under soc_dpcm_debugfs_add() ASoC: soc-core: merge snd_soc_initialize_card_lists() ASoC: soc-core: remove unneeded dai_link check from snd_soc_remove_dai_link() ASoC: soc-core: add NOTE to snd_soc_rtdcom_lookup() ASoC: soc-core: don't call snd_soc_component_set_jack() ASoC: soc-core: don't alloc memory carelessly when creating debugfs ASoC: soc-core: soc_cleanup_card_resources() become void ASoC: soc-core: initialize component list ASoC: soc-core: initialize list at one place ASoC: soc-dai: use bit field for bus_control ASoC: soc-topology: use for_each_component_dais() at remove_dai() include/sound/soc-dai.h | 3 +- include/sound/soc-dpcm.h | 9 ++- include/sound/soc.h | 15 +--- sound/soc/soc-core.c | 197 ++++++++++++++++++++++++----------------------- sound/soc/soc-pcm.c | 3 + sound/soc/soc-topology.c | 2 +- 6 files changed, 118 insertions(+), 111 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda - Fix potential endless loop at applying quirks
by Takashi Iwai 29 Aug '19

29 Aug '19

Since the chained quirks via chained_before flag is applied before the depth check, it may lead to the endless recursive calls, when the chain were set up incorrectly. Fix it by moving the depth check at the beginning of the loop. Fixes: 1f57825077dc ("ALSA: hda - Add chained_before flag to the fixup entry") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_auto_parser.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 8a134f750bae..2c6d2becfe1a 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -824,6 +824,8 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) while (id >= 0) { const struct hda_fixup *fix = codec->fixup_list + id; + if (++depth > 10) + break; if (fix->chained_before) apply_fixup(codec, fix->chain_id, action, depth + 1); @@ -863,8 +865,6 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) } if (!fix->chained || fix->chained_before) break; - if (++depth > 10) - break; id = fix->chain_id; } } -- 2.16.4

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[alsa-devel] Options for ASYNC feedback source in USB-audio gadget (USB OTG)?
by Pavel Hofman 28 Aug '19

28 Aug '19

Hi, The existing usb-audio v.2 gadget defines its EP OUT as asynchronous https://github.com/torvalds/linux/blob/master/drivers/usb/gadget/function/f… , but does not define any feedback IN EP (which is against USB audio class 2.0 specs). Windows 10 USB audio class 2 driver requires presence of the feedback EP. By changing the EP OUT to adaptive (which in fact it is), we could fix the driver issue and windows plays through such device (RPi4) correctly now (linux/OSX already worked OK before). We will post a patch soon. RPi can now be used as an "intelligent" USB soundcard, allowing further processing in linux on RPi, before outputting to some final sound device. However, there are two clock domains involved - the host USB controller sending stream to the USB gadget, and the output soundcard of the gadget. While adaptive resampling is possible (jackd, PA do that), IMO the correct solution is going back to ASYNC EP OUT in the audio gadget and implementing a proper feedback EP. But the feedback EP needs input data. The whole chain behind the virtual capture alsa device defined by the gadget driver is clocked by the playback soundcard. May I ask for your opinions on possible options for 1) gathering the feedback data from the playback chain 2) passing the gathered samplerate-diff data to the gadget module for the feedback EP to send to the USB host A general solution would be very useful, at best without using alsa-lib defined PCM devices (e.g. Java still supports only hardware devices (hw:X), not PCM devices defined by alsa-lib). I very much appreciate your suggestions and help. Best regards, Pavel.

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[alsa-devel] [PATCH] ALSA: hda - Update descriptions about new position_fix values
by Takashi Iwai 28 Aug '19

28 Aug '19

The new values for the recent Intel and AMD chips are missing in the documentation. Add the new descriptions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa-configuration.rst | 2 ++ Documentation/sound/hd-audio/notes.rst | 5 +++++ 2 files changed, 7 insertions(+) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 4a3cecc8ad38..02aacd69ab96 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -1001,6 +1001,8 @@ position_fix 2 = POSBUF: use position buffer, 3 = VIACOMBO: VIA-specific workaround for capture, 4 = COMBO: use LPIB for playback, auto for capture stream + 5 = SKL+: apply the delay calculation available on recent Intel chips + 6 = FIFO: correct the position with the fixed FIFO size, for recent AMD chips probe_mask Bitmask to probe codecs (default = -1, meaning all slots); When the bit 8 (0x100) is set, the lower 8 bits are used diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst index 9f7347830ba4..0f3109d9abc8 100644 --- a/Documentation/sound/hd-audio/notes.rst +++ b/Documentation/sound/hd-audio/notes.rst @@ -66,6 +66,11 @@ by comparing both LPIB and position-buffer values. ``position_fix=4`` is another combination available for all controllers, and uses LPIB for the playback and the position-buffer for the capture streams. +``position_fix=5`` is specific to Intel platforms, so far, for Skylake +and onward. It applies the delay calculation for the precise position +reporting. +``position_fix=6`` is to correct the position with the fixed FIFO +size, mainly targeted for the recent AMD controllers. 0 is the default value for all other controllers, the automatic check and fallback to LPIB as described in the above. If you get a problem of repeated sounds, this option might -- 2.16.4

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[alsa-devel] [PATCH v2 0/3] ASoC: mediatek: mt8183: fix mt8183-mt6358 tdm out issues
by Jiaxin Yu 28 Aug '19

28 Aug '19

v2: 1. Base on branch asoc-next that merged branch 'asoc-5.4'. 2. Split the PATCH 2/2 that in v1 into three patches. v1: 1. Base on branch asoc-next that merged branch 'asoc-5.3'. 2. Mark suggest the patch would be split up into separate patches, each fixing single issue. 3. PATCH 1/2 has been applied to the asoc tree 'asoc-5.4'. Jiaxin Yu (3): ASoC: mediatek: mt8183: fix tdm out data align issue ASoC: mediatek: mt8183: tdm hw support tdm out and 8ch i2s out ASoC: mediatek: mt8183: fix tdm out data is valid on rising edge sound/soc/mediatek/common/mtk-afe-fe-dai.c | 3 +- sound/soc/mediatek/mt8183/mt8183-dai-tdm.c | 177 ++++++++++++++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 3 + sound/soc/mediatek/mt8183/mt8183-reg.h | 8 +- 4 files changed, 152 insertions(+), 39 deletions(-) -- 2.18.0

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[alsa-devel] [PATCH v2 0/4] ALSA: pcm: add support for 352.8KHz and 384KHz sample rate
by Srinivas Kandagatla 28 Aug '19

28 Aug '19

This patchset adds missing support to 352.8KHz and 384KHz sample rates in Qualcomm WCD9335 codec and QDSP dais. First patch adds these new rates to known list of rates in pcm core and also adds defines in pcm.h so that drivers can use it. Changes since v1: - Added wcd and qdsp users of these new rates. Srinivas Kandagatla (3): ASoC: wcd9335: Fix primary interpolator max rate ASoC: qdsp6: q6afe-dai: Update max rate for slim and tdm dais ASoC: qdsp6: q6asm-dai: fix max rates on q6asm dais Vidyakumar Athota (1): ALSA: pcm: add support for 352.8KHz and 384KHz sample rate include/sound/pcm.h | 5 ++ sound/core/pcm_native.c | 2 +- sound/soc/codecs/wcd9335.c | 20 ++++--- sound/soc/qcom/qdsp6/q6afe-dai.c | 92 +++++++++++--------------------- sound/soc/qcom/qdsp6/q6asm-dai.c | 18 +++---- 5 files changed, 59 insertions(+), 78 deletions(-) -- 2.21.0

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[alsa-devel] [PATCH 1/5] dt-bindings: sound: sun4i-spdif: Fix dma-names warning
by Maxime Ripard 28 Aug '19

28 Aug '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> Even though the H6 compatible has been properly added, the exeption for the number of DMA channels hasn't been updated, leading in a validation warning. Fix this. Fixes: b20453031472 ("dt-bindings: sound: sun4i-spdif: Add Allwinner H6 compatible") Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- .../devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml index e0284d8c3b63..38d4cede0860 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml @@ -70,7 +70,9 @@ allOf: properties: compatible: contains: - const: allwinner,sun8i-h3-spdif + enum: + - allwinner,sun8i-h3-spdif + - allwinner,sun50i-h6-spdif then: properties: -- 2.21.0

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[alsa-devel] [PATCH 2/2] ASoC: rt1011: ADCDAT pin config modification
by shumingf＠realtek.com 28 Aug '19

28 Aug '19

From: Shuming Fan <shumingf(a)realtek.com> ADCDAT pin config sets to output mode in default. And the driver creates the kcontrol to control ADCDAT pin configuration. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index f9235764ad32..52c17450d5a0 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1029,6 +1029,14 @@ static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc1_1_enum, RT1011_TDM1_SET_3, 6, static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc2_1_enum, RT1011_TDM1_SET_3, 4, rt1011_tdm_adc_swap_select); +static const char * const rt1011_adcdat_pin_config_select[] = { + "Output", "Input" +}; +static SOC_ENUM_SINGLE_DECL(rt1011_adcdat1_pin_config_enum, RT1011_TDM_TOTAL_SET, 4, + rt1011_adcdat_pin_config_select); +static SOC_ENUM_SINGLE_DECL(rt1011_adcdat2_pin_config_enum, RT1011_TDM_TOTAL_SET, 3, + rt1011_adcdat_pin_config_select); + static void rt1011_reset(struct regmap *regmap) { regmap_write(regmap, RT1011_RESET, 0); @@ -1343,6 +1351,8 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = { SOC_ENUM("I2S ADC DOUT Mode", rt1011_adc_dout_mode_enum), SOC_ENUM("TDM1 DOUT Length", rt1011_tdm1_dout_len_enum), SOC_ENUM("TDM2 DOUT Length", rt1011_tdm2_dout_len_enum), + SOC_ENUM("ADCDAT1 Pin Config", rt1011_adcdat1_pin_config_enum), + SOC_ENUM("ADCDAT2 Pin Config", rt1011_adcdat2_pin_config_enum), /* Speaker/Receiver Mode */ SOC_SINGLE_EXT("RECV SPK Mode", SND_SOC_NOPM, 0, 1, 0, @@ -1859,9 +1869,6 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); snd_soc_component_update_bits(component, RT1011_TDM2_SET_2, RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); - snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, - RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, - RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); snd_soc_dapm_mutex_unlock(dapm); return ret; @@ -2207,6 +2214,11 @@ static void rt1011_calibration_work(struct work_struct *work) /* initial */ rt1011_reg_init(component); + + /* set ADCDAT to output mode */ + snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, + RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, + RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); } static int rt1011_i2c_probe(struct i2c_client *i2c, -- 2.23.0

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