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

[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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[alsa-devel] [PATCH] ASoC: Intel: Baytrail: Fix implicit fallthrough warning
by Cezary Rojewski 28 Aug '19

28 Aug '19

Separate _SUSPEND from _PAUSE_PUSH to ensure code is handled correctly while fixing warning reported during compilation. Fixes: b80d19c166c4 ("ASoC: Intel: Restore Baytrail ADSP streams only when ADSP was in reset") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index 9cbc982d46a9..eddb24ab8072 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -193,6 +193,8 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_SUSPEND: pdata->restore_stream = false; + sst_byt_stream_pause(byt, pcm_data->stream); + break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: sst_byt_stream_pause(byt, pcm_data->stream); break; -- 2.17.1

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[alsa-devel] [PATCH v2 1/2] ASoC: sun4i-i2s: Revert "ASoC: sun4i-i2s: Remove duplicated quirks structure"
by Maxime Ripard 28 Aug '19

28 Aug '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> This reverts commit 3e9acd7ac6933cdc20c441bbf9a38ed9e42e1490. It turns out that while one I2S controller is described in the A83t datasheet, the driver supports another, undocumented, one that has been inherited from the older SoCs, while the documented one uses the new design. Fixes: 3e9acd7ac693 ("ASoC: sun4i-i2s: Remove duplicated quirks structure") Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- Changes from v1: - Add the proper prefix to the commit title --- sound/soc/sunxi/sun4i-i2s.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 57bf2a33753e..a6a3f772fdf0 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1097,6 +1097,11 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = { .set_fmt = sun4i_i2s_set_soc_fmt, }; +/* + * This doesn't describe the TDM controller documented in the A83t + * datasheet, but the three undocumented I2S controller that use the + * older design. + */ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, @@ -1115,6 +1120,24 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .set_fmt = sun8i_i2s_set_soc_fmt, }; +static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = { + .has_reset = true, + .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, + .sun4i_i2s_regmap = &sun8i_i2s_regmap_config, + .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8), + .field_fmt_wss = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2), + .field_fmt_sr = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6), + .bclk_dividers = sun8i_i2s_clk_div, + .num_bclk_dividers = ARRAY_SIZE(sun8i_i2s_clk_div), + .mclk_dividers = sun8i_i2s_clk_div, + .num_mclk_dividers = ARRAY_SIZE(sun8i_i2s_clk_div), + .get_bclk_parent_rate = sun8i_i2s_get_bclk_parent_rate, + .get_sr = sun8i_i2s_get_sr_wss, + .get_wss = sun8i_i2s_get_sr_wss, + .set_chan_cfg = sun8i_i2s_set_chan_cfg, + .set_fmt = sun8i_i2s_set_soc_fmt, +}; + static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, @@ -1296,7 +1319,7 @@ static const struct of_device_id sun4i_i2s_match[] = { }, { .compatible = "allwinner,sun8i-h3-i2s", - .data = &sun8i_a83t_i2s_quirks, + .data = &sun8i_h3_i2s_quirks, }, { .compatible = "allwinner,sun50i-a64-codec-i2s", -- 2.21.0

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[alsa-devel] [PATCH 1/2] ALSA: usb-audio: DSD auto-detection for Playback Designs
by Jussi Laako 28 Aug '19

28 Aug '19

Add DSD support auto-detection for newer Playback Designs devices. Older device generations have a different USB interface implementation. Keep the auto-detection VID whitelist sorted. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7e9735aa7ac9..79fb24843d5b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1566,7 +1566,8 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, struct usb_interface *iface; /* Playback Designs */ - if (USB_ID_VENDOR(chip->usb_id) == 0x23ba) { + if (USB_ID_VENDOR(chip->usb_id) == 0x23ba && + USB_ID_PRODUCT(chip->usb_id) < 0x0110) { switch (fp->altsetting) { case 1: fp->dsd_dop = true; @@ -1654,8 +1655,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, * from XMOS/Thesycon */ switch (USB_ID_VENDOR(chip->usb_id)) { - case 0x20b1: /* XMOS based devices */ case 0x152a: /* Thesycon devices */ + case 0x20b1: /* XMOS based devices */ + case 0x23ba: /* Playback Designs */ case 0x25ce: /* Mytek devices */ case 0x2ab6: /* T+A devices */ if (fp->dsd_raw) -- 2.17.1

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[alsa-devel] [BUG] bdw-rt5650 DSP boot timeout
by Jon Flatley 28 Aug '19

28 Aug '19

I've been working on upstreaming the bdw-rt5650 machine driver for the Acer Chromebase 24 (buddy). There seems to be an issue when first setting the hardware controls that appears to be crashing the DSP: [ 51.424554] haswell-pcm-audio haswell-pcm-audio: FW loaded, mailbox readback FW info: type 01, - version: 00.00, build 77, source commit id: 876ac6906f31a43b6772b23c7c983ce9dcb18a19 ... [ 84.924666] haswell-pcm-audio haswell-pcm-audio: error: audio DSP boot timeout IPCD 0x0 IPCX 0x0 [ 85.260655] haswell-pcm-audio haswell-pcm-audio: ipc: --message timeout-- ipcx 0x83000000 isr 0x00000000 ipcd 0x00000000 imrx 0x7fff0000 [ 85.273609] haswell-pcm-audio haswell-pcm-audio: error: stream commit failed [ 85.279746] System PCM: error: failed to commit stream -110 [ 85.285388] haswell-pcm-audio haswell-pcm-audio: ASoC: haswell-pcm-audio hw params failed: -110 [ 85.293963] System PCM: ASoC: hw_params FE failed -110 This happens roughly 50% of the time when first setting hardware controls after a reboot. The other 50% of the time the DSP comes up just fine and audio works fine thereafter. Adding "#define DEBUG 1" to sound/soc/intel/haswell/sst-haswell-ipc.c makes the issue occur much less frequently in my testing. Seems like a subtle timing issue. There were timing issues encountered during the bringup of the 2015 chromebook pixel (samus) which uses the bdw-rt5677 machine driver. Those were slightly different, and manifested during repeated arecords. Both devices use the same revision of the sst2 firmware. Any ideas for how to debug this? Thanks, Jon

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