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

[alsa-devel] [PATCH] ASoC: sun4i-i2s: incorrect regmap for A83t
by codekipper＠gmail.com 02 Sep '19

02 Sep '19

From: Marcus Cooper <codekipper(a)gmail.com> Fixes: 21faaea1343f ("ASoC: sun4i-i2s: Add support for A83T") Signed-off-by: Marcus Cooper <codekipper(a)gmail.com> --- sound/soc/sunxi/sun4i-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index cbc6e59aa089..056a299c03fb 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1109,7 +1109,7 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = { static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, - .sun4i_i2s_regmap = &sun4i_i2s_regmap_config, + .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), -- 2.23.0

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[alsa-devel] [PATCH] MAINTAINERS: update entry of firewire audio drivers
by Takashi Sakamoto 02 Sep '19

02 Sep '19

This commit adds myself as one of maintainers for firewire audio drivers and IEC 61883-1/6 packet streaming engine. I call them ALSA firewire stack as a whole. 6 years ago I joined in development for this category of drivers with heavy reverse-engineering tasks and over 100 models are now available from ALSA applications. IEEE 1394 bus itself and units on the bus are enough legacy but the development still continues. I have a plan to add drastic enhancement in kernel v5.5 and v5.6 period. This commit adds myself into MAINTAINERS so that developers and users can easily find active developer to post their issues, especially for regression. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- MAINTAINERS | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 24e29b2e53c9..8929a2ec75f7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6264,8 +6264,9 @@ S: Maintained F: drivers/hwmon/f75375s.c F: include/linux/f75375s.h -FIREWIRE AUDIO DRIVERS +FIREWIRE AUDIO DRIVERS and IEC 61883-1/6 PACKET STREAMING ENGINE M: Clemens Ladisch <clemens(a)ladisch.de> ++M: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained -- 2.20.1

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[alsa-devel] [PATCH] drm: dw-hdmi-i2s: enable audio clock in audio_startup
by Cheng-Yi Chiang 02 Sep '19

02 Sep '19

In the designware databook, the sequence of enabling audio clock and setting format is not clearly specified. Currently, audio clock is enabled in the end of hw_param ops after setting format. On some monitors, there is a possibility that audio does not come out. Fix this by enabling audio clock in audio_startup ops before hw_param ops setting format. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index 5cbb71a866d5..08b4adbb1ddc 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -69,6 +69,14 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + return 0; +} + +static int dw_hdmi_i2s_audio_startup(struct device *dev, void *data) +{ + struct dw_hdmi_i2s_audio_data *audio = data; + struct dw_hdmi *hdmi = audio->hdmi; + dw_hdmi_audio_enable(hdmi); return 0; @@ -105,6 +113,7 @@ static int dw_hdmi_i2s_get_dai_id(struct snd_soc_component *component, } static struct hdmi_codec_ops dw_hdmi_i2s_ops = { + .audio_startup = dw_hdmi_i2s_audio_startup, .hw_params = dw_hdmi_i2s_hw_params, .audio_shutdown = dw_hdmi_i2s_audio_shutdown, .get_dai_id = dw_hdmi_i2s_get_dai_id, -- 2.23.0.187.g17f5b7556c-goog

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[alsa-devel] [PATCH 1/3] ASoC: es8316: judge PCM rate at later timing
by Katsuhiro Suzuki 31 Aug '19

31 Aug '19

This patch change the judge timing about playing/capturing PCM rate. Original code set constraints list of PCM rate limits at set_sysclk. This strategy works well if system is using fixed rate clock. But some boards and SoC (such as RockPro64 and RockChip I2S) has connected SoC MCLK out to ES8316 MCLK in. In this case, SoC side I2S will choose suitable frequency of MCLK such as fs * mclk-fs when user starts playing or capturing. Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - ES8316 set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 30.720kHz (1/400), 24kHz (1/512), 16kHz (1/768), 12kHz (1/1024) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk by 8.192MHz (= fs * mclk-fs = 32k * 256) - ES8316 set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 20.48kHz (1/400), 16kHz (1/512), 10.66kHz (1/768), 8kHz (1/1024) - Play 48kHz again, but it's NOT acceptable because constraints list does not allow 48kHz Root cause of this strange behavior is changing constraints list at set_sysclk timing. It seems that is too early to determine. So this patch does not use constraints list and check PCM rate limit more later timing at hw_params. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/codecs/es8316.c | 37 +++++++++++++++---------------------- 1 file changed, 15 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index ed2959dbe1fb..229808fa627c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -363,27 +363,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); - int i; - int count = 0; es8316->sysclk = freq; if (freq == 0) return 0; - /* Limit supported sample rates to ones that can be autodetected - * by the codec running in slave mode. - */ - for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { - const unsigned int ratio = supported_mclk_lrck_ratios[i]; - - if (freq % ratio == 0) - es8316->allowed_rates[count++] = freq / ratio; - } - - es8316->sysclk_constraints.list = es8316->allowed_rates; - es8316->sysclk_constraints.count = count; - return 0; } @@ -449,13 +434,6 @@ static int es8316_pcm_startup(struct snd_pcm_substream *substream, return -EINVAL; } - /* The set of sample rates that can be supported depends on the - * MCLK supplied to the CODEC. - */ - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &es8316->sysclk_constraints); - return 0; } @@ -466,12 +444,27 @@ static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); u8 wordlen = 0; + int i; if (!es8316->sysclk) { dev_err(component->dev, "No MCLK configured\n"); return -EINVAL; } + /* Limit supported sample rates to ones that can be autodetected + * by the codec running in slave mode. + */ + for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { + const unsigned int ratio = supported_mclk_lrck_ratios[i]; + + if (es8316->sysclk % ratio != 0) + continue; + if (es8316->sysclk / ratio == params_rate(params)) + break; + } + if (i == NR_SUPPORTED_MCLK_LRCK_RATIOS) + return -EINVAL; + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: wordlen = ES8316_SERDATA2_LEN_16; -- 2.23.0.rc1

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[alsa-devel] [PATCH 0/4] ASoC: mediatek: mt8183: some cleanups
by Tzung-Bi Shih 30 Aug '19

30 Aug '19

The series contains some cleanup refactors and should not change any behavior. Tzung-Bi Shih (4): ASoC: mediatek: mt8183: move private structure ASoC: mediatek: mt8183: move headset jack to card-specific storage ASoC: mediatek: mt8183: remove forward declaration of headset_init ASoC: mediatek: mt8183: fix space issues .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 55 +++++++++---------- 1 file changed, 26 insertions(+), 29 deletions(-) -- 2.23.0.187.g17f5b7556c-goog

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[alsa-devel] [PATCH v2 1/5] ASoC: dt-bindings: sun4i-spdif: Fix dma-names warning
by Maxime Ripard 30 Aug '19

30 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> --- Changes from v1: - Fix subject prefix --- .../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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Re: [alsa-devel] [PATCH 2/2] ASoC: rt1011: ADCDAT pin config modification
by Mark Brown 30 Aug '19

30 Aug '19

On Wed, Aug 28, 2019 at 11:01:21AM +0000, Shuming [范書銘] wrote: > > > ADCDAT pin config sets to output mode in default. > > > And the driver creates the kcontrol to control ADCDAT pin configuration. > > Why would this change at runtime? I'd expect this to be controlled by the > > machine driver or a DT property. > It will not change at runtime in the normal case. > I think we may use it to test the feedback signal in factory mode. So really this is setting up a loopback for testing?

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[alsa-devel] [PATCH 0/4] ASoC: ti: davinci: Fixes for daVinci boards with McBSP
by Peter Ujfalusi 30 Aug '19

30 Aug '19

Hi, Morimoto-san pointed out in a private thread that the davinci-i2s driver is using snd_soc_rtdcom_lookup(), which is should not really. It took a while to get access to a daVinci board which uses McBSP, but eventually I got access to a remote dm644x-evm and dma365-evm. Initially the audio was not even probing on them (fixes sent for mach-davinci), but then the driver was still crashing and not behaving correctly. With the 4 patch dm644x-evm have working audio and the offending snd_soc_rtdcom_lookup() is gone from the driver. dm365-evm still have some issues regarding to EDMA, but it looks like a system issue which can be fixed later. Regards, Peter --- Peter Ujfalusi (4): ASoC: ti: davinci-i2s: Add S32_LE as support format ASoC: ti: davinci-i2s: Move the XSYNCERR workaround to .prepare callback ASoC: ti: edma-pcm: Fix for legacy dma_slave_map based channel lookup ASoC: ti: davinci-evm: Do not fail if the dai_set_sysclk returns -ENOTSUPP sound/soc/ti/davinci-evm.c | 2 +- sound/soc/ti/davinci-i2s.c | 90 +++++++++++++++++--------------------- sound/soc/ti/edma-pcm.c | 17 ++++++- 3 files changed, 57 insertions(+), 52 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH v2] ASoC: Intel: Baytrail: Fix implicit fallthrough warning
by Cezary Rojewski 30 Aug '19

30 Aug '19

Append fallthrough statement to fix 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> --- Changes from v1: - made use of fallthrough statement as suggested by Pierre - updated commit message sound/soc/intel/baytrail/sst-baytrail-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index 9cbc982d46a9..54f2ee3010ee 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -193,6 +193,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_SUSPEND: pdata->restore_stream = false; + /* fallthrough */ 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] ASoC: fsl_ssi: Fix clock control issue in master mode
by Shengjiu Wang 30 Aug '19

30 Aug '19

The test case is arecord -Dhw:0 -d 10 -f S16_LE -r 48000 -c 2 temp.wav & aplay -Dhw:0 -d 30 -f S16_LE -r 48000 -c 2 test.wav There will be error after end of arecord: aplay: pcm_write:2051: write error: Input/output error Capture and Playback work in parallel in master mode, one substream stops, the other substream is impacted, the reason is that clock is disabled wrongly. The clock's reference count is not increased when second substream starts, the hw_param() function returns in the beginning because first substream is enabled, then in end of first substream, the hw_free() disables the clock. This patch is to move the clock enablement to the place before checking of the device enablement in hw_param(). Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_ssi.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b0a6fead1a6a..537dc69256f0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -799,15 +799,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, u32 wl = SSI_SxCCR_WL(sample_size); int ret; - /* - * SSI is properly configured if it is enabled and running in - * the synchronous mode; Note that AC97 mode is an exception - * that should set separate configurations for STCCR and SRCCR - * despite running in the synchronous mode. - */ - if (ssi->streams && ssi->synchronous) - return 0; - if (fsl_ssi_is_i2s_master(ssi)) { ret = fsl_ssi_set_bclk(substream, dai, hw_params); if (ret) @@ -823,6 +814,15 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } } + /* + * SSI is properly configured if it is enabled and running in + * the synchronous mode; Note that AC97 mode is an exception + * that should set separate configurations for STCCR and SRCCR + * despite running in the synchronous mode. + */ + if (ssi->streams && ssi->synchronous) + return 0; + if (!fsl_ssi_is_ac97(ssi)) { /* * Keep the ssi->i2s_net intact while having a local variable -- 2.21.0

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