April 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoc: nau8810: add AUX related dapm widgets and routes
by Seven Lee 29 Apr '20

29 Apr '20

This patch implements the following features: - AUX input for recording. - An input AUX output to SPK/MOUT. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 35 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/nau8810.h | 8 ++++++++ 2 files changed, 42 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index de26758c30a8..efe9fb3fb225 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -355,6 +355,8 @@ static const struct snd_kcontrol_new nau8810_snd_controls[] = { /* Speaker Output Mixer */ static const struct snd_kcontrol_new nau8810_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("AUX Bypass Switch", NAU8810_REG_SPKMIX, + NAU8810_AUXSPK_SFT, 1, 0), SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_SPKMIX, NAU8810_BYPSPK_SFT, 1, 0), SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_SPKMIX, @@ -363,6 +365,8 @@ static const struct snd_kcontrol_new nau8810_speaker_mixer_controls[] = { /* Mono Output Mixer */ static const struct snd_kcontrol_new nau8810_mono_mixer_controls[] = { + SOC_DAPM_SINGLE("AUX Bypass Switch", NAU8810_REG_MONOMIX, + NAU8810_AUXMOUT_SFT, 1, 0), SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_MONOMIX, NAU8810_BYPMOUT_SFT, 1, 0), SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_MONOMIX, @@ -371,6 +375,8 @@ static const struct snd_kcontrol_new nau8810_mono_mixer_controls[] = { /* PGA Mute */ static const struct snd_kcontrol_new nau8810_pgaboost_mixer_controls[] = { + SOC_DAPM_SINGLE("AUX PGA Switch", NAU8810_REG_ADCBOOST, + NAU8810_AUXBSTGAIN_SFT, 0x7, 0), SOC_DAPM_SINGLE("PGA Mute Switch", NAU8810_REG_PGAGAIN, NAU8810_PGAMT_SFT, 1, 1), SOC_DAPM_SINGLE("PMIC PGA Switch", NAU8810_REG_ADCBOOST, @@ -379,6 +385,8 @@ static const struct snd_kcontrol_new nau8810_pgaboost_mixer_controls[] = { /* Input PGA */ static const struct snd_kcontrol_new nau8810_inpga[] = { + SOC_DAPM_SINGLE("AUX Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_AUXPGA_SFT, 1, 0), SOC_DAPM_SINGLE("MicN Switch", NAU8810_REG_INPUT_SIGNAL, NAU8810_NMICPGA_SFT, 1, 0), SOC_DAPM_SINGLE("MicP Switch", NAU8810_REG_INPUT_SIGNAL, @@ -401,6 +409,23 @@ static int check_mclk_select_pll(struct snd_soc_dapm_widget *source, return (value & NAU8810_CLKM_MASK); } +static int check_mic_enabled(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(source->dapm); + struct nau8810 *nau8810 = snd_soc_component_get_drvdata(component); + unsigned int value; + + regmap_read(nau8810->regmap, NAU8810_REG_INPUT_SIGNAL, &value); + if (value & NAU8810_PMICPGA_EN || value & NAU8810_NMICPGA_EN) + return 1; + regmap_read(nau8810->regmap, NAU8810_REG_ADCBOOST, &value); + if (value & NAU8810_PMICBSTGAIN_MASK) + return 1; + return 0; +} + static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Speaker Mixer", NAU8810_REG_POWER3, NAU8810_SPKMX_EN_SFT, 0, &nau8810_speaker_mixer_controls[0], @@ -425,6 +450,8 @@ static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Input Boost Stage", NAU8810_REG_POWER2, NAU8810_BST_EN_SFT, 0, nau8810_pgaboost_mixer_controls, ARRAY_SIZE(nau8810_pgaboost_mixer_controls)), + SND_SOC_DAPM_PGA("AUX Input", NAU8810_REG_POWER1, + NAU8810_AUX_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Bias", NAU8810_REG_POWER1, NAU8810_MICBIAS_EN_SFT, 0, NULL, 0), @@ -434,6 +461,7 @@ static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, &nau8810_loopback), + SND_SOC_DAPM_INPUT("AUX"), SND_SOC_DAPM_INPUT("MICN"), SND_SOC_DAPM_INPUT("MICP"), SND_SOC_DAPM_OUTPUT("MONOOUT"), @@ -445,10 +473,12 @@ static const struct snd_soc_dapm_route nau8810_dapm_routes[] = { {"DAC", NULL, "PLL", check_mclk_select_pll}, /* Mono output mixer */ + {"Mono Mixer", "AUX Bypass Switch", "AUX Input"}, {"Mono Mixer", "PCM Playback Switch", "DAC"}, {"Mono Mixer", "Line Bypass Switch", "Input Boost Stage"}, /* Speaker output mixer */ + {"Speaker Mixer", "AUX Bypass Switch", "AUX Input"}, {"Speaker Mixer", "PCM Playback Switch", "DAC"}, {"Speaker Mixer", "Line Bypass Switch", "Input Boost Stage"}, @@ -463,13 +493,16 @@ static const struct snd_soc_dapm_route nau8810_dapm_routes[] = { /* Input Boost Stage */ {"ADC", NULL, "Input Boost Stage"}, {"ADC", NULL, "PLL", check_mclk_select_pll}, + {"Input Boost Stage", "AUX PGA Switch", "AUX Input"}, {"Input Boost Stage", "PGA Mute Switch", "Input PGA"}, {"Input Boost Stage", "PMIC PGA Switch", "MICP"}, /* Input PGA */ - {"Input PGA", NULL, "Mic Bias"}, + {"Input PGA", NULL, "Mic Bias", check_mic_enabled}, + {"Input PGA", "AUX Switch", "AUX Input"}, {"Input PGA", "MicN Switch", "MICN"}, {"Input PGA", "MicP Switch", "MICP"}, + {"AUX Input", NULL, "AUX"}, /* Digital Looptack */ {"Digital Loopback", "Switch", "ADC"}, diff --git a/sound/soc/codecs/nau8810.h b/sound/soc/codecs/nau8810.h index 1ada31883dc6..6a7cacbe044a 100644 --- a/sound/soc/codecs/nau8810.h +++ b/sound/soc/codecs/nau8810.h @@ -69,6 +69,7 @@ /* NAU8810_REG_POWER1 (0x1) */ #define NAU8810_DCBUF_EN (0x1 << 8) +#define NAU8810_AUX_EN_SFT 6 #define NAU8810_PLL_EN_SFT 5 #define NAU8810_MICBIAS_EN_SFT 4 #define NAU8810_ABIAS_EN (0x1 << 3) @@ -228,7 +229,10 @@ /* NAU8810_REG_INPUT_SIGNAL (0x2C) */ #define NAU8810_PMICPGA_SFT 0 +#define NAU8810_PMICPGA_EN (0x1 << NAU8810_PMICPGA_SFT) #define NAU8810_NMICPGA_SFT 1 +#define NAU8810_NMICPGA_EN (0x1 << NAU8810_NMICPGA_SFT) +#define NAU8810_AUXPGA_SFT 2 /* NAU8810_REG_PGAGAIN (0x2D) */ #define NAU8810_PGAGAIN_SFT 0 @@ -236,12 +240,15 @@ #define NAU8810_PGAZC_SFT 7 /* NAU8810_REG_ADCBOOST (0x2F) */ +#define NAU8810_AUXBSTGAIN_SFT 0 #define NAU8810_PMICBSTGAIN_SFT 4 +#define NAU8810_PMICBSTGAIN_MASK (0x7 << NAU8810_PMICBSTGAIN_SFT) #define NAU8810_PGABST_SFT 8 /* NAU8810_REG_SPKMIX (0x32) */ #define NAU8810_DACSPK_SFT 0 #define NAU8810_BYPSPK_SFT 1 +#define NAU8810_AUXSPK_SFT 5 /* NAU8810_REG_SPKGAIN (0x36) */ #define NAU8810_SPKGAIN_SFT 0 @@ -251,6 +258,7 @@ /* NAU8810_REG_MONOMIX (0x38) */ #define NAU8810_DACMOUT_SFT 0 #define NAU8810_BYPMOUT_SFT 1 +#define NAU8810_AUXMOUT_SFT 2 #define NAU8810_MOUTMXMT_SFT 6 -- 2.25.2

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[PATCH] ASoC: mxs-saif: Fix unused assignment
by Tang Bin 29 Apr '20

29 Apr '20

Delete unused initialized value, because 'ret' will be assigined by the function of_alias_get_id(). Signed-off-by: Zhang Shengju <zhangshengju(a)cmss.chinamobile.com> Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/mxs/mxs-saif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 1e38ce858..dc197883e 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -733,7 +733,7 @@ static int mxs_saif_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct mxs_saif *saif; - int irq, ret = 0; + int irq, ret; struct device_node *master; if (!np) -- 2.20.1.windows.1

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[PATCH] ASoC: fsl_easrc: mark PM functions __maybe_unused
by Arnd Bergmann 29 Apr '20

29 Apr '20

ifdefs are hard, and in this driver the suspend/resume functions are the only callers of some other helpers that trigger a harmless warning when CONFIG_PM is disabled: sound/soc/fsl/fsl_easrc.c:1807:12: warning: 'fsl_easrc_get_firmware' defined but not used [-Wunused-function] 1807 | static int fsl_easrc_get_firmware(struct fsl_asrc *easrc) | ^~~~~~~~~~~~~~~~~~~~~~ sound/soc/fsl/fsl_easrc.c:303:12: warning: 'fsl_easrc_resampler_config' defined but not used [-Wunused-function] 303 | static int fsl_easrc_resampler_config(struct fsl_asrc *easrc) | ^~~~~~~~~~~~~~~~~~~~~~~~~~ Remove the #ifdef and just mark the callers as __maybe_unused to suppress the warnings altogether. Fixes: 955ac624058f ("ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/fsl/fsl_easrc.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 20326bffab64..c6b5eb2d2af7 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1997,8 +1997,7 @@ static int fsl_easrc_remove(struct platform_device *pdev) return 0; } -#ifdef CONFIG_PM -static int fsl_easrc_runtime_suspend(struct device *dev) +static __maybe_unused int fsl_easrc_runtime_suspend(struct device *dev) { struct fsl_asrc *easrc = dev_get_drvdata(dev); struct fsl_easrc_priv *easrc_priv = easrc->private; @@ -2015,7 +2014,7 @@ static int fsl_easrc_runtime_suspend(struct device *dev) return 0; } -static int fsl_easrc_runtime_resume(struct device *dev) +static __maybe_unused int fsl_easrc_runtime_resume(struct device *dev) { struct fsl_asrc *easrc = dev_get_drvdata(dev); struct fsl_easrc_priv *easrc_priv = easrc->private; @@ -2094,7 +2093,6 @@ static int fsl_easrc_runtime_resume(struct device *dev) clk_disable_unprepare(easrc->mem_clk); return ret; } -#endif /* CONFIG_PM */ static const struct dev_pm_ops fsl_easrc_pm_ops = { SET_RUNTIME_PM_OPS(fsl_easrc_runtime_suspend, -- 2.26.0

3 2

[PATCH] ASoC: component: suppress uninitialized-variable warning
by Arnd Bergmann 29 Apr '20

29 Apr '20

Old versions of gcc (tested on gcc-4.8) produce a warning for correct code: sound/soc/soc-compress.c: In function 'soc_compr_open': sound/soc/soc-compress.c:75:28: error: 'component' is used uninitialized in this function [-Werror=uninitialized] struct snd_soc_component *component, *save = NULL; Change the for_each_rtd_components() macro to ensure 'component' gets initialized to a value the compiler does not complain about. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 907afbc652f2..69a82487fa9b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1179,7 +1179,7 @@ struct snd_soc_pcm_runtime { #define asoc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->num_cpus] #define for_each_rtd_components(rtd, i, component) \ - for ((i) = 0; \ + for ((i) = 0, component = NULL; \ ((i) < rtd->num_components) && ((component) = rtd->components[i]);\ (i)++) #define for_each_rtd_cpu_dais(rtd, i, dai) \ -- 2.26.0

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[PATCH -next] ALSA: seq: oss: remove unused inline function snd_seq_oss_timer_is_realtime
by YueHaibing 29 Apr '20

29 Apr '20

There's no callers in-tree. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/core/seq/oss/seq_oss_timer.h | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_timer.h b/sound/core/seq/oss/seq_oss_timer.h index 2d86125b5d0f..dee190b4ec6b 100644 --- a/sound/core/seq/oss/seq_oss_timer.h +++ b/sound/core/seq/oss/seq_oss_timer.h @@ -44,14 +44,4 @@ snd_seq_oss_timer_cur_tick(struct seq_oss_timer *timer) return timer->cur_tick; } - -/* - * is realtime event? - */ -static inline int -snd_seq_oss_timer_is_realtime(struct seq_oss_timer *timer) -{ - return timer->realtime; -} - #endif -- 2.17.1

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[RFT PATCH 0/7] Add Allwinner H3/H5/A64 HDMI audio
by Clément Péron 29 Apr '20

29 Apr '20

Hi, This is some work done by Marcus Cooper and Jernej Škrabec. These patches are present in LibreElec kernel and have been tested by LE users. I have rework them to follow kernel rules but I can't test them as I don't have H3, H5 or A64 boards. If some sunxi users could confirm they worked as expected! This serie goes in top off "Add H6 I2S support"[0]. You can find a branch for testing here[1]. Thanks, Clement 0: https://lwn.net/Articles/817914/ 1: https://github.com/clementperon/linux/tree/allwinner_hdmi Marcus Cooper (7): arm: dts: sunxi: h3/h5: Add DAI node for HDMI arm: dts: sunxi: h3/h5: Add HDMI audio arm64: dts: allwinner: a64: Add DAI node for HDMI arm64: dts: allwinner: a64: Add HDMI audio arm: sun8i: h3: Add HDMI audio to Orange Pi 2 arm: sun8i: h3: Add HDMI audio to Beelink X2 arm64: dts: allwinner: a64: Add HDMI audio to Pine64 arch/arm/boot/dts/sun8i-h3-beelink-x2.dts | 8 +++++ arch/arm/boot/dts/sun8i-h3-orangepi-2.dts | 8 +++++ arch/arm/boot/dts/sunxi-h3-h5.dtsi | 34 ++++++++++++++++++ .../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 +++++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 35 +++++++++++++++++++ 5 files changed, 93 insertions(+) -- 2.20.1

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underrun problems after setting parameters with snd_pcm_set_params()
by robert rozee 29 Apr '20

29 Apr '20

hi, i'm having some problems using ALSA to stream simple sounds on one particular netbook running EasyOS, a cut-down 64-bit linux distro that is rather slow. the problem only occurs with SHORT sound clips. i found that clips longer than 1111ms played without flaw. clips less that 500ms generally failed to play to the end, but there was no error code returned. clips between 501ms and 1111ms generated the error message: "ALSA lib pcm.c:8424:(snd_pcm_recover) underrun occurred". my buffer size is equivalent to 500ms. the code essentially does the following: ----------------------------------------------------------------- snd_pcm_open(@handle, "default" , SND_PCM_STREAM_PLAYBACK, 0); snd_pcm_set_params(handle, SND_PCM_FORMAT_U8, SND_PCM_ACCESS_RW_INTERLEAVED, 1, // number of channels 48000, // bitrate (bps) 1, // resampling on/off min(500000, sample_length)); // latency while (frames_left_to_write>0) { // // CODE IN HERE TO: refill buffer, and set frames_to_write_now appropriately // // send out current buffer content frames=snd_pcm_writei(handle, @buffer, frames_to_write_now); if frames<0 then frames=snd_pcm_recover(handle, frames, 0); if frames<0 then break; } ----------------------------------------------------------------- ******************************************************************************** the problem, as far as i can tell, is with snd_pcm_writei() failing to play the portion of the sound clip in the buffer, but returning NO error code. the error -EFILE is only generated on the attempt to write a SECOND buffer load. the sound sample i'm using is a 440Hz sine wave tone, amplitude of +/-100, centred around 128. ends are feathered in/out, so any short playback is very obvious. after doing a bit of a search on similar problems, i'd like to speculate that snd_pcm_set_params() is choosing an inappropriate small period size. for low-end computers this is causing the underrun issue. if snd_pcm_set_params() is used for configuration, is there any way to change the period size after the call? can it be set between opening the handle and setting the other parameters? cheers, rob :-)

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[PATCH v2] soc: imx8m: Make imx8m_dsp_ops static
by ChenTao 29 Apr '20

29 Apr '20

Fix the following warning: sound/soc/sof/imx/imx8m.c:95:20: warning: symbol 'imx8m_dsp_ops' was not declared. Should it be static? Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: ChenTao <chentao107(a)huawei.com> Acked-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- v1->v2: - add recipient broonie(a)kernel.org alsa-devel(a)alsa-project.org sound/soc/sof/imx/imx8m.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index 07451ba4efae..1a5b0f9ebac1 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -92,7 +92,7 @@ static void imx8m_dsp_handle_request(struct imx_dsp_ipc *ipc) snd_sof_ipc_msgs_rx(priv->sdev); } -struct imx_dsp_ops imx8m_dsp_ops = { +static struct imx_dsp_ops imx8m_dsp_ops = { .handle_reply = imx8m_dsp_handle_reply, .handle_request = imx8m_dsp_handle_request, }; -- 2.22.0

1 0

[PATCH v2 0/3] add channel constraint for BDW machine drivers
by Brent Lu 28 Apr '20

28 Apr '20

The machine driver bdw-rt5650 (for Google buddy) supports 2 or 4-channel recording while other two drivers support only 2-channel recording. HW constraints are implemented to reflect the hardware limitation on BDW platform. Changes since v1: - Change the patch title. - Remove the DUAL_CHANNEL and QUAD_CHANNEL macros which are too obvious. - Follow the naming convertion, using 'bdw_rt5650_' and 'bdw_rt5677_' to name startup functions. - Refine the comments in startup functions. - Redesign the bdw_rt5650_fe_startup() function for readability. - Add an assignment to initialize runtime->hw.channels_max variable. Brent Lu (3): ASoC: bdw-rt5677: add channel constraint ASoC: bdw-rt5650: add channel constraint ASoC: broadwell: add channel constraint sound/soc/intel/boards/bdw-rt5650.c | 29 +++++++++++++++++++++++++++++ sound/soc/intel/boards/bdw-rt5677.c | 26 ++++++++++++++++++++++++++ sound/soc/intel/boards/broadwell.c | 26 ++++++++++++++++++++++++++ 3 files changed, 81 insertions(+) -- 2.7.4

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[PATCH -next] ASoC: tegra: tegra_wm8903: Use devm_snd_soc_register_card()
by Wei Yongjun 28 Apr '20

28 Apr '20

Using devm_snd_soc_register_card() can make the code shorter and cleaner. Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/tegra/tegra_wm8903.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 7bf159965c4d..d3ead0213cef 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -351,9 +351,9 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev) if (ret) return ret; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + dev_err(&pdev->dev, "devm_snd_soc_register_card failed (%d)\n", ret); return ret; } @@ -361,15 +361,6 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev) return 0; } -static int tegra_wm8903_driver_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static const struct of_device_id tegra_wm8903_of_match[] = { { .compatible = "nvidia,tegra-audio-wm8903", }, {}, @@ -382,7 +373,6 @@ static struct platform_driver tegra_wm8903_driver = { .of_match_table = tegra_wm8903_of_match, }, .probe = tegra_wm8903_driver_probe, - .remove = tegra_wm8903_driver_remove, }; module_platform_driver(tegra_wm8903_driver);

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