February 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: x86: hdmi: Add single_port option for compatible behavior
by Takashi Iwai 21 Feb '18

21 Feb '18

The recent support for the multiple PCM devices allowed user to use multiple HDMI/DP outputs, but at the same time, the PCM stream assignment has been changed, too. Due to that, the former PCM#0 (there was only one stream in the past) is likely assigned to a different one (e.g. PCM#2), and it ends up with the regression when user sticks with the fixed configuration using the device#0. Although the multiple monitor support shouldn't matter when user deploys the backend like PulseAudio that checks the jack detection state, the behavior change isn't always acceptable for some users. As a mitigation, this patch introduces an option to switch the behavior back to the old-good-days: when the new option, single_port=1, is passed, the driver creates only a single PCM device, and it's assigned to the first connected one, like the earlier versions did. The option is turned off as default still to support the multiple monitors. Fixes: 8a2d6ae1f737 ("ALSA: x86: Register multiple PCM devices for the LPE audio card") Reported-and-tested-by: Hubert Mantel <mantel(a)metadox.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/x86/intel_hdmi_audio.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index a0951505c7f5..96115c401292 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -50,6 +50,7 @@ /*standard module options for ALSA. This module supports only one card*/ static int hdmi_card_index = SNDRV_DEFAULT_IDX1; static char *hdmi_card_id = SNDRV_DEFAULT_STR1; +static bool single_port; module_param_named(index, hdmi_card_index, int, 0444); MODULE_PARM_DESC(index, @@ -57,6 +58,9 @@ MODULE_PARM_DESC(index, module_param_named(id, hdmi_card_id, charp, 0444); MODULE_PARM_DESC(id, "ID string for INTEL Intel HDMI Audio controller."); +module_param(single_port, bool, 0444); +MODULE_PARM_DESC(single_port, + "Single-port mode (for compatibility)"); /* * ELD SA bits in the CEA Speaker Allocation data block @@ -1579,7 +1583,11 @@ static irqreturn_t display_pipe_interrupt_handler(int irq, void *dev_id) static void notify_audio_lpe(struct platform_device *pdev, int port) { struct snd_intelhad_card *card_ctx = platform_get_drvdata(pdev); - struct snd_intelhad *ctx = &card_ctx->pcm_ctx[port]; + struct snd_intelhad *ctx; + + ctx = &card_ctx->pcm_ctx[single_port ? 0 : port]; + if (single_port) + ctx->port = port; schedule_work(&ctx->hdmi_audio_wq); } @@ -1816,7 +1824,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) init_channel_allocations(); card_ctx->num_pipes = pdata->num_pipes; - card_ctx->num_ports = pdata->num_ports; + card_ctx->num_ports = single_port ? 1 : pdata->num_ports; for_each_port(card_ctx, port) { struct snd_intelhad *ctx = &card_ctx->pcm_ctx[port]; @@ -1824,7 +1832,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) ctx->card_ctx = card_ctx; ctx->dev = card_ctx->dev; - ctx->port = port; + ctx->port = single_port ? -1 : port; ctx->pipe = -1; INIT_WORK(&ctx->hdmi_audio_wq, had_audio_wq); -- 2.16.1

2 2

[alsa-devel] [PATCH v2] ASoC: soc-generic-dmaengine-pcm: Add a DMA debugfs_prefix entry
by Fabio Estevam 21 Feb '18

21 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> After the conversion from platform to component the following warnings are seen: fsl-ssi-dai 2028000.ssi: ASoC: Failed to create component debugfs directory fsl-spdif-dai 2004000.spdif: ASoC: Failed to create component debugfs directory As explained by Mark Brown: "It's a legit warning - we shouldn't really be creating two components for the same device. However this is a bit of a corner case as it's the dmaengine platform driver which is kind of a virtual device, it's not really the device that we use with DMA that's being represented but rather the link between that and the DMA controller." Disambiguate the DMA component name by adding a "dma" debugfs_prefix entry. Suggested-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Fixed typo in commit log "createcomponent" sound/soc/soc-generic-dmaengine-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index c07d5c7..768247f 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -442,6 +442,9 @@ int snd_dmaengine_pcm_register(struct device *dev, if (!pcm) return -ENOMEM; +#ifdef CONFIG_DEBUG_FS + pcm->component.debugfs_prefix = "dma"; +#endif pcm->config = config; pcm->flags = flags; -- 2.7.4

1 0

[alsa-devel] ASoC: Failed to create component debugfs directory
by Fabio Estevam 21 Feb '18

21 Feb '18

Hi, On linux-next I see the following errors on imx6 wandboard: [ 3.040897] fsl-ssi-dai 2028000.ssi: ASoC: Failed to create component debugfs directory [ 3.065722] fsl-spdif-dai 2004000.spdif: ASoC: Failed to create component debugfs directory I haven't started debugging this yet and not sure if this is something fsl driver specific. Any ideas? Thanks

5 14

[alsa-devel] [PATCH 0/2] ASoC: add support for ROHM BD28623 codec
by Katsuhiro Suzuki 21 Feb '18

21 Feb '18

This patch adds support for ROHM BD28623MUV class D speaker amplifier codec driver. This driver only refers information of HW specification document that can be derivered at website of ROHM. http://www.rohm.com/web/global/products/-/product/BD28623MUV Katsuhiro Suzuki (2): ASoC: add DT bindings documentation for ROHM BD28623 codec ASoC: support ROHM BD28623 codec .../devicetree/bindings/sound/rohm,bd28623.txt | 26 +++ sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/bd28623.c | 258 +++++++++++++++++++++ 4 files changed, 294 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.txt create mode 100644 sound/soc/codecs/bd28623.c -- 2.16.1

2 4

[alsa-devel] [PATCH] ASoC: dmic: don't use codec anymore
by Kuninori Morimoto 21 Feb '18

21 Feb '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 6d6c3946d877 ("ASoC: dmic: replace codec to component") replaced codec to component, but commit 05c9b302eda7 ("ASoC: dmic: Add optional wakeup delay") used codec again. This patch fix it up. Fixes: 05c9b302eda7 ("ASoC: dmic: Add optional wakeup delay") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/dmic.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index bac5bbb..8c4926d 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -37,8 +37,8 @@ struct dmic { static int dmic_aif_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - struct dmic *dmic = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct dmic *dmic = snd_soc_component_get_drvdata(component); switch (event) { case SND_SOC_DAPM_POST_PMU: -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: arizona: Fixup some minor formatting issues
by Charles Keepax 21 Feb '18

21 Feb '18

Repair any formatting/style issues that can be fixed without major code refactoring. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/arizona.c | 59 ++++++++++++++++++++++++------------------- sound/soc/codecs/cs47l24.c | 22 ++++++++-------- sound/soc/codecs/wm5102.c | 20 +++++++-------- sound/soc/codecs/wm5110.c | 62 +++++++++++++++++++++++++--------------------- sound/soc/codecs/wm8997.c | 14 +++++------ sound/soc/codecs/wm8998.c | 33 +++++++++++++----------- 6 files changed, 113 insertions(+), 97 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ffcfe8ea1848..5727ea079ad7 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -90,7 +90,8 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: - val = snd_soc_component_read32(component, ARIZONA_INTERRUPT_RAW_STATUS_3); + val = snd_soc_component_read32(component, + ARIZONA_INTERRUPT_RAW_STATUS_3); if (val & ARIZONA_SPK_OVERHEAT_STS) { dev_crit(arizona->dev, "Speaker not enabled due to temperature\n"); @@ -641,7 +642,6 @@ const unsigned int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = { }; EXPORT_SYMBOL_GPL(arizona_rate_val); - const struct soc_enum arizona_isrc_fsh[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_1, ARIZONA_ISRC1_FSH_SHIFT, 0xf, @@ -923,7 +923,8 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, priv->in_pending++; break; case SND_SOC_DAPM_POST_PMU: - snd_soc_component_update_bits(component, reg, ARIZONA_IN1L_MUTE, 0); + snd_soc_component_update_bits(component, reg, + ARIZONA_IN1L_MUTE, 0); /* If this is the last input pending then allow VU */ priv->in_pending--; @@ -1070,9 +1071,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(arizona_out_ev); -int arizona_hp_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, - int event) +int arizona_hp_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, + int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct arizona_priv *priv = snd_soc_component_get_drvdata(component); @@ -1275,8 +1275,8 @@ static unsigned int arizona_opclk_ref_44k1_rates[] = { 45158400, }; -static int arizona_set_opclk(struct snd_soc_component *component, unsigned int clk, - unsigned int freq) +static int arizona_set_opclk(struct snd_soc_component *component, + unsigned int clk, unsigned int freq) { struct arizona_priv *priv = snd_soc_component_get_drvdata(component); unsigned int reg; @@ -1302,7 +1302,7 @@ static int arizona_set_opclk(struct snd_soc_component *component, unsigned int c rates = arizona_opclk_ref_48k_rates; for (ref = 0; ref < ARRAY_SIZE(arizona_opclk_ref_48k_rates) && - rates[ref] <= refclk; ref++) { + rates[ref] <= refclk; ref++) { div = 1; while (rates[ref] / div >= freq && div < 32) { if (rates[ref] / div == freq) { @@ -1727,18 +1727,23 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, } snd_soc_component_update_bits(component, ARIZONA_SAMPLE_RATE_1, - ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + ARIZONA_SAMPLE_RATE_1_MASK, + sr_val); if (base) - snd_soc_component_update_bits(component, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 0); + snd_soc_component_update_bits(component, + base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 0); break; case ARIZONA_CLK_ASYNCCLK: - snd_soc_component_update_bits(component, ARIZONA_ASYNC_SAMPLE_RATE_1, - ARIZONA_ASYNC_SAMPLE_RATE_1_MASK, sr_val); + snd_soc_component_update_bits(component, + ARIZONA_ASYNC_SAMPLE_RATE_1, + ARIZONA_ASYNC_SAMPLE_RATE_1_MASK, + sr_val); if (base) - snd_soc_component_update_bits(component, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, - 8 << ARIZONA_AIF1_RATE_SHIFT); + snd_soc_component_update_bits(component, + base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, + 8 << ARIZONA_AIF1_RATE_SHIFT); break; default: arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); @@ -1849,9 +1854,10 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, base + ARIZONA_AIF_RX_ENABLES); /* Disable AIF TX/RX before reconfiguring it */ regmap_update_bits_async(arizona->regmap, - base + ARIZONA_AIF_TX_ENABLES, 0xff, 0x0); + base + ARIZONA_AIF_TX_ENABLES, + 0xff, 0x0); regmap_update_bits(arizona->regmap, - base + ARIZONA_AIF_RX_ENABLES, 0xff, 0x0); + base + ARIZONA_AIF_RX_ENABLES, 0xff, 0x0); } ret = arizona_hw_params_rate(substream, params, dai); @@ -1960,8 +1966,9 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) else reg = 0; - return snd_soc_component_update_bits(component, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_TRI, reg); + return snd_soc_component_update_bits(component, + base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_TRI, reg); } static void arizona_set_channels_to_mask(struct snd_soc_dai *dai, @@ -2319,7 +2326,6 @@ static int arizona_calc_fll(struct arizona_fll *fll, arizona_fll_dbg(fll, "GAIN=0x%x(%d)\n", cfg->gain, 1 << cfg->gain); return 0; - } static void arizona_apply_fll(struct arizona *arizona, unsigned int base, @@ -2565,9 +2571,8 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, fll->ref_src = source; fll->ref_freq = Fref; - if (fll->fout && Fref > 0) { + if (fll->fout && Fref > 0) ret = arizona_enable_fll(fll); - } return ret; } @@ -2656,7 +2661,8 @@ EXPORT_SYMBOL_GPL(arizona_init_fll); * Most systems have a single static configuration and should use * platform data instead. */ -int arizona_set_output_mode(struct snd_soc_component *component, int output, bool diff) +int arizona_set_output_mode(struct snd_soc_component *component, int output, + bool diff) { unsigned int reg, val; @@ -2670,7 +2676,8 @@ int arizona_set_output_mode(struct snd_soc_component *component, int output, boo else val = 0; - return snd_soc_component_update_bits(component, reg, ARIZONA_OUT1_MONO, val); + return snd_soc_component_update_bits(component, reg, + ARIZONA_OUT1_MONO, val); } EXPORT_SYMBOL_GPL(arizona_set_output_mode); diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index c93d3f7e8e36..196e9c343aeb 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -196,14 +196,13 @@ SOC_SINGLE("HPOUT1 SC Protect Switch", ARIZONA_HP1_SHORT_CIRCUIT_CTRL, SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), SOC_SINGLE("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, - ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), + ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, - ARIZONA_OUT4L_VOL_SHIFT, - 0xbf, 0, digital_tlv), + ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), @@ -494,8 +493,7 @@ SND_SOC_DAPM_PGA("ISRC3DEC4", ARIZONA_ISRC_3_CTRL_3, ARIZONA_ISRC3_DEC3_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, - &cs47l24_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, &cs47l24_aec_loopback_mux), SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), @@ -931,8 +929,8 @@ static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = { { "DSP3 Voice Trigger", "Switch", "DSP3" }, }; -static int cs47l24_set_fll(struct snd_soc_component *component, int fll_id, int source, - unsigned int Fref, unsigned int Fout) +static int cs47l24_set_fll(struct snd_soc_component *component, int fll_id, + int source, unsigned int Fref, unsigned int Fout) { struct cs47l24_priv *cs47l24 = snd_soc_component_get_drvdata(component); @@ -1144,7 +1142,8 @@ static int cs47l24_component_probe(struct snd_soc_component *component) goto err_adsp2_codec_probe; ret = snd_soc_add_component_controls(component, - &arizona_adsp2_rate_controls[1], 2); + &arizona_adsp2_rate_controls[1], + 2); if (ret) goto err_adsp2_codec_probe; @@ -1214,7 +1213,7 @@ static int cs47l24_probe(struct platform_device *pdev) BUILD_BUG_ON(ARRAY_SIZE(cs47l24_dai) > ARIZONA_MAX_DAI); cs47l24 = devm_kzalloc(&pdev->dev, sizeof(struct cs47l24_priv), - GFP_KERNEL); + GFP_KERNEL); if (!cs47l24) return -ENOMEM; @@ -1294,8 +1293,9 @@ static int cs47l24_probe(struct platform_device *pdev) goto err_dsp_irq; ret = devm_snd_soc_register_component(&pdev->dev, - &soc_component_dev_cs47l24, - cs47l24_dai, ARRAY_SIZE(cs47l24_dai)); + &soc_component_dev_cs47l24, + cs47l24_dai, + ARRAY_SIZE(cs47l24_dai)); if (ret < 0) { dev_err(&pdev->dev, "Failed to register component: %d\n", ret); goto err_spk_irqs; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 24f19cfc98f5..1ac83388d1b8 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -620,7 +620,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, } static int wm5102_adsp_power_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct arizona *arizona = dev_get_drvdata(component->dev->parent); @@ -717,7 +717,7 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, return 0; } -static const char *wm5102_osr_text[] = { +static const char * const wm5102_osr_text[] = { "Low power", "Normal", "High performance", }; @@ -1062,7 +1062,7 @@ ARIZONA_MIXER_ENUMS(DSP1R, ARIZONA_DSP1RMIX_INPUT_1_SOURCE); ARIZONA_DSP_AUX_ENUMS(DSP1, ARIZONA_DSP1AUX1MIX_INPUT_1_SOURCE); -static const char *wm5102_aec_loopback_texts[] = { +static const char * const wm5102_aec_loopback_texts[] = { "HPOUT1L", "HPOUT1R", "HPOUT2L", "HPOUT2R", "EPOUT", "SPKOUTL", "SPKOUTR", "SPKDAT1L", "SPKDAT1R", }; @@ -1317,8 +1317,7 @@ SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, - &wm5102_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, &wm5102_aec_loopback_mux), SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, @@ -1736,8 +1735,8 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "DRC1 Signal Activity", NULL, "DRC1R" }, }; -static int wm5102_set_fll(struct snd_soc_component *component, int fll_id, int source, - unsigned int Fref, unsigned int Fout) +static int wm5102_set_fll(struct snd_soc_component *component, int fll_id, + int source, unsigned int Fref, unsigned int Fout) { struct wm5102_priv *wm5102 = snd_soc_component_get_drvdata(component); @@ -1947,7 +1946,7 @@ static int wm5102_component_probe(struct snd_soc_component *component) return ret; ret = snd_soc_add_component_controls(component, - arizona_adsp2_rate_controls, 1); + arizona_adsp2_rate_controls, 1); if (ret) goto err_adsp2_codec_probe; @@ -2105,8 +2104,9 @@ static int wm5102_probe(struct platform_device *pdev) goto err_dsp_irq; ret = devm_snd_soc_register_component(&pdev->dev, - &soc_component_dev_wm5102, - wm5102_dai, ARRAY_SIZE(wm5102_dai)); + &soc_component_dev_wm5102, + wm5102_dai, + ARRAY_SIZE(wm5102_dai)); if (ret < 0) { dev_err(&pdev->dev, "Failed to register component: %d\n", ret); goto err_spk_irqs; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index b33cbd0161d3..fb9835dcd836 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -332,19 +332,25 @@ static int wm5110_hp_pre_disable(struct snd_soc_dapm_widget *w) switch (w->shift) { case ARIZONA_OUT1L_ENA_SHIFT: if (!(val & ARIZONA_DRE1L_ENA_MASK)) { - snd_soc_component_update_bits(component, ARIZONA_SPARE_TRIGGERS, - ARIZONA_WS_TRG1, ARIZONA_WS_TRG1); - snd_soc_component_update_bits(component, ARIZONA_SPARE_TRIGGERS, - ARIZONA_WS_TRG1, 0); + snd_soc_component_update_bits(component, + ARIZONA_SPARE_TRIGGERS, + ARIZONA_WS_TRG1, + ARIZONA_WS_TRG1); + snd_soc_component_update_bits(component, + ARIZONA_SPARE_TRIGGERS, + ARIZONA_WS_TRG1, 0); priv->out_down_delay += 27; } break; case ARIZONA_OUT1R_ENA_SHIFT: if (!(val & ARIZONA_DRE1R_ENA_MASK)) { - snd_soc_component_update_bits(component, ARIZONA_SPARE_TRIGGERS, - ARIZONA_WS_TRG2, ARIZONA_WS_TRG2); - snd_soc_component_update_bits(component, ARIZONA_SPARE_TRIGGERS, - ARIZONA_WS_TRG2, 0); + snd_soc_component_update_bits(component, + ARIZONA_SPARE_TRIGGERS, + ARIZONA_WS_TRG2, + ARIZONA_WS_TRG2); + snd_soc_component_update_bits(component, + ARIZONA_SPARE_TRIGGERS, + ARIZONA_WS_TRG2, 0); priv->out_down_delay += 27; } break; @@ -539,7 +545,7 @@ static int wm5110_in_analog_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: snd_soc_component_update_bits(component, reg, mask, - wm5110->in_pga_cache[w->shift]); + wm5110->in_pga_cache[w->shift]); wm5110->in_post_pending--; if (wm5110->in_post_pending == 0) @@ -863,14 +869,14 @@ SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, ARIZONA_SPK2R_MUTE_SHIFT, 1, 1), SOC_DOUBLE_EXT("HPOUT1 DRE Switch", ARIZONA_DRE_ENABLE, - ARIZONA_DRE1L_ENA_SHIFT, ARIZONA_DRE1R_ENA_SHIFT, 1, 0, - snd_soc_get_volsw, wm5110_put_dre), + ARIZONA_DRE1L_ENA_SHIFT, ARIZONA_DRE1R_ENA_SHIFT, 1, 0, + snd_soc_get_volsw, wm5110_put_dre), SOC_DOUBLE_EXT("HPOUT2 DRE Switch", ARIZONA_DRE_ENABLE, - ARIZONA_DRE2L_ENA_SHIFT, ARIZONA_DRE2R_ENA_SHIFT, 1, 0, - snd_soc_get_volsw, wm5110_put_dre), + ARIZONA_DRE2L_ENA_SHIFT, ARIZONA_DRE2R_ENA_SHIFT, 1, 0, + snd_soc_get_volsw, wm5110_put_dre), SOC_DOUBLE_EXT("HPOUT3 DRE Switch", ARIZONA_DRE_ENABLE, - ARIZONA_DRE3L_ENA_SHIFT, ARIZONA_DRE3R_ENA_SHIFT, 1, 0, - snd_soc_get_volsw, wm5110_put_dre), + ARIZONA_DRE3L_ENA_SHIFT, ARIZONA_DRE3R_ENA_SHIFT, 1, 0, + snd_soc_get_volsw, wm5110_put_dre), SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), @@ -1036,7 +1042,7 @@ ARIZONA_MUX_ENUMS(ISRC3DEC2, ARIZONA_ISRC3DEC2MIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ISRC3DEC3, ARIZONA_ISRC3DEC3MIX_INPUT_1_SOURCE); ARIZONA_MUX_ENUMS(ISRC3DEC4, ARIZONA_ISRC3DEC4MIX_INPUT_1_SOURCE); -static const char *wm5110_aec_loopback_texts[] = { +static const char * const wm5110_aec_loopback_texts[] = { "HPOUT1L", "HPOUT1R", "HPOUT2L", "HPOUT2R", "HPOUT3L", "HPOUT3R", "SPKOUTL", "SPKOUTR", "SPKDAT1L", "SPKDAT1R", "SPKDAT2L", "SPKDAT2R", }; @@ -1274,18 +1280,17 @@ SND_SOC_DAPM_PGA("ISRC3DEC4", ARIZONA_ISRC_3_CTRL_3, ARIZONA_ISRC3_DEC3_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, - &wm5110_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, &wm5110_aec_loopback_mux), SND_SOC_DAPM_SUPPLY("RXANC NG External Clock", SND_SOC_NOPM, - ARIZONA_EXT_NG_SEL_SET_SHIFT, 0, arizona_anc_ev, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + ARIZONA_EXT_NG_SEL_SET_SHIFT, 0, arizona_anc_ev, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("RXANCL NG External", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("RXANCR NG External", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("RXANC NG Clock", SND_SOC_NOPM, - ARIZONA_CLK_NG_ENA_SET_SHIFT, 0, arizona_anc_ev, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + ARIZONA_CLK_NG_ENA_SET_SHIFT, 0, arizona_anc_ev, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("RXANCL NG Internal", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("RXANCR NG Internal", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -2034,8 +2039,8 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "DSP3 Voice Trigger", "Switch", "DSP3" }, }; -static int wm5110_set_fll(struct snd_soc_component *component, int fll_id, int source, - unsigned int Fref, unsigned int Fout) +static int wm5110_set_fll(struct snd_soc_component *component, int fll_id, + int source, unsigned int Fref, unsigned int Fout) { struct wm5110_priv *wm5110 = snd_soc_component_get_drvdata(component); @@ -2302,8 +2307,8 @@ static int wm5110_component_probe(struct snd_soc_component *component) } ret = snd_soc_add_component_controls(component, - arizona_adsp2_rate_controls, - WM5110_NUM_ADSP); + arizona_adsp2_rate_controls, + WM5110_NUM_ADSP); if (ret) goto err_adsp2_codec_probe; @@ -2460,8 +2465,9 @@ static int wm5110_probe(struct platform_device *pdev) goto err_dsp_irq; ret = devm_snd_soc_register_component(&pdev->dev, - &soc_component_dev_wm5110, - wm5110_dai, ARRAY_SIZE(wm5110_dai)); + &soc_component_dev_wm5110, + wm5110_dai, + ARRAY_SIZE(wm5110_dai)); if (ret < 0) { dev_err(&pdev->dev, "Failed to register component: %d\n", ret); goto err_spk_irqs; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 45ee1e6bf715..df5b36b8fc5a 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -118,7 +118,7 @@ static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, return arizona_dvfs_sysclk_ev(w, kcontrol, event); } -static const char *wm8997_osr_text[] = { +static const char * const wm8997_osr_text[] = { "Low power", "Normal", "High performance", }; @@ -609,8 +609,7 @@ SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, ARIZONA_SLIMRX8_ENA_SHIFT, 0), SND_SOC_DAPM_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, - &wm8997_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, &wm8997_aec_loopback_mux), SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, @@ -927,8 +926,8 @@ static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { { "MICSUPP", NULL, "SYSCLK" }, }; -static int wm8997_set_fll(struct snd_soc_component *component, int fll_id, int source, - unsigned int Fref, unsigned int Fout) +static int wm8997_set_fll(struct snd_soc_component *component, int fll_id, + int source, unsigned int Fref, unsigned int Fout) { struct wm8997_priv *wm8997 = snd_soc_component_get_drvdata(component); @@ -1173,8 +1172,9 @@ static int wm8997_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_component(&pdev->dev, - &soc_component_dev_wm8997, - wm8997_dai, ARRAY_SIZE(wm8997_dai)); + &soc_component_dev_wm8997, + wm8997_dai, + ARRAY_SIZE(wm8997_dai)); if (ret < 0) { dev_err(&pdev->dev, "Failed to register component: %d\n", ret); goto err_spk_irqs; diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 6f3fdd7b095e..61294c787f27 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -137,11 +137,13 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol, if (inmode & ARIZONA_INMODE_SE) src_val |= 1 << ARIZONA_IN1L_SRC_SE_SHIFT; - snd_soc_component_update_bits(component, mode_reg, ARIZONA_IN1_MODE_MASK, mode_val); + snd_soc_component_update_bits(component, mode_reg, + ARIZONA_IN1_MODE_MASK, mode_val); snd_soc_component_update_bits(component, e->reg, - ARIZONA_IN1L_SRC_MASK | ARIZONA_IN1L_SRC_SE_MASK, - src_val); + ARIZONA_IN1L_SRC_MASK | + ARIZONA_IN1L_SRC_SE_MASK, + src_val); return snd_soc_dapm_mux_update_power(dapm, kcontrol, ucontrol->value.enumerated.item[0], @@ -322,7 +324,7 @@ SOC_DOUBLE_R("HPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, SOC_DOUBLE_R("LINEOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, - ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R("SPKDAT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, @@ -335,7 +337,7 @@ SOC_DOUBLE_R_TLV("LINEOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, - ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv), @@ -615,12 +617,12 @@ SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3, ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_MUX("AEC1 Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, - &wm8998_aec_loopback_mux[0]), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm8998_aec_loopback_mux[0]), SND_SOC_DAPM_MUX("AEC2 Loopback", ARIZONA_DAC_AEC_CONTROL_2, - ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, - &wm8998_aec_loopback_mux[1]), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm8998_aec_loopback_mux[1]), SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), @@ -739,9 +741,9 @@ SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA("SPD1TX1", ARIZONA_SPD1_TX_CONTROL, - ARIZONA_SPD1_VAL1_SHIFT, 0, NULL, 0), + ARIZONA_SPD1_VAL1_SHIFT, 0, NULL, 0), SND_SOC_DAPM_PGA("SPD1TX2", ARIZONA_SPD1_TX_CONTROL, - ARIZONA_SPD1_VAL2_SHIFT, 0, NULL, 0), + ARIZONA_SPD1_VAL2_SHIFT, 0, NULL, 0), SND_SOC_DAPM_OUT_DRV("SPD1", ARIZONA_SPD1_TX_CONTROL, ARIZONA_SPD1_ENA_SHIFT, 0, NULL, 0), @@ -1249,8 +1251,8 @@ static struct snd_soc_dai_driver wm8998_dai[] = { }, }; -static int wm8998_set_fll(struct snd_soc_component *component, int fll_id, int source, - unsigned int Fref, unsigned int Fout) +static int wm8998_set_fll(struct snd_soc_component *component, int fll_id, + int source, unsigned int Fref, unsigned int Fout) { struct wm8998_priv *wm8998 = snd_soc_component_get_drvdata(component); @@ -1379,8 +1381,9 @@ static int wm8998_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_component(&pdev->dev, - &soc_component_dev_wm8998, - wm8998_dai, ARRAY_SIZE(wm8998_dai)); + &soc_component_dev_wm8998, + wm8998_dai, + ARRAY_SIZE(wm8998_dai)); if (ret < 0) { dev_err(&pdev->dev, "Failed to register component: %d\n", ret); goto err_spk_irqs; -- 2.11.0

2 1

[alsa-devel] audio CODEC(CS42888) test
by Muni Sekhar 21 Feb '18

21 Feb '18

[ Please keep me in CC as I'm not subscribed to the list] Hi All, I’m having a sound hardware which contains an analogue audio CODEC(CS42888) and configuring the CODEC is performed over SPI bus. I noticed that linux kernel already had a driver(snd-soc-cs42xx8.ko) for this, I planned to use this driver. Basically I’d like to perform loopback test and compare the sent and received files to verify data fidelity. Also the test to be repeated for various data rates and data formats. I’d like to know how to test this loopback scenario and compare the sent\received files? I appreciate any help. -- Thanks, Sekhar

1 0

[alsa-devel] [PATCH] drm/i915: Remove unused IRQ chip data of HDMI LPE audio
by Augustine.Chen 20 Feb '18

20 Feb '18

The chip data of HDMI LPE audio is set to drm_i915_private which is not consistent with the expectation by x86 APIC driver. In the case of not enabling CONFIG_CPUMASK_OFFSTACK, this would cause kernel panic while doing CPU hotplug. Since the dependency of IRQ chip data was removed from HDMI LPE audio by Commit 9bd9590997b92fbd79fd028f704f6c584b4439d7 ("drm/i915: Stop pretending to mask/unmask LPE audio interrupts"), remove the code of setting IRQ chip data to resolve this issue. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=103731 Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)intel.com> Cc: Jerome Anand <jerome.anand(a)intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Augustine.Chen <augustine.chen(a)intel.com> --- drivers/gpu/drm/i915/intel_lpe_audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/i915/intel_lpe_audio.c b/drivers/gpu/drm/i915/intel_lpe_audio.c index 3bf6528..56176f9 100644 --- a/drivers/gpu/drm/i915/intel_lpe_audio.c +++ b/drivers/gpu/drm/i915/intel_lpe_audio.c @@ -176,7 +176,7 @@ static int lpe_audio_irq_init(struct drm_i915_private *dev_priv) handle_simple_irq, "hdmi_lpe_audio_irq_handler"); - return irq_set_chip_data(irq, dev_priv); + return 0; } static bool lpe_audio_detect(struct drm_i915_private *dev_priv) -- 1.9.1

6 15

[alsa-devel] [PATCH, alsa-lib 1/2] topology: Fix parsing config with multiple hw_configs
by Kirill Marinushkin 20 Feb '18

20 Feb '18

Currently, if the config file includes several hw_configs sections, parse_hw_config_refs() returns after parsing only the first section. For example, the following config, based on alsa-lib/src/conf/topology/broadwell/broadwell.conf, is parsed incorrectly: ~~~~ SectionHWConfig."CodecHWConfig" { id "1" format "I2S" # physical audio format. bclk "master" # Platform is master of bit clock fsync "master" # platform is master of fsync } SectionHWConfig."CodecHWConfig2" { id "2" format "AC97" } SectionLink."Codec" { # used for binding to the physical link id "0" hw_configs [ "CodecHWConfig" "CodecHWConfig2" ] default_hw_conf_id "2" } ~~~~ Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: patch(a)alsa-project.org --- src/topology/pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/src/topology/pcm.c b/src/topology/pcm.c index 58cee31d..d3836677 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -882,6 +882,7 @@ static int parse_hw_config_refs(snd_tplg_t *tplg, snd_config_t *cfg, /* refer to a list of HW configs */ snd_config_for_each(i, next, cfg) { const char *val; + int err; n = snd_config_iterator_entry(i); if (snd_config_get_string(n, &val) < 0) @@ -893,7 +894,9 @@ static int parse_hw_config_refs(snd_tplg_t *tplg, snd_config_t *cfg, } link->num_hw_configs++; - return tplg_ref_add(elem, SND_TPLG_TYPE_HW_CONFIG, val); + err = tplg_ref_add(elem, SND_TPLG_TYPE_HW_CONFIG, val); + if (err < 0) + return err; } return 0; -- 2.13.6

2 6

[alsa-devel] [PATCH 1/2] ASoC: topology: Rename clock_gated to clock_cont in snd_soc_tplg_hw_config
by Kirill Marinushkin 20 Feb '18

20 Feb '18

In kernel `soc-dai.h`, DAI clock gating is defined as following: ~~~~ \#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */ \#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ ~~~~ Therefore, the corresponding field of struct snd_soc_tplg_hw_config should be inverted compared to the current logic: clock_count = 1 => SND_SOC_DAIFMT_CONT clock_count = 0 => SND_SOC_DAIFMT_GATED Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- include/uapi/sound/asoc.h | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 69c37ecbff7e..10188850dede 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -312,7 +312,9 @@ struct snd_soc_tplg_hw_config { __le32 size; /* in bytes of this structure */ __le32 id; /* unique ID - - used to match */ __le32 fmt; /* SND_SOC_DAI_FORMAT_ format value */ - __u8 clock_gated; /* 1 if clock can be gated to save power */ + __u8 clock_cont; /* 1 if clock is continuous, and can not be + * gated to save power + */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ __u8 bclk_master; /* 1 for master of BCLK, 0 for slave */ -- 2.13.6

3 10