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

[PATCH] sound: remove duplicate "the the" phrase in Kconfig text
by Colin King 18 Aug '20

18 Aug '20

From: Colin Ian King <colin.king(a)canonical.com> There are a couple of occurrences of "the the" in the Kconfig text. Fix these. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/xilinx/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 1d3586b68db7..5bd2730aab76 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -9,14 +9,14 @@ config SND_SOC_XILINX_I2S encapsulates PCM in AES format and sends AES data. config SND_SOC_XILINX_AUDIO_FORMATTER - tristate "Audio support for the the Xilinx audio formatter" + tristate "Audio support for the Xilinx audio formatter" help Select this option to enable Xilinx audio formatter support. This provides DMA platform device support for audio functionality. config SND_SOC_XILINX_SPDIF - tristate "Audio support for the the Xilinx SPDIF" + tristate "Audio support for the Xilinx SPDIF" help Select this option to enable Xilinx SPDIF Audio. This provides playback and capture of SPDIF audio in -- 2.27.0

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[PATCH v3 00/10] ASoC: qdsp6: add gapless compressed audio support
by Srinivas Kandagatla 18 Aug '20

18 Aug '20

This patchset adds gapless compressed audio support on q6asm. Gapless on q6asm is implemented using 2 streams in a single q6asm session. First few patches such as stream id per each command, gapless flags and silence meta data are for preparedness for adding gapless support. Last patch implements copy callback to allow finer control over buffer offsets, specially in partial drain cases. This patchset is tested on RB3 aka DB845c platform. This patchset as it is will support gapless however QDSP can also support switching decoders on a single stream. Patches to support such feature are send in different patchset which involves adding generic interfaces. Thanks, srini Changes since v2:(mostly suggested by Pierre) - removed unnessary kernel style comments, - moved TIMESTAMP flag to respective patch. - move preparatory code from gapless support patch to new one. - fix subject prefix of one patch. - add comments to clarify valid stream_ids Srinivas Kandagatla (10): ASoC: q6asm: rename misleading session id variable ASoC: q6asm: make commands specific to streams ASoC: q6asm: use flags directly from q6asm-dai ASoC: q6asm: add length to write command token ASoC: q6asm: add support to remove intial and trailing silence ASoC: q6asm: add support to gapless flag in q6asm open ASoC: q6asm-dai: add next track metadata support ASoC: q6asm-dai: prepare set params to accept profile change ASoC: q6asm-dai: add gapless support ASoC: q6asm-dai: add support to copy callback sound/soc/qcom/qdsp6/q6asm-dai.c | 414 +++++++++++++++++++++++-------- sound/soc/qcom/qdsp6/q6asm.c | 169 +++++++++---- sound/soc/qcom/qdsp6/q6asm.h | 49 ++-- 3 files changed, 469 insertions(+), 163 deletions(-) -- 2.21.0

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[PATCH] sound/soc/intel: Fix spelling mistake "cant" --> "can't"
by Youling Tang 18 Aug '20

18 Aug '20

There is some spelling mistakes in a dev_err message. Fix it. Signed-off-by: Youling Tang <tangyouling(a)loongson.cn> --- sound/soc/intel/common/sst-firmware.c | 4 ++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 6 +++--- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 0594f89..1189ec3 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -130,7 +130,7 @@ static void block_list_remove(struct sst_dsp *dsp, err = block->ops->disable(block); if (err < 0) dev_err(dsp->dev, - "error: cant disable block %d:%d\n", + "error: can't disable block %d:%d\n", block->type, block->index); } } @@ -158,7 +158,7 @@ static int block_list_prepare(struct sst_dsp *dsp, ret = block->ops->enable(block); if (ret < 0) { dev_err(dsp->dev, - "error: cant disable block %d:%d\n", + "error: can't disable block %d:%d\n", block->type, block->index); goto err; } diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 0ff89ea..773688b 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -1507,7 +1507,7 @@ static int sst_hsw_dx_state_dump(struct sst_hsw *hsw) ret = sst_dsp_dma_get_channel(sst, 0); if (ret < 0) { - dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + dev_err(hsw->dev, "error: can't allocate dma channel %d\n", ret); return ret; } @@ -1587,7 +1587,7 @@ int sst_hsw_dsp_load(struct sst_hsw *hsw) ret = sst_dsp_dma_get_channel(dsp, 0); if (ret < 0) { - dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + dev_err(hsw->dev, "error: can't allocate dma channel %d\n", ret); return ret; } @@ -1616,7 +1616,7 @@ static int sst_hsw_dsp_restore(struct sst_hsw *hsw) ret = sst_dsp_dma_get_channel(dsp, 0); if (ret < 0) { - dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + dev_err(hsw->dev, "error: can't allocate dma channel %d\n", ret); return ret; } -- 2.1.0

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[PATCH] ASoC: wm8962: Export DAC/ADC monomix switches
by Sebastian Krzyszkowiak 18 Aug '20

18 Aug '20

This allows solutions like ALSA UCM to utilize hardware mono downmix for cases where mono output to a single speaker is desired only in specific situations (like on a mobile phone). Signed-off-by: Sebastian Krzyszkowiak <sebastian.krzyszkowiak(a)puri.sm> --- sound/soc/codecs/wm8962.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 317916cb4e27..d36aa24da7e0 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1702,6 +1702,8 @@ SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8962_LEFT_DAC_VOLUME, SOC_SINGLE("DAC High Performance Switch", WM8962_ADC_DAC_CONTROL_2, 0, 1, 0), SOC_SINGLE("DAC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 5, 1, 0), SOC_SINGLE("ADC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 8, 1, 0), +SOC_SINGLE("DAC Monomix Switch", WM8962_DAC_DSP_MIXING_1, WM8962_DAC_MONOMIX_SHIFT, 1, 0), +SOC_SINGLE("ADC Monomix Switch", WM8962_THREED1, WM8962_ADC_MONOMIX_SHIFT, 1, 0), SOC_SINGLE("ADC High Performance Switch", WM8962_ADDITIONAL_CONTROL_1, 5, 1, 0), -- 2.28.0

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[PATCH v2 0/8] ASoC: sun50i-codec-analog: Cleanup and power management
by Samuel Holland 18 Aug '20

18 Aug '20

This series performs some minor cleanup on the driver for the analog codec in the Allwinner A64, and hooks up the existing mute switches to DAPM widgets, in order to provide improved power management. Changes since v1: - Collected Acked-by/Reviewed-by tags - Used SOC_MIXER_NAMED_CTL_ARRAY to avoid naming a widget "Earpiece" Samuel Holland (8): ASoC: sun50i-codec-analog: Fix duplicate use of ADC enable bits ASoC: sun50i-codec-analog: Gate the amplifier clock during suspend ASoC: sun50i-codec-analog: Group and sort mixer routes ASoC: sun50i-codec-analog: Make headphone routes stereo ASoC: sun50i-codec-analog: Enable DAPM for headphone switch ASoC: sun50i-codec-analog: Make line out routes stereo ASoC: sun50i-codec-analog: Enable DAPM for line out switch ASoC: sun50i-codec-analog: Enable DAPM for earpiece switch sound/soc/sunxi/sun50i-codec-analog.c | 176 ++++++++++++++++---------- 1 file changed, 111 insertions(+), 65 deletions(-) -- 2.26.2

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[PATCH v3 0/3] refine and clean code for synchronous mode
by Shengjiu Wang 18 Aug '20

18 Aug '20

refine and clean code for synchronous mode Shengjiu Wang (3): ASoC: fsl_sai: Refine enable/disable TE/RE sequence in trigger() ASoC: fsl_sai: Drop TMR/RMR settings for synchronous mode ASoC: fsl_sai: Replace synchronous check with fsl_sai_dir_is_synced changes in v3: - Add reviewed-by Nicolin - refine the commit log. - Add one more patch #3 changes in v2: - Split the commit - refine the sequence in trigger stop sound/soc/fsl/fsl_sai.c | 173 +++++++++++++++++++++++----------------- 1 file changed, 102 insertions(+), 71 deletions(-) -- 2.27.0

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[PATCH] ASoC: wm8994: Prevent access to invalid VU register bits on WM1811
by Sylwester Nawrocki 18 Aug '20

18 Aug '20

The ADC2 and DAC2 are not available on WM1811 device. This patch moves the ADC2, DAC2 VU bitfields to a separate array so we can skip accessing them and avoid unreadable register access on WM1811. This allows to get rid of warnings during boot like: wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5 Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/codecs/wm8994.c | 60 ++++++++++++++++++++++++++++----------- 1 file changed, 44 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a84ae879d37e..038be667c1a6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -43,10 +43,12 @@ #define WM8994_NUM_DRC 3 #define WM8994_NUM_EQ 3 -static struct { +struct wm8994_reg_mask { unsigned int reg; unsigned int mask; -} wm8994_vu_bits[] = { +}; + +static struct wm8994_reg_mask wm8994_vu_bits[] = { { WM8994_LEFT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_RIGHT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_LEFT_LINE_INPUT_3_4_VOLUME, WM8994_IN2_VU }, @@ -60,14 +62,10 @@ static struct { { WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU }, { WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU }, - { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, - { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, { WM8994_AIF2_DAC_LEFT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1ADC1_VU }, { WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU }, - { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, - { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2ADC_VU }, { WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_VU }, @@ -76,6 +74,14 @@ static struct { { WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU }, }; +/* VU bitfields for ADC2, DAC2 not available on WM1811 */ +static struct wm8994_reg_mask wm8994_adc2_dac2_vu_bits[] = { + { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, + { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, +}; + static int wm8994_drc_base[] = { WM8994_AIF1_DRC1_1, WM8994_AIF1_DRC2_1, @@ -1030,6 +1036,26 @@ static bool wm8994_check_class_w_digital(struct snd_soc_component *component) return true; } +static void wm8994_update_vu_bits(struct snd_soc_component *component) +{ + struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); + struct wm8994 *control = wm8994->wm8994; + int i; + + for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) + snd_soc_component_write(component, wm8994_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_vu_bits[i].reg)); + if (control->type == WM1811) + return; + + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_write(component, + wm8994_adc2_dac2_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_adc2_dac2_vu_bits[i].reg)); +} + static int aif_mclk_set(struct snd_soc_component *component, int aif, bool enable) { struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); @@ -1076,7 +1102,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; - int ret, i; + int ret; int dac; int adc; int val; @@ -1144,10 +1170,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -1181,7 +1204,7 @@ static int aif2clk_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); - int ret, i; + int ret; int dac; int adc; int val; @@ -1237,10 +1260,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -4346,6 +4366,14 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994_vu_bits[i].mask, wm8994_vu_bits[i].mask); + if (control->type != WM1811) { + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_update_bits(component, + wm8994_adc2_dac2_vu_bits[i].reg, + wm8994_adc2_dac2_vu_bits[i].mask, + wm8994_adc2_dac2_vu_bits[i].mask); + } + /* Set the low bit of the 3D stereo depth so TLV matches */ snd_soc_component_update_bits(component, WM8994_AIF1_DAC1_FILTERS_2, 1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT, -- 2.17.1

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[PATCH v2 0/3] Codec workaround
by Michael Sit Wei Hong 18 Aug '20

18 Aug '20

This patch series enables some features on the tlv3204 codec and also fixes some issues faced while testing v2: Fixed the build error from snd_soc_component_read32 v1: initial ASoC: codec: tlv3204: Codec workaround series Michael Sit Wei Hong (3): ASoC: codec: tlv3204: Enable 24 bit audio support ASoC: codec: tlv3204: Increased maximum supported channels ASoC: codec: tlv3204: Moving GPIO reset and add ADC reset sound/soc/codecs/tlv320aic32x4.c | 60 +++++++++++++++++++++++--------- 1 file changed, 44 insertions(+), 16 deletions(-) -- 2.17.1

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[PATCH] ASoC: codecs: wm0010: use DECLARE_COMPLETION_ONSTACK() macro
by Pierre-Louis Bossart 18 Aug '20

18 Aug '20

Follow recommendation in Documentation/scheduler/completion.rst and use macro to declare local 'struct completion' Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/wm0010.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 2f2b2f5d55e4..28b4656c4e14 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -346,7 +346,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_component *comp struct list_head xfer_list; struct wm0010_boot_xfer *xfer; int ret; - struct completion done; + DECLARE_COMPLETION_ONSTACK(done); const struct firmware *fw; const struct dfw_binrec *rec; const struct dfw_inforec *inforec; @@ -370,7 +370,6 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_component *comp wm0010->boot_failed = false; if (WARN_ON(!list_empty(&xfer_list))) return -EINVAL; - init_completion(&done); /* First record should be INFO */ if (rec->command != DFW_CMD_INFO) { -- 2.25.1

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[PATCH v2 0/2] This patch series is to enable multiple features on the Keembay Platform
by Michael Sit Wei Hong 18 Aug '20

18 Aug '20

v2: Change channel-max device tree property to use new compatible string v1: Initial patch series Michael Sit Wei Hong (2): ASoC: Intel: KMB: Enable TDM audio capture dt-bindings: sound: intel,keembay-i2s: Add new compatible string .../bindings/sound/intel,keembay-i2s.yaml | 1 + sound/soc/intel/keembay/kmb_platform.c | 145 +++++++++++++----- 2 files changed, 109 insertions(+), 37 deletions(-) -- 2.17.1

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