June 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: max98396: Fix TDM mode BSEL settings
by Daniel Mack 29 Jun '22

29 Jun '22

In TDM mode, the BSEL register value must be set according to table 5 in the datasheet. This patch adds a lookup function and uses it in max98396_dai_tdm_slot(). As the first 3 entries can also be used for non-TDM setups, the code re-uses the same table for such scenarios. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/codecs/max98396.c | 138 +++++++++++++++++++++++++----------- sound/soc/codecs/max98396.h | 2 +- 2 files changed, 96 insertions(+), 44 deletions(-) diff --git a/sound/soc/codecs/max98396.c b/sound/soc/codecs/max98396.c index f28831f4e74b..1b6f053adba2 100644 --- a/sound/soc/codecs/max98396.c +++ b/sound/soc/codecs/max98396.c @@ -438,47 +438,68 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } -/* BCLKs per LRCLK */ -static const int bclk_sel_table[] = { - 32, 48, 64, 96, 128, 192, 256, 384, 512, 320, +#define MAX98396_BSEL_32 0x2 +#define MAX98396_BSEL_48 0x3 +#define MAX98396_BSEL_64 0x4 +#define MAX98396_BSEL_96 0x5 +#define MAX98396_BSEL_128 0x6 +#define MAX98396_BSEL_192 0x7 +#define MAX98396_BSEL_256 0x8 +#define MAX98396_BSEL_384 0x9 +#define MAX98396_BSEL_512 0xa +#define MAX98396_BSEL_320 0xb +#define MAX98396_BSEL_250 0xc +#define MAX98396_BSEL_125 0xd + +/* Refer to table 5 in the datasheet */ +static const struct max98396_pcm_config { + int in, out, width, bsel, max_sr; +} max98396_pcm_configs[] = { + { .in = 2, .out = 4, .width = 16, .bsel = MAX98396_BSEL_32, .max_sr = 192000 }, + { .in = 2, .out = 6, .width = 24, .bsel = MAX98396_BSEL_48, .max_sr = 192000 }, + { .in = 2, .out = 8, .width = 32, .bsel = MAX98396_BSEL_64, .max_sr = 192000 }, + { .in = 3, .out = 15, .width = 32, .bsel = MAX98396_BSEL_125, .max_sr = 192000 }, + { .in = 4, .out = 8, .width = 16, .bsel = MAX98396_BSEL_64, .max_sr = 192000 }, + { .in = 4, .out = 12, .width = 24, .bsel = MAX98396_BSEL_96, .max_sr = 192000 }, + { .in = 4, .out = 16, .width = 32, .bsel = MAX98396_BSEL_128, .max_sr = 192000 }, + { .in = 5, .out = 15, .width = 24, .bsel = MAX98396_BSEL_125, .max_sr = 192000 }, + { .in = 7, .out = 15, .width = 16, .bsel = MAX98396_BSEL_125, .max_sr = 192000 }, + { .in = 2, .out = 4, .width = 16, .bsel = MAX98396_BSEL_32, .max_sr = 96000 }, + { .in = 2, .out = 6, .width = 24, .bsel = MAX98396_BSEL_48, .max_sr = 96000 }, + { .in = 2, .out = 8, .width = 32, .bsel = MAX98396_BSEL_64, .max_sr = 96000 }, + { .in = 3, .out = 15, .width = 32, .bsel = MAX98396_BSEL_125, .max_sr = 96000 }, + { .in = 4, .out = 8, .width = 16, .bsel = MAX98396_BSEL_64, .max_sr = 96000 }, + { .in = 4, .out = 12, .width = 24, .bsel = MAX98396_BSEL_96, .max_sr = 96000 }, + { .in = 4, .out = 16, .width = 32, .bsel = MAX98396_BSEL_128, .max_sr = 96000 }, + { .in = 5, .out = 15, .width = 24, .bsel = MAX98396_BSEL_125, .max_sr = 96000 }, + { .in = 7, .out = 15, .width = 16, .bsel = MAX98396_BSEL_125, .max_sr = 96000 }, + { .in = 7, .out = 31, .width = 32, .bsel = MAX98396_BSEL_250, .max_sr = 96000 }, + { .in = 8, .out = 16, .width = 16, .bsel = MAX98396_BSEL_128, .max_sr = 96000 }, + { .in = 8, .out = 24, .width = 24, .bsel = MAX98396_BSEL_192, .max_sr = 96000 }, + { .in = 8, .out = 32, .width = 32, .bsel = MAX98396_BSEL_256, .max_sr = 96000 }, + { .in = 10, .out = 31, .width = 24, .bsel = MAX98396_BSEL_250, .max_sr = 96000 }, + { .in = 15, .out = 31, .width = 16, .bsel = MAX98396_BSEL_250, .max_sr = 96000 }, + { .in = 16, .out = 32, .width = 16, .bsel = MAX98396_BSEL_256, .max_sr = 96000 }, + { .in = 7, .out = 31, .width = 32, .bsel = MAX98396_BSEL_250, .max_sr = 48000 }, + { .in = 10, .out = 31, .width = 24, .bsel = MAX98396_BSEL_250, .max_sr = 48000 }, + { .in = 10, .out = 40, .width = 32, .bsel = MAX98396_BSEL_320, .max_sr = 48000 }, + { .in = 15, .out = 31, .width = 16, .bsel = MAX98396_BSEL_250, .max_sr = 48000 }, + { .in = 16, .out = 48, .width = 24, .bsel = MAX98396_BSEL_384, .max_sr = 48000 }, + { .in = 16, .out = 64, .width = 32, .bsel = MAX98396_BSEL_512, .max_sr = 48000 }, }; -static int max98396_get_bclk_sel(int bclk) +static int max98396_pcm_config_index(int in_slots, int out_slots, int width) { int i; - /* match BCLKs per LRCLK */ - for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { - if (bclk_sel_table[i] == bclk) - return i + 2; - } - return 0; -} - -static int max98396_set_clock(struct snd_soc_component *component, - struct snd_pcm_hw_params *params) -{ - struct max98396_priv *max98396 = snd_soc_component_get_drvdata(component); - /* BCLK/LRCLK ratio calculation */ - int blr_clk_ratio = params_channels(params) * max98396->ch_size; - int value; - if (!max98396->tdm_mode) { - /* BCLK configuration */ - value = max98396_get_bclk_sel(blr_clk_ratio); - if (!value) { - dev_err(component->dev, - "blr_clk_ratio %d unsupported, format %d\n", - blr_clk_ratio, params_format(params)); - return -EINVAL; - } + for (i = 0; i < ARRAY_SIZE(max98396_pcm_configs); i++) { + const struct max98396_pcm_config *c = &max98396_pcm_configs[i]; - regmap_update_bits(max98396->regmap, - MAX98396_R2042_PCM_CLK_SETUP, - MAX98396_PCM_CLK_SETUP_BSEL_MASK, - value); + if (in_slots == c->in && out_slots <= c->out && width == c->width) + return i; } - return 0; + return -1; } static int max98396_dai_hw_params(struct snd_pcm_substream *substream, @@ -489,8 +510,7 @@ static int max98396_dai_hw_params(struct snd_pcm_substream *substream, struct max98396_priv *max98396 = snd_soc_component_get_drvdata(component); unsigned int sampling_rate = 0; unsigned int chan_sz = 0; - int ret, reg; - int status; + int ret, reg, status, bsel = 0; bool update = false; /* pcm mode configuration */ @@ -510,8 +530,6 @@ static int max98396_dai_hw_params(struct snd_pcm_substream *substream, goto err; } - max98396->ch_size = snd_pcm_format_width(params_format(params)); - dev_dbg(component->dev, "format supported %d", params_format(params)); @@ -559,6 +577,33 @@ static int max98396_dai_hw_params(struct snd_pcm_substream *substream, goto err; } + if (max98396->tdm_mode) { + if (params_rate(params) > max98396->tdm_max_samplerate) { + dev_err(component->dev, "TDM sample rate %d too high", + params_rate(params)); + goto err; + } + } else { + /* BCLK configuration */ + ret = max98396_pcm_config_index(params_channels(params), + params_channels(params), + snd_pcm_format_width(params_format(params))); + if (ret < 0) { + dev_err(component->dev, + "no PCM config for %d channels, format %d\n", + params_channels(params), params_format(params)); + goto err; + } + + bsel = max98396_pcm_configs[ret].bsel; + + if (params_rate(params) > max98396_pcm_configs[ret].max_sr) { + dev_err(component->dev, "sample rate %d too high", + params_rate(params)); + goto err; + } + } + ret = regmap_read(max98396->regmap, MAX98396_R210F_GLOBAL_EN, &status); if (ret < 0) goto err; @@ -604,12 +649,16 @@ static int max98396_dai_hw_params(struct snd_pcm_substream *substream, MAX98396_IVADC_SR_MASK, sampling_rate << MAX98396_IVADC_SR_SHIFT); - ret = max98396_set_clock(component, params); + if (bsel) + regmap_update_bits(max98396->regmap, + MAX98396_R2042_PCM_CLK_SETUP, + MAX98396_PCM_CLK_SETUP_BSEL_MASK, + bsel); if (status && update) max98396_global_enable_onoff(max98396->regmap, true); - return ret; + return 0; err: return -EINVAL; @@ -634,13 +683,16 @@ static int max98396_dai_tdm_slot(struct snd_soc_dai *dai, max98396->tdm_mode = true; /* BCLK configuration */ - bsel = max98396_get_bclk_sel(slots * slot_width); - if (bsel == 0) { - dev_err(component->dev, "BCLK %d not supported\n", - slots * slot_width); + ret = max98396_pcm_config_index(slots, slots, slot_width); + if (ret < 0) { + dev_err(component->dev, "no TDM config for %d slots %d bits\n", + slots, slot_width); return -EINVAL; } + bsel = max98396_pcm_configs[ret].bsel; + max98396->tdm_max_samplerate = max98396_pcm_configs[ret].max_sr; + /* Channel size configuration */ switch (slot_width) { case 16: diff --git a/sound/soc/codecs/max98396.h b/sound/soc/codecs/max98396.h index ff330ef61568..7278c779989a 100644 --- a/sound/soc/codecs/max98396.h +++ b/sound/soc/codecs/max98396.h @@ -306,8 +306,8 @@ struct max98396_priv { unsigned int spkfb_slot; unsigned int bypass_slot; bool interleave_mode; - unsigned int ch_size; bool tdm_mode; + int tdm_max_samplerate; int device_id; }; #endif -- 2.36.1

2 1

[PATCH] ASoC: fsl_micfil: change micfil default settings
by Shengjiu Wang 29 Jun '22

29 Jun '22

Previous default settings resulted in loose dynamic range and low sound level. New default configuration changes: - outgain = 2 - quality mode = VLOW0 - dc remover = bypass Signed-off-by: Irina Patru <ioana-irina.patru(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_micfil.c | 22 ++++++++++++++++++---- sound/soc/fsl/fsl_micfil.h | 9 +++++++++ 2 files changed, 27 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 25c647ae080a..18ab80b68752 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -48,6 +48,7 @@ struct fsl_micfil { char name[32]; int irq[MICFIL_IRQ_LINES]; enum quality quality; + int dc_remover; }; struct fsl_micfil_soc_data { @@ -336,12 +337,25 @@ static const struct snd_soc_dai_ops fsl_micfil_dai_ops = { static int fsl_micfil_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_micfil *micfil = dev_get_drvdata(cpu_dai->dev); - int ret; + struct device *dev = cpu_dai->dev; + unsigned int val = 0; + int ret, i; + + micfil->quality = QUALITY_VLOW0; - micfil->quality = QUALITY_MEDIUM; + /* set default gain to 2 */ + regmap_write(micfil->regmap, REG_MICFIL_OUT_CTRL, 0x22222222); - /* set default gain to max_gain */ - regmap_write(micfil->regmap, REG_MICFIL_OUT_CTRL, 0x77777777); + /* set DC Remover in bypass mode*/ + for (i = 0; i < MICFIL_OUTPUT_CHANNELS; i++) + val |= MICFIL_DC_BYPASS << MICFIL_DC_CHX_SHIFT(i); + ret = regmap_update_bits(micfil->regmap, REG_MICFIL_DC_CTRL, + MICFIL_DC_CTRL_CONFIG, val); + if (ret) { + dev_err(dev, "failed to set DC Remover mode bits\n"); + return ret; + } + micfil->dc_remover = MICFIL_DC_BYPASS; snd_soc_dai_init_dma_data(cpu_dai, NULL, &micfil->dma_params_rx); diff --git a/sound/soc/fsl/fsl_micfil.h b/sound/soc/fsl/fsl_micfil.h index 053caba3caf3..d60285dd07bc 100644 --- a/sound/soc/fsl/fsl_micfil.h +++ b/sound/soc/fsl/fsl_micfil.h @@ -73,6 +73,15 @@ #define MICFIL_FIFO_STAT_FIFOX_OVER(ch) BIT(ch) #define MICFIL_FIFO_STAT_FIFOX_UNDER(ch) BIT((ch) + 8) +/* MICFIL DC Remover Control Register -- REG_MICFIL_DC_CTRL */ +#define MICFIL_DC_CTRL_CONFIG GENMASK(15, 0) +#define MICFIL_DC_CHX_SHIFT(ch) ((ch) << 1) +#define MICFIL_DC_CHX(ch) GENMASK((((ch) << 1) + 1), ((ch) << 1)) +#define MICFIL_DC_CUTOFF_21HZ 0 +#define MICFIL_DC_CUTOFF_83HZ 1 +#define MICFIL_DC_CUTOFF_152Hz 2 +#define MICFIL_DC_BYPASS 3 + /* MICFIL HWVAD0 Control 1 Register -- REG_MICFIL_VAD0_CTRL1*/ #define MICFIL_VAD0_CTRL1_CHSEL GENMASK(26, 24) #define MICFIL_VAD0_CTRL1_CICOSR GENMASK(19, 16) -- 2.17.1

2 1

[PATCH] ASoC: dt-bindings: fsl,micfil: Convert format to json-schema
by Shengjiu Wang 29 Jun '22

29 Jun '22

Convert the NXP MICFIL binding to DT schema format using json-schema. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- .../devicetree/bindings/sound/fsl,micfil.txt | 33 --------- .../devicetree/bindings/sound/fsl,micfil.yaml | 73 +++++++++++++++++++ 2 files changed, 73 insertions(+), 33 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,micfil.txt b/Documentation/devicetree/bindings/sound/fsl,micfil.txt deleted file mode 100644 index 1ea05d4996c7..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,micfil.txt +++ /dev/null @@ -1,33 +0,0 @@ -NXP MICFIL Digital Audio Interface (MICFIL). - -The MICFIL digital interface provides a 16-bit audio signal from a PDM -microphone bitstream in a configurable output sampling rate. - -Required properties: - - - compatible : Compatible list, contains "fsl,imx8mm-micfil" - or "fsl,imx8mp-micfil" - - - reg : Offset and length of the register set for the device. - - - interrupts : Contains the micfil interrupts. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "ipg_clk" for register access and - "ipg_clk_app" for internal micfil clock. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - -Example: -micfil: micfil@30080000 { - compatible = "fsl,imx8mm-micfil"; - reg = <0x0 0x30080000 0x0 0x10000>; - interrupts = <GIC_SPI 109 IRQ_TYPE_LEVEL_HIGH>, - <GIC_SPI 110 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&clk IMX8MM_CLK_PDM_IPG>, - <&clk IMX8MM_CLK_PDM_ROOT>; - clock-names = "ipg_clk", "ipg_clk_app"; - dmas = <&sdma2 24 26 0x80000000>; -}; diff --git a/Documentation/devicetree/bindings/sound/fsl,micfil.yaml b/Documentation/devicetree/bindings/sound/fsl,micfil.yaml new file mode 100644 index 000000000000..74c77f4cf7a4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,micfil.yaml @@ -0,0 +1,73 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,micfil.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP MICFIL Digital Audio Interface (MICFIL) + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The MICFIL digital interface provides a 16-bit or 24-bit audio signal + from a PDM microphone bitstream in a configurable output sampling rate. + +properties: + compatible: + enum: + - fsl,imx8mm-micfil + - fsl,imx8mp-micfil + + reg: + maxItems: 1 + + interrupts: + maxItems: 4 + + dmas: + items: + - description: DMA controller phandle and request line for RX + + dma-names: + items: + - const: rx + + clocks: + items: + - description: The ipg clock for register access + - description: internal micfil clock + + clock-names: + items: + - const: ipg_clk + - const: ipg_clk_app + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mm-clock.h> + micfil: audio-controller@30080000 { + compatible = "fsl,imx8mm-micfil"; + reg = <0x30080000 0x10000>; + interrupts = <GIC_SPI 109 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 110 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 44 IRQ_TYPE_LEVEL_HIGH>, + <GIC_SPI 45 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk IMX8MM_CLK_PDM_IPG>, + <&clk IMX8MM_CLK_PDM_ROOT>; + clock-names = "ipg_clk", "ipg_clk_app"; + dmas = <&sdma2 24 25 0>; + dma-names = "rx"; + }; -- 2.17.1

4 3

[PATCH] ASoC: doc: Capitalize RESET line name
by Marek Vasut 29 Jun '22

29 Jun '22

Make sure all AC97 interface lines are spelled in capitals, to avoid confusing readers about where the 5th line is. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/soc/dai.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/soc/dai.rst b/Documentation/sound/soc/dai.rst index 009b07e5a0f38..bf8431386d269 100644 --- a/Documentation/sound/soc/dai.rst +++ b/Documentation/sound/soc/dai.rst @@ -10,7 +10,7 @@ AC97 ==== AC97 is a five wire interface commonly found on many PC sound cards. It is -now also popular in many portable devices. This DAI has a reset line and time +now also popular in many portable devices. This DAI has a RESET line and time multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines. The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97 -- 2.35.1

2 1

[PATCH 1/4] ASoC: wm5102: Fix event generation for output compensation
by Charles Keepax 29 Jun '22

29 Jun '22

The output compensation controls always returns zero regardless of if the control value was updated. This results in missing notifications to user-space of the control change. Update the handling to return 1 when the value is changed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm5102.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index da2f8998df87a..b034df47a5ef1 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -680,12 +680,17 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct arizona *arizona = dev_get_drvdata(component->dev->parent); + uint16_t dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); + int ret = 0; mutex_lock(&arizona->dac_comp_lock); - arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); + if (arizona->dac_comp_coeff != dac_comp_coeff) { + arizona->dac_comp_coeff = dac_comp_coeff; + ret = 1; + } mutex_unlock(&arizona->dac_comp_lock); - return 0; + return ret; } static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, @@ -706,12 +711,20 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct arizona *arizona = dev_get_drvdata(component->dev->parent); + struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int ret = 0; + + if (ucontrol->value.integer.value[0] > mc->max) + return -EINVAL; mutex_lock(&arizona->dac_comp_lock); - arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + if (arizona->dac_comp_enabled != ucontrol->value.integer.value[0]) { + arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + ret = 1; + } mutex_unlock(&arizona->dac_comp_lock); - return 0; + return ret; } static const char * const wm5102_osr_text[] = { -- 2.30.2

2 4

[PATCH] pinctrl: qcom: sc7280: Fix compile bug
by Srinivasa Rao Mandadapu 29 Jun '22

29 Jun '22

Fix the compilation error, caused by updtating constant variable. Hence remove redundant constant variable, which is no more useful as per new design. The issue is due to some unstaged changes. Fix it up. Fixes: 36fe26843d6d ("pinctrl: qcom: sc7280: Add clock optional check for ADSP bypass targets") Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 5 +---- drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 1 - 2 files changed, 1 insertion(+), 5 deletions(-) diff --git a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c index b5d1b99..e97ce45 100644 --- a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c +++ b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c @@ -388,9 +388,6 @@ int lpi_pinctrl_probe(struct platform_device *pdev) pctrl->data = data; pctrl->dev = &pdev->dev; - data->is_clk_optional = of_property_read_bool(dev->of_node, - "qcom,adsp-bypass-mode"); - pctrl->clks[0].id = "core"; pctrl->clks[1].id = "audio"; @@ -404,7 +401,7 @@ int lpi_pinctrl_probe(struct platform_device *pdev) return dev_err_probe(dev, PTR_ERR(pctrl->slew_base), "Slew resource not provided\n"); - if (data->is_clk_optional) + if (of_property_read_bool(dev->of_node, "qcom,adsp-bypass-mode")) ret = devm_clk_bulk_get_optional(dev, MAX_LPI_NUM_CLKS, pctrl->clks); else ret = devm_clk_bulk_get(dev, MAX_LPI_NUM_CLKS, pctrl->clks); diff --git a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.h b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.h index 759d5d8..afbac2a 100644 --- a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.h +++ b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.h @@ -77,7 +77,6 @@ struct lpi_pinctrl_variant_data { int ngroups; const struct lpi_function *functions; int nfunctions; - bool is_clk_optional; }; int lpi_pinctrl_probe(struct platform_device *pdev); -- 2.7.4

2 1

[PATCH v5 0/2] Add software clock gating requirement check
by Srinivasa Rao Mandadapu 29 Jun '22

29 Jun '22

This patch set is to add software clock gating requirement check Changes Since V4: -- Fix error check, after a reset control get api return -- Fix typo in commit message Changes Since V3: -- Remove redundant check before reset control call -- Reorganiaze patches. -- Fix typos. Changes Since V2: -- Fix if check before reset control call Changes Since V1: -- Use boolean flag for bool variable initialization instead of hard coding. Srinivasa Rao Mandadapu (2): soundwire: qcom: Add flag for software clock gating check ASoC: qcom: soundwire: Enable software clock gating requirement flag drivers/soundwire/qcom.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) -- 2.7.4

2 6

[PATCH 0/8] ctl: elem-value: rewrite getter methods
by GitHub pull_request - opened 29 Jun '22

29 Jun '22

alsa-project/alsa-gobject pull request #82 was opened from takaswie: In UAPI of Linux sound subsystem, 'struct snd_ctl_elem_values' has fixed-sized array. It's natural to retrieve the array as is. This patchset rewrites getter methods of ALSACtl.ElemValue to retrieve pointer to arrays in internal storage with annotation for fixed-sized array. ``` Takashi Sakamoto (8): ctl: elem-value: fix casting for byte data ctl: elem-value: retrieve fixed-sized array by getter method for boolean values ctl: elem-value: retrieve fixed-sized array by getter method for integer values ctl: elem-value: retrieve fixed-sized array by getter method for enumerated values ctl: elem-value: retrieve fixed-sized array by getter method for byte values ctl: elem-value: retrieve fixed-sized array by getter method for IEC 60958 values ctl: elem-value: retrieve fixed-sized array by getter method for integer 64 values update README about change of ALSACtl.ElemValue README.rst | 2 +- src/ctl/elem-value.c | 55 ++++++++++++-------------------------------- src/ctl/elem-value.h | 21 ++++++----------- 3 files changed, 23 insertions(+), 55 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/82 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/82.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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Re: Sound not working after commit bbf7d3b1c4f40eb02dd1dffb500ba00b0bff0303 on Amlogic A311D device
by Pierre-Louis Bossart 29 Jun '22

29 Jun '22

On 6/28/22 17:10, Alex Natalsson wrote: >> Any chance you could try with an updated kernel? It's not clear to me if >> this is a real issue or just a warning that's been fixed since by commit. > > I luanching 5.19-rc2 and finding this problem. Then I used git bisect > and found this commit, which broke my audio system. I don't know maybe > I have wrong settings, but for me this real problem because sound not > working.... > >> You may also want to open a bugzilla issue > Yes, I can. If problem not in my config settings. It's great that you've been able to bisect. What's surprising is that you have a long list of errors reported before the kernel oops, e.g. [ 243.188688] fe.dai-link-1: ASoC: no backend DAIs enabled for fe.dai-link-1 [ 243.188702] fe.dai-link-1: ASoC: dpcm_fe_dai_prepare() failed (-22) [ 243.188919] fe.dai-link-1: ASoC: no backend DAIs enabled for fe.dai-link-1 [ 243.188922] fe.dai-link-1: ASoC: dpcm_fe_dai_prepare() failed (-22) Did those appear with this commit? That's usually a sign of missing mixer settings to connect FE and BE, not sure how this would appear because of change in the locking. It would also help if you could add dev_err logs in dpcm_be_connect() to see in which code section the oops occurs. In theory we have all kinds of checks before calling dpcm_be_connect() but we've obviously missing something.

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[PATCH v2] ASoC: qcom: Add driver support for audioreach solution
by Srinivasa Rao Mandadapu 28 Jun '22

28 Jun '22

Add Machine driver support for audioreach solution, which uses ADSP in SC7280 based paltforms. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- Changes Since V1: -- Remove audioreach compatible name. -- Remove dt-binding patch. sound/soc/qcom/sc7280.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/qcom/sc7280.c b/sound/soc/qcom/sc7280.c index 34cdb99..da7469a 100644 --- a/sound/soc/qcom/sc7280.c +++ b/sound/soc/qcom/sc7280.c @@ -19,9 +19,11 @@ #include "../codecs/rt5682s.h" #include "common.h" #include "lpass.h" +#include "qdsp6/q6afe.h" #define DEFAULT_MCLK_RATE 19200000 #define RT5682_PLL_FREQ (48000 * 512) +#define MI2S_BCLK_RATE 1536000 struct sc7280_snd_data { struct snd_soc_card card; @@ -79,6 +81,7 @@ static int sc7280_headset_init(struct snd_soc_pcm_runtime *rtd) case MI2S_PRIMARY: case LPASS_CDC_DMA_RX0: case LPASS_CDC_DMA_TX3: + case TX_CODEC_DMA_TX_3: for_each_rtd_codec_dais(rtd, i, codec_dai) { rval = snd_soc_component_set_jack(component, &pdata->hs_jack, NULL); if (rval != 0 && rval != -ENOTSUPP) { @@ -164,10 +167,14 @@ static int sc7280_init(struct snd_soc_pcm_runtime *rtd) switch (cpu_dai->id) { case MI2S_PRIMARY: case LPASS_CDC_DMA_TX3: + case TX_CODEC_DMA_TX_3: return sc7280_headset_init(rtd); case LPASS_CDC_DMA_RX0: case LPASS_CDC_DMA_VA_TX0: case MI2S_SECONDARY: + case RX_CODEC_DMA_RX_0: + case SECONDARY_MI2S_RX: + case VA_CODEC_DMA_TX_0: return 0; case LPASS_DP_RX: return sc7280_hdmi_init(rtd); @@ -195,6 +202,10 @@ static int sc7280_snd_hw_params(struct snd_pcm_substream *substream, switch (cpu_dai->id) { case LPASS_CDC_DMA_TX3: case LPASS_CDC_DMA_RX0: + case RX_CODEC_DMA_RX_0: + case SECONDARY_MI2S_RX: + case TX_CODEC_DMA_TX_3: + case VA_CODEC_DMA_TX_0: for_each_rtd_codec_dais(rtd, i, codec_dai) { sruntime = snd_soc_dai_get_stream(codec_dai, substream->stream); if (sruntime != ERR_PTR(-ENOTSUPP)) @@ -245,6 +256,9 @@ static int sc7280_snd_prepare(struct snd_pcm_substream *substream) switch (cpu_dai->id) { case LPASS_CDC_DMA_RX0: case LPASS_CDC_DMA_TX3: + case RX_CODEC_DMA_RX_0: + case TX_CODEC_DMA_TX_3: + case VA_CODEC_DMA_TX_0: return sc7280_snd_swr_prepare(substream); default: break; @@ -263,6 +277,9 @@ static int sc7280_snd_hw_free(struct snd_pcm_substream *substream) switch (cpu_dai->id) { case LPASS_CDC_DMA_RX0: case LPASS_CDC_DMA_TX3: + case RX_CODEC_DMA_RX_0: + case TX_CODEC_DMA_TX_3: + case VA_CODEC_DMA_TX_0: if (sruntime && data->stream_prepared[cpu_dai->id]) { sdw_disable_stream(sruntime); sdw_deprepare_stream(sruntime); @@ -291,6 +308,10 @@ static void sc7280_snd_shutdown(struct snd_pcm_substream *substream) SNDRV_PCM_STREAM_PLAYBACK); } break; + case SECONDARY_MI2S_RX: + snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT, + 0, SNDRV_PCM_STREAM_PLAYBACK); + break; default: break; } @@ -298,14 +319,26 @@ static void sc7280_snd_shutdown(struct snd_pcm_substream *substream) static int sc7280_snd_startup(struct snd_pcm_substream *substream) { + unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; + unsigned int codec_dai_fmt = SND_SOC_DAIFMT_CBS_CFS; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret = 0; switch (cpu_dai->id) { case MI2S_PRIMARY: ret = sc7280_rt5682_init(rtd); break; + case SECONDARY_MI2S_RX: + codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S; + + snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT, + MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + + snd_soc_dai_set_fmt(cpu_dai, fmt); + snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); + break; default: break; } -- 2.7.4

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