May 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 6.3 32/67] ASoC: SOF: pcm: fix pm_runtime imbalance in error handling
by Sasha Levin 25 May '23

25 May '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit da0fe8fd515a471d373acc3682bfb5522cca4d55 ] When an error occurs, we need to make sure the device can pm_runtime suspend instead of keeping it active. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-3-peter.ujfalusi@linux.intel.… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/pcm.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 445acb5c3a21b..2570f33db9f3e 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -616,16 +616,17 @@ static int sof_pcm_probe(struct snd_soc_component *component) "%s/%s", plat_data->tplg_filename_prefix, plat_data->tplg_filename); - if (!tplg_filename) - return -ENOMEM; + if (!tplg_filename) { + ret = -ENOMEM; + goto pm_error; + } ret = snd_sof_load_topology(component, tplg_filename); - if (ret < 0) { + if (ret < 0) dev_err(component->dev, "error: failed to load DSP topology %d\n", ret); - return ret; - } +pm_error: pm_runtime_mark_last_busy(component->dev); pm_runtime_put_autosuspend(component->dev); -- 2.39.2

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[PATCH AUTOSEL 6.3 31/67] ASoC: SOF: debug: conditionally bump runtime_pm counter on exceptions
by Sasha Levin 25 May '23

25 May '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 3de975862f985f1c9e225a0d13aa3d501373f7c3 ] When a firmware IPC error happens during a pm_runtime suspend, we ignore the error and suspend anyways. However, the code unconditionally increases the runtime_pm counter. This results in a confusing configuration where the code will suspend, resume but never suspend again due to the use of pm_runtime_get_noresume(). The intent of the counter increase was to prevent entry in D3, but if that transition to D3 is already started it cannot be stopped. In addition, there's no point in that case in trying to prevent anything, the firmware error is handled and the next resume will re-initialize the firmware completely. This patch changes the logic to prevent suspend when the device is pm_runtime active and has a use_count > 0. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-2-peter.ujfalusi@linux.intel.… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/debug.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index ade0507328af4..5042312b1b98d 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -437,8 +437,8 @@ void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg) /* should we prevent DSP entering D3 ? */ if (!sdev->ipc_dump_printed) dev_info(sdev->dev, - "preventing DSP entering D3 state to preserve context\n"); - pm_runtime_get_noresume(sdev->dev); + "Attempting to prevent DSP from entering D3 state to preserve context\n"); + pm_runtime_get_if_in_use(sdev->dev); } /* dump vital information to the logs */ -- 2.39.2

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[PATCH AUTOSEL 6.3 14/67] ASoC: ssm2602: Add workaround for playback distortions
by Sasha Levin 25 May '23

25 May '23

From: Paweł Anikiel <pan(a)semihalf.com> [ Upstream commit f63550e2b165208a2f382afcaf5551df9569e1d4 ] Apply a workaround for what appears to be a hardware quirk. The problem seems to happen when enabling "whole chip power" (bit D7 register R6) for the very first time after the chip receives power. If either "output" (D4) or "DAC" (D3) aren't powered on at that time, playback becomes very distorted later on. This happens on the Google Chameleon v3, as well as on a ZYBO Z7-10: https://ez.analog.com/audio/f/q-a/543726/solved-ssm2603-right-output-offset… I suspect this happens only when using an external MCLK signal (which is the case for both of these boards). Here are some experiments run on a Google Chameleon v3. These were run in userspace using a wrapper around the i2cset utility: ssmset() { i2cset -y 0 0x1a $(($1*2)) $2 } For each of the following sequences, we apply power to the ssm2603 chip, set the configuration registers R0-R5 and R7-R8, run the selected sequence, and check for distortions on playback. ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac OK ssmset 0x09 0x01 # core ssmset 0x06 0x87 # out, dac ssmset 0x06 0x07 # chip OK (disable MCLK) ssmset 0x09 0x01 # core ssmset 0x06 0x1f # chip ssmset 0x06 0x07 # out, dac (enable MCLK) OK ssmset 0x09 0x01 # core ssmset 0x06 0x1f # chip ssmset 0x06 0x07 # out, dac NOT OK ssmset 0x06 0x1f # chip ssmset 0x09 0x01 # core ssmset 0x06 0x07 # out, dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x0f # chip, out ssmset 0x06 0x07 # dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x17 # chip, dac ssmset 0x06 0x07 # out NOT OK For each of the following sequences, we apply power to the ssm2603 chip, run the selected sequence, issue a reset with R15, configure R0-R5 and R7-R8, run one of the NOT OK sequences from above, and check for distortions. ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac OK (disable MCLK) ssmset 0x09 0x01 # core ssmset 0x06 0x07 # chip, out, dac (enable MCLK after reset) NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x17 # chip, dac NOT OK ssmset 0x09 0x01 # core ssmset 0x06 0x0f # chip, out NOT OK ssmset 0x06 0x07 # chip, out, dac NOT OK Signed-off-by: Paweł Anikiel <pan(a)semihalf.com Link: https://lore.kernel.org/r/20230508113037.137627-8-pan@semihalf.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/ssm2602.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index cbbe83b85adaf..cf7927222be15 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -53,6 +53,18 @@ static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = { { .reg = 0x09, .def = 0x0000 } }; +/* + * ssm2602 register patch + * Workaround for playback distortions after power up: activates digital + * core, and then powers on output, DAC, and whole chip at the same time + */ + +static const struct reg_sequence ssm2602_patch[] = { + { SSM2602_ACTIVE, 0x01 }, + { SSM2602_PWR, 0x07 }, + { SSM2602_RESET, 0x00 }, +}; + /*Appending several "None"s just for OSS mixer use*/ static const char *ssm2602_input_select[] = { @@ -589,6 +601,9 @@ static int ssm260x_component_probe(struct snd_soc_component *component) return ret; } + regmap_register_patch(ssm2602->regmap, ssm2602_patch, + ARRAY_SIZE(ssm2602_patch)); + /* set the update bits */ regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL, LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH); -- 2.39.2

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[PATCH AUTOSEL 6.3 13/67] ALSA: hda/realtek: Add quirks for ASUS GU604V and GU603V
by Sasha Levin 25 May '23

25 May '23

From: Alexandru Sorodoc <ealex95(a)gmail.com> [ Upstream commit 4b963ae1df6426f0e51de64133d379d9bde50c48 ] These models use 2 CS35L41 amplifiers using SPI for down-facing speakers. alc285_fixup_speaker2_to_dac1 is needed to fix volume control of the down-facing speakers. Pin configs are needed to enable headset mic detection. Note that these models lack the ACPI _DSD properties needed to initialize the amplifiers. They can be added during boot to get working sound out of the speakers: https://gist.github.com/lamperez/862763881c0e1c812392b5574727f6ff Signed-off-by: Alexandru Sorodoc <ealex95(a)gmail.com> Link: https://lore.kernel.org/r/20230511161510.315170-1-ealex95@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 172ffc2c332b7..543e0adde3faf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7063,6 +7063,8 @@ enum { ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, ALC285_FIXUP_SPEAKER2_TO_DAC1, + ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1, + ALC285_FIXUP_ASUS_HEADSET_MIC, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, @@ -8033,6 +8035,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, + [ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + .chained = true, + .chain_id = ALC245_FIXUP_CS35L41_SPI_2 + }, + [ALC285_FIXUP_ASUS_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, + { 0x1b, 0x03a11c30 }, + { } + }, + .chained = true, + .chain_id = ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1 + }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -9500,6 +9518,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), + SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), -- 2.39.2

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[PATCH AUTOSEL 6.3 12/67] ASoC: dt-bindings: Adjust #sound-dai-cells on TI's single-DAI codecs
by Sasha Levin 25 May '23

25 May '23

From: Martin Povišer <povik+lin(a)cutebit.org> [ Upstream commit efb2bfd7b3d210c479b9361c176d7426e5eb8663 ] A bunch of TI's codecs have binding schemas which force #sound-dai-cells to one despite those codecs only having a single DAI. Allow for bindings with zero DAI cells and deprecate the former non-zero value. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org Link: https://lore.kernel.org/r/20230509153412.62847-1-povik+lin@cutebit.org Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- Documentation/devicetree/bindings/sound/tas2562.yaml | 6 ++++-- Documentation/devicetree/bindings/sound/tas2770.yaml | 6 ++++-- Documentation/devicetree/bindings/sound/tas27xx.yaml | 6 ++++-- 3 files changed, 12 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index 1085592cefccc..81218c07079a8 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -55,7 +55,9 @@ properties: description: TDM TX current sense time slot. '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -72,7 +74,7 @@ examples: codec: codec@4c { compatible = "ti,tas2562"; reg = <0x4c>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; shutdown-gpios = <&gpio1 15 0>; diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml index 982949ba8a4be..cdb493db47f9b 100644 --- a/Documentation/devicetree/bindings/sound/tas2770.yaml +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -57,7 +57,9 @@ properties: - 1 # Falling edge '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -74,7 +76,7 @@ examples: codec: codec@41 { compatible = "ti,tas2770"; reg = <0x41>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpio = <&gpio1 15 0>; diff --git a/Documentation/devicetree/bindings/sound/tas27xx.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml index 0957dd435bb4b..2ef05aacc167a 100644 --- a/Documentation/devicetree/bindings/sound/tas27xx.yaml +++ b/Documentation/devicetree/bindings/sound/tas27xx.yaml @@ -50,7 +50,9 @@ properties: description: TDM TX voltage sense time slot. '#sound-dai-cells': - const: 1 + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] required: - compatible @@ -67,7 +69,7 @@ examples: codec: codec@38 { compatible = "ti,tas2764"; reg = <0x38>; - #sound-dai-cells = <1>; + #sound-dai-cells = <0>; interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpios = <&gpio1 15 0>; -- 2.39.2

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[PATCH AUTOSEL 6.3 11/67] ASoC: jz4740-i2s: Make I2S divider calculations more robust
by Sasha Levin 25 May '23

25 May '23

From: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com> [ Upstream commit ad721bc919edfd8b4b06977458a412011e2f0c50 ] When the CPU supplies bit/frame clocks, the system clock (clk_i2s) is divided to produce the bit clock. This is a simple 1/N divider with a fairly limited range, so for a given system clock frequency only a few sample rates can be produced. Usually a wider range of sample rates is supported by varying the system clock frequency. The old calculation method was not very robust and could easily produce the wrong clock rate, especially with non-standard rates. For example, if the system clock is 1.99x the target bit clock rate, the divider would be calculated as 1 instead of the more accurate 2. Instead, use a more accurate method that considers two adjacent divider settings and selects the one that produces the least error versus the requested rate. If the error is 5% or higher then the rate setting is rejected to prevent garbled audio. Skip divider calculation when the codec is supplying both the bit and frame clock; in that case, the divider outputs are unused and we don't want to constrain the sample rate. Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com Link: https://lore.kernel.org/r/20230509125134.208129-1-aidanmacdonald.0x0@gmail.… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/jz4740/jz4740-i2s.c | 54 ++++++++++++++++++++++++++++++++--- 1 file changed, 50 insertions(+), 4 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 6d9cfe0a50411..d0f6c945d9aee 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -218,18 +218,48 @@ static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int jz4740_i2s_get_i2sdiv(unsigned long mclk, unsigned long rate, + unsigned long i2sdiv_max) +{ + unsigned long div, rate1, rate2, err1, err2; + + div = mclk / (64 * rate); + if (div == 0) + div = 1; + + rate1 = mclk / (64 * div); + rate2 = mclk / (64 * (div + 1)); + + err1 = abs(rate1 - rate); + err2 = abs(rate2 - rate); + + /* + * Choose the divider that produces the smallest error in the + * output rate and reject dividers with a 5% or higher error. + * In the event that both dividers are outside the acceptable + * error margin, reject the rate to prevent distorted audio. + * (The number 5% is arbitrary.) + */ + if (div <= i2sdiv_max && err1 <= err2 && err1 < rate/20) + return div; + if (div < i2sdiv_max && err2 < rate/20) + return div + 1; + + return -EINVAL; +} + static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); struct regmap_field *div_field; + unsigned long i2sdiv_max; unsigned int sample_size; - uint32_t ctrl; - int div; + uint32_t ctrl, conf; + int div = 1; regmap_read(i2s->regmap, JZ_REG_AIC_CTRL, &ctrl); - - div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params)); + regmap_read(i2s->regmap, JZ_REG_AIC_CONF, &conf); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: @@ -258,11 +288,27 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO; div_field = i2s->field_i2sdiv_playback; + i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_playback.msb, + i2s->soc_info->field_i2sdiv_playback.lsb); } else { ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE; ctrl |= FIELD_PREP(JZ_AIC_CTRL_INPUT_SAMPLE_SIZE, sample_size); div_field = i2s->field_i2sdiv_capture; + i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_capture.msb, + i2s->soc_info->field_i2sdiv_capture.lsb); + } + + /* + * Only calculate I2SDIV if we're supplying the bit or frame clock. + * If the codec is supplying both clocks then the divider output is + * unused, and we don't want it to limit the allowed sample rates. + */ + if (conf & (JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER)) { + div = jz4740_i2s_get_i2sdiv(clk_get_rate(i2s->clk_i2s), + params_rate(params), i2sdiv_max); + if (div < 0) + return div; } regmap_write(i2s->regmap, JZ_REG_AIC_CTRL, ctrl); -- 2.39.2

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[PATCH AUTOSEL 6.3 08/67] ASoC: SOF: amd: Fix NULL pointer crash in acp_sof_ipc_msg_data function
by Sasha Levin 25 May '23

25 May '23

From: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> [ Upstream commit 051d71e073614a72ad423d6dacba37a7eeff274d ] Check substream and runtime variables before assigning. Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com Link: https://lore.kernel.org/r/20230508070510.6100-1-Vsujithkumar.Reddy@amd.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/amd/acp-ipc.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp-ipc.c b/sound/soc/sof/amd/acp-ipc.c index 4e0c48a361599..749e856dc6011 100644 --- a/sound/soc/sof/amd/acp-ipc.c +++ b/sound/soc/sof/amd/acp-ipc.c @@ -209,7 +209,12 @@ int acp_sof_ipc_msg_data(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *sp acp_mailbox_read(sdev, offset, p, sz); } else { struct snd_pcm_substream *substream = sps->substream; - struct acp_dsp_stream *stream = substream->runtime->private_data; + struct acp_dsp_stream *stream; + + if (!substream || !substream->runtime) + return -ESTRPIPE; + + stream = substream->runtime->private_data; if (!stream) return -ESTRPIPE; -- 2.39.2

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[PATCH AUTOSEL 6.3 05/67] ASoC: dwc: limit the number of overrun messages
by Sasha Levin 25 May '23

25 May '23

From: Maxim Kochetkov <fido_max(a)inbox.ru> [ Upstream commit ab6ecfbf40fccf74b6ec2ba7ed6dd2fc024c3af2 ] On slow CPU (FPGA/QEMU emulated) printing overrun messages from interrupt handler to uart console may leads to more overrun errors. So use dev_err_ratelimited to limit the number of error messages. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru Link: https://lore.kernel.org/r/20230505062820.21840-1-fido_max@inbox.ru Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 7f7dd07c63b2f..3496301582b22 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -132,13 +132,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) /* Error Handling: TX */ if (isr[i] & ISR_TXFO) { - dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i); irq_valid = true; } /* Error Handling: TX */ if (isr[i] & ISR_RXFO) { - dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i); + dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; } } -- 2.39.2

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[PATCH AUTOSEL 6.3 04/67] ASoC: amd: yc: Add DMI entry to support System76 Pangolin 12
by Sasha Levin 25 May '23

25 May '23

From: Jeremy Soller <jeremy(a)system76.com> [ Upstream commit 7b9891ad25246b18b5ccc19518da7abc7763aa0a ] Add pang12 quirk to enable the internal microphone. Signed-off-by: Jeremy Soller <jeremy(a)system76.com Signed-off-by: Tim Crawford <tcrawford(a)system76.com Link: https://lore.kernel.org/r/20230505161458.19676-1-tcrawford@system76.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 0acdf0156f075..b844f321d139e 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -276,6 +276,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "8A22"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "System76"), + DMI_MATCH(DMI_PRODUCT_VERSION, "pang12"), + } + }, {} }; -- 2.39.2

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[PATCH v2 00/12] ALSA: hda: Adding support for CS35L56 on HDA systems
by Richard Fitzgerald 25 May '23

25 May '23

This set of patches adds support for using the CS35L56 boosted smart amplifier on HDA systems. In these systems the CS35L56 audio is routed through a HDA-to-I2S bridge codec. This doesn't include the changes to the Realtek driver to actually hook up the CS35L56 driver, because we don't yet have the QUIRK IDs to associate it with. But we want to publish the driver now so that it is available for bringing up hardware with the CS35L56. The first 9 patches are moving code out of the ASoC driver and into the shared library so that it can be shared with the HDA driver. Patch #10 fixes missing #includes in the HDA headers so that the CS35L56 driver doesn't have to #include headers that it doesn't use. Finally, #11 and #12 actually add the support for CS35L56 on HDA. CHANGES SINCE V1 Squash the fix for passing the correct pointer to cs35l56_irq() into patch #1, where the pointer change to cs35l56_irq() was made. Richard Fitzgerald (4): ASoC: cs35l56: Move runtime suspend/resume to shared library ASoC: cs35l56: Move cs_dsp init into shared library ASoC: cs35l56: Move part of cs35l56_init() to shared library ALSA: hda: Fix missing header dependencies Simon Trimmer (8): ASoC: cs35l56: Move shared data into a common data structure ASoC: cs35l56: Make cs35l56_system_reset() code more generic ASoC: cs35l56: Convert utility functions to use common data structure ASoC: cs35l56: Move utility functions to shared file ASoC: cs35l56: Make common function for control port wait ASoC: cs35l56: Make a common function to shutdown the DSP ALSA: hda: Add mute_hook to hda_component ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier include/sound/cs35l56.h | 29 +- sound/pci/hda/Kconfig | 31 + sound/pci/hda/Makefile | 6 + sound/pci/hda/cs35l56_hda.c | 995 ++++++++++++++++++++++++++++++ sound/pci/hda/cs35l56_hda.h | 48 ++ sound/pci/hda/cs35l56_hda_i2c.c | 69 +++ sound/pci/hda/cs35l56_hda_spi.c | 68 ++ sound/pci/hda/hda_auto_parser.h | 2 + sound/pci/hda/hda_component.h | 1 + sound/pci/hda/hda_generic.h | 3 + sound/soc/codecs/cs35l56-i2c.c | 14 +- sound/soc/codecs/cs35l56-sdw.c | 72 +-- sound/soc/codecs/cs35l56-shared.c | 459 +++++++++++++- sound/soc/codecs/cs35l56-spi.c | 10 +- sound/soc/codecs/cs35l56.c | 648 ++++--------------- sound/soc/codecs/cs35l56.h | 15 +- 16 files changed, 1878 insertions(+), 592 deletions(-) create mode 100644 sound/pci/hda/cs35l56_hda.c create mode 100644 sound/pci/hda/cs35l56_hda.h create mode 100644 sound/pci/hda/cs35l56_hda_i2c.c create mode 100644 sound/pci/hda/cs35l56_hda_spi.c -- 2.30.2

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