[PATCH AUTOSEL 6.3 04/67] ASoC: amd: yc: Add DMI entry to support System76 Pangolin 12

[PATCH AUTOSEL 6.3 04/67] ASoC: amd: yc: Add DMI entry to support System76 Pangolin 12

[Sasha Levin]

From: Jeremy Soller <jeremy@system76.com>

[ Upstream commit 7b9891ad25246b18b5ccc19518da7abc7763aa0a ]

Add pang12 quirk to enable the internal microphone.

Signed-off-by: Jeremy Soller <jeremy@system76.com
Signed-off-by: Tim Crawford <tcrawford@system76.com
Link: https://lore.kernel.org/r/20230505161458.19676-1-tcrawford@system76.com
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@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

[PATCH AUTOSEL 6.3 05/67] ASoC: dwc: limit the number of overrun messages

[Sasha Levin]

From: Maxim Kochetkov <fido_max@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@inbox.ru
Link: https://lore.kernel.org/r/20230505062820.21840-1-fido_max@inbox.ru
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@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

[PATCH AUTOSEL 6.3 08/67] ASoC: SOF: amd: Fix NULL pointer crash in acp_sof_ipc_msg_data function

[Sasha Levin]

From: V sujith kumar Reddy <Vsujithkumar.Reddy@amd.com>

[ Upstream commit 051d71e073614a72ad423d6dacba37a7eeff274d ]

Check substream and runtime variables before assigning.

Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy@amd.com
Link: https://lore.kernel.org/r/20230508070510.6100-1-Vsujithkumar.Reddy@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@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

[PATCH AUTOSEL 6.3 11/67] ASoC: jz4740-i2s: Make I2S divider calculations more robust

[Sasha Levin]

From: Aidan MacDonald <aidanmacdonald.0x0@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@gmail.com
Link: https://lore.kernel.org/r/20230509125134.208129-1-aidanmacdonald.0x0@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@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

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

[Sasha Levin]

From: Martin Povišer <povik+lin@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@cutebit.org
Link: https://lore.kernel.org/r/20230509153412.62847-1-povik+lin@cutebit.org
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@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

[PATCH AUTOSEL 6.3 13/67] ALSA: hda/realtek: Add quirks for ASUS GU604V and GU603V

[Sasha Levin]

From: Alexandru Sorodoc <ealex95@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@gmail.com>
Link: https://lore.kernel.org/r/20230511161510.315170-1-ealex95@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@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

[PATCH AUTOSEL 6.3 14/67] ASoC: ssm2602: Add workaround for playback distortions

[Sasha Levin]

From: Paweł Anikiel <pan@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-issue/480229
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@semihalf.com
Link: https://lore.kernel.org/r/20230508113037.137627-8-pan@semihalf.com
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@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

[PATCH AUTOSEL 6.3 31/67] ASoC: SOF: debug: conditionally bump runtime_pm counter on exceptions

[Sasha Levin]

From: Pierre-Louis Bossart <pierre-louis.bossart@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@linux.intel.com
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Link: https://lore.kernel.org/r/20230512103315.8921-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@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

[PATCH AUTOSEL 6.3 32/67] ASoC: SOF: pcm: fix pm_runtime imbalance in error handling

[Sasha Levin]

From: Pierre-Louis Bossart <pierre-louis.bossart@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@linux.intel.com
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Link: https://lore.kernel.org/r/20230512103315.8921-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@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

[PATCH AUTOSEL 6.3 33/67] ASoC: SOF: sof-client-probes: fix pm_runtime imbalance in error handling

[Sasha Levin]

From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

[ Upstream commit bc424273c74c1565c459c8f2a6ed95caee368d0a ]

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@linux.intel.com
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Link: https://lore.kernel.org/r/20230512103315.8921-4-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/sof-client-probes.c | 14 ++++++++------
 1 file changed, 8 insertions(+), 6 deletions(-)

diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c
index fff126808bc04..8d9e9d5f40e45 100644
--- a/sound/soc/sof/sof-client-probes.c
+++ b/sound/soc/sof/sof-client-probes.c
@@ -218,12 +218,7 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to,
 
 	ret = ipc->points_info(cdev, &desc, &num_desc);
 	if (ret < 0)
-		goto exit;
-
-	pm_runtime_mark_last_busy(dev);
-	err = pm_runtime_put_autosuspend(dev);
-	if (err < 0)
-		dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err);
+		goto pm_error;
 
 	for (i = 0; i < num_desc; i++) {
 		offset = strlen(buf);
@@ -241,6 +236,13 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to,
 	ret = simple_read_from_buffer(to, count, ppos, buf, strlen(buf));
 
 	kfree(desc);
+
+pm_error:
+	pm_runtime_mark_last_busy(dev);
+	err = pm_runtime_put_autosuspend(dev);
+	if (err < 0)
+		dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err);
+
 exit:
 	kfree(buf);
 	return ret;
-- 
2.39.2

[PATCH AUTOSEL 6.3 34/67] ASoC: SOF: pm: save io region state in case of errors in resume

[Sasha Levin]

From: Kai Vehmanen <kai.vehmanen@linux.intel.com>

[ Upstream commit 171b53be635ac15d4feafeb33946035649b1ca14 ]

If there are failures in DSP runtime resume, the device state will not
reach active and this makes it impossible e.g. to retrieve a possible
DSP panic dump via "exception" debugfs node. If
CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE=y is set, the data in
cache is stale. If debugfs cache is not used, the region simply cannot
be read.

To allow debugging these scenarios, update the debugfs cache contents in
resume error handler. User-space can then later retrieve DSP panic and
other state via debugfs (requires SOF debugfs cache to be enabled in
build).

Reported-by: Curtis Malainey <cujomalainey@chromium.org
Link: https://github.com/thesofproject/linux/issues/4274
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Reviewed-by: Curtis Malainey <cujomalainey@chromium.org
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Link: https://lore.kernel.org/r/20230512104638.21376-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sof/pm.c | 14 +++++++++++++-
 1 file changed, 13 insertions(+), 1 deletion(-)

diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index 85412aeb1ca16..40f392efd8246 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -159,7 +159,7 @@ static int sof_resume(struct device *dev, bool runtime_resume)
 		ret = tplg_ops->set_up_all_pipelines(sdev, false);
 		if (ret < 0) {
 			dev_err(sdev->dev, "Failed to restore pipeline after resume %d\n", ret);
-			return ret;
+			goto setup_fail;
 		}
 	}
 
@@ -173,6 +173,18 @@ static int sof_resume(struct device *dev, bool runtime_resume)
 			dev_err(sdev->dev, "ctx_restore IPC error during resume: %d\n", ret);
 	}
 
+setup_fail:
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE)
+	if (ret < 0) {
+		/*
+		 * Debugfs cannot be read in runtime suspend, so cache
+		 * the contents upon failure. This allows to capture
+		 * possible DSP coredump information.
+		 */
+		sof_cache_debugfs(sdev);
+	}
+#endif
+
 	return ret;
 }
 
-- 
2.39.2

[PATCH AUTOSEL 6.3 59/67] ALSA: oss: avoid missing-prototype warnings

[Sasha Levin]

From: Arnd Bergmann <arnd@arndb.de>

[ Upstream commit 040b5a046a9e18098580d3ccd029e2318fca7859 ]

Two functions are defined and used in pcm_oss.c but also optionally
used from io.c, with an optional prototype. If CONFIG_SND_PCM_OSS_PLUGINS
is disabled, this causes a warning as the functions are not static
and have no prototype:

sound/core/oss/pcm_oss.c:1235:19: error: no previous prototype for 'snd_pcm_oss_write3' [-Werror=missing-prototypes]
sound/core/oss/pcm_oss.c:1266:19: error: no previous prototype for 'snd_pcm_oss_read3' [-Werror=missing-prototypes]

Avoid this by making the prototypes unconditional.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20230516195046.550584-2-arnd@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/core/oss/pcm_plugin.h | 16 ++++++++--------
 1 file changed, 8 insertions(+), 8 deletions(-)

diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h
index 46e273bd4a786..50a6b50f5db4c 100644
--- a/sound/core/oss/pcm_plugin.h
+++ b/sound/core/oss/pcm_plugin.h
@@ -141,6 +141,14 @@ int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_channel,
 
 void *snd_pcm_plug_buf_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t size);
 void snd_pcm_plug_buf_unlock(struct snd_pcm_substream *plug, void *ptr);
+#else
+
+static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; }
+static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; }
+static inline int snd_pcm_plug_slave_format(int format, const struct snd_mask *format_mask) { return format; }
+
+#endif
+
 snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream,
 				     const char *ptr, snd_pcm_uframes_t size,
 				     int in_kernel);
@@ -151,14 +159,6 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream,
 snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream,
 				     void **bufs, snd_pcm_uframes_t frames);
 
-#else
-
-static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; }
-static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; }
-static inline int snd_pcm_plug_slave_format(int format, const struct snd_mask *format_mask) { return format; }
-
-#endif
-
 #ifdef PLUGIN_DEBUG
 #define pdprintf(fmt, args...) printk(KERN_DEBUG "plugin: " fmt, ##args)
 #else
-- 
2.39.2