From e0e8e4bce61cac674fdabd85d070e7bab1634a8b Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Mon, 3 Jun 2024 10:32:23 +0300
Subject: ASoC: SOF: Intel: hda-dai: skip tlv for dspless mode
MIME-Version: 1.0
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sof_ipc4_dma_config_tlv{} is for Audio DSP firmware only.
Don't set it in dspless mode.
Fixes: 17386cb1b48b ("ASoC: SOF: Intel: hda-dai: set dma_stream_channel_map device")
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240603073224.14726-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dai.c | 3 +++
1 file changed, 3 insertions(+)
(limited to 'sound')
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index ce675c22a5ab..a2b6dbcfa918 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -525,6 +525,9 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
return ret;
}
+ if (sdev->dspless_mode_selected)
+ return 0;
+
ipc4_copier = widget_to_copier(w);
dma_config_tlv = &ipc4_copier->dma_config_tlv[cpu_dai_id];
dma_config = &dma_config_tlv->dma_config;
--
cgit
From 3b06e137089fc0beb5ffa6a869de9a93df984072 Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Mon, 3 Jun 2024 10:32:24 +0300
Subject: ASoC: SOF: Intel: hda-dai: remove skip_tlv label
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
We just return 0 after the skip_tlv label. No need to use a label.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240603073224.14726-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dai.c | 3 +--
1 file changed, 1 insertion(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index a2b6dbcfa918..c61d298ea6b3 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -379,7 +379,7 @@ static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream,
sdev = widget_to_sdev(w);
if (sdev->dspless_mode_selected)
- goto skip_tlv;
+ return 0;
/* get stream_id */
hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream);
@@ -423,7 +423,6 @@ static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream,
dma_config->dma_stream_channel_map.device_count = 1;
dma_config->dma_priv_config_size = 0;
-skip_tlv:
return 0;
}
--
cgit
From a73a83021ae136ab6b0d08eb196d84b1d02814e9 Mon Sep 17 00:00:00 2001
From: Jeff Johnson <quic_jjohnson@quicinc.com>
Date: Sun, 2 Jun 2024 09:30:13 -0700
Subject: ASoC: mxs: add missing MODULE_DESCRIPTION() macro
make allmodconfig && make W=1 C=1 reports:
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/mxs/snd-soc-mxs-pcm.o
Add the missing invocation of the MODULE_DESCRIPTION() macro.
Signed-off-by: Jeff Johnson <quic_jjohnson@quicinc.com>
Link: https://msgid.link/r/20240602-md-snd-soc-mxs-pcm-v1-1-1e663d11328d@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mxs/mxs-pcm.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index df2e4be992d2..9bb08cadeb18 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -43,4 +43,5 @@ int mxs_pcm_platform_register(struct device *dev)
}
EXPORT_SYMBOL_GPL(mxs_pcm_platform_register);
+MODULE_DESCRIPTION("MXS ASoC PCM driver");
MODULE_LICENSE("GPL");
--
cgit
From 7478e15bcc16cbc0fa1b8c431163bf651033c088 Mon Sep 17 00:00:00 2001
From: Jeff Johnson <quic_jjohnson@quicinc.com>
Date: Sun, 2 Jun 2024 10:00:27 -0700
Subject: ASoC: fsl: add missing MODULE_DESCRIPTION() macro
make allmodconfig && make W=1 C=1 reports:
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/fsl/imx-pcm-dma.o
Add the missing invocation of the MODULE_DESCRIPTION() macro.
Signed-off-by: Jeff Johnson <quic_jjohnson@quicinc.com>
Link: https://msgid.link/r/20240602-md-snd-fsl-imx-pcm-dma-v1-1-e7efc33c6bf3@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/imx-pcm-dma.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index 14e94270911c..4fa208d6a032 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -50,4 +50,5 @@ int imx_pcm_dma_init(struct platform_device *pdev)
}
EXPORT_SYMBOL_GPL(imx_pcm_dma_init);
+MODULE_DESCRIPTION("Freescale i.MX PCM DMA interface");
MODULE_LICENSE("GPL");
--
cgit
From 968c974c08106fcf911d8d390d0f049af855d348 Mon Sep 17 00:00:00 2001
From: Jack Yu <jack.yu@realtek.com>
Date: Mon, 3 Jun 2024 10:47:16 +0000
Subject: ASoC: rt722-sdca-sdw: add silence detection register as volatile
Including silence detection register as volatile.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://msgid.link/r/c66a6bd6d220426793096b42baf85437@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt722-sdca-sdw.c | 2 ++
1 file changed, 2 insertions(+)
(limited to 'sound')
diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c
index b33da2215ade..f73ee3bf90f5 100644
--- a/sound/soc/codecs/rt722-sdca-sdw.c
+++ b/sound/soc/codecs/rt722-sdca-sdw.c
@@ -68,6 +68,7 @@ static bool rt722_sdca_mbq_readable_register(struct device *dev, unsigned int re
case 0x200007f:
case 0x2000082 ... 0x200008e:
case 0x2000090 ... 0x2000094:
+ case 0x3110000:
case 0x5300000 ... 0x5300002:
case 0x5400002:
case 0x5600000 ... 0x5600007:
@@ -125,6 +126,7 @@ static bool rt722_sdca_mbq_volatile_register(struct device *dev, unsigned int re
case 0x2000067:
case 0x2000084:
case 0x2000086:
+ case 0x3110000:
return true;
default:
return false;
--
cgit
From 65909a7e7aa8b25c9cc5f04c1fd5d6f0f1d76fcd Mon Sep 17 00:00:00 2001
From: Jeff Johnson <quic_jjohnson@quicinc.com>
Date: Mon, 3 Jun 2024 17:16:07 -0700
Subject: ASoC: qcom: add missing MODULE_DESCRIPTION() macro
make allmodconfig && make W=1 C=1 reports:
WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/qcom/snd-soc-qcom-sdw.o
Add the missing invocation of the MODULE_DESCRIPTION() macro.
Signed-off-by: Jeff Johnson <quic_jjohnson@quicinc.com>
Link: https://msgid.link/r/20240603-md-snd-soc-qcom-sdw-v1-1-101ea8bcdd38@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/sdw.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/soc/qcom/sdw.c b/sound/soc/qcom/sdw.c
index eaa8bb016e50..f2eda2ff46c0 100644
--- a/sound/soc/qcom/sdw.c
+++ b/sound/soc/qcom/sdw.c
@@ -160,4 +160,5 @@ int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream,
return 0;
}
EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_free);
+MODULE_DESCRIPTION("Qualcomm ASoC SoundWire helper functions");
MODULE_LICENSE("GPL");
--
cgit
From afe377286ad49e0b69071d2a767e2c6553f4094b Mon Sep 17 00:00:00 2001
From: Maciej Strozek <mstrozek@opensource.cirrus.com>
Date: Tue, 4 Jun 2024 14:28:43 +0100
Subject: ASoC: cs42l43: Increase default type detect time and button delay
Some problematic headsets have been discovered, to help with correctly
identifying these, the detect time must be increased. Also improve the
reliability of the impedance value from the button detect by slightly
increasing the button detect delay.
Fixes: 686b8f711b99 ("ASoC: cs42l43: Lower default type detect time")
Signed-off-by: Maciej Strozek <mstrozek@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://msgid.link/r/20240604132843.3309114-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/cs42l43-jack.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c
index 901b9dbcf585..d9ab003e166b 100644
--- a/sound/soc/codecs/cs42l43-jack.c
+++ b/sound/soc/codecs/cs42l43-jack.c
@@ -121,7 +121,7 @@ int cs42l43_set_jack(struct snd_soc_component *component,
priv->buttons[3] = 735;
}
- ret = cs42l43_find_index(priv, "cirrus,detect-us", 1000, &priv->detect_us,
+ ret = cs42l43_find_index(priv, "cirrus,detect-us", 50000, &priv->detect_us,
cs42l43_accdet_us, ARRAY_SIZE(cs42l43_accdet_us));
if (ret < 0)
goto error;
@@ -433,7 +433,7 @@ irqreturn_t cs42l43_button_press(int irq, void *data)
// Wait for 2 full cycles of comb filter to ensure good reading
queue_delayed_work(system_wq, &priv->button_press_work,
- msecs_to_jiffies(10));
+ msecs_to_jiffies(20));
return IRQ_HANDLED;
}
--
cgit
From b7c40988808f8d7426dee1e4d96a4e204de4a8bc Mon Sep 17 00:00:00 2001
From: Zhang Yi <zhangyi@everest-semi.com>
Date: Tue, 4 Jun 2024 10:19:46 +0800
Subject: ASoC: codecs: ES8326: Solve headphone detection issue
When switching between OMTP and CTIA headset, we can hear pop noise.
To solve this issue, We modified the configuration for headphone detection
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240604021946.2911-1-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/es8326.c | 8 ++++++--
1 file changed, 6 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c
index 03b539ba540f..6a4e42e5e35b 100644
--- a/sound/soc/codecs/es8326.c
+++ b/sound/soc/codecs/es8326.c
@@ -857,12 +857,16 @@ static void es8326_jack_detect_handler(struct work_struct *work)
* set auto-check mode, then restart jack_detect_work after 400ms.
* Don't report jack status.
*/
- regmap_write(es8326->regmap, ES8326_INT_SOURCE,
- (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON));
+ regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x00);
regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01);
+ regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x00);
es8326_enable_micbias(es8326->component);
usleep_range(50000, 70000);
regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00);
+ regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x10);
+ usleep_range(50000, 70000);
+ regmap_write(es8326->regmap, ES8326_INT_SOURCE,
+ (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON));
regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x1f);
regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x08);
queue_delayed_work(system_wq, &es8326->jack_detect_work,
--
cgit
From ccd8d753f0fe8f16745fa2b6be5946349731d901 Mon Sep 17 00:00:00 2001
From: Alibek Omarov <a1ba.omarov@gmail.com>
Date: Tue, 4 Jun 2024 21:47:52 +0300
Subject: ASoC: rockchip: i2s-tdm: Fix trcm mode by setting clock on right mclk
When TRCM mode is enabled, I2S RX and TX clocks are synchronized through
selected clock source. Without this fix BCLK and LRCK might get parented
to an uninitialized MCLK and the DAI will receive data at wrong pace.
However, unlike in original i2s-tdm driver, there is no need to manually
synchronize mclk_rx and mclk_tx, as only one gets used anyway.
Tested on a board with RK3568 SoC and Silergy SY24145S codec with enabled and
disabled TRCM mode.
Fixes: 9e2ab4b18ebd ("ASoC: rockchip: i2s-tdm: Fix inaccurate sampling rates")
Signed-off-by: Alibek Omarov <a1ba.omarov@gmail.com>
Reviewed-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240604184752.697313-1-a1ba.omarov@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/rockchip/rockchip_i2s_tdm.c | 13 +++++++++++--
1 file changed, 11 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c
index 9fa020ef7eab..ee517d7b5b7b 100644
--- a/sound/soc/rockchip/rockchip_i2s_tdm.c
+++ b/sound/soc/rockchip/rockchip_i2s_tdm.c
@@ -655,8 +655,17 @@ static int rockchip_i2s_tdm_hw_params(struct snd_pcm_substream *substream,
int err;
if (i2s_tdm->is_master_mode) {
- struct clk *mclk = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
- i2s_tdm->mclk_tx : i2s_tdm->mclk_rx;
+ struct clk *mclk;
+
+ if (i2s_tdm->clk_trcm == TRCM_TX) {
+ mclk = i2s_tdm->mclk_tx;
+ } else if (i2s_tdm->clk_trcm == TRCM_RX) {
+ mclk = i2s_tdm->mclk_rx;
+ } else if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ mclk = i2s_tdm->mclk_tx;
+ } else {
+ mclk = i2s_tdm->mclk_rx;
+ }
err = clk_set_rate(mclk, DEFAULT_MCLK_FS * params_rate(params));
if (err)
--
cgit
From 97d8613679eb53bd0c07d0fbd3d8471e46ba46c1 Mon Sep 17 00:00:00 2001
From: Hsin-Te Yuan <yuanhsinte@chromium.org>
Date: Fri, 31 May 2024 08:37:54 +0000
Subject: ASoC: mediatek: mt8183-da7219-max98357: Fix kcontrol name collision
Since "Headphone Switch" kcontrol name has already been used by da7219,
rename the control name from "Headphone" to "Headphones" to prevent the
colision. Also, this change makes kcontrol name align with the one in
mt8186-mt6366-da7219-max98357.c.
Fixes: 9c7388baa2053 ("ASoC: mediatek: mt8183-da7219-max98357: Map missing jack kcontrols")
Change-Id: I9ae69a4673cd04786b247cc514fdd20f878ef009
Signed-off-by: Hsin-Te Yuan <yuanhsinte@chromium.org>
Reviewed-by: Chen-Yu Tsai <wenst@chromium.org>
Link: https://msgid.link/r/20240531-da7219-v1-1-ac3343f3ae6a@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 10 +++++-----
1 file changed, 5 insertions(+), 5 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
index acaf81fd6c9b..f848e14b091a 100644
--- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
@@ -31,7 +31,7 @@ struct mt8183_da7219_max98357_priv {
static struct snd_soc_jack_pin mt8183_da7219_max98357_jack_pins[] = {
{
- .pin = "Headphone",
+ .pin = "Headphones",
.mask = SND_JACK_HEADPHONE,
},
{
@@ -626,7 +626,7 @@ static struct snd_soc_codec_conf mt6358_codec_conf[] = {
};
static const struct snd_kcontrol_new mt8183_da7219_max98357_snd_controls[] = {
- SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headphones"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("Speakers"),
SOC_DAPM_PIN_SWITCH("Line Out"),
@@ -634,7 +634,7 @@ static const struct snd_kcontrol_new mt8183_da7219_max98357_snd_controls[] = {
static const
struct snd_soc_dapm_widget mt8183_da7219_max98357_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_HP("Headphones", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_SPK("Speakers", NULL),
SND_SOC_DAPM_SPK("Line Out", NULL),
@@ -680,7 +680,7 @@ static struct snd_soc_codec_conf mt8183_da7219_rt1015_codec_conf[] = {
};
static const struct snd_kcontrol_new mt8183_da7219_rt1015_snd_controls[] = {
- SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headphones"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("Left Spk"),
SOC_DAPM_PIN_SWITCH("Right Spk"),
@@ -689,7 +689,7 @@ static const struct snd_kcontrol_new mt8183_da7219_rt1015_snd_controls[] = {
static const
struct snd_soc_dapm_widget mt8183_da7219_rt1015_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_HP("Headphones", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_SPK("Left Spk", NULL),
SND_SOC_DAPM_SPK("Right Spk", NULL),
--
cgit
From 2ed22161b19b11239aa742804549f63edd7c91e3 Mon Sep 17 00:00:00 2001
From: Andrei Simion <andrei.simion@microchip.com>
Date: Tue, 4 Jun 2024 13:10:30 +0300
Subject: ASoC: atmel: atmel-classd: Re-add dai_link->platform to fix card init
The removed dai_link->platform component cause a fail which
is exposed at runtime. (ex: when a sound tool is used)
This patch re-adds the dai_link->platform component to have
a full card registered.
Before this patch:
:~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: CLASSD [CLASSD], device 0: CLASSD PCM snd-soc-dummy-dai-0 []
Subdevices: 1/1
Subdevice #0: subdevice #0
:~$ speaker-test -t sine
speaker-test 1.2.6
Playback device is default
Stream parameters are 48000Hz, S16_LE, 1 channels
Sine wave rate is 440.0000Hz
Playback open error: -22,Invalid argument
After this patch which restores the platform component:
:~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: CLASSD [CLASSD], device 0: CLASSD PCM snd-soc-dummy-dai-0
[CLASSD PCM snd-soc-dummy-dai-0]
Subdevices: 1/1
Subdevice #0: subdevice #0
-> Resolve the playback error.
Fixes: 2f650f87c03c ("ASoC: atmel: remove unnecessary dai_link->platform")
Signed-off-by: Andrei Simion <andrei.simion@microchip.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://msgid.link/r/20240604101030.237792-1-andrei.simion@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/atmel/atmel-classd.c | 7 +++++--
1 file changed, 5 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c
index 6aed1ee443b4..ba314b279919 100644
--- a/sound/soc/atmel/atmel-classd.c
+++ b/sound/soc/atmel/atmel-classd.c
@@ -473,19 +473,22 @@ static int atmel_classd_asoc_card_init(struct device *dev,
if (!dai_link)
return -ENOMEM;
- comp = devm_kzalloc(dev, sizeof(*comp), GFP_KERNEL);
+ comp = devm_kzalloc(dev, 2 * sizeof(*comp), GFP_KERNEL);
if (!comp)
return -ENOMEM;
- dai_link->cpus = comp;
+ dai_link->cpus = &comp[0];
dai_link->codecs = &snd_soc_dummy_dlc;
+ dai_link->platforms = &comp[1];
dai_link->num_cpus = 1;
dai_link->num_codecs = 1;
+ dai_link->num_platforms = 1;
dai_link->name = "CLASSD";
dai_link->stream_name = "CLASSD PCM";
dai_link->cpus->dai_name = dev_name(dev);
+ dai_link->platforms->name = dev_name(dev);
card->dai_link = dai_link;
card->num_links = 1;
--
cgit
From 97ab304ecd95c0b1703ff8c8c3956dc6e2afe8e1 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Mon, 3 Jun 2024 12:28:15 +0200
Subject: ASoC: topology: Fix references to freed memory
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Most users after parsing a topology file, release memory used by it, so
having pointer references directly into topology file contents is wrong.
Use devm_kmemdup(), to allocate memory as needed.
Reported-by: Jason Montleon <jmontleo@redhat.com>
Link: https://github.com/thesofproject/avs-topology-xml/issues/22#issuecomment-2127892605
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240603102818.36165-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-topology.c | 27 ++++++++++++++++++++++-----
1 file changed, 22 insertions(+), 5 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 90ca37e008b3..75d9395a18ed 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1060,15 +1060,32 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
break;
}
- route->source = elem->source;
- route->sink = elem->sink;
+ route->source = devm_kmemdup(tplg->dev, elem->source,
+ min(strlen(elem->source),
+ SNDRV_CTL_ELEM_ID_NAME_MAXLEN),
+ GFP_KERNEL);
+ route->sink = devm_kmemdup(tplg->dev, elem->sink,
+ min(strlen(elem->sink), SNDRV_CTL_ELEM_ID_NAME_MAXLEN),
+ GFP_KERNEL);
+ if (!route->source || !route->sink) {
+ ret = -ENOMEM;
+ break;
+ }
/* set to NULL atm for tplg users */
route->connected = NULL;
- if (strnlen(elem->control, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 0)
+ if (strnlen(elem->control, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 0) {
route->control = NULL;
- else
- route->control = elem->control;
+ } else {
+ route->control = devm_kmemdup(tplg->dev, elem->control,
+ min(strlen(elem->control),
+ SNDRV_CTL_ELEM_ID_NAME_MAXLEN),
+ GFP_KERNEL);
+ if (!route->control) {
+ ret = -ENOMEM;
+ break;
+ }
+ }
/* add route dobj to dobj_list */
route->dobj.type = SND_SOC_DOBJ_GRAPH;
--
cgit
From fd660b1bd015e5aa9a558ee04088f2431010548d Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Mon, 3 Jun 2024 12:28:16 +0200
Subject: ASoC: Intel: avs: Fix route override
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Instead of overriding existing memory strings that may be too short,
just allocate needed memory and point the route at it.
Reported-by: Jason Montleon <jmontleo@redhat.com>
Link: https://github.com/thesofproject/avs-topology-xml/issues/22#issuecomment-2127892605
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240603102818.36165-3-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/avs/topology.c | 19 +++++++++++++++----
1 file changed, 15 insertions(+), 4 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c
index 02bae207f6ec..b6c5d94a1554 100644
--- a/sound/soc/intel/avs/topology.c
+++ b/sound/soc/intel/avs/topology.c
@@ -1545,8 +1545,8 @@ static int avs_route_load(struct snd_soc_component *comp, int index,
{
struct snd_soc_acpi_mach *mach = dev_get_platdata(comp->card->dev);
size_t len = SNDRV_CTL_ELEM_ID_NAME_MAXLEN;
- char buf[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
int ssp_port, tdm_slot;
+ char *buf;
/* See parse_link_formatted_string() for dynamic naming when(s). */
if (!avs_mach_singular_ssp(mach))
@@ -1557,13 +1557,24 @@ static int avs_route_load(struct snd_soc_component *comp, int index,
return 0;
tdm_slot = avs_mach_ssp_tdm(mach, ssp_port);
+ buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
avs_ssp_sprint(buf, len, route->source, ssp_port, tdm_slot);
- strscpy((char *)route->source, buf, len);
+ route->source = buf;
+
+ buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
avs_ssp_sprint(buf, len, route->sink, ssp_port, tdm_slot);
- strscpy((char *)route->sink, buf, len);
+ route->sink = buf;
+
if (route->control) {
+ buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
avs_ssp_sprint(buf, len, route->control, ssp_port, tdm_slot);
- strscpy((char *)route->control, buf, len);
+ route->control = buf;
}
return 0;
--
cgit
From daf0b99d4720c9f05bdb81c73b2efdb43fa9def3 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Mon, 3 Jun 2024 12:28:17 +0200
Subject: ASoC: topology: Do not assign fields that are already set
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
The routes are allocated with kzalloc(), so all fields are zeroed by
default, skip unnecessary assignments.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240603102818.36165-4-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-topology.c | 6 +-----
1 file changed, 1 insertion(+), 5 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 75d9395a18ed..1db540aaad45 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1072,11 +1072,7 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
break;
}
- /* set to NULL atm for tplg users */
- route->connected = NULL;
- if (strnlen(elem->control, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 0) {
- route->control = NULL;
- } else {
+ if (strnlen(elem->control, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) != 0) {
route->control = devm_kmemdup(tplg->dev, elem->control,
min(strlen(elem->control),
SNDRV_CTL_ELEM_ID_NAME_MAXLEN),
--
cgit
From e0e7bc2cbee93778c4ad7d9a792d425ffb5af6f7 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Mon, 3 Jun 2024 12:28:18 +0200
Subject: ASoC: topology: Clean up route loading
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Instead of using very long macro name, assign it to shorter variable
and use it instead. While doing that, we can reduce multiple if checks
using this define to one.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240603102818.36165-5-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-topology.c | 26 ++++++++------------------
1 file changed, 8 insertions(+), 18 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 1db540aaad45..2ac442644ed4 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1021,6 +1021,7 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
struct snd_soc_tplg_hdr *hdr)
{
struct snd_soc_dapm_context *dapm = &tplg->comp->dapm;
+ const size_t maxlen = SNDRV_CTL_ELEM_ID_NAME_MAXLEN;
struct snd_soc_tplg_dapm_graph_elem *elem;
struct snd_soc_dapm_route *route;
int count, i;
@@ -1044,38 +1045,27 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
tplg->pos += sizeof(struct snd_soc_tplg_dapm_graph_elem);
/* validate routes */
- if (strnlen(elem->source, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) ==
- SNDRV_CTL_ELEM_ID_NAME_MAXLEN) {
- ret = -EINVAL;
- break;
- }
- if (strnlen(elem->sink, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) ==
- SNDRV_CTL_ELEM_ID_NAME_MAXLEN) {
- ret = -EINVAL;
- break;
- }
- if (strnlen(elem->control, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) ==
- SNDRV_CTL_ELEM_ID_NAME_MAXLEN) {
+ if ((strnlen(elem->source, maxlen) == maxlen) ||
+ (strnlen(elem->sink, maxlen) == maxlen) ||
+ (strnlen(elem->control, maxlen) == maxlen)) {
ret = -EINVAL;
break;
}
route->source = devm_kmemdup(tplg->dev, elem->source,
- min(strlen(elem->source),
- SNDRV_CTL_ELEM_ID_NAME_MAXLEN),
+ min(strlen(elem->source), maxlen),
GFP_KERNEL);
route->sink = devm_kmemdup(tplg->dev, elem->sink,
- min(strlen(elem->sink), SNDRV_CTL_ELEM_ID_NAME_MAXLEN),
+ min(strlen(elem->sink), maxlen),
GFP_KERNEL);
if (!route->source || !route->sink) {
ret = -ENOMEM;
break;
}
- if (strnlen(elem->control, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) != 0) {
+ if (strnlen(elem->control, maxlen) != 0) {
route->control = devm_kmemdup(tplg->dev, elem->control,
- min(strlen(elem->control),
- SNDRV_CTL_ELEM_ID_NAME_MAXLEN),
+ min(strlen(elem->control), maxlen),
GFP_KERNEL);
if (!route->control) {
ret = -ENOMEM;
--
cgit
From e3209a1827646daaab744aa6a5767b1f57fb5385 Mon Sep 17 00:00:00 2001
From: Thomas GENTY <tomlohave@gmail.com>
Date: Sat, 8 Jun 2024 19:02:51 +0200
Subject: bytcr_rt5640 : inverse jack detect for Archos 101 cesium
When headphones are plugged in, they appear absent; when they are removed,
they appear present.
Add a specific entry in bytcr_rt5640 for this device
Signed-off-by: Thomas GENTY <tomlohave@gmail.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++
1 file changed, 11 insertions(+)
(limited to 'sound')
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index b41a1147f1c3..a64d1989e28a 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -610,6 +610,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
BYT_RT5640_SSP0_AIF1 |
BYT_RT5640_MCLK_EN),
},
+ {
+ .matches = {
+ DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"),
+ },
+ .driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+ BYT_RT5640_JD_NOT_INV |
+ BYT_RT5640_DIFF_MIC |
+ BYT_RT5640_SSP0_AIF1 |
+ BYT_RT5640_MCLK_EN),
+ },
{
.matches = {
DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"),
--
cgit
From e8343410ddf08fc36a9b9cc7c51a4e53a262d4c6 Mon Sep 17 00:00:00 2001
From: Jai Luthra <j-luthra@ti.com>
Date: Tue, 11 Jun 2024 18:02:55 +0530
Subject: ALSA: dmaengine: Synchronize dma channel after drop()
Sometimes the stream may be stopped due to XRUN events, in which case
the userspace can call snd_pcm_drop() and snd_pcm_prepare() to stop and
start the stream again.
In these cases, we must wait for the DMA channel to synchronize before
marking the stream as prepared for playback, as the DMA channel gets
stopped by drop() without any synchronization. Make sure the ALSA core
synchronizes the DMA channel by adding a sync_stop() hook.
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Signed-off-by: Jai Luthra <j-luthra@ti.com>
Link: https://lore.kernel.org/r/20240611-asoc_next-v3-1-fcfd84b12164@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/core/pcm_dmaengine.c | 10 ++++++++++
sound/soc/soc-generic-dmaengine-pcm.c | 8 ++++++++
2 files changed, 18 insertions(+)
(limited to 'sound')
diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c
index 12aa1cef11a1..ed07fa5693d2 100644
--- a/sound/core/pcm_dmaengine.c
+++ b/sound/core/pcm_dmaengine.c
@@ -349,6 +349,16 @@ int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream,
}
EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open_request_chan);
+int snd_dmaengine_pcm_sync_stop(struct snd_pcm_substream *substream)
+{
+ struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
+
+ dmaengine_synchronize(prtd->dma_chan);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_sync_stop);
+
/**
* snd_dmaengine_pcm_close - Close a dmaengine based PCM substream
* @substream: PCM substream
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index ea3bc9318412..a63e942fdc0b 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -318,6 +318,12 @@ static int dmaengine_copy(struct snd_soc_component *component,
return 0;
}
+static int dmaengine_pcm_sync_stop(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ return snd_dmaengine_pcm_sync_stop(substream);
+}
+
static const struct snd_soc_component_driver dmaengine_pcm_component = {
.name = SND_DMAENGINE_PCM_DRV_NAME,
.probe_order = SND_SOC_COMP_ORDER_LATE,
@@ -327,6 +333,7 @@ static const struct snd_soc_component_driver dmaengine_pcm_component = {
.trigger = dmaengine_pcm_trigger,
.pointer = dmaengine_pcm_pointer,
.pcm_construct = dmaengine_pcm_new,
+ .sync_stop = dmaengine_pcm_sync_stop,
};
static const struct snd_soc_component_driver dmaengine_pcm_component_process = {
@@ -339,6 +346,7 @@ static const struct snd_soc_component_driver dmaengine_pcm_component_process = {
.pointer = dmaengine_pcm_pointer,
.copy = dmaengine_copy,
.pcm_construct = dmaengine_pcm_new,
+ .sync_stop = dmaengine_pcm_sync_stop,
};
static const char * const dmaengine_pcm_dma_channel_names[] = {
--
cgit
From c5dcf8ab10606e76c1d8a0ec77f27d84a392e874 Mon Sep 17 00:00:00 2001
From: Jai Luthra <j-luthra@ti.com>
Date: Tue, 11 Jun 2024 18:02:56 +0530
Subject: ASoC: ti: davinci-mcasp: Set min period size using FIFO config
The minimum period size was enforced to 64 as older devices integrating
McASP with EDMA used an internal FIFO of 64 samples.
With UDMA based platforms this internal McASP FIFO is optional, as the
DMA engine internally does some buffering which is already accounted for
when registering the platform. So we should read the actual FIFO
configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to
64.
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Signed-off-by: Jai Luthra <j-luthra@ti.com>
Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/davinci-mcasp.c | 9 +++++++--
1 file changed, 7 insertions(+), 2 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 1e760c315521..2b1ed91a736c 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -1472,10 +1472,11 @@ static int davinci_mcasp_hw_rule_min_periodsize(
{
struct snd_interval *period_size = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+ u8 numevt = *((u8 *)rule->private);
struct snd_interval frames;
snd_interval_any(&frames);
- frames.min = 64;
+ frames.min = numevt;
frames.integer = 1;
return snd_interval_refine(period_size, &frames);
@@ -1490,6 +1491,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
u32 max_channels = 0;
int i, dir, ret;
int tdm_slots = mcasp->tdm_slots;
+ u8 *numevt;
/* Do not allow more then one stream per direction */
if (mcasp->substreams[substream->stream])
@@ -1589,9 +1591,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
return ret;
}
+ numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ &mcasp->txnumevt :
+ &mcasp->rxnumevt;
snd_pcm_hw_rule_add(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
- davinci_mcasp_hw_rule_min_periodsize, NULL,
+ davinci_mcasp_hw_rule_min_periodsize, numevt,
SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1);
return 0;
--
cgit
From 524d3f126362b6033e92cbe107ae2158d7fbff94 Mon Sep 17 00:00:00 2001
From: Primoz Fiser <primoz.fiser@norik.com>
Date: Mon, 10 Jun 2024 14:58:47 +0200
Subject: ASoC: ti: omap-hdmi: Fix too long driver name
Set driver name to "HDMI". This simplifies the code and gets rid of
the following error messages:
ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e'
Signed-off-by: Primoz Fiser <primoz.fiser@norik.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/ti/omap-hdmi.c | 6 +-----
1 file changed, 1 insertion(+), 5 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c
index 639bc83f4263..cf43ac19c4a6 100644
--- a/sound/soc/ti/omap-hdmi.c
+++ b/sound/soc/ti/omap-hdmi.c
@@ -354,11 +354,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
if (!card)
return -ENOMEM;
- card->name = devm_kasprintf(dev, GFP_KERNEL,
- "HDMI %s", dev_name(ad->dssdev));
- if (!card->name)
- return -ENOMEM;
-
+ card->name = "HDMI";
card->owner = THIS_MODULE;
card->dai_link =
devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL);
--
cgit
From 6f2a43e3d14f6e31a3b041a1043195d02c54d615 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Wed, 12 Jun 2024 15:12:03 +0300
Subject: ASoC: SOF: sof-audio: Skip unprepare for in-use widgets on error
rollback
If the ipc_prepare() callback fails for a module instance, on error rewind
we must skip the ipc_unprepare() call for ones that has positive use count.
The positive use count means that the module instance is in active use, it
cannot be unprepared.
The issue affects capture direction paths with branches (single dai with
multiple PCMs), the affected widgets are in the shared part of the paths.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20240612121203.15468-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/sof-audio.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
index b3ac040811e7..ef9318947d74 100644
--- a/sound/soc/sof/sof-audio.c
+++ b/sound/soc/sof/sof-audio.c
@@ -485,7 +485,7 @@ sink_prepare:
if (ret < 0) {
/* unprepare the source widget */
if (widget_ops[widget->id].ipc_unprepare &&
- swidget && swidget->prepared) {
+ swidget && swidget->prepared && swidget->use_count == 0) {
widget_ops[widget->id].ipc_unprepare(swidget);
swidget->prepared = false;
}
--
cgit
From f3b198e4788fcc8d03ed0c8bd5e3856c6a5760c5 Mon Sep 17 00:00:00 2001
From: Jack Yu <jack.yu@realtek.com>
Date: Wed, 12 Jun 2024 09:01:07 +0000
Subject: ASoC: rt722-sdca-sdw: add debounce time for type detection
Add debounce time in headset type detection for better performance.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/7e502e9a9dd94122a1b60deb5ceb60fb@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt722-sdca-sdw.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c
index f73ee3bf90f5..87354bb1564e 100644
--- a/sound/soc/codecs/rt722-sdca-sdw.c
+++ b/sound/soc/codecs/rt722-sdca-sdw.c
@@ -352,7 +352,7 @@ static int rt722_sdca_interrupt_callback(struct sdw_slave *slave,
if (status->sdca_cascade && !rt722->disable_irq)
mod_delayed_work(system_power_efficient_wq,
- &rt722->jack_detect_work, msecs_to_jiffies(30));
+ &rt722->jack_detect_work, msecs_to_jiffies(280));
mutex_unlock(&rt722->disable_irq_lock);
--
cgit
From 0298f51652be47b79780833e0b63194e1231fa34 Mon Sep 17 00:00:00 2001
From: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Date: Thu, 13 Jun 2024 11:01:26 +0200
Subject: ASoC: topology: Fix route memory corruption
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
It was reported that recent fix for memory corruption during topology
load, causes corruption in other cases. Instead of being overeager with
checking topology, assume that it is properly formatted and just
duplicate strings.
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Closes: https://lore.kernel.org/linux-sound/171812236450.201359.3019210915105428447.b4-ty@kernel.org/T/#m8c4bd5abf453960fde6f826c4b7f84881da63e9d
Suggested-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240613090126.841189-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-topology.c | 12 +++---------
1 file changed, 3 insertions(+), 9 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 2ac442644ed4..6951ff7bc61e 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1052,21 +1052,15 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
break;
}
- route->source = devm_kmemdup(tplg->dev, elem->source,
- min(strlen(elem->source), maxlen),
- GFP_KERNEL);
- route->sink = devm_kmemdup(tplg->dev, elem->sink,
- min(strlen(elem->sink), maxlen),
- GFP_KERNEL);
+ route->source = devm_kstrdup(tplg->dev, elem->source, GFP_KERNEL);
+ route->sink = devm_kstrdup(tplg->dev, elem->sink, GFP_KERNEL);
if (!route->source || !route->sink) {
ret = -ENOMEM;
break;
}
if (strnlen(elem->control, maxlen) != 0) {
- route->control = devm_kmemdup(tplg->dev, elem->control,
- min(strlen(elem->control), maxlen),
- GFP_KERNEL);
+ route->control = devm_kstrdup(tplg->dev, elem->control, GFP_KERNEL);
if (!route->control) {
ret = -ENOMEM;
break;
--
cgit
From 8af49868e51ed1ba117b74728af12abe1eda82e5 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.cirrus.com>
Date: Thu, 13 Jun 2024 14:25:27 +0100
Subject: ASoC: cs35l56: Disconnect ASP1 TX sources when ASP1 DAI is hooked up
If the ASP1 DAI is hooked up by the machine driver the ASP TX mixer
sources should be initialized to disconnected. There aren't currently
any available products using the ASP so this doesn't affect any
existing systems.
The cs35l56 does not have any fixed default for the mixer source
registers. When the cs35l56 boots, its firmware patches these registers
to setup a system-specific routing; this is so that Windows can use
generic SDCA drivers instead of needing knowledge of chip-specific
registers. The setup varies between end-products, which each have
customized firmware, and so the default register state varies between
end-products. It can also change if the firmware on an end-product is
upgraded - for example if a change was needed to the routing for Windows
use-cases. It must be emphasized that the settings applied by the
firmware are not internal magic tuning; they are statically implementing
use-case setup that on Linux would be done via ALSA controls.
The driver is currently syncing the mixer controls with whatever
initial state the firmware wrote to the registers, so that they report
the actual audio routing. But if the ASP DAI is hooked up this can create
a powered-up DAPM graph without anything intentionally setting up a path.
This can lead to parts of the audio system powering up unexpectedly.
For example when cs35l56 is connected to cs42l43 using a codec-codec link,
this can create a complete DAPM graph which then powers-up cs42l43. But
the cs42l43 can only be clocked from its SoundWire bus so this causes a
bunch of errors in the kernel log where cs42l43 is unexpectedly powered-up
without a clock.
If the host is taking ownership of the ASP (either directly or as a
codec-to-codec link) there is no need to keep the mixer settings that the
firmware wrote. The driver has ALSA controls for setting these using
standard Linux mechanisms. So if the machine driver hooks up the ASP the
ASP mixers are initialized to "None" (no input). This prevents unintended
DAPM-graph power-ups, and means the initial state of the mixers is
always going to be None.
Since the initial state of the mixers can vary from system to system and
potentially between firmware upgrades, no use-case manager can currently
assume that cs35l56 has a known initial state. The firmware could just as
easily default them to "None" as to any input source. So defaulting them
to "None" in the driver is not increasing the entropy of the system.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240613132527.46537-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/cs35l56-shared.c | 4 ++++
1 file changed, 4 insertions(+)
(limited to 'sound')
diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c
index 8af89a263594..30497152e02a 100644
--- a/sound/soc/codecs/cs35l56-shared.c
+++ b/sound/soc/codecs/cs35l56-shared.c
@@ -215,6 +215,10 @@ static const struct reg_sequence cs35l56_asp1_defaults[] = {
REG_SEQ0(CS35L56_ASP1_FRAME_CONTROL5, 0x00020100),
REG_SEQ0(CS35L56_ASP1_DATA_CONTROL1, 0x00000018),
REG_SEQ0(CS35L56_ASP1_DATA_CONTROL5, 0x00000018),
+ REG_SEQ0(CS35L56_ASP1TX1_INPUT, 0x00000000),
+ REG_SEQ0(CS35L56_ASP1TX2_INPUT, 0x00000000),
+ REG_SEQ0(CS35L56_ASP1TX3_INPUT, 0x00000000),
+ REG_SEQ0(CS35L56_ASP1TX4_INPUT, 0x00000000),
};
/*
--
cgit
From be1fae62cf253a5b67526cee9fbc07689b97c125 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Thu, 13 Jun 2024 13:13:05 +0100
Subject: ASoC: q6apm-lpass-dai: close graph on prepare errors
There is an issue around with error handling and graph management with
the exising code, none of the error paths close the graph, which result in
leaving the loaded graph in dsp, however the driver thinks otherwise.
This can have a nasty side effect specially when we try to load the same
graph to dsp, dsp returns error which leaves the board with no sound and
requires restart.
Fix this by properly closing the graph when we hit errors between
open and close.
Fixes: 30ad723b93ad ("ASoC: qdsp6: audioreach: add q6apm lpass dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org>
Tested-by: Dmitry Baryshkov <dmitry.baryshkov@linaro.org> # X13s
Link: https://lore.kernel.org/r/20240613-q6apm-fixes-v1-1-d88953675ab3@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 32 ++++++++++++++++++++------------
1 file changed, 20 insertions(+), 12 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
index 68a38f63a2db..66b911b49e3f 100644
--- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
+++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
@@ -141,14 +141,17 @@ static void q6apm_lpass_dai_shutdown(struct snd_pcm_substream *substream, struct
struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev);
int rc;
- if (!dai_data->is_port_started[dai->id])
- return;
- rc = q6apm_graph_stop(dai_data->graph[dai->id]);
- if (rc < 0)
- dev_err(dai->dev, "fail to close APM port (%d)\n", rc);
+ if (dai_data->is_port_started[dai->id]) {
+ rc = q6apm_graph_stop(dai_data->graph[dai->id]);
+ dai_data->is_port_started[dai->id] = false;
+ if (rc < 0)
+ dev_err(dai->dev, "fail to close APM port (%d)\n", rc);
+ }
- q6apm_graph_close(dai_data->graph[dai->id]);
- dai_data->is_port_started[dai->id] = false;
+ if (dai_data->graph[dai->id]) {
+ q6apm_graph_close(dai_data->graph[dai->id]);
+ dai_data->graph[dai->id] = NULL;
+ }
}
static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
@@ -163,8 +166,10 @@ static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct s
q6apm_graph_stop(dai_data->graph[dai->id]);
dai_data->is_port_started[dai->id] = false;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
q6apm_graph_close(dai_data->graph[dai->id]);
+ dai_data->graph[dai->id] = NULL;
+ }
}
/**
@@ -183,26 +188,29 @@ static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct s
cfg->direction = substream->stream;
rc = q6apm_graph_media_format_pcm(dai_data->graph[dai->id], cfg);
-
if (rc) {
dev_err(dai->dev, "Failed to set media format %d\n", rc);
- return rc;
+ goto err;
}
rc = q6apm_graph_prepare(dai_data->graph[dai->id]);
if (rc) {
dev_err(dai->dev, "Failed to prepare Graph %d\n", rc);
- return rc;
+ goto err;
}
rc = q6apm_graph_start(dai_data->graph[dai->id]);
if (rc < 0) {
dev_err(dai->dev, "fail to start APM port %x\n", dai->id);
- return rc;
+ goto err;
}
dai_data->is_port_started[dai->id] = true;
return 0;
+err:
+ q6apm_graph_close(dai_data->graph[dai->id]);
+ dai_data->graph[dai->id] = NULL;
+ return rc;
}
static int q6apm_lpass_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
--
cgit
From 60ff540a1d476c2d48b96f7bc8ac8581b820e878 Mon Sep 17 00:00:00 2001
From: Shuming Fan <shumingf@realtek.com>
Date: Wed, 12 Jun 2024 15:57:40 +0800
Subject: ASoC: Intel: soc-acpi: mtl: fix speaker no sound on Dell SKU 0C64
Dell SKU 0C64 has a single rt1318 amplifier.
The prefix name of control still needs to be set rt1318-1 corresponding to UCM config.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240612075740.1678082-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index 48252fa9e39e..8e0ae3635a35 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
@@ -293,7 +293,7 @@ static const struct snd_soc_acpi_adr_device rt1318_1_single_adr[] = {
.adr = 0x000130025D131801,
.num_endpoints = 1,
.endpoints = &single_endpoint,
- .name_prefix = "rt1318"
+ .name_prefix = "rt1318-1"
}
};
--
cgit
From 98d919dfee1cc402ca29d45da642852d7c9a2301 Mon Sep 17 00:00:00 2001
From: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Date: Mon, 17 Jun 2024 12:58:34 +0530
Subject: ASoC: amd: acp: add a null check for chip_pdev structure
When acp platform device creation is skipped, chip->chip_pdev value will
remain NULL. Add NULL check for chip->chip_pdev structure in
snd_acp_resume() function to avoid null pointer dereference.
Fixes: 088a40980efb ("ASoC: amd: acp: add pm ops support for acp pci driver")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240617072844.871468-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/amd/acp/acp-pci.c | 10 ++++++----
1 file changed, 6 insertions(+), 4 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c
index ad320b29e87d..aa3e72d13451 100644
--- a/sound/soc/amd/acp/acp-pci.c
+++ b/sound/soc/amd/acp/acp-pci.c
@@ -199,10 +199,12 @@ static int __maybe_unused snd_acp_resume(struct device *dev)
ret = acp_init(chip);
if (ret)
dev_err(dev, "ACP init failed\n");
- child = chip->chip_pdev->dev;
- adata = dev_get_drvdata(&child);
- if (adata)
- acp_enable_interrupts(adata);
+ if (chip->chip_pdev) {
+ child = chip->chip_pdev->dev;
+ adata = dev_get_drvdata(&child);
+ if (adata)
+ acp_enable_interrupts(adata);
+ }
return ret;
}
--
cgit
From 70fa3900c3ed92158628710e81d274e5cb52f92b Mon Sep 17 00:00:00 2001
From: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Date: Mon, 17 Jun 2024 12:58:35 +0530
Subject: ASoC: amd: acp: remove i2s configuration check in acp_i2s_probe()
ACP supports different pin configurations for I2S IO. Checking ACP pin
configuration value against specific value breaks the functionality for
other I2S pin configurations. This check is no longer required in i2s dai
driver probe call as i2s configuration check will be verified during acp
platform device creation sequence.
Remove i2s_mode check in acp_i2s_probe() function.
Fixes: b24484c18b10 ("ASoC: amd: acp: ACP code generic to support newer platforms")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240617072844.871468-2-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/amd/acp/acp-i2s.c | 8 --------
1 file changed, 8 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c
index 60cbc881be6e..ef12f97ddc69 100644
--- a/sound/soc/amd/acp/acp-i2s.c
+++ b/sound/soc/amd/acp/acp-i2s.c
@@ -588,20 +588,12 @@ static int acp_i2s_probe(struct snd_soc_dai *dai)
{
struct device *dev = dai->component->dev;
struct acp_dev_data *adata = dev_get_drvdata(dev);
- struct acp_resource *rsrc = adata->rsrc;
- unsigned int val;
if (!adata->acp_base) {
dev_err(dev, "I2S base is NULL\n");
return -EINVAL;
}
- val = readl(adata->acp_base + rsrc->i2s_pin_cfg_offset);
- if (val != rsrc->i2s_mode) {
- dev_err(dev, "I2S Mode not supported val %x\n", val);
- return -EINVAL;
- }
-
return 0;
}
--
cgit
From 379bcd2c9197bf2c429434e8a01cea0ee1852316 Mon Sep 17 00:00:00 2001
From: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Date: Mon, 17 Jun 2024 12:58:36 +0530
Subject: ASoC: amd: acp: move chip->flag variable assignment
chip->flag variable assignment will be skipped when acp platform device
creation is skipped. In this case chip>flag value will not be set.
chip->flag variable should be assigned along with other structure
variables for 'chip' structure. Move chip->flag variable assignment
prior to acp platform device creation.
Fixes: 3a94c8ad0aae ("ASoC: amd: acp: add code for scanning acp pdm controller")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240617072844.871468-3-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/amd/acp/acp-pci.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c
index aa3e72d13451..777b5a78d8a9 100644
--- a/sound/soc/amd/acp/acp-pci.c
+++ b/sound/soc/amd/acp/acp-pci.c
@@ -100,6 +100,7 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
ret = -EINVAL;
goto release_regions;
}
+ chip->flag = flag;
dmic_dev = platform_device_register_data(dev, "dmic-codec", PLATFORM_DEVID_NONE, NULL, 0);
if (IS_ERR(dmic_dev)) {
dev_err(dev, "failed to create DMIC device\n");
@@ -139,7 +140,6 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
}
}
- chip->flag = flag;
memset(&pdevinfo, 0, sizeof(pdevinfo));
pdevinfo.name = chip->name;
--
cgit
From 90f3feb24172185f1832636264943e8b5e289245 Mon Sep 17 00:00:00 2001
From: Elinor Montmasson <elinor.montmasson@savoirfairelinux.com>
Date: Thu, 20 Jun 2024 15:25:03 +0200
Subject: ASoC: fsl-asoc-card: set priv->pdev before using it
priv->pdev pointer was set after being used in
fsl_asoc_card_audmux_init().
Move this assignment at the start of the probe function, so
sub-functions can correctly use pdev through priv.
fsl_asoc_card_audmux_init() dereferences priv->pdev to get access to the
dev struct, used with dev_err macros.
As priv is zero-initialised, there would be a NULL pointer dereference.
Note that if priv->dev is dereferenced before assignment but never used,
for example if there is no error to be printed, the driver won't crash
probably due to compiler optimisations.
Fixes: 708b4351f08c ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support")
Signed-off-by: Elinor Montmasson <elinor.montmasson@savoirfairelinux.com>
Link: https://patch.msgid.link/20240620132511.4291-2-elinor.montmasson@savoirfairelinux.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/fsl-asoc-card.c | 3 ++-
1 file changed, 2 insertions(+), 1 deletion(-)
(limited to 'sound')
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 5ddc0c2fe53f..eb67689dcd6e 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -559,6 +559,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
if (!priv)
return -ENOMEM;
+ priv->pdev = pdev;
+
cpu_np = of_parse_phandle(np, "audio-cpu", 0);
/* Give a chance to old DT binding */
if (!cpu_np)
@@ -787,7 +789,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
}
/* Initialize sound card */
- priv->pdev = pdev;
priv->card.dev = &pdev->dev;
priv->card.owner = THIS_MODULE;
ret = snd_soc_of_parse_card_name(&priv->card, "model");
--
cgit
From 282a4482e198e03781c152c88aac8aa382ef9a55 Mon Sep 17 00:00:00 2001
From: Chen-Yu Tsai <wenst@chromium.org>
Date: Mon, 24 Jun 2024 14:12:56 +0800
Subject: ASoC: mediatek: mt8195: Add platform entry for ETDM1_OUT_BE dai link
Commit e70b8dd26711 ("ASoC: mediatek: mt8195: Remove afe-dai component
and rework codec link") removed the codec entry for the ETDM1_OUT_BE
dai link entirely instead of replacing it with COMP_EMPTY(). This worked
by accident as the remaining COMP_EMPTY() platform entry became the codec
entry, and the platform entry became completely empty, effectively the
same as COMP_DUMMY() since snd_soc_fill_dummy_dai() doesn't do anything
for platform entries.
This causes a KASAN out-of-bounds warning in mtk_soundcard_common_probe()
in sound/soc/mediatek/common/mtk-soundcard-driver.c:
for_each_card_prelinks(card, i, dai_link) {
if (adsp_node && !strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")))
dai_link->platforms->of_node = adsp_node;
else if (!dai_link->platforms->name && !dai_link->platforms->of_node)
dai_link->platforms->of_node = platform_node;
}
where the code expects the platforms array to have space for at least one entry.
Add an COMP_EMPTY() entry so that dai_link->platforms has space.
Fixes: e70b8dd26711 ("ASoC: mediatek: mt8195: Remove afe-dai component and rework codec link")
Signed-off-by: Chen-Yu Tsai <wenst@chromium.org>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://patch.msgid.link/20240624061257.3115467-1-wenst@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/mt8195/mt8195-mt6359.c | 1 +
1 file changed, 1 insertion(+)
(limited to 'sound')
diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c
index ca8751190520..2832ef78eaed 100644
--- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c
+++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c
@@ -827,6 +827,7 @@ SND_SOC_DAILINK_DEFS(ETDM2_IN_BE,
SND_SOC_DAILINK_DEFS(ETDM1_OUT_BE,
DAILINK_COMP_ARRAY(COMP_CPU("ETDM1_OUT")),
+ DAILINK_COMP_ARRAY(COMP_EMPTY()),
DAILINK_COMP_ARRAY(COMP_EMPTY()));
SND_SOC_DAILINK_DEFS(ETDM2_OUT_BE,
--
cgit
From 63b47f026cc841bd3d3438dd6fccbc394dfead87 Mon Sep 17 00:00:00 2001
From: Vyacheslav Frantsishko <itmymaill@gmail.com>
Date: Wed, 26 Jun 2024 10:03:34 +0300
Subject: ASoC: amd: yc: Fix non-functional mic on ASUS M5602RA
The Vivobook S 16X IPS needs a quirks-table entry for the internal microphone to function properly.
Signed-off-by: Vyacheslav Frantsishko <itmymaill@gmail.com>
Reviewed-by: Mario Limonciello <mario.limonciello@amd.com>
Link: https://patch.msgid.link/20240626070334.45633-1-itmymaill@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/amd/yc/acp6x-mach.c | 7 +++++++
1 file changed, 7 insertions(+)
(limited to 'sound')
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index 1760b5d42460..4e3a8ce690a4 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -283,6 +283,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
DMI_MATCH(DMI_PRODUCT_NAME, "M5402RA"),
}
},
+ {
+ .driver_data = &acp6x_card,
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
+ DMI_MATCH(DMI_PRODUCT_NAME, "M5602RA"),
+ }
+ },
{
.driver_data = &acp6x_card,
.matches = {
--
cgit
From 68f97fe330e01450ace63da0ce5cab676fc97f9a Mon Sep 17 00:00:00 2001
From: Jack Yu <jack.yu@realtek.com>
Date: Wed, 26 Jun 2024 08:25:34 +0000
Subject: ASoC: rt5645: fix issue of random interrupt from push-button
Modify register setting sequence of enabling inline command
to fix issue of random interrupt from push-button.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://patch.msgid.link/9a7a3a66cbcb426487ca6f558f45e922@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt5645.c | 24 ++++++++++++++++++------
sound/soc/codecs/rt5645.h | 6 ++++++
2 files changed, 24 insertions(+), 6 deletions(-)
(limited to 'sound')
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index cdb7ff7020e9..51187b1e0ed2 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -81,7 +81,7 @@ static const struct reg_sequence init_list[] = {
static const struct reg_sequence rt5650_init_list[] = {
{0xf6, 0x0100},
{RT5645_PWR_ANLG1, 0x02},
- {RT5645_IL_CMD3, 0x0018},
+ {RT5645_IL_CMD3, 0x6728},
};
static const struct reg_default rt5645_reg[] = {
@@ -3130,20 +3130,32 @@ static void rt5645_enable_push_button_irq(struct snd_soc_component *component,
bool enable)
{
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+ int ret;
if (enable) {
snd_soc_dapm_force_enable_pin(dapm, "ADC L power");
snd_soc_dapm_force_enable_pin(dapm, "ADC R power");
snd_soc_dapm_sync(dapm);
+ snd_soc_component_update_bits(component, RT5650_4BTN_IL_CMD2,
+ RT5645_EN_4BTN_IL_MASK | RT5645_RST_4BTN_IL_MASK,
+ RT5645_EN_4BTN_IL_EN | RT5645_RST_4BTN_IL_RST);
+ usleep_range(10000, 15000);
+ snd_soc_component_update_bits(component, RT5650_4BTN_IL_CMD2,
+ RT5645_EN_4BTN_IL_MASK | RT5645_RST_4BTN_IL_MASK,
+ RT5645_EN_4BTN_IL_EN | RT5645_RST_4BTN_IL_NORM);
+ msleep(50);
+ ret = snd_soc_component_read(component, RT5645_INT_IRQ_ST);
+ pr_debug("%s read %x = %x\n", __func__, RT5645_INT_IRQ_ST,
+ snd_soc_component_read(component, RT5645_INT_IRQ_ST));
+ snd_soc_component_write(component, RT5645_INT_IRQ_ST, ret);
+ ret = snd_soc_component_read(component, RT5650_4BTN_IL_CMD1);
+ pr_debug("%s read %x = %x\n", __func__, RT5650_4BTN_IL_CMD1,
+ snd_soc_component_read(component, RT5650_4BTN_IL_CMD1));
+ snd_soc_component_write(component, RT5650_4BTN_IL_CMD1, ret);
snd_soc_component_update_bits(component, RT5650_4BTN_IL_CMD1, 0x3, 0x3);
snd_soc_component_update_bits(component,
RT5645_INT_IRQ_ST, 0x8, 0x8);
- snd_soc_component_update_bits(component,
- RT5650_4BTN_IL_CMD2, 0x8000, 0x8000);
- snd_soc_component_read(component, RT5650_4BTN_IL_CMD1);
- pr_debug("%s read %x = %x\n", __func__, RT5650_4BTN_IL_CMD1,
- snd_soc_component_read(component, RT5650_4BTN_IL_CMD1));
} else {
snd_soc_component_update_bits(component, RT5650_4BTN_IL_CMD2, 0x8000, 0x0);
snd_soc_component_update_bits(component, RT5645_INT_IRQ_ST, 0x8, 0x0);
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index 90816b2c5489..bef74b29fd54 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -2011,6 +2011,12 @@
#define RT5645_ZCD_HP_DIS (0x0 << 15)
#define RT5645_ZCD_HP_EN (0x1 << 15)
+/* Buttons Inline Command Function 2 (0xe0) */
+#define RT5645_EN_4BTN_IL_MASK (0x1 << 15)
+#define RT5645_EN_4BTN_IL_EN (0x1 << 15)
+#define RT5645_RST_4BTN_IL_MASK (0x1 << 14)
+#define RT5645_RST_4BTN_IL_RST (0x0 << 14)
+#define RT5645_RST_4BTN_IL_NORM (0x1 << 14)
/* Codec Private Register definition */
/* DAC ADC Digital Volume (0x00) */
--
cgit