From 4008b29eb47433c15ed3f242ee0132ba27dbdb67 Mon Sep 17 00:00:00 2001
From: Steve Lee <steves.lee@maximintegrated.com>
Date: Thu, 11 Jun 2020 18:48:00 +0900
Subject: ASoC: max98390: Update regmap readable reg and volatile

 Update max98390_readable_register and max98390_volatile_reg

Signed-off-by: Steve Lee <steves.lee@maximintegrated.com>
Link: https://lore.kernel.org/r/20200611094800.18422-1-steves.lee@maximintegrated.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98390.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c
index 0d63ebfbff2f..e6613b52bd78 100644
--- a/sound/soc/codecs/max98390.c
+++ b/sound/soc/codecs/max98390.c
@@ -700,8 +700,8 @@ static bool max98390_readable_register(struct device *dev, unsigned int reg)
 	case MAX98390_IRQ_CTRL ... MAX98390_WDOG_CTRL:
 	case MAX98390_MEAS_ADC_THERM_WARN_THRESH
 		... MAX98390_BROWNOUT_INFINITE_HOLD:
-	case MAX98390_BROWNOUT_LVL_HOLD ... THERMAL_COILTEMP_RD_BACK_BYTE0:
-	case DSMIG_DEBUZZER_THRESHOLD ... MAX98390_R24FF_REV_ID:
+	case MAX98390_BROWNOUT_LVL_HOLD ... DSMIG_DEBUZZER_THRESHOLD:
+	case DSM_VOL_ENA ... MAX98390_R24FF_REV_ID:
 		return true;
 	default:
 		return false;
@@ -717,7 +717,7 @@ static bool max98390_volatile_reg(struct device *dev, unsigned int reg)
 	case MAX98390_BROWNOUT_LOWEST_STATUS:
 	case MAX98390_ENV_TRACK_BOOST_VOUT_READ:
 	case DSM_STBASS_HPF_B0_BYTE0 ... DSM_DEBUZZER_ATTACK_TIME_BYTE2:
-	case THERMAL_RDC_RD_BACK_BYTE1 ... THERMAL_COILTEMP_RD_BACK_BYTE0:
+	case THERMAL_RDC_RD_BACK_BYTE1 ... DSMIG_DEBUZZER_THRESHOLD:
 	case DSM_THERMAL_GAIN ... DSM_WBDRC_GAIN:
 		return true;
 	default:
-- 
cgit v1.2.3


From 6476b60f32866be49d05e2e0163f337374c55b06 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Thu, 11 Jun 2020 13:41:53 +0100
Subject: ASoC: q6asm: handle EOS correctly

Successful send of EOS command does not indicate that EOS is actually
finished, correct event to wait EOS is finished is EOS_RENDERED event.
EOS_RENDERED means that the DSP has finished processing all the buffers
for that particular session and stream.

This patch fixes EOS handling!

Fixes: 68fd8480bb7b ("ASoC: qdsp6: q6asm: Add support to audio stream apis")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200611124159.20742-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm.c | 7 ++++---
 1 file changed, 4 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 0e0e8f7a460a..ae4b2cabdf2d 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -25,6 +25,7 @@
 #define ASM_STREAM_CMD_FLUSH			0x00010BCE
 #define ASM_SESSION_CMD_PAUSE			0x00010BD3
 #define ASM_DATA_CMD_EOS			0x00010BDB
+#define ASM_DATA_EVENT_RENDERED_EOS		0x00010C1C
 #define ASM_NULL_POPP_TOPOLOGY			0x00010C68
 #define ASM_STREAM_CMD_FLUSH_READBUFS		0x00010C09
 #define ASM_STREAM_CMD_SET_ENCDEC_PARAM		0x00010C10
@@ -622,9 +623,6 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 		case ASM_SESSION_CMD_SUSPEND:
 			client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE;
 			break;
-		case ASM_DATA_CMD_EOS:
-			client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
-			break;
 		case ASM_STREAM_CMD_FLUSH:
 			client_event = ASM_CLIENT_EVENT_CMD_FLUSH_DONE;
 			break;
@@ -727,6 +725,9 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
 			spin_unlock_irqrestore(&ac->lock, flags);
 		}
 
+		break;
+	case ASM_DATA_EVENT_RENDERED_EOS:
+		client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
 		break;
 	}
 
-- 
cgit v1.2.3


From adb36a8203831e40494a92095dacd566b2ad4a69 Mon Sep 17 00:00:00 2001
From: Aaron Plattner <aplattner@nvidia.com>
Date: Thu, 11 Jun 2020 11:08:45 -0700
Subject: ALSA: hda: Add NVIDIA codec IDs 9a & 9d through a0 to patch table

These IDs are for upcoming NVIDIA chips with audio functions that are largely
similar to the existing ones.

Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200611180845.39942-1-aplattner@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index fbd7cc6026d8..e2b21ef5d7d1 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -4145,6 +4145,11 @@ HDA_CODEC_ENTRY(0x10de0095, "GPU 95 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0097, "GPU 97 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0098, "GPU 98 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0099, "GPU 99 HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009a, "GPU 9a HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009d, "GPU 9d HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009e, "GPU 9e HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP",	patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI",	patch_nvhdmi_2ch),
 HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI",	patch_nvhdmi_2ch),
 HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP",	patch_via_hdmi),
-- 
cgit v1.2.3


From e7585db1b0b5b4e4eb1967bb1472df308f7ffcbf Mon Sep 17 00:00:00 2001
From: Laurence Tratt <laurie@tratt.net>
Date: Fri, 12 Jun 2020 12:18:07 +0100
Subject: ALSA: usb-audio: Add implicit feedback quirk for SSL2+.

This uses the same quirk as the Motu M2 and M4 to ensure the driver uses the
audio interface's clock. Tested on an SSL2+.

Signed-off-by: Laurence Tratt <laurie@tratt.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200612111807.dgnig6rwhmsl2bod@overdrive.tratt.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/pcm.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 8a05dcb1344f..84c0ae431936 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -367,6 +367,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 		ifnum = 0;
 		goto add_sync_ep_from_ifnum;
 	case USB_ID(0x07fd, 0x0008): /* MOTU M Series */
+	case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */
 		ep = 0x81;
 		ifnum = 2;
 		goto add_sync_ep_from_ifnum;
-- 
cgit v1.2.3


From 6fbea6b6a838f9aa941fe53a3637fd8d8aab1eba Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Fri, 12 Jun 2020 15:37:48 +0800
Subject: ASoC: soc-card: export snd_soc_lookup_component_nolocked

snd_soc_lookup_component_nolocked can be used for the DPCM case
that Front-End needs to get the unused platform component but
added by Back-End cpu dai driver.

If the component is gotten, then we can get the dma chan created
by Back-End component and reused it in Front-End.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/55f6e0d76f67a517b9a44136d790ff2a06b5caa8.1591947428.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  | 2 ++
 sound/soc/soc-core.c | 3 ++-
 2 files changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 74868436ac79..565612a8d690 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -444,6 +444,8 @@ int devm_snd_soc_register_component(struct device *dev,
 			 const struct snd_soc_component_driver *component_driver,
 			 struct snd_soc_dai_driver *dai_drv, int num_dai);
 void snd_soc_unregister_component(struct device *dev);
+struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev,
+							    const char *driver_name);
 struct snd_soc_component *snd_soc_lookup_component(struct device *dev,
 						   const char *driver_name);
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 7b387202c5db..0f30f5aabaa8 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -310,7 +310,7 @@ struct snd_soc_component *snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd,
 }
 EXPORT_SYMBOL_GPL(snd_soc_rtdcom_lookup);
 
-static struct snd_soc_component
+struct snd_soc_component
 *snd_soc_lookup_component_nolocked(struct device *dev, const char *driver_name)
 {
 	struct snd_soc_component *component;
@@ -329,6 +329,7 @@ static struct snd_soc_component
 
 	return found_component;
 }
+EXPORT_SYMBOL_GPL(snd_soc_lookup_component_nolocked);
 
 struct snd_soc_component *snd_soc_lookup_component(struct device *dev,
 						   const char *driver_name)
-- 
cgit v1.2.3


From a9a21e1eafc94b79502cab8272b392f7f63ef7bb Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Fri, 12 Jun 2020 15:37:49 +0800
Subject: ASoC: dmaengine_pcm: export soc_component_to_pcm

In DPCM case, Front-End needs to get the dma chan which has
been requested by Back-End and reuse it.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/429c6ae1f3c5b47eb893f475d531d71cdcfe34c0.1591947428.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/dmaengine_pcm.h         | 11 +++++++++++
 sound/soc/soc-generic-dmaengine-pcm.c | 12 ------------
 2 files changed, 11 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index b65220685920..8c5e38180fb0 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -161,4 +161,15 @@ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
 
 #define SND_DMAENGINE_PCM_DRV_NAME "snd_dmaengine_pcm"
 
+struct dmaengine_pcm {
+	struct dma_chan *chan[SNDRV_PCM_STREAM_LAST + 1];
+	const struct snd_dmaengine_pcm_config *config;
+	struct snd_soc_component component;
+	unsigned int flags;
+};
+
+static inline struct dmaengine_pcm *soc_component_to_pcm(struct snd_soc_component *p)
+{
+	return container_of(p, struct dmaengine_pcm, component);
+}
 #endif
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index f728309a0833..80a4e71f2d95 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -21,18 +21,6 @@
  */
 #define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(31)
 
-struct dmaengine_pcm {
-	struct dma_chan *chan[SNDRV_PCM_STREAM_LAST + 1];
-	const struct snd_dmaengine_pcm_config *config;
-	struct snd_soc_component component;
-	unsigned int flags;
-};
-
-static struct dmaengine_pcm *soc_component_to_pcm(struct snd_soc_component *p)
-{
-	return container_of(p, struct dmaengine_pcm, component);
-}
-
 static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm,
 	struct snd_pcm_substream *substream)
 {
-- 
cgit v1.2.3


From 706e2c8811585f42612b6cff218ab3adbe63a4ee Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Fri, 12 Jun 2020 15:37:50 +0800
Subject: ASoC: fsl_asrc_dma: Reuse the dma channel if available in Back-End

The dma channel has been requested by Back-End cpu dai driver already.
If fsl_asrc_dma requests dma chan with same dma:tx symlink, then
there will be below warning with SDMA.

[   48.174236] fsl-esai-dai 2024000.esai: Cannot create DMA dma:tx symlink

So if we can reuse the dma channel of Back-End, then the issue can be
fixed.

In order to get the dma channel which is already requested in Back-End.
we use the exported two functions (snd_soc_lookup_component_nolocked
and soc_component_to_pcm). If we can get the dma channel, then reuse it,
if can't, then request a new one.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/3a79f0442cb4930c633cf72145cfe95a45b9c78e.1591947428.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_asrc_dma.c | 25 ++++++++++++++++++++-----
 1 file changed, 20 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index d6a3fc5f87e5..d88e6343e0a2 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -135,6 +135,8 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
 	struct snd_dmaengine_dai_dma_data *dma_params_be = NULL;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct fsl_asrc_pair *pair = runtime->private_data;
+	struct dma_chan *tmp_chan = NULL, *be_chan = NULL;
+	struct snd_soc_component *component_be = NULL;
 	struct fsl_asrc *asrc = pair->asrc;
 	struct dma_slave_config config_fe, config_be;
 	enum asrc_pair_index index = pair->index;
@@ -142,7 +144,6 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
 	int stream = substream->stream;
 	struct imx_dma_data *tmp_data;
 	struct snd_soc_dpcm *dpcm;
-	struct dma_chan *tmp_chan;
 	struct device *dev_be;
 	u8 dir = tx ? OUT : IN;
 	dma_cap_mask_t mask;
@@ -197,18 +198,30 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
 	dma_cap_set(DMA_SLAVE, mask);
 	dma_cap_set(DMA_CYCLIC, mask);
 
+	/*
+	 * The Back-End device might have already requested a DMA channel,
+	 * so try to reuse it first, and then request a new one upon NULL.
+	 */
+	component_be = snd_soc_lookup_component_nolocked(dev_be, SND_DMAENGINE_PCM_DRV_NAME);
+	if (component_be) {
+		be_chan = soc_component_to_pcm(component_be)->chan[substream->stream];
+		tmp_chan = be_chan;
+	}
+	if (!tmp_chan)
+		tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx");
+
 	/*
 	 * An EDMA DEV_TO_DEV channel is fixed and bound with DMA event of each
 	 * peripheral, unlike SDMA channel that is allocated dynamically. So no
-	 * need to configure dma_request and dma_request2, but get dma_chan via
-	 * dma_request_slave_channel directly with dma name of Front-End device
+	 * need to configure dma_request and dma_request2, but get dma_chan of
+	 * Back-End device directly via dma_request_slave_channel.
 	 */
 	if (!asrc->use_edma) {
 		/* Get DMA request of Back-End */
-		tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx");
 		tmp_data = tmp_chan->private;
 		pair->dma_data.dma_request = tmp_data->dma_request;
-		dma_release_channel(tmp_chan);
+		if (!be_chan)
+			dma_release_channel(tmp_chan);
 
 		/* Get DMA request of Front-End */
 		tmp_chan = asrc->get_dma_channel(pair, dir);
@@ -221,6 +234,8 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
 		pair->dma_chan[dir] =
 			dma_request_channel(mask, filter, &pair->dma_data);
 	} else {
+		if (!be_chan)
+			dma_release_channel(tmp_chan);
 		pair->dma_chan[dir] =
 			asrc->get_dma_channel(pair, dir);
 	}
-- 
cgit v1.2.3


From b287a6d9723c601dd947f1c27d4cc0192e384a5a Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Fri, 12 Jun 2020 15:37:51 +0800
Subject: ASoC: fsl_asrc_dma: Fix data copying speed issue with EDMA

With EDMA, there is two dma channels can be used for dev_to_dev,
one is from ASRC, one is from another peripheral (ESAI or SAI).

If we select the dma channel of ASRC, there is an issue for ideal
ratio case, the speed of copy data is faster than sample
frequency, because ASRC output data is very fast in ideal ratio
mode.

So it is reasonable to use the dma channel of Back-End peripheral.
then copying speed of DMA is controlled by data consumption
speed in the peripheral FIFO,

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/424ed6c249bafcbe30791c9de0352821c5ea67e2.1591947428.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_asrc_common.h |  2 ++
 sound/soc/fsl/fsl_asrc_dma.c    | 26 +++++++++++++++-----------
 2 files changed, 17 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_asrc_common.h b/sound/soc/fsl/fsl_asrc_common.h
index 77665b15c8db..7e1c13ca37f1 100644
--- a/sound/soc/fsl/fsl_asrc_common.h
+++ b/sound/soc/fsl/fsl_asrc_common.h
@@ -32,6 +32,7 @@ enum asrc_pair_index {
  * @dma_chan: inputer and output DMA channels
  * @dma_data: private dma data
  * @pos: hardware pointer position
+ * @req_dma_chan: flag to release dev_to_dev chan
  * @private: pair private area
  */
 struct fsl_asrc_pair {
@@ -45,6 +46,7 @@ struct fsl_asrc_pair {
 	struct dma_chan *dma_chan[2];
 	struct imx_dma_data dma_data;
 	unsigned int pos;
+	bool req_dma_chan;
 
 	void *private;
 };
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index d88e6343e0a2..5f01a58f422a 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -233,11 +233,11 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
 
 		pair->dma_chan[dir] =
 			dma_request_channel(mask, filter, &pair->dma_data);
+		pair->req_dma_chan = true;
 	} else {
-		if (!be_chan)
-			dma_release_channel(tmp_chan);
-		pair->dma_chan[dir] =
-			asrc->get_dma_channel(pair, dir);
+		pair->dma_chan[dir] = tmp_chan;
+		/* Do not flag to release if we are reusing the Back-End one */
+		pair->req_dma_chan = !be_chan;
 	}
 
 	if (!pair->dma_chan[dir]) {
@@ -276,7 +276,8 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
 	ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be);
 	if (ret) {
 		dev_err(dev, "failed to config DMA channel for Back-End\n");
-		dma_release_channel(pair->dma_chan[dir]);
+		if (pair->req_dma_chan)
+			dma_release_channel(pair->dma_chan[dir]);
 		return ret;
 	}
 
@@ -288,19 +289,22 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
 static int fsl_asrc_dma_hw_free(struct snd_soc_component *component,
 				struct snd_pcm_substream *substream)
 {
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct fsl_asrc_pair *pair = runtime->private_data;
+	u8 dir = tx ? OUT : IN;
 
 	snd_pcm_set_runtime_buffer(substream, NULL);
 
-	if (pair->dma_chan[IN])
-		dma_release_channel(pair->dma_chan[IN]);
+	if (pair->dma_chan[!dir])
+		dma_release_channel(pair->dma_chan[!dir]);
 
-	if (pair->dma_chan[OUT])
-		dma_release_channel(pair->dma_chan[OUT]);
+	/* release dev_to_dev chan if we aren't reusing the Back-End one */
+	if (pair->dma_chan[dir] && pair->req_dma_chan)
+		dma_release_channel(pair->dma_chan[dir]);
 
-	pair->dma_chan[IN] = NULL;
-	pair->dma_chan[OUT] = NULL;
+	pair->dma_chan[!dir] = NULL;
+	pair->dma_chan[dir] = NULL;
 
 	return 0;
 }
-- 
cgit v1.2.3


From c9808bbfed3cfc911ecb60fe8e80c0c27876c657 Mon Sep 17 00:00:00 2001
From: "Yick W. Tse" <y_w_tse@yahoo.com.hk>
Date: Sat, 13 Jun 2020 11:40:06 +0000
Subject: ALSA: usb-audio: add quirk for Denon DCD-1500RE

fix error "clock source 41 is not valid, cannot use"

[] New USB device found, idVendor=154e, idProduct=1002, bcdDevice= 1.00
[] New USB device strings: Mfr=1, Product=2, SerialNumber=0
[] Product: DCD-1500RE
[] Manufacturer: D & M Holdings Inc.
[]
[] clock source 41 is not valid, cannot use
[] usbcore: registered new interface driver snd-usb-audio

Signed-off-by: Yick W. Tse <y_w_tse@yahoo.com.hk>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1373857985.210365.1592048406997@mail.yahoo.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index bca0179a0ef8..c495e720e2f1 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1532,6 +1532,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
 static bool is_itf_usb_dsd_dac(unsigned int id)
 {
 	switch (id) {
+	case USB_ID(0x154e, 0x1002): /* Denon DCD-1500RE */
 	case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */
 	case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */
 	case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */
-- 
cgit v1.2.3


From 8abf41dcd1bcdda0d09905fb59d18f45c035c752 Mon Sep 17 00:00:00 2001
From: Christopher Swenson <swenson@swenson.io>
Date: Sun, 14 Jun 2020 17:11:47 -0700
Subject: ALSA: usb-audio: Set 48 kHz rate for Rodecaster

Like the Line6 devices, the Rode Rodecaster Pro does not support
UAC2_CS_RANGE and only supports a sample rate of 48 kHz.

Tested against a Rode Rodecaster Pro.

Tested-by: Christopher Swenson <swenson@swenson.io>
Signed-off-by: Christopher Swenson <swenson@swenson.io>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/ebdb9e72-9649-0b5e-b9b9-d757dbf26927@swenson.io
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/format.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/format.c b/sound/usb/format.c
index 5ffb457cc88c..1b28d01d1f4c 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -394,8 +394,9 @@ skip_rate:
 	return nr_rates;
 }
 
-/* Line6 Helix series don't support the UAC2_CS_RANGE usb function
- * call. Return a static table of known clock rates.
+/* Line6 Helix series and the Rode Rodecaster Pro don't support the
+ * UAC2_CS_RANGE usb function call. Return a static table of known
+ * clock rates.
  */
 static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip,
 						struct audioformat *fp)
@@ -408,6 +409,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip,
 	case USB_ID(0x0e41, 0x4248): /* Line6 Helix >= fw 2.82 */
 	case USB_ID(0x0e41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */
 	case USB_ID(0x0e41, 0x424a): /* Line6 Helix LT >= fw 2.82 */
+	case USB_ID(0x19f7, 0x0011): /* Rode Rodecaster Pro */
 		return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000);
 	}
 
-- 
cgit v1.2.3


From 0fae253af563cf5d1f5dc651d520c3eafd74f183 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 12 Jun 2020 15:59:37 -0500
Subject: ASoC: soc-devres: add devm_snd_soc_register_dai()

The registration of DAIs may be done at two distinct times, once
during a component registration and later when loading a
topology. Since devm_ managed resources are freed in the reverse order
they were allocated, when a component starts unregistering DAIs by
walking through the DAI list, the memory allocated for the
topology-registered DAIs was freed already, which leads to 100%
reproducible KASAN use-after-free reports.

This patch suggests a new devm_ function to force the DAI list to be
updated prior to freeing the memory chunks referenced by the list
pointers.

Suggested-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2186
Link: https://lore.kernel.org/r/20200612205938.26415-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h    |  4 ++++
 sound/soc/soc-devres.c | 37 +++++++++++++++++++++++++++++++++++++
 2 files changed, 41 insertions(+)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 565612a8d690..fddab504c227 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1363,6 +1363,10 @@ void snd_soc_remove_pcm_runtime(struct snd_soc_card *card,
 struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component,
 					 struct snd_soc_dai_driver *dai_drv,
 					 bool legacy_dai_naming);
+struct snd_soc_dai *devm_snd_soc_register_dai(struct device *dev,
+					      struct snd_soc_component *component,
+					      struct snd_soc_dai_driver *dai_drv,
+					      bool legacy_dai_naming);
 void snd_soc_unregister_dai(struct snd_soc_dai *dai);
 
 struct snd_soc_dai *snd_soc_find_dai(
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index a9ea172a66a7..11e5d7962370 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -9,6 +9,43 @@
 #include <sound/soc.h>
 #include <sound/dmaengine_pcm.h>
 
+static void devm_dai_release(struct device *dev, void *res)
+{
+	snd_soc_unregister_dai(*(struct snd_soc_dai **)res);
+}
+
+/**
+ * devm_snd_soc_register_dai - resource-managed dai registration
+ * @dev: Device used to manage component
+ * @component: The component the DAIs are registered for
+ * @dai_drv: DAI driver to use for the DAI
+ * @legacy_dai_naming: if %true, use legacy single-name format;
+ *	if %false, use multiple-name format;
+ */
+struct snd_soc_dai *devm_snd_soc_register_dai(struct device *dev,
+					      struct snd_soc_component *component,
+					      struct snd_soc_dai_driver *dai_drv,
+					      bool legacy_dai_naming)
+{
+	struct snd_soc_dai **ptr;
+	struct snd_soc_dai *dai;
+
+	ptr = devres_alloc(devm_dai_release, sizeof(*ptr), GFP_KERNEL);
+	if (!ptr)
+		return NULL;
+
+	dai = snd_soc_register_dai(component, dai_drv, legacy_dai_naming);
+	if (dai) {
+		*ptr = dai;
+		devres_add(dev, ptr);
+	} else {
+		devres_free(ptr);
+	}
+
+	return dai;
+}
+EXPORT_SYMBOL_GPL(devm_snd_soc_register_dai);
+
 static void devm_component_release(struct device *dev, void *res)
 {
 	snd_soc_unregister_component(*(struct device **)res);
-- 
cgit v1.2.3


From 6ae4902f2f3400503f9b78e87e8371e4ffde1e0c Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 12 Jun 2020 15:59:38 -0500
Subject: ASoC: soc-topology: use devm_snd_soc_register_dai()

Use devm_ to avoid use-after-free KASAN reports and simplify error
handling.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/2186
Link: https://lore.kernel.org/r/20200612205938.26415-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-topology.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 9e89633676b7..43e5745b06aa 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1851,7 +1851,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 	list_add(&dai_drv->dobj.list, &tplg->comp->dobj_list);
 
 	/* register the DAI to the component */
-	dai = snd_soc_register_dai(tplg->comp, dai_drv, false);
+	dai = devm_snd_soc_register_dai(tplg->comp->dev, tplg->comp, dai_drv, false);
 	if (!dai)
 		return -ENOMEM;
 
@@ -1859,7 +1859,6 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 	ret = snd_soc_dapm_new_dai_widgets(dapm, dai);
 	if (ret != 0) {
 		dev_err(dai->dev, "Failed to create DAI widgets %d\n", ret);
-		snd_soc_unregister_dai(dai);
 		return ret;
 	}
 
-- 
cgit v1.2.3


From 19ab0f005b165146ea4a93f71e9cb5e71de9c0ce Mon Sep 17 00:00:00 2001
From: "derek.fang" <derek.fang@realtek.com>
Date: Fri, 12 Jun 2020 13:15:25 +0800
Subject: ASoC: rt5682: Let dai clks be registered whether mclk exists or not

According to ideal rt5682 CCF, the root clk is mclk.
But in some platforms, mclk is not exported to CCF.
In this condition, rt5682_register_dai_clks will not be called.
This patch lets dai clks could be registered whether mclk exists or not.

Signed-off-by: derek.fang <derek.fang@realtek.com>
Link: https://lore.kernel.org/r/1591938925-1070-5-git-send-email-derek.fang@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5682.c | 11 ++++++-----
 1 file changed, 6 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index d3245123101d..3e9d2c6c51f9 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -2829,12 +2829,13 @@ static int rt5682_probe(struct snd_soc_component *component)
 				return ret;
 			}
 			rt5682->mclk = NULL;
-		} else {
-			/* Register CCF DAI clock control */
-			ret = rt5682_register_dai_clks(component);
-			if (ret)
-				return ret;
 		}
+
+		/* Register CCF DAI clock control */
+		ret = rt5682_register_dai_clks(component);
+		if (ret)
+			return ret;
+
 		/* Initial setup for CCF */
 		rt5682->lrck[RT5682_AIF1] = CLK_48;
 #endif
-- 
cgit v1.2.3


From 96bf62f018f40cb5d4e4bed95e50fd990a2354af Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Fri, 12 Jun 2020 15:35:07 -0500
Subject: ASoC: soc-pcm: fix checks for multi-cpu FE dailinks

soc_dpcm_fe_runtime_update() is called for all dailinks, and we want
to first discard all back-ends, then deal with front-ends.

The existing code first reports an error with multi-cpu front-ends,
and that check needs to be moved after we know that we are dealing
with a front-end.

Fixes: 6e1276a5e613d ('ASoC: Return error if the function does not support multi-cpu')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
BugLink: https://github.com/thesofproject/linux/issues/1970
Link: https://lore.kernel.org/r/20200612203507.25621-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-pcm.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 2c114b4542ce..c517064f5391 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2630,15 +2630,15 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
 	int count, paths;
 	int ret;
 
+	if (!fe->dai_link->dynamic)
+		return 0;
+
 	if (fe->num_cpus > 1) {
 		dev_err(fe->dev,
 			"%s doesn't support Multi CPU yet\n", __func__);
 		return -EINVAL;
 	}
 
-	if (!fe->dai_link->dynamic)
-		return 0;
-
 	/* only check active links */
 	if (!snd_soc_dai_active(asoc_rtd_to_cpu(fe, 0)))
 		return 0;
-- 
cgit v1.2.3


From 4a95737440d426e93441d49d11abf4c6526d4666 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Fri, 12 Jun 2020 13:37:10 +0100
Subject: ASoc: q6afe: add support to get port direction

This patch adds support to q6afe_is_rx_port() to get direction
of DSP BE dai port, this is useful for setting dailink
directions correctly.

Fixes: c25e295cd77b (ASoC: qcom: Add support to parse common audio device nodes)
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200612123711.29130-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/qdsp6/q6afe.c | 8 ++++++++
 sound/soc/qcom/qdsp6/q6afe.h | 1 +
 2 files changed, 9 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index e0945f7a58c8..0ce4eb60f984 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -800,6 +800,14 @@ int q6afe_get_port_id(int index)
 }
 EXPORT_SYMBOL_GPL(q6afe_get_port_id);
 
+int q6afe_is_rx_port(int index)
+{
+	if (index < 0 || index >= AFE_PORT_MAX)
+		return -EINVAL;
+
+	return port_maps[index].is_rx;
+}
+EXPORT_SYMBOL_GPL(q6afe_is_rx_port);
 static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt,
 			    struct q6afe_port *port)
 {
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index c7ed5422baff..1a0f80a14afe 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -198,6 +198,7 @@ int q6afe_port_start(struct q6afe_port *port);
 int q6afe_port_stop(struct q6afe_port *port);
 void q6afe_port_put(struct q6afe_port *port);
 int q6afe_get_port_id(int index);
+int q6afe_is_rx_port(int index);
 void q6afe_hdmi_port_prepare(struct q6afe_port *port,
 			    struct q6afe_hdmi_cfg *cfg);
 void q6afe_slim_port_prepare(struct q6afe_port *port,
-- 
cgit v1.2.3


From a2120089251f1fe221305e88df99af16f940e236 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Fri, 12 Jun 2020 13:37:11 +0100
Subject: ASoC: qcom: common: set correct directions for dailinks

Currently both FE and BE dai-links are configured bi-directional,
However the DSP BE dais are only single directional,
so set the directions as supported by the BE dais.

Fixes: c25e295cd77b (ASoC: qcom: Add support to parse common audio device nodes)
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: John Stultz <john.stultz@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200612123711.29130-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/common.c | 14 ++++++++++++--
 1 file changed, 12 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
index 6c20bdd850f3..8ada4ecba847 100644
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -4,6 +4,7 @@
 
 #include <linux/module.h>
 #include "common.h"
+#include "qdsp6/q6afe.h"
 
 int qcom_snd_parse_of(struct snd_soc_card *card)
 {
@@ -101,6 +102,15 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 			}
 			link->no_pcm = 1;
 			link->ignore_pmdown_time = 1;
+
+			if (q6afe_is_rx_port(link->id)) {
+				link->dpcm_playback = 1;
+				link->dpcm_capture = 0;
+			} else {
+				link->dpcm_playback = 0;
+				link->dpcm_capture = 1;
+			}
+
 		} else {
 			dlc = devm_kzalloc(dev, sizeof(*dlc), GFP_KERNEL);
 			if (!dlc)
@@ -113,12 +123,12 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 			link->codecs->dai_name = "snd-soc-dummy-dai";
 			link->codecs->name = "snd-soc-dummy";
 			link->dynamic = 1;
+			link->dpcm_playback = 1;
+			link->dpcm_capture = 1;
 		}
 
 		link->ignore_suspend = 1;
 		link->nonatomic = 1;
-		link->dpcm_playback = 1;
-		link->dpcm_capture = 1;
 		link->stream_name = link->name;
 		link++;
 
-- 
cgit v1.2.3


From e74a1e7eaed95f2c6422e7cf9ed70154f17a6db3 Mon Sep 17 00:00:00 2001
From: Jack Yu <jack.yu@realtek.com>
Date: Mon, 15 Jun 2020 11:24:32 +0800
Subject: ASoC: rt1015: Update rt1015 default register value according to spec
 modification.

Update rt1015 default register value according to spec modification.

Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/20200615032433.31061-1-jack.yu@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1015.c | 124 ++++++++++++++++++++++++++++------------------
 sound/soc/codecs/rt1015.h |  15 +++++-
 2 files changed, 89 insertions(+), 50 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c
index 67e2e944d21b..2cccb310fa96 100644
--- a/sound/soc/codecs/rt1015.c
+++ b/sound/soc/codecs/rt1015.c
@@ -34,30 +34,32 @@ static const struct reg_default rt1015_reg[] = {
 	{ 0x0000, 0x0000 },
 	{ 0x0004, 0xa000 },
 	{ 0x0006, 0x0003 },
-	{ 0x000a, 0x0802 },
-	{ 0x000c, 0x0020 },
+	{ 0x000a, 0x081e },
+	{ 0x000c, 0x0006 },
 	{ 0x000e, 0x0000 },
 	{ 0x0010, 0x0000 },
 	{ 0x0012, 0x0000 },
+	{ 0x0014, 0x0000 },
+	{ 0x0016, 0x0000 },
+	{ 0x0018, 0x0000 },
 	{ 0x0020, 0x8000 },
-	{ 0x0022, 0x471b },
-	{ 0x006a, 0x0000 },
-	{ 0x006c, 0x4020 },
+	{ 0x0022, 0x8043 },
 	{ 0x0076, 0x0000 },
 	{ 0x0078, 0x0000 },
-	{ 0x007a, 0x0000 },
+	{ 0x007a, 0x0002 },
 	{ 0x007c, 0x10ec },
 	{ 0x007d, 0x1015 },
 	{ 0x00f0, 0x5000 },
-	{ 0x00f2, 0x0774 },
-	{ 0x00f3, 0x8400 },
+	{ 0x00f2, 0x004c },
+	{ 0x00f3, 0xecfe },
 	{ 0x00f4, 0x0000 },
+	{ 0x00f6, 0x0400 },
 	{ 0x0100, 0x0028 },
 	{ 0x0102, 0xff02 },
-	{ 0x0104, 0x8232 },
+	{ 0x0104, 0xa213 },
 	{ 0x0106, 0x200c },
-	{ 0x010c, 0x002f },
-	{ 0x010e, 0xc000 },
+	{ 0x010c, 0x0000 },
+	{ 0x010e, 0x0058 },
 	{ 0x0111, 0x0200 },
 	{ 0x0112, 0x0400 },
 	{ 0x0114, 0x0022 },
@@ -65,38 +67,46 @@ static const struct reg_default rt1015_reg[] = {
 	{ 0x0118, 0x0000 },
 	{ 0x011a, 0x0123 },
 	{ 0x011c, 0x4567 },
-	{ 0x0300, 0xdddd },
-	{ 0x0302, 0x0000 },
-	{ 0x0311, 0x9330 },
-	{ 0x0313, 0x0000 },
-	{ 0x0314, 0x0000 },
+	{ 0x0300, 0x203d },
+	{ 0x0302, 0x001e },
+	{ 0x0311, 0x0000 },
+	{ 0x0313, 0x6014 },
+	{ 0x0314, 0x00a2 },
 	{ 0x031a, 0x00a0 },
 	{ 0x031c, 0x001f },
 	{ 0x031d, 0xffff },
 	{ 0x031e, 0x0000 },
 	{ 0x031f, 0x0000 },
+	{ 0x0320, 0x0000 },
 	{ 0x0321, 0x0000 },
-	{ 0x0322, 0x0000 },
-	{ 0x0328, 0x0000 },
-	{ 0x0329, 0x0000 },
-	{ 0x032a, 0x0000 },
-	{ 0x032b, 0x0000 },
-	{ 0x032c, 0x0000 },
-	{ 0x032d, 0x0000 },
-	{ 0x032e, 0x030e },
-	{ 0x0330, 0x0080 },
+	{ 0x0322, 0xd7df },
+	{ 0x0328, 0x10b2 },
+	{ 0x0329, 0x0175 },
+	{ 0x032a, 0x36ad },
+	{ 0x032b, 0x7e55 },
+	{ 0x032c, 0x0520 },
+	{ 0x032d, 0xaa00 },
+	{ 0x032e, 0x570e },
+	{ 0x0330, 0xe180 },
 	{ 0x0332, 0x0034 },
-	{ 0x0334, 0x0000 },
-	{ 0x0336, 0x0000 },
+	{ 0x0334, 0x0001 },
+	{ 0x0336, 0x0010 },
+	{ 0x0338, 0x0000 },
+	{ 0x04fa, 0x0030 },
+	{ 0x04fc, 0x35c8 },
+	{ 0x04fe, 0x0800 },
+	{ 0x0500, 0x0400 },
+	{ 0x0502, 0x1000 },
+	{ 0x0504, 0x0000 },
 	{ 0x0506, 0x04ff },
-	{ 0x0508, 0x0030 },
-	{ 0x050a, 0x0018 },
-	{ 0x0519, 0x307f },
-	{ 0x051a, 0xffff },
-	{ 0x051b, 0x4000 },
+	{ 0x0508, 0x0010 },
+	{ 0x050a, 0x001a },
+	{ 0x0519, 0x1c68 },
+	{ 0x051a, 0x0ccc },
+	{ 0x051b, 0x0666 },
 	{ 0x051d, 0x0000 },
 	{ 0x051f, 0x0000 },
-	{ 0x0536, 0x1000 },
+	{ 0x0536, 0x061c },
 	{ 0x0538, 0x0000 },
 	{ 0x053a, 0x0000 },
 	{ 0x053c, 0x0000 },
@@ -110,19 +120,18 @@ static const struct reg_default rt1015_reg[] = {
 	{ 0x0544, 0x0000 },
 	{ 0x0568, 0x0000 },
 	{ 0x056a, 0x0000 },
-	{ 0x1000, 0x0000 },
-	{ 0x1002, 0x6505 },
+	{ 0x1000, 0x0040 },
+	{ 0x1002, 0x5405 },
 	{ 0x1006, 0x5515 },
-	{ 0x1007, 0x003f },
-	{ 0x1009, 0x770f },
-	{ 0x100a, 0x01ff },
-	{ 0x100c, 0x0000 },
+	{ 0x1007, 0x05f7 },
+	{ 0x1009, 0x0b0a },
+	{ 0x100a, 0x00ef },
 	{ 0x100d, 0x0003 },
 	{ 0x1010, 0xa433 },
 	{ 0x1020, 0x0000 },
-	{ 0x1200, 0x3d02 },
-	{ 0x1202, 0x0813 },
-	{ 0x1204, 0x0211 },
+	{ 0x1200, 0x5a01 },
+	{ 0x1202, 0x6524 },
+	{ 0x1204, 0x1f00 },
 	{ 0x1206, 0x0000 },
 	{ 0x1208, 0x0000 },
 	{ 0x120a, 0x0000 },
@@ -130,16 +139,16 @@ static const struct reg_default rt1015_reg[] = {
 	{ 0x120e, 0x0000 },
 	{ 0x1210, 0x0000 },
 	{ 0x1212, 0x0000 },
-	{ 0x1300, 0x0701 },
-	{ 0x1302, 0x12f9 },
-	{ 0x1304, 0x3405 },
+	{ 0x1300, 0x10a1 },
+	{ 0x1302, 0x12ff },
+	{ 0x1304, 0x0400 },
 	{ 0x1305, 0x0844 },
-	{ 0x1306, 0x1611 },
+	{ 0x1306, 0x4611 },
 	{ 0x1308, 0x555e },
 	{ 0x130a, 0x0000 },
-	{ 0x130c, 0x2400},
-	{ 0x130e, 0x7700 },
-	{ 0x130f, 0x0000 },
+	{ 0x130c, 0x2000 },
+	{ 0x130e, 0x0100 },
+	{ 0x130f, 0x0001 },
 	{ 0x1310, 0x0000 },
 	{ 0x1312, 0x0000 },
 	{ 0x1314, 0x0000 },
@@ -209,6 +218,9 @@ static bool rt1015_volatile_register(struct device *dev, unsigned int reg)
 	case RT1015_DC_CALIB_CLSD7:
 	case RT1015_DC_CALIB_CLSD8:
 	case RT1015_S_BST_TIMING_INTER1:
+	case RT1015_OSCK_STA:
+	case RT1015_MONO_DYNA_CTRL1:
+	case RT1015_MONO_DYNA_CTRL5:
 		return true;
 
 	default:
@@ -224,6 +236,12 @@ static bool rt1015_readable_register(struct device *dev, unsigned int reg)
 	case RT1015_CLK3:
 	case RT1015_PLL1:
 	case RT1015_PLL2:
+	case RT1015_DUM_RW1:
+	case RT1015_DUM_RW2:
+	case RT1015_DUM_RW3:
+	case RT1015_DUM_RW4:
+	case RT1015_DUM_RW5:
+	case RT1015_DUM_RW6:
 	case RT1015_CLK_DET:
 	case RT1015_SIL_DET:
 	case RT1015_CUSTOMER_ID:
@@ -235,6 +253,7 @@ static bool rt1015_readable_register(struct device *dev, unsigned int reg)
 	case RT1015_PAD_DRV2:
 	case RT1015_GAT_BOOST:
 	case RT1015_PRO_ALT:
+	case RT1015_OSCK_STA:
 	case RT1015_MAN_I2C:
 	case RT1015_DAC1:
 	case RT1015_DAC2:
@@ -272,6 +291,13 @@ static bool rt1015_readable_register(struct device *dev, unsigned int reg)
 	case RT1015_SMART_BST_CTRL2:
 	case RT1015_ANA_CTRL1:
 	case RT1015_ANA_CTRL2:
+	case RT1015_PWR_STATE_CTRL:
+	case RT1015_MONO_DYNA_CTRL:
+	case RT1015_MONO_DYNA_CTRL1:
+	case RT1015_MONO_DYNA_CTRL2:
+	case RT1015_MONO_DYNA_CTRL3:
+	case RT1015_MONO_DYNA_CTRL4:
+	case RT1015_MONO_DYNA_CTRL5:
 	case RT1015_SPK_VOL:
 	case RT1015_SHORT_DETTOP1:
 	case RT1015_SHORT_DETTOP2:
diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h
index 6fbe802082c4..8169962935a5 100644
--- a/sound/soc/codecs/rt1015.h
+++ b/sound/soc/codecs/rt1015.h
@@ -21,6 +21,12 @@
 #define RT1015_CLK3				0x0006
 #define RT1015_PLL1				0x000a
 #define RT1015_PLL2				0x000c
+#define RT1015_DUM_RW1				0x000e
+#define RT1015_DUM_RW2				0x0010
+#define RT1015_DUM_RW3				0x0012
+#define RT1015_DUM_RW4				0x0014
+#define RT1015_DUM_RW5				0x0016
+#define RT1015_DUM_RW6				0x0018
 #define RT1015_CLK_DET				0x0020
 #define RT1015_SIL_DET				0x0022
 #define RT1015_CUSTOMER_ID			0x0076
@@ -32,6 +38,7 @@
 #define RT1015_PAD_DRV2				0x00f2
 #define RT1015_GAT_BOOST			0x00f3
 #define RT1015_PRO_ALT				0x00f4
+#define RT1015_OSCK_STA				0x00f6
 #define RT1015_MAN_I2C				0x0100
 #define RT1015_DAC1				0x0102
 #define RT1015_DAC2				0x0104
@@ -70,7 +77,13 @@
 #define RT1015_ANA_CTRL1			0x0334
 #define RT1015_ANA_CTRL2			0x0336
 #define RT1015_PWR_STATE_CTRL			0x0338
-#define RT1015_SPK_VOL				0x0506
+#define RT1015_MONO_DYNA_CTRL			0x04fa
+#define RT1015_MONO_DYNA_CTRL1			0x04fc
+#define RT1015_MONO_DYNA_CTRL2			0x04fe
+#define RT1015_MONO_DYNA_CTRL3			0x0500
+#define RT1015_MONO_DYNA_CTRL4			0x0502
+#define RT1015_MONO_DYNA_CTRL5			0x0504
+#define RT1015_SPK_VOL					0x0506
 #define RT1015_SHORT_DETTOP1			0x0508
 #define RT1015_SHORT_DETTOP2			0x050a
 #define RT1015_SPK_DC_DETECT1			0x0519
-- 
cgit v1.2.3


From 40e2c465894e5b79b49f55d9574dbcda4ac0f08f Mon Sep 17 00:00:00 2001
From: Brent Lu <brent.lu@intel.com>
Date: Fri, 12 Jun 2020 18:50:48 +0800
Subject: ASoC: SOF: Intel: hda: Clear RIRB status before reading WP

Port commit 6d011d5057ff ("ALSA: hda: Clear RIRB status before reading
WP") from legacy HDA driver to fix the get response timeout issue.
Current SOF driver does not suffer from this issue because sync write
is enabled in hda_init. The issue will come back if the sync write is
disabled for some reason.

Signed-off-by: Brent Lu <brent.lu@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/1591959048-15813-1-git-send-email-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/hda-stream.c | 9 +++++++--
 1 file changed, 7 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index 7f65dcc95811..1bda14c3590c 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -653,11 +653,16 @@ irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context)
 		if (status & AZX_INT_CTRL_EN) {
 			rirb_status = snd_hdac_chip_readb(bus, RIRBSTS);
 			if (rirb_status & RIRB_INT_MASK) {
+				/*
+				 * Clearing the interrupt status here ensures
+				 * that no interrupt gets masked after the RIRB
+				 * wp is read in snd_hdac_bus_update_rirb.
+				 */
+				snd_hdac_chip_writeb(bus, RIRBSTS,
+						     RIRB_INT_MASK);
 				active = true;
 				if (rirb_status & RIRB_INT_RESPONSE)
 					snd_hdac_bus_update_rirb(bus);
-				snd_hdac_chip_writeb(bus, RIRBSTS,
-						     RIRB_INT_MASK);
 			}
 		}
 #endif
-- 
cgit v1.2.3


From ed1220df6e666500ebf58c4f2fccc681941646fb Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Tue, 16 Jun 2020 10:53:48 +0800
Subject: ASoC: fsl_ssi: Fix bclk calculation for mono channel

For mono channel, SSI will switch to Normal mode.

In Normal mode and Network mode, the Word Length Control bits
control the word length divider in clock generator, which is
different with I2S Master mode (the word length is fixed to
32bit), it should be the value of params_width(hw_params).

The condition "slots == 2" is not good for I2S Master mode,
because for Network mode and Normal mode, the slots can also
be 2. Then we need to use (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK)
to check if it is I2S Master mode.

So we refine the formula for mono channel, otherwise there
will be sound issue for S24_LE.

Fixes: b0a7043d5c2c ("ASoC: fsl_ssi: Caculate bit clock rate using slot number and width")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/034eff1435ff6ce300b6c781130cefd9db22ab9a.1592276147.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/fsl_ssi.c | 13 +++++++++----
 1 file changed, 9 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index bad89b0d129e..1a2fa7f18142 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -678,8 +678,9 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 	struct regmap *regs = ssi->regs;
 	u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i;
 	unsigned long clkrate, baudrate, tmprate;
-	unsigned int slots = params_channels(hw_params);
-	unsigned int slot_width = 32;
+	unsigned int channels = params_channels(hw_params);
+	unsigned int slot_width = params_width(hw_params);
+	unsigned int slots = 2;
 	u64 sub, savesub = 100000;
 	unsigned int freq;
 	bool baudclk_is_used;
@@ -688,10 +689,14 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 	/* Override slots and slot_width if being specifically set... */
 	if (ssi->slots)
 		slots = ssi->slots;
-	/* ...but keep 32 bits if slots is 2 -- I2S Master mode */
-	if (ssi->slot_width && slots != 2)
+	if (ssi->slot_width)
 		slot_width = ssi->slot_width;
 
+	/* ...but force 32 bits for stereo audio using I2S Master Mode */
+	if (channels == 2 &&
+	    (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) == SSI_SCR_I2S_MODE_MASTER)
+		slot_width = 32;
+
 	/* Generate bit clock based on the slot number and slot width */
 	freq = slots * slot_width * params_rate(hw_params);
 
-- 
cgit v1.2.3


From a0b03952a797591d4b6d6fa7b9b7872e27783729 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 16 Jun 2020 15:21:50 +0200
Subject: ALSA: hda/realtek - Add quirk for MSI GE63 laptop

MSI GE63 laptop with ALC1220 codec requires the very same quirk
(ALC1220_FIXUP_CLEVO_P950) as other MSI devices for the proper sound
output.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208057
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200616132150.8778-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6d73f8beadb6..2713560e0c4b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2461,6 +2461,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
 	SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950),
+	SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950),
-- 
cgit v1.2.3


From ff58bbc7b9704a5869204176f804eff57307fef0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 16 Jun 2020 14:09:21 +0200
Subject: ALSA: usb-audio: Fix potential use-after-free of streams

With the recent full-duplex support of implicit feedback streams, an
endpoint can be still running after closing the capture stream as long
as the playback stream with the sync-endpoint is running.  In such a
state, the URBs are still be handled and they may call retire_data_urb
callback, which tries to transfer the data from the PCM buffer.  Since
the PCM stream gets closed, this may lead to use-after-free.

This patch adds the proper clearance of the callback at stopping the
capture stream for addressing the possible UAF above.

Fixes: 10ce77e4817f ("ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback")
Link: https://lore.kernel.org/r/20200616120921.12249-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/pcm.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 84c0ae431936..a777d36c4f5a 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -1787,6 +1787,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream
 		return 0;
 	case SNDRV_PCM_TRIGGER_STOP:
 		stop_endpoints(subs);
+		subs->data_endpoint->retire_data_urb = NULL;
 		subs->running = 0;
 		return 0;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-- 
cgit v1.2.3


From b2c22910fe5aae10b7e17b0721e63a3edf0c9553 Mon Sep 17 00:00:00 2001
From: Kai-Heng Feng <kai.heng.feng@canonical.com>
Date: Wed, 17 Jun 2020 18:29:02 +0800
Subject: ALSA: hda/realtek: Add mute LED and micmute LED support for HP
 systems

There are two more HP systems control mute LED from HDA codec and need
to expose micmute led class so SoF can control micmute LED.

Add quirks to support them.

Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200617102906.16156-2-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2713560e0c4b..737ef82a75fd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7471,6 +7471,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3),
+	SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED),
+	SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED),
-- 
cgit v1.2.3


From 4228668eb936357657046b486207b167caea5175 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 17 Jun 2020 11:47:53 -0500
Subject: ASoC: Intel: SOF: merge COMETLAKE_LP and COMETLAKE_H

We already have two configurations for CometLake, and a third one
coming. On other platforms, we used a single Kconfig option, so we
should follow the same trend by merging the two cases in a backwards
compatible way.

The backwards compatibility is handled by overloading the COMETLAKE_LP
kconfig as COMETLAKE. In practice we've never seen a case where
COMETLAKE_H is not selected along with COMETLAKE_LP, so keeping one
of the two is enough.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200617164755.18104-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/hda/intel-dsp-config.c   |  4 +---
 sound/soc/intel/boards/Kconfig |  4 ++--
 sound/soc/sof/intel/Kconfig    | 29 ++++++++---------------------
 sound/soc/sof/sof-pci-dev.c    | 12 ++++--------
 4 files changed, 15 insertions(+), 34 deletions(-)

(limited to 'sound')

diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c
index be1df80ed013..5df0d2253844 100644
--- a/sound/hda/intel-dsp-config.c
+++ b/sound/hda/intel-dsp-config.c
@@ -192,8 +192,8 @@ static const struct config_entry config_table[] = {
 	},
 #endif
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE)
 /* Cometlake-LP */
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP)
 	{
 		.flags = FLAG_SOF,
 		.device = 0x02c8,
@@ -211,9 +211,7 @@ static const struct config_entry config_table[] = {
 		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
 		.device = 0x02c8,
 	},
-#endif
 /* Cometlake-H */
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H)
 	{
 		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
 		.device = 0x06c8,
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index a2a5798c9139..5dc489a79454 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -492,7 +492,7 @@ config SND_SOC_INTEL_SOF_PCM512x_MACH
 
 endif ## SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL
 
-if (SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK)
+if (SND_SOC_SOF_COMETLAKE && SND_SOC_SOF_HDA_LINK)
 
 config SND_SOC_INTEL_CML_LP_DA7219_MAX98357A_MACH
 	tristate "CML_LP with DA7219 and MAX98357A in I2S Mode"
@@ -520,7 +520,7 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH
 	  Say Y if you have such a device.
 	  If unsure select "N".
 
-endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK
+endif ## SND_SOC_SOF_COMETLAKE && SND_SOC_SOF_HDA_LINK
 
 if SND_SOC_SOF_JASPERLAKE
 
diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index c9a2bee4b55c..3aaf25e4f766 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -25,8 +25,7 @@ config SND_SOC_SOF_INTEL_PCI
 	select SND_SOC_SOF_CANNONLAKE  if SND_SOC_SOF_CANNONLAKE_SUPPORT
 	select SND_SOC_SOF_COFFEELAKE  if SND_SOC_SOF_COFFEELAKE_SUPPORT
 	select SND_SOC_SOF_ICELAKE     if SND_SOC_SOF_ICELAKE_SUPPORT
-	select SND_SOC_SOF_COMETLAKE_LP if SND_SOC_SOF_COMETLAKE_LP_SUPPORT
-	select SND_SOC_SOF_COMETLAKE_H if SND_SOC_SOF_COMETLAKE_H_SUPPORT
+	select SND_SOC_SOF_COMETLAKE   if SND_SOC_SOF_COMETLAKE_SUPPORT
 	select SND_SOC_SOF_TIGERLAKE   if SND_SOC_SOF_TIGERLAKE_SUPPORT
 	select SND_SOC_SOF_ELKHARTLAKE if SND_SOC_SOF_ELKHARTLAKE_SUPPORT
 	select SND_SOC_SOF_JASPERLAKE  if SND_SOC_SOF_JASPERLAKE_SUPPORT
@@ -201,34 +200,22 @@ config SND_SOC_SOF_ICELAKE
 	  This option is not user-selectable but automagically handled by
 	  'select' statements at a higher level
 
-config SND_SOC_SOF_COMETLAKE_LP
+config SND_SOC_SOF_COMETLAKE
 	tristate
 	select SND_SOC_SOF_HDA_COMMON
 	help
 	  This option is not user-selectable but automagically handled by
 	  'select' statements at a higher level
 
-config SND_SOC_SOF_COMETLAKE_LP_SUPPORT
-	bool "SOF support for CometLake-LP"
-	help
-	  This adds support for Sound Open Firmware for Intel(R) platforms
-	  using the Cometlake-LP processors.
-	  Say Y if you have such a device.
-	  If unsure select "N".
+config SND_SOC_SOF_COMETLAKE_SUPPORT
+	bool
 
-config SND_SOC_SOF_COMETLAKE_H
-	tristate
-	select SND_SOC_SOF_HDA_COMMON
-	help
-	  This option is not user-selectable but automagically handled by
-	  'select' statements at a higher level
-
-config SND_SOC_SOF_COMETLAKE_H_SUPPORT
-	bool "SOF support for CometLake-H"
+config SND_SOC_SOF_COMETLAKE_LP_SUPPORT
+	bool "SOF support for CometLake"
+	select SND_SOC_SOF_COMETLAKE_SUPPORT
 	help
 	  This adds support for Sound Open Firmware for Intel(R) platforms
-	  using the Cometlake-H processors.
-	  Say Y if you have such a device.
+	  using the Cometlake processors.
 	  If unsure select "N".
 
 config SND_SOC_SOF_TIGERLAKE_SUPPORT
diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c
index b13697dab7c0..7b5f6e17b05f 100644
--- a/sound/soc/sof/sof-pci-dev.c
+++ b/sound/soc/sof/sof-pci-dev.c
@@ -151,9 +151,7 @@ static const struct sof_dev_desc cfl_desc = {
 };
 #endif
 
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) || \
-	IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H)
-
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE)
 static const struct sof_dev_desc cml_desc = {
 	.machines		= snd_soc_acpi_intel_cml_machines,
 	.alt_machines		= snd_soc_acpi_intel_cml_sdw_machines,
@@ -420,12 +418,10 @@ static const struct pci_device_id sof_pci_ids[] = {
 	{ PCI_DEVICE(0x8086, 0x4dc8),
 		.driver_data = (unsigned long)&jsl_desc},
 #endif
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP)
-	{ PCI_DEVICE(0x8086, 0x02c8),
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE)
+	{ PCI_DEVICE(0x8086, 0x02c8), /* CML-LP */
 		.driver_data = (unsigned long)&cml_desc},
-#endif
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H)
-	{ PCI_DEVICE(0x8086, 0x06c8),
+	{ PCI_DEVICE(0x8086, 0x06c8), /* CML-H */
 		.driver_data = (unsigned long)&cml_desc},
 #endif
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE)
-- 
cgit v1.2.3


From 258fb4f4c34a0db9d3834aba6784d7b322176bb9 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 17 Jun 2020 11:47:54 -0500
Subject: ASoC: SOF: Intel: add PCI ID for CometLake-S

Mirror ID added for legacy HDaudio

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200617164755.18104-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/sof-pci-dev.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c
index 7b5f6e17b05f..f3cb2db67130 100644
--- a/sound/soc/sof/sof-pci-dev.c
+++ b/sound/soc/sof/sof-pci-dev.c
@@ -423,6 +423,8 @@ static const struct pci_device_id sof_pci_ids[] = {
 		.driver_data = (unsigned long)&cml_desc},
 	{ PCI_DEVICE(0x8086, 0x06c8), /* CML-H */
 		.driver_data = (unsigned long)&cml_desc},
+	{ PCI_DEVICE(0x8086, 0xa3f0), /* CML-S */
+		.driver_data = (unsigned long)&cml_desc},
 #endif
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE)
 	{ PCI_DEVICE(0x8086, 0xa0c8),
-- 
cgit v1.2.3


From c8d2e2bfaeffa0f914330e8b4e45b986c8d30b58 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 17 Jun 2020 11:47:55 -0500
Subject: ASoC: SOF: Intel: add PCI IDs for ICL-H and TGL-H

Usually the DSP is not traditionally enabled on H skews but this might
be used moving forward.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200617164755.18104-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/sof-pci-dev.c | 10 ++++++++--
 1 file changed, 8 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c
index f3cb2db67130..aa3532ba1434 100644
--- a/sound/soc/sof/sof-pci-dev.c
+++ b/sound/soc/sof/sof-pci-dev.c
@@ -409,8 +409,11 @@ static const struct pci_device_id sof_pci_ids[] = {
 		.driver_data = (unsigned long)&cfl_desc},
 #endif
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE)
-	{ PCI_DEVICE(0x8086, 0x34C8),
+	{ PCI_DEVICE(0x8086, 0x34C8), /* ICL-LP */
 		.driver_data = (unsigned long)&icl_desc},
+	{ PCI_DEVICE(0x8086, 0x3dc8), /* ICL-H */
+		.driver_data = (unsigned long)&icl_desc},
+
 #endif
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_JASPERLAKE)
 	{ PCI_DEVICE(0x8086, 0x38c8),
@@ -427,8 +430,11 @@ static const struct pci_device_id sof_pci_ids[] = {
 		.driver_data = (unsigned long)&cml_desc},
 #endif
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE)
-	{ PCI_DEVICE(0x8086, 0xa0c8),
+	{ PCI_DEVICE(0x8086, 0xa0c8), /* TGL-LP */
 		.driver_data = (unsigned long)&tgl_desc},
+	{ PCI_DEVICE(0x8086, 0x43c8), /* TGL-H */
+		.driver_data = (unsigned long)&tgl_desc},
+
 #endif
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE)
 	{ PCI_DEVICE(0x8086, 0x4b55),
-- 
cgit v1.2.3


From a94eaccefea1186947c5c5451fcae2245dd7e714 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 17 Jun 2020 11:41:44 -0500
Subject: ASoC: hdac_hda: fix memleak with regmap not freed on remove

kmemleak throws error reports on module load/unload tests, add
snd_hdac_regmap_exit() in .remove().

While we are at it, also fix the error handling flow in .probe() to
use snd_hdac_regmap_exit() if needed.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200617164144.17859-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/hdac_hda.c | 16 ++++++++++------
 1 file changed, 10 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c
index de003acb1951..473efe9ef998 100644
--- a/sound/soc/codecs/hdac_hda.c
+++ b/sound/soc/codecs/hdac_hda.c
@@ -441,13 +441,13 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component)
 	ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name);
 	if (ret < 0) {
 		dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name);
-		goto error;
+		goto error_pm;
 	}
 
 	ret = snd_hdac_regmap_init(&hcodec->core);
 	if (ret < 0) {
 		dev_err(&hdev->dev, "regmap init failed\n");
-		goto error;
+		goto error_pm;
 	}
 
 	patch = (hda_codec_patch_t)hcodec->preset->driver_data;
@@ -455,7 +455,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component)
 		ret = patch(hcodec);
 		if (ret < 0) {
 			dev_err(&hdev->dev, "patch failed %d\n", ret);
-			goto error;
+			goto error_regmap;
 		}
 	} else {
 		dev_dbg(&hdev->dev, "no patch file found\n");
@@ -467,7 +467,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component)
 	ret = snd_hda_codec_parse_pcms(hcodec);
 	if (ret < 0) {
 		dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret);
-		goto error;
+		goto error_regmap;
 	}
 
 	/* HDMI controls need to be created in machine drivers */
@@ -476,7 +476,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component)
 		if (ret < 0) {
 			dev_err(&hdev->dev, "unable to create controls %d\n",
 				ret);
-			goto error;
+			goto error_regmap;
 		}
 	}
 
@@ -496,7 +496,9 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component)
 
 	return 0;
 
-error:
+error_regmap:
+	snd_hdac_regmap_exit(hdev);
+error_pm:
 	pm_runtime_put(&hdev->dev);
 error_no_pm:
 	snd_hdac_ext_bus_link_put(hdev->bus, hlink);
@@ -518,6 +520,8 @@ static void hdac_hda_codec_remove(struct snd_soc_component *component)
 
 	pm_runtime_disable(&hdev->dev);
 	snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+
+	snd_hdac_regmap_exit(hdev);
 }
 
 static const struct snd_soc_dapm_route hdac_hda_dapm_routes[] = {
-- 
cgit v1.2.3


From d50313a5a0d803bcf55121a2b82086633060d05e Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 17 Jun 2020 11:49:09 -0500
Subject: ALSA: hda: Intel: add missing PCI IDs for ICL-H, TGL-H and EKL

Mirror PCI ids used for SOF.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200617164909.18225-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 41a03c61a74b..11ec5c56c80e 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2470,6 +2470,9 @@ static const struct pci_device_id azx_ids[] = {
 	/* Icelake */
 	{ PCI_DEVICE(0x8086, 0x34c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	/* Icelake-H */
+	{ PCI_DEVICE(0x8086, 0x3dc8),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Jasperlake */
 	{ PCI_DEVICE(0x8086, 0x38c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
@@ -2478,9 +2481,14 @@ static const struct pci_device_id azx_ids[] = {
 	/* Tigerlake */
 	{ PCI_DEVICE(0x8086, 0xa0c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	/* Tigerlake-H */
+	{ PCI_DEVICE(0x8086, 0x43c8),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Elkhart Lake */
 	{ PCI_DEVICE(0x8086, 0x4b55),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	{ PCI_DEVICE(0x8086, 0x4b58),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Broxton-P(Apollolake) */
 	{ PCI_DEVICE(0x8086, 0x5a98),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON },
-- 
cgit v1.2.3


From 9f7041b71a2f5defc1629368e7dbe83a9c6ea388 Mon Sep 17 00:00:00 2001
From: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Date: Thu, 18 Jun 2020 12:56:52 +0530
Subject: ASoC: amd: closing specific instance.

The steps to reproduce:

Record from the internal mic :
(arecord -D hw:1,2 -f dat /dev/null -V stereos)

Record from the headphone mic:
(arecord -D hw:1,0 -f dat /dev/null -V stereos)

Kill the recording from internal mic.
We can see the recording from the headphone mic is broken.

This patch rectifies the issue reported.

Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200618072653.27103-1-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/amd/raven/acp3x-pcm-dma.c | 30 +++++++++++++++++++++++-------
 1 file changed, 23 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c
index d8f554f369a8..e6386de20ac7 100644
--- a/sound/soc/amd/raven/acp3x-pcm-dma.c
+++ b/sound/soc/amd/raven/acp3x-pcm-dma.c
@@ -342,11 +342,34 @@ static int acp3x_dma_close(struct snd_soc_component *component,
 {
 	struct snd_soc_pcm_runtime *prtd;
 	struct i2s_dev_data *adata;
+	struct i2s_stream_instance *ins;
 
 	prtd = substream->private_data;
 	component = snd_soc_rtdcom_lookup(prtd, DRV_NAME);
 	adata = dev_get_drvdata(component->dev);
+	ins = substream->runtime->private_data;
+	if (!ins)
+		return -EINVAL;
 
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		switch (ins->i2s_instance) {
+		case I2S_BT_INSTANCE:
+			adata->play_stream = NULL;
+			break;
+		case I2S_SP_INSTANCE:
+		default:
+			adata->i2ssp_play_stream = NULL;
+		}
+	} else {
+		switch (ins->i2s_instance) {
+		case I2S_BT_INSTANCE:
+			adata->capture_stream = NULL;
+			break;
+		case I2S_SP_INSTANCE:
+		default:
+			adata->i2ssp_capture_stream = NULL;
+		}
+	}
 
 	/* Disable ACP irq, when the current stream is being closed and
 	 * another stream is also not active.
@@ -354,13 +377,6 @@ static int acp3x_dma_close(struct snd_soc_component *component,
 	if (!adata->play_stream && !adata->capture_stream &&
 		!adata->i2ssp_play_stream && !adata->i2ssp_capture_stream)
 		rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB);
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		adata->play_stream = NULL;
-		adata->i2ssp_play_stream = NULL;
-	} else {
-		adata->capture_stream = NULL;
-		adata->i2ssp_capture_stream = NULL;
-	}
 	return 0;
 }
 
-- 
cgit v1.2.3


From f141a422159a199f4c8dedb7e0df55b3b2cf16cd Mon Sep 17 00:00:00 2001
From: Qiushi Wu <wu000273@umn.edu>
Date: Sat, 13 Jun 2020 15:51:58 -0500
Subject: ASoC: rockchip: Fix a reference count leak.

Calling pm_runtime_get_sync increments the counter even in case of
failure, causing incorrect ref count if pm_runtime_put is not called in
error handling paths. Call pm_runtime_put if pm_runtime_get_sync fails.

Fixes: fc05a5b22253 ("ASoC: rockchip: add support for pdm controller")
Signed-off-by: Qiushi Wu <wu000273@umn.edu>
Reviewed-by: Heiko Stuebner <heiko@sntech.de>
Link: https://lore.kernel.org/r/20200613205158.27296-1-wu000273@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rockchip_pdm.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c
index 7cd42fcfcf38..1707414cfa92 100644
--- a/sound/soc/rockchip/rockchip_pdm.c
+++ b/sound/soc/rockchip/rockchip_pdm.c
@@ -590,8 +590,10 @@ static int rockchip_pdm_resume(struct device *dev)
 	int ret;
 
 	ret = pm_runtime_get_sync(dev);
-	if (ret < 0)
+	if (ret < 0) {
+		pm_runtime_put(dev);
 		return ret;
+	}
 
 	ret = regcache_sync(pdm->regmap);
 
-- 
cgit v1.2.3


From 73094608b8e214952444fb104651704c98a37aeb Mon Sep 17 00:00:00 2001
From: Christoffer Nielsen <cn@obviux.dk>
Date: Fri, 19 Jun 2020 13:48:22 +0200
Subject: ALSA: usb-audio: Add registration quirk for Kingston HyperX Cloud
 Flight S

Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud
Alpha S (0951:0x16ea) uses two interfaces, but only the second
interface contains the capture stream. This patch delays the
registration until the second interface appears.

Signed-off-by: Christoffer Nielsen <cn@obviux.dk>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAOtG2YHOM3zy+ed9KS-J4HkZo_QGzcUG9MigSp4e4_-13r6B=Q@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index c495e720e2f1..54d4b4b5bd11 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1857,6 +1857,7 @@ struct registration_quirk {
 static const struct registration_quirk registration_quirks[] = {
 	REG_QUIRK_ENTRY(0x0951, 0x16d8, 2),	/* Kingston HyperX AMP */
 	REG_QUIRK_ENTRY(0x0951, 0x16ed, 2),	/* Kingston HyperX Cloud Alpha S */
+	REG_QUIRK_ENTRY(0x0951, 0x16ea, 2),	/* Kingston HyperX Cloud Flight S */
 	{ 0 }					/* terminator */
 };
 
-- 
cgit v1.2.3


From a32a1fc99807244d920d274adc46ba04b538cc8a Mon Sep 17 00:00:00 2001
From: Macpaul Lin <macpaul.lin@mediatek.com>
Date: Tue, 23 Jun 2020 19:03:23 +0800
Subject: ALSA: usb-audio: add quirk for Samsung USBC Headset (AKG)

We've found Samsung USBC Headset (AKG) (VID: 0x04e8, PID: 0xa051)
need a tiny delay after each class compliant request.
Otherwise the device might not be able to be recognized each times.

Signed-off-by: Chihhao Chen <chihhao.chen@mediatek.com>
Signed-off-by: Macpaul Lin <macpaul.lin@mediatek.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/1592910203-24035-1-git-send-email-macpaul.lin@mediatek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound')

diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 54d4b4b5bd11..fca72730a802 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1674,6 +1674,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 	     chip->usb_id == USB_ID(0x0951, 0x16ad)) &&
 	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
 		usleep_range(1000, 2000);
+
+	/*
+	 * Samsung USBC Headset (AKG) need a tiny delay after each
+	 * class compliant request. (Model number: AAM625R or AAM627R)
+	 */
+	if (chip->usb_id == USB_ID(0x04e8, 0xa051) &&
+	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
+		usleep_range(5000, 6000);
 }
 
 /*
-- 
cgit v1.2.3


From 220345e98f1cdc768eeb6e3364a0fa7ab9647fe7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 24 Jun 2020 14:23:40 +0200
Subject: ALSA: usb-audio: Fix OOB access of mixer element list

The USB-audio mixer code holds a linked list of usb_mixer_elem_list,
and several operations are performed for each mixer element.  A few of
them (snd_usb_mixer_notify_id() and snd_usb_mixer_interrupt_v2())
assume each mixer element being a usb_mixer_elem_info object that is a
subclass of usb_mixer_elem_list, cast via container_of() and access it
members.  This may result in an out-of-bound access when a
non-standard list element has been added, as spotted by syzkaller
recently.

This patch adds a new field, is_std_info, in usb_mixer_elem_list to
indicate that the element is the usb_mixer_elem_info type or not, and
skip the access to such an element if needed.

Reported-by: syzbot+fb14314433463ad51625@syzkaller.appspotmail.com
Reported-by: syzbot+2405ca3401e943c538b5@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200624122340.9615-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c        | 15 +++++++++++----
 sound/usb/mixer.h        |  9 +++++++--
 sound/usb/mixer_quirks.c |  3 ++-
 3 files changed, 20 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 15769f266790..eab0fd4fd7c3 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -581,8 +581,9 @@ static int check_matrix_bitmap(unsigned char *bmap,
  * if failed, give up and free the control instance.
  */
 
-int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
-			      struct snd_kcontrol *kctl)
+int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list,
+			   struct snd_kcontrol *kctl,
+			   bool is_std_info)
 {
 	struct usb_mixer_interface *mixer = list->mixer;
 	int err;
@@ -596,6 +597,7 @@ int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
 		return err;
 	}
 	list->kctl = kctl;
+	list->is_std_info = is_std_info;
 	list->next_id_elem = mixer->id_elems[list->id];
 	mixer->id_elems[list->id] = list;
 	return 0;
@@ -3234,8 +3236,11 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid)
 	unitid = delegate_notify(mixer, unitid, NULL, NULL);
 
 	for_each_mixer_elem(list, mixer, unitid) {
-		struct usb_mixer_elem_info *info =
-			mixer_elem_list_to_info(list);
+		struct usb_mixer_elem_info *info;
+
+		if (!list->is_std_info)
+			continue;
+		info = mixer_elem_list_to_info(list);
 		/* invalidate cache, so the value is read from the device */
 		info->cached = 0;
 		snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
@@ -3315,6 +3320,8 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer,
 
 		if (!list->kctl)
 			continue;
+		if (!list->is_std_info)
+			continue;
 
 		info = mixer_elem_list_to_info(list);
 		if (count > 1 && info->control != control)
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 41ec9dc4139b..c29e27ac43a7 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -66,6 +66,7 @@ struct usb_mixer_elem_list {
 	struct usb_mixer_elem_list *next_id_elem; /* list of controls with same id */
 	struct snd_kcontrol *kctl;
 	unsigned int id;
+	bool is_std_info;
 	usb_mixer_elem_dump_func_t dump;
 	usb_mixer_elem_resume_func_t resume;
 };
@@ -103,8 +104,12 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid);
 int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval,
 				int request, int validx, int value_set);
 
-int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
-			      struct snd_kcontrol *kctl);
+int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list,
+			   struct snd_kcontrol *kctl,
+			   bool is_std_info);
+
+#define snd_usb_mixer_add_control(list, kctl) \
+	snd_usb_mixer_add_list(list, kctl, true)
 
 void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list,
 				 struct usb_mixer_interface *mixer,
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index b6bcf2f92383..cec1cfd7edb7 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -158,7 +158,8 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer,
 		return -ENOMEM;
 	}
 	kctl->private_free = snd_usb_mixer_elem_free;
-	return snd_usb_mixer_add_control(list, kctl);
+	/* don't use snd_usb_mixer_add_control() here, this is a special list element */
+	return snd_usb_mixer_add_list(list, kctl, false);
 }
 
 /*
-- 
cgit v1.2.3