17 files changed, 913 insertions, 133 deletions

diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 3963bf234664..67267304e7e9 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -1,6 +1,6 @@
 # SPDX-License-Identifier: GPL-2.0-only
-snd-q6dsp-common-objs := q6dsp-common.o q6dsp-lpass-ports.o q6dsp-lpass-clocks.o
-snd-q6apm-objs := q6apm.o audioreach.o topology.o
+snd-q6dsp-common-y := q6dsp-common.o q6dsp-lpass-ports.o q6dsp-lpass-clocks.o
+snd-q6apm-y := q6apm.o audioreach.o topology.o
 
 obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += snd-q6dsp-common.o
 obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
@@ -17,3 +17,4 @@ obj-$(CONFIG_SND_SOC_QDSP6_APM_DAI) += q6apm-dai.o
 obj-$(CONFIG_SND_SOC_QDSP6_APM_LPASS_DAI) += q6apm-lpass-dais.o
 obj-$(CONFIG_SND_SOC_QDSP6_PRM) += q6prm.o
 obj-$(CONFIG_SND_SOC_QDSP6_PRM_LPASS_CLOCKS) += q6prm-clocks.o
+obj-$(CONFIG_SND_SOC_QDSP6_USB) += q6usb.o
diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index 5291deac0a0b..4ebaaf736fb9 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -267,7 +267,7 @@ void *audioreach_alloc_apm_cmd_pkt(int pkt_size, uint32_t opcode, uint32_t token
 }
 EXPORT_SYMBOL_GPL(audioreach_alloc_apm_cmd_pkt);
 
-static void audioreach_set_channel_mapping(u8 *ch_map, int num_channels)
+void audioreach_set_default_channel_mapping(u8 *ch_map, int num_channels)
 {
 	if (num_channels == 1) {
 		ch_map[0] =  PCM_CHANNEL_FL;
@@ -281,6 +281,7 @@ static void audioreach_set_channel_mapping(u8 *ch_map, int num_channels)
 		ch_map[3] =  PCM_CHANNEL_RS;
 	}
 }
+EXPORT_SYMBOL_GPL(audioreach_set_default_channel_mapping);
 
 static void apm_populate_container_config(struct apm_container_obj *cfg,
 					  struct audioreach_container *cont)
@@ -819,7 +820,7 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph,
 	uint32_t num_channels = cfg->num_channels;
 	int payload_size;
 	struct gpr_pkt *pkt;
-	int rc;
+	int rc, i;
 	void *p;
 
 	payload_size = APM_MFC_CFG_PSIZE(media_format, num_channels) +
@@ -842,18 +843,8 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph,
 	media_format->sample_rate = cfg->sample_rate;
 	media_format->bit_width = cfg->bit_width;
 	media_format->num_channels = cfg->num_channels;
-
-	if (num_channels == 1) {
-		media_format->channel_mapping[0] = PCM_CHANNEL_FL;
-	} else if (num_channels == 2) {
-		media_format->channel_mapping[0] = PCM_CHANNEL_FL;
-		media_format->channel_mapping[1] = PCM_CHANNEL_FR;
-	} else if (num_channels == 4) {
-		media_format->channel_mapping[0] = PCM_CHANNEL_FL;
-		media_format->channel_mapping[1] = PCM_CHANNEL_FR;
-		media_format->channel_mapping[2] = PCM_CHANNEL_LS;
-		media_format->channel_mapping[3] = PCM_CHANNEL_RS;
-	}
+	for (i = 0; i < num_channels; i++)
+		media_format->channel_mapping[i] = cfg->channel_map[i];
 
 	rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);
 
@@ -883,9 +874,6 @@ static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr,
 		mp3_cfg->q_factor = mcfg->bit_width - 1;
 		mp3_cfg->endianness = PCM_LITTLE_ENDIAN;
 		mp3_cfg->num_channels = mcfg->num_channels;
-
-		audioreach_set_channel_mapping(mp3_cfg->channel_mapping,
-					       mcfg->num_channels);
 		break;
 	case SND_AUDIOCODEC_AAC:
 		media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
@@ -1104,9 +1092,7 @@ static int audioreach_pcm_set_media_format(struct q6apm_graph *graph,
 	media_cfg->num_channels = mcfg->num_channels;
 	media_cfg->q_factor = mcfg->bit_width - 1;
 	media_cfg->bits_per_sample = mcfg->bit_width;
-
-	audioreach_set_channel_mapping(media_cfg->channel_mapping,
-				       num_channels);
+	memcpy(media_cfg->channel_mapping, mcfg->channel_map, mcfg->num_channels);
 
 	rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);
 
@@ -1163,9 +1149,7 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph,
 		cfg->q_factor = mcfg->bit_width - 1;
 		cfg->endianness = PCM_LITTLE_ENDIAN;
 		cfg->num_channels = mcfg->num_channels;
-
-		audioreach_set_channel_mapping(cfg->channel_mapping,
-					       num_channels);
+		memcpy(cfg->channel_mapping, mcfg->channel_map, mcfg->num_channels);
 	} else {
 		rc = audioreach_set_compr_media_format(header, p, mcfg);
 		if (rc) {
diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h
index 2c82917b7162..61a69df4f50f 100644
--- a/sound/soc/qcom/qdsp6/audioreach.h
+++ b/sound/soc/qcom/qdsp6/audioreach.h
@@ -755,7 +755,6 @@ struct audioreach_module_config {
 
 	u16	data_format;
 	u16	num_channels;
-	u16	active_channels_mask;
 	u16	dp_idx;
 	u32	channel_allocation;
 	u32	sd_line_mask;
@@ -767,6 +766,7 @@ struct audioreach_module_config {
 /* Packet Allocation routines */
 void *audioreach_alloc_apm_cmd_pkt(int pkt_size, uint32_t opcode, uint32_t
 				    token);
+void audioreach_set_default_channel_mapping(u8 *ch_map, int num_channels);
 void *audioreach_alloc_cmd_pkt(int payload_size, uint32_t opcode,
 			       uint32_t token, uint32_t src_port,
 			       uint32_t dest_port);
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index a9c4f896a7df..0f47aadaabe1 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -92,6 +92,39 @@ static int q6hdmi_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int q6afe_usb_hw_params(struct snd_pcm_substream *substream,
+			       struct snd_pcm_hw_params *params,
+			       struct snd_soc_dai *dai)
+{
+	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+	int channels = params_channels(params);
+	int rate = params_rate(params);
+	struct q6afe_usb_cfg *usb = &dai_data->port_config[dai->id].usb_audio;
+
+	usb->sample_rate = rate;
+	usb->num_channels = channels;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_U16_LE:
+	case SNDRV_PCM_FORMAT_S16_LE:
+		usb->bit_width = 16;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+	case SNDRV_PCM_FORMAT_S24_3LE:
+		usb->bit_width = 24;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		usb->bit_width = 32;
+		break;
+	default:
+		dev_err(dai->dev, "%s: invalid format %d\n",
+			__func__, params_format(params));
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
 static int q6i2s_hw_params(struct snd_pcm_substream *substream,
 			   struct snd_pcm_hw_params *params,
 			   struct snd_soc_dai *dai)
@@ -172,8 +205,8 @@ static int q6tdm_set_tdm_slot(struct snd_soc_dai *dai,
 }
 
 static int q6tdm_set_channel_map(struct snd_soc_dai *dai,
-				unsigned int tx_num, unsigned int *tx_slot,
-				unsigned int rx_num, unsigned int *rx_slot)
+				unsigned int tx_num, const unsigned int *tx_slot,
+				unsigned int rx_num, const unsigned int *rx_slot)
 {
 
 	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
@@ -250,8 +283,10 @@ static int q6tdm_hw_params(struct snd_pcm_substream *substream,
 }
 
 static int q6dma_set_channel_map(struct snd_soc_dai *dai,
-				 unsigned int tx_num, unsigned int *tx_ch_mask,
-				 unsigned int rx_num, unsigned int *rx_ch_mask)
+				 unsigned int tx_num,
+				 const unsigned int *tx_ch_mask,
+				 unsigned int rx_num,
+				 const unsigned int *rx_ch_mask)
 {
 
 	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
@@ -392,6 +427,10 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream,
 		q6afe_cdc_dma_port_prepare(dai_data->port[dai->id],
 					   &dai_data->port_config[dai->id].dma_cfg);
 		break;
+	case USB_RX:
+		q6afe_usb_port_prepare(dai_data->port[dai->id],
+				       &dai_data->port_config[dai->id].usb_audio);
+		break;
 	default:
 		return -EINVAL;
 	}
@@ -407,8 +446,10 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream,
 }
 
 static int q6slim_set_channel_map(struct snd_soc_dai *dai,
-				unsigned int tx_num, unsigned int *tx_slot,
-				unsigned int rx_num, unsigned int *rx_slot)
+				  unsigned int tx_num,
+				  const unsigned int *tx_slot,
+				  unsigned int rx_num,
+				  const unsigned int *rx_slot)
 {
 	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
 	struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id];
@@ -618,6 +659,9 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
 	{"TX_CODEC_DMA_TX_5", NULL, "TX_CODEC_DMA_TX_5 Capture"},
 	{"RX_CODEC_DMA_RX_6 Playback", NULL, "RX_CODEC_DMA_RX_6"},
 	{"RX_CODEC_DMA_RX_7 Playback", NULL, "RX_CODEC_DMA_RX_7"},
+
+	/* USB playback AFE port receives data for playback, hence use the RX port */
+	{"USB Playback", NULL, "USB_RX"},
 };
 
 static int msm_dai_q6_dai_probe(struct snd_soc_dai *dai)
@@ -645,6 +689,23 @@ static int msm_dai_q6_dai_remove(struct snd_soc_dai *dai)
 	return 0;
 }
 
+static const struct snd_soc_dai_ops q6afe_usb_ops = {
+	.probe		= msm_dai_q6_dai_probe,
+	.prepare	= q6afe_dai_prepare,
+	.hw_params	= q6afe_usb_hw_params,
+	/*
+	 * Shutdown callback required to stop the USB AFE port, which is enabled
+	 * by the prepare() stage.  This stops the audio traffic on the USB AFE
+	 * port on the Q6DSP.
+	 */
+	.shutdown	= q6afe_dai_shutdown,
+	/*
+	 * Startup callback not needed, as AFE port start command passes the PCM
+	 * parameters within the AFE command, which is provided by the PCM core
+	 * during the prepare() stage.
+	 */
+};
+
 static const struct snd_soc_dai_ops q6hdmi_ops = {
 	.probe			= msm_dai_q6_dai_probe,
 	.remove			= msm_dai_q6_dai_remove,
@@ -943,6 +1004,8 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
 		0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("RX_CODEC_DMA_RX_7", "NULL",
 		0, SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_AIF_IN("USB_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
 };
 
 static const struct snd_soc_component_driver q6afe_dai_component = {
@@ -1057,6 +1120,7 @@ static int q6afe_dai_dev_probe(struct platform_device *pdev)
 	cfg.q6i2s_ops = &q6i2s_ops;
 	cfg.q6tdm_ops = &q6tdm_ops;
 	cfg.q6dma_ops = &q6dma_ops;
+	cfg.q6usb_ops = &q6afe_usb_ops;
 	dais = q6dsp_audio_ports_set_config(dev, &cfg, &num_dais);
 
 	return devm_snd_soc_register_component(dev, &q6afe_dai_component, dais, num_dais);
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index ef7557be5d66..7b59d514b432 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -35,6 +35,8 @@
 #define AFE_MODULE_TDM			0x0001028A
 
 #define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG 0x00010235
+#define AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS    0x000102A5
+#define AFE_PARAM_ID_USB_AUDIO_DEV_LPCM_FMT 0x000102AA
 
 #define AFE_PARAM_ID_LPAIF_CLK_CONFIG	0x00010238
 #define AFE_PARAM_ID_INT_DIGITAL_CDC_CLK_CONFIG	0x00010239
@@ -44,6 +46,7 @@
 #define AFE_PARAM_ID_TDM_CONFIG	0x0001029D
 #define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG	0x00010297
 #define AFE_PARAM_ID_CODEC_DMA_CONFIG	0x000102B8
+#define AFE_PARAM_ID_USB_AUDIO_CONFIG    0x000102A4
 #define AFE_CMD_REMOTE_LPASS_CORE_HW_VOTE_REQUEST	0x000100f4
 #define AFE_CMD_RSP_REMOTE_LPASS_CORE_HW_VOTE_REQUEST   0x000100f5
 #define AFE_CMD_REMOTE_LPASS_CORE_HW_DEVOTE_REQUEST	0x000100f6
@@ -72,12 +75,16 @@
 #define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL	0x1
 #define AFE_LINEAR_PCM_DATA				0x0
 
+#define AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG 0x1
 
 /* Port IDs */
 #define AFE_API_VERSION_HDMI_CONFIG	0x1
 #define AFE_PORT_ID_MULTICHAN_HDMI_RX	0x100E
 #define AFE_PORT_ID_HDMI_OVER_DP_RX	0x6020
 
+/* USB AFE port */
+#define AFE_PORT_ID_USB_RX                       0x7000
+
 #define AFE_API_VERSION_SLIMBUS_CONFIG 0x1
 /* Clock set API version */
 #define AFE_API_VERSION_CLOCK_SET 1
@@ -359,7 +366,7 @@
 #define AFE_API_VERSION_SLOT_MAPPING_CONFIG	1
 #define AFE_API_VERSION_CODEC_DMA_CONFIG	1
 
-#define TIMEOUT_MS 1000
+#define TIMEOUT_MS 3000
 #define AFE_CMD_RESP_AVAIL	0
 #define AFE_CMD_RESP_NONE	1
 #define AFE_CLK_TOKEN		1024
@@ -513,12 +520,96 @@ struct afe_param_id_cdc_dma_cfg {
 	u16	active_channels_mask;
 } __packed;
 
+struct afe_param_id_usb_cfg {
+/* Minor version used for tracking USB audio device configuration.
+ * Supported values: AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
+ */
+	u32	cfg_minor_version;
+/* Sampling rate of the port.
+ * Supported values:
+ * - AFE_PORT_SAMPLE_RATE_8K
+ * - AFE_PORT_SAMPLE_RATE_11025
+ * - AFE_PORT_SAMPLE_RATE_12K
+ * - AFE_PORT_SAMPLE_RATE_16K
+ * - AFE_PORT_SAMPLE_RATE_22050
+ * - AFE_PORT_SAMPLE_RATE_24K
+ * - AFE_PORT_SAMPLE_RATE_32K
+ * - AFE_PORT_SAMPLE_RATE_44P1K
+ * - AFE_PORT_SAMPLE_RATE_48K
+ * - AFE_PORT_SAMPLE_RATE_96K
+ * - AFE_PORT_SAMPLE_RATE_192K
+ */
+	u32	sample_rate;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+	u16	bit_width;
+/* Number of channels.
+ * Supported values: 1 and 2
+ */
+	u16	num_channels;
+/* Data format supported by the USB. The supported value is
+ * 0 (#AFE_USB_AUDIO_DATA_FORMAT_LINEAR_PCM).
+ */
+	u16	data_format;
+/* this field must be 0 */
+	u16	reserved;
+/* device token of actual end USB audio device */
+	u32	dev_token;
+/* endianness of this interface */
+	u32	endian;
+/* service interval */
+	u32	service_interval;
+} __packed;
+
+/**
+ * struct afe_param_id_usb_audio_dev_params
+ * @cfg_minor_version: Minor version used for tracking USB audio device
+ * configuration.
+ * Supported values:
+ *     AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
+ * @dev_token: device token of actual end USB audio device
+ **/
+struct afe_param_id_usb_audio_dev_params {
+	u32	cfg_minor_version;
+	u32	dev_token;
+} __packed;
+
+/**
+ * struct afe_param_id_usb_audio_dev_lpcm_fmt
+ * @cfg_minor_version: Minor version used for tracking USB audio device
+ * configuration.
+ * Supported values:
+ *     AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
+ * @endian: endianness of this interface
+ **/
+struct afe_param_id_usb_audio_dev_lpcm_fmt {
+	u32	cfg_minor_version;
+	u32	endian;
+} __packed;
+
+#define AFE_PARAM_ID_USB_AUDIO_SVC_INTERVAL     0x000102B7
+
+/**
+ * struct afe_param_id_usb_audio_svc_interval
+ * @cfg_minor_version: Minor version used for tracking USB audio device
+ * configuration.
+ * Supported values:
+ *     AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
+ * @svc_interval: service interval
+ **/
+struct afe_param_id_usb_audio_svc_interval {
+	u32	cfg_minor_version;
+	u32	svc_interval;
+} __packed;
+
 union afe_port_config {
 	struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch;
 	struct afe_param_id_slimbus_cfg           slim_cfg;
 	struct afe_param_id_i2s_cfg	i2s_cfg;
 	struct afe_param_id_tdm_cfg	tdm_cfg;
 	struct afe_param_id_cdc_dma_cfg	dma_cfg;
+	struct afe_param_id_usb_cfg usb_cfg;
 } __packed;
 
 
@@ -833,6 +924,7 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = {
 				RX_CODEC_DMA_RX_6, 1, 1},
 	[RX_CODEC_DMA_RX_7] = { AFE_PORT_ID_RX_CODEC_DMA_RX_7,
 				RX_CODEC_DMA_RX_7, 1, 1},
+	[USB_RX] = { AFE_PORT_ID_USB_RX, USB_RX, 1, 1},
 };
 
 static void q6afe_port_free(struct kref *ref)
@@ -1291,6 +1383,99 @@ void q6afe_tdm_port_prepare(struct q6afe_port *port,
 EXPORT_SYMBOL_GPL(q6afe_tdm_port_prepare);
 
 /**
+ * afe_port_send_usb_dev_param() - Send USB dev token
+ *
+ * @port: Instance of afe port
+ * @cardidx: USB SND card index to reference
+ * @pcmidx: USB SND PCM device index to reference
+ *
+ * The USB dev token carries information about which USB SND card instance and
+ * PCM device to execute the offload on.  This information is carried through
+ * to the stream enable QMI request, which is handled by the offload class
+ * driver.  The information is parsed to determine which USB device to query
+ * the required resources for.
+ */
+int afe_port_send_usb_dev_param(struct q6afe_port *port, int cardidx, int pcmidx)
+{
+	struct afe_param_id_usb_audio_dev_params usb_dev;
+	int ret;
+
+	memset(&usb_dev, 0, sizeof(usb_dev));
+
+	usb_dev.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG;
+	usb_dev.dev_token = (cardidx << 16) | (pcmidx << 8);
+	ret = q6afe_port_set_param_v2(port, &usb_dev,
+				      AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS,
+				      AFE_MODULE_AUDIO_DEV_INTERFACE,
+				      sizeof(usb_dev));
+	if (ret)
+		dev_err(port->afe->dev, "%s: AFE device param cmd failed %d\n",
+			__func__, ret);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(afe_port_send_usb_dev_param);
+
+static int afe_port_send_usb_params(struct q6afe_port *port, struct q6afe_usb_cfg *cfg)
+{
+	union afe_port_config *pcfg = &port->port_cfg;
+	struct afe_param_id_usb_audio_dev_lpcm_fmt lpcm_fmt;
+	struct afe_param_id_usb_audio_svc_interval svc_int;
+	int ret;
+
+	if (!pcfg) {
+		dev_err(port->afe->dev, "%s: Error, no configuration data\n", __func__);
+		return -EINVAL;
+	}
+
+	memset(&lpcm_fmt, 0, sizeof(lpcm_fmt));
+	memset(&svc_int, 0, sizeof(svc_int));
+
+	lpcm_fmt.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG;
+	lpcm_fmt.endian = pcfg->usb_cfg.endian;
+	ret = q6afe_port_set_param_v2(port, &lpcm_fmt,
+				      AFE_PARAM_ID_USB_AUDIO_DEV_LPCM_FMT,
+				      AFE_MODULE_AUDIO_DEV_INTERFACE, sizeof(lpcm_fmt));
+	if (ret) {
+		dev_err(port->afe->dev, "%s: AFE device param cmd LPCM_FMT failed %d\n",
+			__func__, ret);
+		return ret;
+	}
+
+	svc_int.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG;
+	svc_int.svc_interval = pcfg->usb_cfg.service_interval;
+	ret = q6afe_port_set_param_v2(port, &svc_int,
+				      AFE_PARAM_ID_USB_AUDIO_SVC_INTERVAL,
+				      AFE_MODULE_AUDIO_DEV_INTERFACE, sizeof(svc_int));
+	if (ret)
+		dev_err(port->afe->dev, "%s: AFE device param cmd svc_interval failed %d\n",
+			__func__, ret);
+
+	return ret;
+}
+
+/**
+ * q6afe_usb_port_prepare() - Prepare usb afe port.
+ *
+ * @port: Instance of afe port
+ * @cfg: USB configuration for the afe port
+ *
+ */
+void q6afe_usb_port_prepare(struct q6afe_port *port,
+			    struct q6afe_usb_cfg *cfg)
+{
+	union afe_port_config *pcfg = &port->port_cfg;
+
+	pcfg->usb_cfg.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG;
+	pcfg->usb_cfg.sample_rate = cfg->sample_rate;
+	pcfg->usb_cfg.num_channels = cfg->num_channels;
+	pcfg->usb_cfg.bit_width = cfg->bit_width;
+
+	afe_port_send_usb_params(port, cfg);
+}
+EXPORT_SYMBOL_GPL(q6afe_usb_port_prepare);
+
+/**
  * q6afe_hdmi_port_prepare() - Prepare hdmi afe port.
  *
  * @port: Instance of afe port
@@ -1612,7 +1797,10 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
 		break;
 	case AFE_PORT_ID_WSA_CODEC_DMA_RX_0 ... AFE_PORT_ID_RX_CODEC_DMA_RX_7:
 		cfg_type = AFE_PARAM_ID_CODEC_DMA_CONFIG;
-	break;
+		break;
+	case AFE_PORT_ID_USB_RX:
+		cfg_type = AFE_PARAM_ID_USB_AUDIO_CONFIG;
+		break;
 	default:
 		dev_err(dev, "Invalid port id 0x%x\n", port_id);
 		return ERR_PTR(-EINVAL);
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index 65d0676075e1..a29abe4ce436 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -3,7 +3,7 @@
 #ifndef __Q6AFE_H__
 #define __Q6AFE_H__
 
-#define AFE_PORT_MAX		129
+#define AFE_PORT_MAX		137
 
 #define MSM_AFE_PORT_TYPE_RX 0
 #define MSM_AFE_PORT_TYPE_TX 1
@@ -203,6 +203,36 @@ struct q6afe_cdc_dma_cfg {
 	u16	active_channels_mask;
 };
 
+/**
+ * struct q6afe_usb_cfg
+ * @cfg_minor_version: Minor version used for tracking USB audio device
+ * configuration.
+ * Supported values:
+ *     AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG
+ * @sample_rate: Sampling rate of the port
+ *    Supported values:
+ *      AFE_PORT_SAMPLE_RATE_8K
+ *      AFE_PORT_SAMPLE_RATE_11025
+ *      AFE_PORT_SAMPLE_RATE_12K
+ *      AFE_PORT_SAMPLE_RATE_16K
+ *      AFE_PORT_SAMPLE_RATE_22050
+ *      AFE_PORT_SAMPLE_RATE_24K
+ *      AFE_PORT_SAMPLE_RATE_32K
+ *      AFE_PORT_SAMPLE_RATE_44P1K
+ *      AFE_PORT_SAMPLE_RATE_48K
+ *      AFE_PORT_SAMPLE_RATE_96K
+ *      AFE_PORT_SAMPLE_RATE_192K
+ * @bit_width: Bit width of the sample.
+ *    Supported values: 16, 24
+ * @num_channels: Number of channels
+ *    Supported values: 1, 2
+ **/
+struct q6afe_usb_cfg {
+	u32	cfg_minor_version;
+	u32     sample_rate;
+	u16	bit_width;
+	u16	num_channels;
+};
 
 struct q6afe_port_config {
 	struct q6afe_hdmi_cfg hdmi;
@@ -210,6 +240,7 @@ struct q6afe_port_config {
 	struct q6afe_i2s_cfg i2s_cfg;
 	struct q6afe_tdm_cfg tdm;
 	struct q6afe_cdc_dma_cfg dma_cfg;
+	struct q6afe_usb_cfg usb_audio;
 };
 
 struct q6afe_port;
@@ -219,6 +250,8 @@ 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);
+void q6afe_usb_port_prepare(struct q6afe_port *port,
+			    struct q6afe_usb_cfg *cfg);
 void q6afe_hdmi_port_prepare(struct q6afe_port *port,
 			    struct q6afe_hdmi_cfg *cfg);
 void q6afe_slim_port_prepare(struct q6afe_port *port,
@@ -228,6 +261,7 @@ void q6afe_tdm_port_prepare(struct q6afe_port *port, struct q6afe_tdm_cfg *cfg);
 void q6afe_cdc_dma_port_prepare(struct q6afe_port *port,
 				struct q6afe_cdc_dma_cfg *cfg);
 
+int afe_port_send_usb_dev_param(struct q6afe_port *port, int cardidx, int pcmidx);
 int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
 			  int clk_src, int clk_root,
 			  unsigned int freq, int dir);
diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index 00bbd291be5c..2cd522108221 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -24,8 +24,8 @@
 #define PLAYBACK_MIN_PERIOD_SIZE	128
 #define CAPTURE_MIN_NUM_PERIODS		2
 #define CAPTURE_MAX_NUM_PERIODS		8
-#define CAPTURE_MAX_PERIOD_SIZE		4096
-#define CAPTURE_MIN_PERIOD_SIZE		320
+#define CAPTURE_MAX_PERIOD_SIZE		65536
+#define CAPTURE_MIN_PERIOD_SIZE		6144
 #define BUFFER_BYTES_MAX (PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE)
 #define BUFFER_BYTES_MIN (PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE)
 #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
@@ -64,20 +64,16 @@ struct q6apm_dai_rtd {
 	phys_addr_t phys;
 	unsigned int pcm_size;
 	unsigned int pcm_count;
-	unsigned int pos;       /* Buffer position */
 	unsigned int periods;
 	unsigned int bytes_sent;
 	unsigned int bytes_received;
 	unsigned int copied_total;
 	uint16_t bits_per_sample;
-	uint16_t source; /* Encoding source bit mask */
-	uint16_t session_id;
+	snd_pcm_uframes_t queue_ptr;
 	bool next_track;
 	enum stream_state state;
 	struct q6apm_graph *graph;
 	spinlock_t lock;
-	uint32_t initial_samples_drop;
-	uint32_t trailing_samples_drop;
 	bool notify_on_drain;
 };
 
@@ -85,7 +81,7 @@ struct q6apm_dai_data {
 	long long sid;
 };
 
-static struct snd_pcm_hardware q6apm_dai_hardware_capture = {
+static const struct snd_pcm_hardware q6apm_dai_hardware_capture = {
 	.info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BLOCK_TRANSFER |
 				 SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
@@ -104,7 +100,7 @@ static struct snd_pcm_hardware q6apm_dai_hardware_capture = {
 	.fifo_size =            0,
 };
 
-static struct snd_pcm_hardware q6apm_dai_hardware_playback = {
+static const struct snd_pcm_hardware q6apm_dai_hardware_playback = {
 	.info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BLOCK_TRANSFER |
 				 SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
@@ -127,25 +123,16 @@ static void event_handler(uint32_t opcode, uint32_t token, void *payload, void *
 {
 	struct q6apm_dai_rtd *prtd = priv;
 	struct snd_pcm_substream *substream = prtd->substream;
-	unsigned long flags;
 
 	switch (opcode) {
 	case APM_CLIENT_EVENT_CMD_EOS_DONE:
 		prtd->state = Q6APM_STREAM_STOPPED;
 		break;
 	case APM_CLIENT_EVENT_DATA_WRITE_DONE:
-		spin_lock_irqsave(&prtd->lock, flags);
-		prtd->pos += prtd->pcm_count;
-		spin_unlock_irqrestore(&prtd->lock, flags);
 		snd_pcm_period_elapsed(substream);
-		if (prtd->state == Q6APM_STREAM_RUNNING)
-			q6apm_write_async(prtd->graph, prtd->pcm_count, 0, 0, 0);
 
 		break;
 	case APM_CLIENT_EVENT_DATA_READ_DONE:
-		spin_lock_irqsave(&prtd->lock, flags);
-		prtd->pos += prtd->pcm_count;
-		spin_unlock_irqrestore(&prtd->lock, flags);
 		snd_pcm_period_elapsed(substream);
 		if (prtd->state == Q6APM_STREAM_RUNNING)
 			q6apm_read(prtd->graph);
@@ -243,6 +230,7 @@ static int q6apm_dai_prepare(struct snd_soc_component *component,
 	cfg.num_channels = runtime->channels;
 	cfg.bit_width = prtd->bits_per_sample;
 	cfg.fmt = SND_AUDIOCODEC_PCM;
+	audioreach_set_default_channel_mapping(cfg.channel_map, runtime->channels);
 
 	if (prtd->state) {
 		/* clear the previous setup if any  */
@@ -251,7 +239,6 @@ static int q6apm_dai_prepare(struct snd_soc_component *component,
 	}
 
 	prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
-	prtd->pos = 0;
 	/* rate and channels are sent to audio driver */
 	ret = q6apm_graph_media_format_shmem(prtd->graph, &cfg);
 	if (ret < 0) {
@@ -297,6 +284,27 @@ static int q6apm_dai_prepare(struct snd_soc_component *component,
 	return 0;
 }
 
+static int q6apm_dai_ack(struct snd_soc_component *component, struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct q6apm_dai_rtd *prtd = runtime->private_data;
+	int i, ret = 0, avail_periods;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		avail_periods = (runtime->control->appl_ptr - prtd->queue_ptr)/runtime->period_size;
+		for (i = 0; i < avail_periods; i++) {
+			ret = q6apm_write_async(prtd->graph, prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+			if (ret < 0) {
+				dev_err(component->dev, "Error queuing playback buffer %d\n", ret);
+				return ret;
+			}
+			prtd->queue_ptr += runtime->period_size;
+		}
+	}
+
+	return ret;
+}
+
 static int q6apm_dai_trigger(struct snd_soc_component *component,
 			     struct snd_pcm_substream *substream, int cmd)
 {
@@ -308,9 +316,6 @@ static int q6apm_dai_trigger(struct snd_soc_component *component,
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		 /* start writing buffers for playback only as we already queued capture buffers */
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			ret = q6apm_write_async(prtd->graph, prtd->pcm_count, 0, 0, 0);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 		/* TODO support be handled via SoftPause Module */
@@ -331,7 +336,7 @@ static int q6apm_dai_open(struct snd_soc_component *component,
 			  struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+	struct snd_soc_pcm_runtime *soc_prtd = snd_soc_substream_to_rtd(substream);
 	struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(soc_prtd, 0);
 	struct device *dev = component->dev;
 	struct q6apm_dai_data *pdata;
@@ -380,13 +385,14 @@ static int q6apm_dai_open(struct snd_soc_component *component,
 		}
 	}
 
-	ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+	/* setup 10ms latency to accommodate DSP restrictions */
+	ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 480);
 	if (ret < 0) {
 		dev_err(dev, "constraint for period bytes step ret = %d\n", ret);
 		goto err;
 	}
 
-	ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+	ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 480);
 	if (ret < 0) {
 		dev_err(dev, "constraint for buffer bytes step ret = %d\n", ret);
 		goto err;
@@ -431,16 +437,12 @@ static snd_pcm_uframes_t q6apm_dai_pointer(struct snd_soc_component *component,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct q6apm_dai_rtd *prtd = runtime->private_data;
 	snd_pcm_uframes_t ptr;
-	unsigned long flags;
 
-	spin_lock_irqsave(&prtd->lock, flags);
-	if (prtd->pos == prtd->pcm_size)
-		prtd->pos = 0;
-
-	ptr =  bytes_to_frames(runtime, prtd->pos);
-	spin_unlock_irqrestore(&prtd->lock, flags);
+	ptr = q6apm_get_hw_pointer(prtd->graph, substream->stream) * runtime->period_size;
+	if (ptr)
+		return ptr - 1;
 
-	return ptr;
+	return 0;
 }
 
 static int q6apm_dai_hw_params(struct snd_soc_component *component,
@@ -655,8 +657,6 @@ static int q6apm_dai_compr_set_params(struct snd_soc_component *component,
 	prtd->pcm_size = runtime->fragments * runtime->fragment_size;
 	prtd->bits_per_sample = 16;
 
-	prtd->pos = 0;
-
 	if (prtd->next_track != true) {
 		memcpy(&prtd->codec, codec, sizeof(*codec));
 
@@ -669,6 +669,8 @@ static int q6apm_dai_compr_set_params(struct snd_soc_component *component,
 		cfg.num_channels = 2;
 		cfg.bit_width = prtd->bits_per_sample;
 		cfg.fmt = codec->id;
+		audioreach_set_default_channel_mapping(cfg.channel_map,
+						       cfg.num_channels);
 		memcpy(&cfg.codec, codec, sizeof(*codec));
 
 		ret = q6apm_graph_media_format_shmem(prtd->graph, &cfg);
@@ -720,14 +722,12 @@ static int q6apm_dai_compr_set_metadata(struct snd_soc_component *component,
 
 	switch (metadata->key) {
 	case SNDRV_COMPRESS_ENCODER_PADDING:
-		prtd->trailing_samples_drop = metadata->value[0];
 		q6apm_remove_trailing_silence(component->dev, prtd->graph,
-					      prtd->trailing_samples_drop);
+					      metadata->value[0]);
 		break;
 	case SNDRV_COMPRESS_ENCODER_DELAY:
-		prtd->initial_samples_drop = metadata->value[0];
 		q6apm_remove_initial_silence(component->dev, prtd->graph,
-					     prtd->initial_samples_drop);
+					     metadata->value[0]);
 		break;
 	default:
 		ret = -EINVAL;
@@ -839,6 +839,7 @@ static const struct snd_soc_component_driver q6apm_fe_dai_component = {
 	.hw_params	= q6apm_dai_hw_params,
 	.pointer	= q6apm_dai_pointer,
 	.trigger	= q6apm_dai_trigger,
+	.ack		= q6apm_dai_ack,
 	.compress_ops	= &q6apm_dai_compress_ops,
 	.use_dai_pcm_id = true,
 };
diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
index 68a38f63a2db..a0d90462fd6a 100644
--- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
+++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
@@ -25,13 +25,15 @@ struct q6apm_lpass_dai_data {
 };
 
 static int q6dma_set_channel_map(struct snd_soc_dai *dai,
-				 unsigned int tx_num, unsigned int *tx_ch_mask,
-				 unsigned int rx_num, unsigned int *rx_ch_mask)
+				 unsigned int tx_num,
+				 const unsigned int *tx_ch_mask,
+				 unsigned int rx_num,
+				 const unsigned int *rx_ch_mask)
 {
 
 	struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev);
 	struct audioreach_module_config *cfg = &dai_data->module_config[dai->id];
-	int ch_mask;
+	int i;
 
 	switch (dai->id) {
 	case WSA_CODEC_DMA_TX_0:
@@ -56,7 +58,8 @@ static int q6dma_set_channel_map(struct snd_soc_dai *dai,
 				tx_num);
 			return -EINVAL;
 		}
-		ch_mask = *tx_ch_mask;
+		for (i = 0; i < tx_num; i++)
+			cfg->channel_map[i] = tx_ch_mask[i];
 
 		break;
 	case WSA_CODEC_DMA_RX_0:
@@ -79,7 +82,8 @@ static int q6dma_set_channel_map(struct snd_soc_dai *dai,
 				rx_num);
 			return -EINVAL;
 		}
-		ch_mask = *rx_ch_mask;
+		for (i = 0; i < rx_num; i++)
+			cfg->channel_map[i] = rx_ch_mask[i];
 
 		break;
 	default:
@@ -88,8 +92,6 @@ static int q6dma_set_channel_map(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	cfg->active_channels_mask = ch_mask;
-
 	return 0;
 }
 
@@ -104,6 +106,7 @@ static int q6hdmi_hw_params(struct snd_pcm_substream *substream,
 	cfg->bit_width = params_width(params);
 	cfg->sample_rate = params_rate(params);
 	cfg->num_channels = channels;
+	audioreach_set_default_channel_mapping(cfg->channel_map, channels);
 
 	switch (dai->id) {
 	case DISPLAY_PORT_RX_0:
@@ -128,10 +131,12 @@ static int q6dma_hw_params(struct snd_pcm_substream *substream,
 {
 	struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev);
 	struct audioreach_module_config *cfg = &dai_data->module_config[dai->id];
+	int channels = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max;
 
 	cfg->bit_width = params_width(params);
 	cfg->sample_rate = params_rate(params);
-	cfg->num_channels = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max;
+	cfg->num_channels = channels;
+	audioreach_set_default_channel_mapping(cfg->channel_map, channels);
 
 	return 0;
 }
@@ -141,14 +146,17 @@ 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 +171,10 @@ 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 +193,29 @@ 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;
+		dev_err(dai->dev, "Failed to start APM port %d\n", dai->id);
+		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)
diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
index 2a2a5bd98110..b4ffa0f0b188 100644
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -230,7 +230,7 @@ int q6apm_map_memory_regions(struct q6apm_graph *graph, unsigned int dir, phys_a
 		return 0;
 	}
 
-	buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_KERNEL);
+	buf = kcalloc(periods, sizeof(struct audio_buffer), GFP_KERNEL);
 	if (!buf) {
 		mutex_unlock(&graph->lock);
 		return -ENOMEM;
@@ -494,6 +494,19 @@ int q6apm_read(struct q6apm_graph *graph)
 }
 EXPORT_SYMBOL_GPL(q6apm_read);
 
+int q6apm_get_hw_pointer(struct q6apm_graph *graph, int dir)
+{
+	struct audioreach_graph_data *data;
+
+	if (dir == SNDRV_PCM_STREAM_PLAYBACK)
+		data = &graph->rx_data;
+	else
+		data = &graph->tx_data;
+
+	return (int)atomic_read(&data->hw_ptr);
+}
+EXPORT_SYMBOL_GPL(q6apm_get_hw_pointer);
+
 static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op)
 {
 	struct data_cmd_rsp_rd_sh_mem_ep_data_buffer_done_v2 *rd_done;
@@ -520,7 +533,8 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op)
 		done = data->payload;
 		phys = graph->rx_data.buf[token].phys;
 		mutex_unlock(&graph->lock);
-
+		/* token numbering starts at 0 */
+		atomic_set(&graph->rx_data.hw_ptr, token + 1);
 		if (lower_32_bits(phys) == done->buf_addr_lsw &&
 		    upper_32_bits(phys) == done->buf_addr_msw) {
 			graph->result.opcode = hdr->opcode;
@@ -553,6 +567,8 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op)
 		rd_done = data->payload;
 		phys = graph->tx_data.buf[hdr->token].phys;
 		mutex_unlock(&graph->lock);
+		/* token numbering starts at 0 */
+		atomic_set(&graph->tx_data.hw_ptr, hdr->token + 1);
 
 		if (upper_32_bits(phys) == rd_done->buf_addr_msw &&
 		    lower_32_bits(phys) == rd_done->buf_addr_lsw) {
diff --git a/sound/soc/qcom/qdsp6/q6apm.h b/sound/soc/qcom/qdsp6/q6apm.h
index c248c8d2b1ab..7ce08b401e31 100644
--- a/sound/soc/qcom/qdsp6/q6apm.h
+++ b/sound/soc/qcom/qdsp6/q6apm.h
@@ -2,6 +2,7 @@
 #ifndef __Q6APM_H__
 #define __Q6APM_H__
 #include <linux/types.h>
+#include <linux/atomic.h>
 #include <linux/slab.h>
 #include <linux/wait.h>
 #include <linux/kernel.h>
@@ -77,6 +78,7 @@ struct audioreach_graph_data {
 	uint32_t num_periods;
 	uint32_t dsp_buf;
 	uint32_t mem_map_handle;
+	atomic_t hw_ptr;
 };
 
 struct audioreach_graph {
@@ -150,4 +152,5 @@ int q6apm_enable_compress_module(struct device *dev, struct q6apm_graph *graph,
 int q6apm_remove_initial_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples);
 int q6apm_remove_trailing_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples);
 int q6apm_set_real_module_id(struct device *dev, struct q6apm_graph *graph, uint32_t codec_id);
+int q6apm_get_hw_pointer(struct q6apm_graph *graph, int dir);
 #endif /* __APM_GRAPH_ */
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index aeb6a9d479ab..a400c9a31fea 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -103,7 +103,7 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_capture = {
 	.fifo_size =            0,
 };
 
-static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
+static const struct snd_pcm_hardware q6asm_dai_hardware_playback = {
 	.info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
 				SNDRV_PCM_INFO_BLOCK_TRANSFER |
 				SNDRV_PCM_INFO_MMAP_VALID |
@@ -128,8 +128,13 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
 #define Q6ASM_FEDAI_DRIVER(num) { \
 		.playback = {						\
 			.stream_name = "MultiMedia"#num" Playback",	\
-			.rates = (SNDRV_PCM_RATE_8000_192000|		\
-					SNDRV_PCM_RATE_KNOT),		\
+			.rates = (SNDRV_PCM_RATE_8000_48000 |		\
+				  SNDRV_PCM_RATE_12000 |		\
+				  SNDRV_PCM_RATE_24000 |		\
+				  SNDRV_PCM_RATE_88200 |		\
+				  SNDRV_PCM_RATE_96000 |		\
+				  SNDRV_PCM_RATE_176400 |		\
+				  SNDRV_PCM_RATE_192000),		\
 			.formats = (SNDRV_PCM_FMTBIT_S16_LE |		\
 					SNDRV_PCM_FMTBIT_S24_LE),	\
 			.channels_min = 1,				\
@@ -139,8 +144,9 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
 		},							\
 		.capture = {						\
 			.stream_name = "MultiMedia"#num" Capture",	\
-			.rates = (SNDRV_PCM_RATE_8000_48000|		\
-					SNDRV_PCM_RATE_KNOT),		\
+			.rates = (SNDRV_PCM_RATE_8000_48000 |		\
+				  SNDRV_PCM_RATE_12000 |		\
+				  SNDRV_PCM_RATE_24000),		\
 			.formats = (SNDRV_PCM_FMTBIT_S16_LE |		\
 				    SNDRV_PCM_FMTBIT_S24_LE),		\
 			.channels_min = 1,				\
@@ -152,18 +158,6 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
 		.id = MSM_FRONTEND_DAI_MULTIMEDIA##num,			\
 	}
 
-/* Conventional and unconventional sample rate supported */
-static unsigned int supported_sample_rates[] = {
-	8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
-	88200, 96000, 176400, 192000
-};
-
-static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
-	.count = ARRAY_SIZE(supported_sample_rates),
-	.list = supported_sample_rates,
-	.mask = 0,
-};
-
 static const struct snd_compr_codec_caps q6asm_compr_caps = {
 	.num_descriptors = 1,
 	.descriptor[0].max_ch = 2,
@@ -390,11 +384,6 @@ static int q6asm_dai_open(struct snd_soc_component *component,
 	else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
 		runtime->hw = q6asm_dai_hardware_capture;
 
-	ret = snd_pcm_hw_constraint_list(runtime, 0,
-				SNDRV_PCM_HW_PARAM_RATE,
-				&constraints_sample_rates);
-	if (ret < 0)
-		dev_info(dev, "snd_pcm_hw_constraint_list failed\n");
 	/* Ensure that buffer size is a multiple of period size */
 	ret = snd_pcm_hw_constraint_integer(runtime,
 					    SNDRV_PCM_HW_PARAM_PERIODS);
@@ -903,9 +892,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 
 		if (ret < 0) {
 			dev_err(dev, "q6asm_open_write failed\n");
-			q6asm_audio_client_free(prtd->audio_client);
-			prtd->audio_client = NULL;
-			return ret;
+			goto open_err;
 		}
 	}
 
@@ -914,7 +901,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 			      prtd->session_id, dir);
 	if (ret) {
 		dev_err(dev, "Stream reg failed ret:%d\n", ret);
-		return ret;
+		goto q6_err;
 	}
 
 	ret = __q6asm_dai_compr_set_codec_params(component, stream,
@@ -922,7 +909,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 						 prtd->stream_id);
 	if (ret) {
 		dev_err(dev, "codec param setup failed ret:%d\n", ret);
-		return ret;
+		goto q6_err;
 	}
 
 	ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
@@ -931,12 +918,21 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
 
 	if (ret < 0) {
 		dev_err(dev, "Buffer Mapping failed ret:%d\n", ret);
-		return -ENOMEM;
+		ret = -ENOMEM;
+		goto q6_err;
 	}
 
 	prtd->state = Q6ASM_STREAM_RUNNING;
 
 	return 0;
+
+q6_err:
+	q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE);
+
+open_err:
+	q6asm_audio_client_free(prtd->audio_client);
+	prtd->audio_client = NULL;
+	return ret;
 }
 
 static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component,
diff --git a/sound/soc/qcom/qdsp6/q6dsp-common.c b/sound/soc/qcom/qdsp6/q6dsp-common.c
index 95585dea2b36..f74585d88bd6 100644
--- a/sound/soc/qcom/qdsp6/q6dsp-common.c
+++ b/sound/soc/qcom/qdsp6/q6dsp-common.c
@@ -98,4 +98,6 @@ int q6dsp_get_channel_allocation(int channels)
 	return channel_allocation;
 }
 EXPORT_SYMBOL_GPL(q6dsp_get_channel_allocation);
+
+MODULE_DESCRIPTION("ASoC MSM QDSP6 helper functions");
 MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c
index 4919001de08b..4eed54b071a5 100644
--- a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c
+++ b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c
@@ -99,6 +99,26 @@
 static struct snd_soc_dai_driver q6dsp_audio_fe_dais[] = {
 	{
 		.playback = {
+			.stream_name = "USB Playback",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+					SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |
+					SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+					SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+					SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
+					SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE |
+					SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |
+					SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE,
+			.channels_min = 1,
+			.channels_max = 2,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
+		.id = USB_RX,
+		.name = "USB_RX",
+	},
+	{
+		.playback = {
 			.stream_name = "HDMI Playback",
 			.rates = SNDRV_PCM_RATE_48000 |
 				 SNDRV_PCM_RATE_96000 |
@@ -624,6 +644,9 @@ struct snd_soc_dai_driver *q6dsp_audio_ports_set_config(struct device *dev,
 		case WSA_CODEC_DMA_RX_0 ... RX_CODEC_DMA_RX_7:
 			q6dsp_audio_fe_dais[i].ops = cfg->q6dma_ops;
 			break;
+		case USB_RX:
+			q6dsp_audio_fe_dais[i].ops = cfg->q6usb_ops;
+			break;
 		default:
 			break;
 		}
diff --git a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h
index 7f052c8a1257..d8dde6dd0aca 100644
--- a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h
+++ b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h
@@ -11,6 +11,7 @@ struct q6dsp_audio_port_dai_driver_config {
 	const struct snd_soc_dai_ops *q6i2s_ops;
 	const struct snd_soc_dai_ops *q6tdm_ops;
 	const struct snd_soc_dai_ops *q6dma_ops;
+	const struct snd_soc_dai_ops *q6usb_ops;
 };
 
 struct snd_soc_dai_driver *q6dsp_audio_ports_set_config(struct device *dev,
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 81fde0681f95..f49243daa517 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -435,6 +435,7 @@ static struct session_data *get_session_from_id(struct msm_routing_data *data,
 
 	return NULL;
 }
+
 /**
  * q6routing_stream_close() - Deregister a stream
  *
@@ -515,6 +516,9 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
+static const struct snd_kcontrol_new usb_rx_mixer_controls[] = {
+	Q6ROUTING_RX_MIXERS(USB_RX) };
+
 static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
 	Q6ROUTING_RX_MIXERS(HDMI_RX) };
 
@@ -933,6 +937,9 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
 	SND_SOC_DAPM_MIXER("RX_CODEC_DMA_RX_7 Audio Mixer", SND_SOC_NOPM, 0, 0,
 		rx_codec_dma_rx_7_mixer_controls,
 		ARRAY_SIZE(rx_codec_dma_rx_7_mixer_controls)),
+	SND_SOC_DAPM_MIXER("USB_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+			   usb_rx_mixer_controls,
+			   ARRAY_SIZE(usb_rx_mixer_controls)),
 	SND_SOC_DAPM_MIXER("MultiMedia1 Mixer", SND_SOC_NOPM, 0, 0,
 		mmul1_mixer_controls, ARRAY_SIZE(mmul1_mixer_controls)),
 	SND_SOC_DAPM_MIXER("MultiMedia2 Mixer", SND_SOC_NOPM, 0, 0,
@@ -949,7 +956,6 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
 		mmul7_mixer_controls, ARRAY_SIZE(mmul7_mixer_controls)),
 	SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0,
 		mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)),
-
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
@@ -1026,6 +1032,7 @@ static const struct snd_soc_dapm_route intercon[] = {
 	Q6ROUTING_RX_DAPM_ROUTE("RX_CODEC_DMA_RX_5 Audio Mixer", "RX_CODEC_DMA_RX_5"),
 	Q6ROUTING_RX_DAPM_ROUTE("RX_CODEC_DMA_RX_6 Audio Mixer", "RX_CODEC_DMA_RX_6"),
 	Q6ROUTING_RX_DAPM_ROUTE("RX_CODEC_DMA_RX_7 Audio Mixer", "RX_CODEC_DMA_RX_7"),
+	Q6ROUTING_RX_DAPM_ROUTE("USB_RX Audio Mixer", "USB_RX"),
 	Q6ROUTING_TX_DAPM_ROUTE("MultiMedia1 Mixer"),
 	Q6ROUTING_TX_DAPM_ROUTE("MultiMedia2 Mixer"),
 	Q6ROUTING_TX_DAPM_ROUTE("MultiMedia3 Mixer"),
@@ -1161,7 +1168,7 @@ static struct platform_driver q6pcm_routing_platform_driver = {
 		.of_match_table = of_match_ptr(q6pcm_routing_device_id),
 	},
 	.probe = q6pcm_routing_probe,
-	.remove_new = q6pcm_routing_remove,
+	.remove = q6pcm_routing_remove,
 };
 module_platform_driver(q6pcm_routing_platform_driver);
 
diff --git a/sound/soc/qcom/qdsp6/q6usb.c b/sound/soc/qcom/qdsp6/q6usb.c
new file mode 100644
index 000000000000..ebe0c2425927
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6usb.c
@@ -0,0 +1,421 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Copyright (c) 2022-2025 Qualcomm Innovation Center, Inc. All rights reserved.
+ */
+
+#include <linux/auxiliary_bus.h>
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+#include <linux/dma-map-ops.h>
+#include <linux/err.h>
+#include <linux/init.h>
+#include <linux/iommu.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+
+#include <sound/asound.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/q6usboffload.h>
+#include <sound/soc.h>
+#include <sound/soc-usb.h>
+
+#include <dt-bindings/sound/qcom,q6afe.h>
+
+#include "q6afe.h"
+#include "q6dsp-lpass-ports.h"
+
+#define Q6_USB_SID_MASK	0xF
+
+struct q6usb_port_data {
+	struct auxiliary_device uauxdev;
+	struct q6afe_usb_cfg usb_cfg;
+	struct snd_soc_usb *usb;
+	struct snd_soc_jack *hs_jack;
+	struct q6usb_offload priv;
+
+	/* Protects against operations between SOC USB and ASoC */
+	struct mutex mutex;
+	struct list_head devices;
+};
+
+static const struct snd_soc_dapm_widget q6usb_dai_widgets[] = {
+	SND_SOC_DAPM_HP("USB_RX_BE", NULL),
+};
+
+static const struct snd_soc_dapm_route q6usb_dapm_routes[] = {
+	{"USB Playback", NULL, "USB_RX_BE"},
+};
+
+static int q6usb_hw_params(struct snd_pcm_substream *substream,
+			   struct snd_pcm_hw_params *params,
+			   struct snd_soc_dai *dai)
+{
+	struct q6usb_port_data *data = dev_get_drvdata(dai->dev);
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
+	int direction = substream->stream;
+	struct q6afe_port *q6usb_afe;
+	struct snd_soc_usb_device *sdev;
+	int ret = -EINVAL;
+
+	mutex_lock(&data->mutex);
+
+	/* No active chip index */
+	if (list_empty(&data->devices))
+		goto out;
+
+	sdev = list_last_entry(&data->devices, struct snd_soc_usb_device, list);
+
+	ret = snd_soc_usb_find_supported_format(sdev->chip_idx, params, direction);
+	if (ret < 0)
+		goto out;
+
+	q6usb_afe = q6afe_port_get_from_id(cpu_dai->dev, USB_RX);
+	if (IS_ERR(q6usb_afe)) {
+		ret = PTR_ERR(q6usb_afe);
+		goto out;
+	}
+
+	/* Notify audio DSP about the devices being offloaded */
+	ret = afe_port_send_usb_dev_param(q6usb_afe, sdev->card_idx,
+					  sdev->ppcm_idx[sdev->num_playback - 1]);
+
+out:
+	mutex_unlock(&data->mutex);
+
+	return ret;
+}
+
+static const struct snd_soc_dai_ops q6usb_ops = {
+	.hw_params = q6usb_hw_params,
+};
+
+static struct snd_soc_dai_driver q6usb_be_dais[] = {
+	{
+		.playback = {
+			.stream_name = "USB BE RX",
+			.rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+				SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |
+				SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+				SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+				SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
+				SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE |
+				SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |
+				SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE,
+			.channels_min = 1,
+			.channels_max = 2,
+			.rate_max =     192000,
+			.rate_min =	8000,
+		},
+		.id = USB_RX,
+		.name = "USB_RX_BE",
+		.ops = &q6usb_ops,
+	},
+};
+
+static int q6usb_audio_ports_of_xlate_dai_name(struct snd_soc_component *component,
+					       const struct of_phandle_args *args,
+					       const char **dai_name)
+{
+	int id = args->args[0];
+	int ret = -EINVAL;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(q6usb_be_dais); i++) {
+		if (q6usb_be_dais[i].id == id) {
+			*dai_name = q6usb_be_dais[i].name;
+			ret = 0;
+			break;
+		}
+	}
+
+	return ret;
+}
+
+static int q6usb_get_pcm_id_from_widget(struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_pcm_runtime *rtd;
+	struct snd_soc_dai *dai;
+
+	for_each_card_rtds(w->dapm->card, rtd) {
+		dai = snd_soc_rtd_to_cpu(rtd, 0);
+		/*
+		 * Only look for playback widget. RTD number carries the assigned
+		 * PCM index.
+		 */
+		if (dai->stream[0].widget == w)
+			return rtd->id;
+	}
+
+	return -1;
+}
+
+static int q6usb_usb_mixer_enabled(struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dapm_path *p;
+
+	/* Checks to ensure USB path is enabled/connected */
+	snd_soc_dapm_widget_for_each_sink_path(w, p)
+		if (!strcmp(p->sink->name, "USB Mixer") && p->connect)
+			return 1;
+
+	return 0;
+}
+
+static int q6usb_get_pcm_id(struct snd_soc_component *component)
+{
+	struct snd_soc_dapm_widget *w;
+	struct snd_soc_dapm_path *p;
+	int pidx;
+
+	/*
+	 * Traverse widgets to find corresponding FE widget.  The DAI links are
+	 * built like the following:
+	 *    MultiMedia* <-> MM_DL* <-> USB Mixer*
+	 */
+	for_each_card_widgets(component->card, w) {
+		if (!strncmp(w->name, "MultiMedia", 10)) {
+			/*
+			 * Look up all paths associated with the FE widget to see if
+			 * the USB BE is enabled.  The sink widget is responsible to
+			 * link with the USB mixers.
+			 */
+			snd_soc_dapm_widget_for_each_sink_path(w, p) {
+				if (q6usb_usb_mixer_enabled(p->sink)) {
+					pidx = q6usb_get_pcm_id_from_widget(w);
+					return pidx;
+				}
+			}
+		}
+	}
+
+	return -1;
+}
+
+static int q6usb_update_offload_route(struct snd_soc_component *component, int card,
+				      int pcm, int direction, enum snd_soc_usb_kctl path,
+				      long *route)
+{
+	struct q6usb_port_data *data = dev_get_drvdata(component->dev);
+	struct snd_soc_usb_device *sdev;
+	int ret = 0;
+	int idx = -1;
+
+	mutex_lock(&data->mutex);
+
+	if (list_empty(&data->devices) ||
+	    direction == SNDRV_PCM_STREAM_CAPTURE) {
+		ret = -ENODEV;
+		goto out;
+	}
+
+	sdev = list_last_entry(&data->devices, struct snd_soc_usb_device, list);
+
+	/*
+	 * Will always look for last PCM device discovered/probed as the
+	 * active offload index.
+	 */
+	if (card == sdev->card_idx &&
+	    pcm == sdev->ppcm_idx[sdev->num_playback - 1]) {
+		idx = path == SND_SOC_USB_KCTL_CARD_ROUTE ?
+				component->card->snd_card->number :
+				q6usb_get_pcm_id(component);
+	}
+
+out:
+	route[0] = idx;
+	mutex_unlock(&data->mutex);
+
+	return ret;
+}
+
+static int q6usb_alsa_connection_cb(struct snd_soc_usb *usb,
+				    struct snd_soc_usb_device *sdev, bool connected)
+{
+	struct q6usb_port_data *data;
+
+	if (!usb->component)
+		return -ENODEV;
+
+	data = dev_get_drvdata(usb->component->dev);
+
+	mutex_lock(&data->mutex);
+	if (connected) {
+		if (data->hs_jack)
+			snd_jack_report(data->hs_jack->jack, SND_JACK_USB);
+
+		/* Selects the latest USB headset plugged in for offloading */
+		list_add_tail(&sdev->list, &data->devices);
+	} else {
+		list_del(&sdev->list);
+
+		if (data->hs_jack)
+			snd_jack_report(data->hs_jack->jack, 0);
+	}
+	mutex_unlock(&data->mutex);
+
+	return 0;
+}
+
+static void q6usb_component_disable_jack(struct q6usb_port_data *data)
+{
+	/* Offload jack has already been disabled */
+	if (!data->hs_jack)
+		return;
+
+	snd_jack_report(data->hs_jack->jack, 0);
+	data->hs_jack = NULL;
+}
+
+static void q6usb_component_enable_jack(struct q6usb_port_data *data,
+					struct snd_soc_jack *jack)
+{
+	snd_jack_report(jack->jack, !list_empty(&data->devices) ? SND_JACK_USB : 0);
+	data->hs_jack = jack;
+}
+
+static int q6usb_component_set_jack(struct snd_soc_component *component,
+				    struct snd_soc_jack *jack, void *priv)
+{
+	struct q6usb_port_data *data = dev_get_drvdata(component->dev);
+
+	mutex_lock(&data->mutex);
+	if (jack)
+		q6usb_component_enable_jack(data, jack);
+	else
+		q6usb_component_disable_jack(data);
+	mutex_unlock(&data->mutex);
+
+	return 0;
+}
+
+static void q6usb_dai_aux_release(struct device *dev) {}
+
+static int q6usb_dai_add_aux_device(struct q6usb_port_data *data,
+				    struct auxiliary_device *auxdev)
+{
+	int ret;
+
+	auxdev->dev.parent = data->priv.dev;
+	auxdev->dev.release = q6usb_dai_aux_release;
+	auxdev->name = "qc-usb-audio-offload";
+
+	ret = auxiliary_device_init(auxdev);
+	if (ret)
+		return ret;
+
+	ret = auxiliary_device_add(auxdev);
+	if (ret)
+		auxiliary_device_uninit(auxdev);
+
+	return ret;
+}
+
+static int q6usb_component_probe(struct snd_soc_component *component)
+{
+	struct q6usb_port_data *data = dev_get_drvdata(component->dev);
+	struct snd_soc_usb *usb;
+	int ret;
+
+	/* Add the QC USB SND aux device */
+	ret = q6usb_dai_add_aux_device(data, &data->uauxdev);
+	if (ret < 0)
+		return ret;
+
+	usb = snd_soc_usb_allocate_port(component, &data->priv);
+	if (IS_ERR(usb))
+		return -ENOMEM;
+
+	usb->connection_status_cb = q6usb_alsa_connection_cb;
+	usb->update_offload_route_info = q6usb_update_offload_route;
+
+	snd_soc_usb_add_port(usb);
+	data->usb = usb;
+
+	return 0;
+}
+
+static void q6usb_component_remove(struct snd_soc_component *component)
+{
+	struct q6usb_port_data *data = dev_get_drvdata(component->dev);
+
+	snd_soc_usb_remove_port(data->usb);
+	auxiliary_device_delete(&data->uauxdev);
+	auxiliary_device_uninit(&data->uauxdev);
+	snd_soc_usb_free_port(data->usb);
+}
+
+static const struct snd_soc_component_driver q6usb_dai_component = {
+	.probe = q6usb_component_probe,
+	.set_jack = q6usb_component_set_jack,
+	.remove = q6usb_component_remove,
+	.name = "q6usb-dai-component",
+	.dapm_widgets = q6usb_dai_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(q6usb_dai_widgets),
+	.dapm_routes = q6usb_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(q6usb_dapm_routes),
+	.of_xlate_dai_name = q6usb_audio_ports_of_xlate_dai_name,
+};
+
+static int q6usb_dai_dev_probe(struct platform_device *pdev)
+{
+	struct device_node *node = pdev->dev.of_node;
+	struct q6usb_port_data *data;
+	struct device *dev = &pdev->dev;
+	struct of_phandle_args args;
+	int ret;
+
+	data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL);
+	if (!data)
+		return -ENOMEM;
+
+	ret = of_property_read_u16(node, "qcom,usb-audio-intr-idx",
+				   &data->priv.intr_num);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to read intr idx.\n");
+		return ret;
+	}
+
+	ret = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);
+	if (!ret)
+		data->priv.sid = args.args[0] & Q6_USB_SID_MASK;
+
+	ret = devm_mutex_init(dev, &data->mutex);
+	if (ret < 0)
+		return ret;
+
+	data->priv.domain = iommu_get_domain_for_dev(&pdev->dev);
+
+	data->priv.dev = dev;
+	INIT_LIST_HEAD(&data->devices);
+	dev_set_drvdata(dev, data);
+
+	return devm_snd_soc_register_component(dev, &q6usb_dai_component,
+					q6usb_be_dais, ARRAY_SIZE(q6usb_be_dais));
+}
+
+static const struct of_device_id q6usb_dai_device_id[] = {
+	{ .compatible = "qcom,q6usb" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, q6usb_dai_device_id);
+
+static struct platform_driver q6usb_dai_platform_driver = {
+	.driver = {
+		.name = "q6usb-dai",
+		.of_match_table = q6usb_dai_device_id,
+	},
+	.probe = q6usb_dai_dev_probe,
+	/*
+	 * Remove not required as resources are cleaned up as part of
+	 * component removal.  Others are device managed resources.
+	 */
+};
+module_platform_driver(q6usb_dai_platform_driver);
+
+MODULE_DESCRIPTION("Q6 USB backend dai driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/qcom/qdsp6/topology.c b/sound/soc/qcom/qdsp6/topology.c
index 70572c83e101..83319a928f29 100644
--- a/sound/soc/qcom/qdsp6/topology.c
+++ b/sound/soc/qcom/qdsp6/topology.c
@@ -1,6 +1,7 @@
 // SPDX-License-Identifier: GPL-2.0
 // Copyright (c) 2020, Linaro Limited
 
+#include <linux/cleanup.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 #include <sound/pcm.h>
@@ -730,6 +731,29 @@ static int audioreach_widget_i2s_module_load(struct audioreach_module *mod,
 	return 0;
 }
 
+static int audioreach_widget_dp_module_load(struct audioreach_module *mod,
+					struct snd_soc_tplg_vendor_array *mod_array)
+{
+	struct snd_soc_tplg_vendor_value_elem *mod_elem;
+	int tkn_count = 0;
+
+	mod_elem = mod_array->value;
+
+	while (tkn_count <= (le32_to_cpu(mod_array->num_elems) - 1)) {
+		switch (le32_to_cpu(mod_elem->token)) {
+		case AR_TKN_U32_MODULE_FMT_DATA:
+			mod->data_format = le32_to_cpu(mod_elem->value);
+			break;
+		default:
+			break;
+		}
+		tkn_count++;
+		mod_elem++;
+	}
+
+	return 0;
+}
+
 static int audioreach_widget_load_buffer(struct snd_soc_component *component,
 					 int index, struct snd_soc_dapm_widget *w,
 					 struct snd_soc_tplg_dapm_widget *tplg_w)
@@ -760,6 +784,9 @@ static int audioreach_widget_load_buffer(struct snd_soc_component *component,
 	case MODULE_ID_I2S_SOURCE:
 		audioreach_widget_i2s_module_load(mod, mod_array);
 		break;
+	case MODULE_ID_DISPLAY_PORT_SINK:
+		audioreach_widget_dp_module_load(mod, mod_array);
+		break;
 	default:
 		return -EINVAL;
 	}
@@ -1240,7 +1267,7 @@ static const struct snd_soc_tplg_kcontrol_ops audioreach_io_ops[] = {
 		audioreach_put_vol_ctrl_audio_mixer, snd_soc_info_volsw},
 };
 
-static struct snd_soc_tplg_ops audioreach_tplg_ops  = {
+static const struct snd_soc_tplg_ops audioreach_tplg_ops = {
 	.io_ops = audioreach_io_ops,
 	.io_ops_count = ARRAY_SIZE(audioreach_io_ops),
 
@@ -1262,18 +1289,19 @@ int audioreach_tplg_init(struct snd_soc_component *component)
 	struct snd_soc_card *card = component->card;
 	struct device *dev = component->dev;
 	const struct firmware *fw;
-	char *tplg_fw_name;
 	int ret;
 
 	/* Inline with Qualcomm UCM configs and linux-firmware path */
-	tplg_fw_name = kasprintf(GFP_KERNEL, "qcom/%s/%s-tplg.bin", card->driver_name, card->name);
+	char *tplg_fw_name __free(kfree) = kasprintf(GFP_KERNEL, "qcom/%s/%s-tplg.bin",
+						     card->driver_name,
+						     card->name);
 	if (!tplg_fw_name)
 		return -ENOMEM;
 
 	ret = request_firmware(&fw, tplg_fw_name, dev);
 	if (ret < 0) {
 		dev_err(dev, "tplg firmware loading %s failed %d\n", tplg_fw_name, ret);
-		goto err;
+		return ret;
 	}
 
 	ret = snd_soc_tplg_component_load(component, &audioreach_tplg_ops, fw);
@@ -1283,8 +1311,6 @@ int audioreach_tplg_init(struct snd_soc_component *component)
 	}
 
 	release_firmware(fw);
-err:
-	kfree(tplg_fw_name);
 
 	return ret;
 }