diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h
index 8d9016e3557c..3464726c408a 100644
--- a/include/sound/apr_audio-v2.h
+++ b/include/sound/apr_audio-v2.h
@@ -2738,6 +2738,333 @@ struct afe_param_id_set_topology_cfg {
 	u32		topology_id;
 } __packed;
 
+
+/*
+ * Generic encoder module ID.
+ * This module supports the following parameter IDs:
+ * #AVS_ENCODER_PARAM_ID_ENC_FMT_ID (cannot be set run time)
+ * #AVS_ENCODER_PARAM_ID_ENC_CFG_BLK (may be set run time)
+ * #AVS_ENCODER_PARAM_ID_ENC_BITRATE (may be set run time)
+ * #AVS_ENCODER_PARAM_ID_PACKETIZER_ID (cannot be set run time)
+ * Opcode - AVS_MODULE_ID_ENCODER
+ * AFE Command AFE_PORT_CMD_SET_PARAM_V2 supports this module ID.
+ */
+#define AFE_MODULE_ID_ENCODER        0x00013229
+
+/* Macro for defining the packetizer ID: COP. */
+#define AFE_MODULE_ID_PACKETIZER_COP 0x0001322A
+
+/*
+ * Packetizer type parameter for the #AVS_MODULE_ID_ENCODER module.
+ * This parameter cannot be set runtime.
+ */
+#define AFE_ENCODER_PARAM_ID_PACKETIZER_ID 0x0001322E
+
+/*
+ * Encoder config block  parameter for the #AVS_MODULE_ID_ENCODER module.
+ * This parameter may be set runtime.
+ */
+#define AFE_ENCODER_PARAM_ID_ENC_CFG_BLK 0x0001322C
+
+/*
+ * Encoder format ID parameter for the #AVS_MODULE_ID_ENCODER module.
+ * This parameter cannot be set runtime.
+ */
+#define AFE_ENCODER_PARAM_ID_ENC_FMT_ID         0x0001322B
+
+/*
+ * Data format to send compressed data
+ * is transmitted/received over Slimbus lines.
+ */
+#define AFE_SB_DATA_FORMAT_GENERIC_COMPRESSED    0x3
+
+/*
+ * ID for AFE port module. This will be used to define port properties.
+ * This module supports following parameter IDs:
+ * #AFE_PARAM_ID_PORT_MEDIA_TYPE
+ * To configure the port property, the client must use the
+ * #AFE_PORT_CMD_SET_PARAM_V2 command,
+ * and fill the module ID with the respective parameter IDs as listed above.
+ * @apr_hdr_fields
+ * Opcode -- AFE_MODULE_PORT
+ */
+#define AFE_MODULE_PORT                          0x000102a6
+
+/*
+ * ID of the parameter used by #AFE_MODULE_PORT to set the port media type.
+ * parameter ID is currently supported using#AFE_PORT_CMD_SET_PARAM_V2 command.
+ */
+#define AFE_PARAM_ID_PORT_MEDIA_TYPE              0x000102a7
+
+/*
+ * Macros for defining the "data_format" field in the
+ * #AFE_PARAM_ID_PORT_MEDIA_TYPE
+ */
+#define AFE_PORT_DATA_FORMAT_PCM                  0x0
+#define AFE_PORT_DATA_FORMAT_GENERIC_COMPRESSED   0x1
+
+/*
+ * Macro for defining the "minor_version" field in the
+ * #AFE_PARAM_ID_PORT_MEDIA_TYPE
+ */
+#define AFE_API_VERSION_PORT_MEDIA_TYPE           0x1
+
+#define ASM_MEDIA_FMT_NONE                        0x0
+
+/*
+ * Media format ID for SBC encode configuration.
+ * @par SBC encode configuration (asm_sbc_enc_cfg_t)
+ * @table{weak__asm__sbc__enc__cfg__t}
+ */
+#define ASM_MEDIA_FMT_SBC                         0x00010BF2
+
+/* SBC channel Mono mode.*/
+#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_MONO                     1
+
+/* SBC channel Stereo mode. */
+#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_STEREO                   2
+
+/* SBC channel Dual Mono mode. */
+#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO                8
+
+/* SBC channel Joint Stereo mode. */
+#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO             9
+
+/* SBC bit allocation method = loudness. */
+#define ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS            0
+
+/* SBC bit allocation method = SNR. */
+#define ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR                 1
+
+
+/*
+ * Payload of the SBC encoder configuration parameters in the
+ * #ASM_MEDIA_FMT_SBC media format.
+ */
+struct asm_sbc_enc_cfg_t {
+	/*
+	 * Number of subbands.
+	 * @values 4, 8
+	 */
+	uint32_t    num_subbands;
+
+	/*
+	 * Size of the encoded block in samples.
+	 * @values 4, 8, 12, 16
+	 */
+	uint32_t    blk_len;
+
+	/*
+	 * Mode used to allocate bits between channels.
+	 * @values
+	 * 0 (Native mode)
+	 * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_MONO
+	 * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_STEREO
+	 * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO
+	 * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO
+	 * Native mode indicates that encoding must be performed with the number
+	 * of channels at the input.
+	 * If postprocessing outputs one-channel data, Mono mode is used. If
+	 * postprocessing outputs two-channel data, Stereo mode is used.
+	 * The number of channels must not change during encoding.
+	 */
+	uint32_t    channel_mode;
+
+	/*
+	 * Encoder bit allocation method.
+	 * @values
+	 * #ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS
+	 * #ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR @tablebulletend
+	 */
+	uint32_t    alloc_method;
+
+	/*
+	 * Number of encoded bits per second.
+	 * @values
+	 * Mono channel -- Maximum of 320 kbps
+	 * Stereo channel -- Maximum of 512 kbps @tablebulletend
+	 */
+	uint32_t    bit_rate;
+
+	/*
+	 * Number of samples per second.
+	 * @values 0 (Native mode), 16000, 32000, 44100, 48000 Hz
+	 * Native mode indicates that encoding must be performed with the
+	 * sampling rate at the input.
+	 * The sampling rate must not change during encoding.
+	 */
+	uint32_t    sample_rate;
+};
+
+#define ASM_MEDIA_FMT_AAC_AOT_LC            2
+#define ASM_MEDIA_FMT_AAC_AOT_SBR           5
+#define ASM_MEDIA_FMT_AAC_AOT_PS            29
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS  0
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW   3
+
+struct asm_aac_enc_cfg_v2_t {
+
+	/* Encoding rate in bits per second.*/
+	uint32_t     bit_rate;
+
+	/*
+	 * Encoding mode.
+	 * Supported values:
+	 * #ASM_MEDIA_FMT_AAC_AOT_LC
+	 * #ASM_MEDIA_FMT_AAC_AOT_SBR
+	 * #ASM_MEDIA_FMT_AAC_AOT_PS
+	 */
+	uint32_t     enc_mode;
+
+	/*
+	 * AAC format flag.
+	 * Supported values:
+	 * #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
+	 * #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
+	 */
+	uint16_t     aac_fmt_flag;
+
+	/*
+	 * Number of channels to encode.
+	 * Supported values:
+	 * 0 - Native mode
+	 * 1 - Mono
+	 * 2 - Stereo
+	 * Other values are not supported.
+	 * @note1hang The eAAC+ encoder mode supports only stereo.
+	 * Native mode indicates that encoding must be performed with the
+	 * number of channels at the input.
+	 * The number of channels must not change during encoding.
+	 */
+	uint32_t     channel_cfg;
+
+	/*
+	 * Number of samples per second.
+	 * Supported values: - 0 -- Native mode - For other values,
+	 * Native mode indicates that encoding must be performed with the
+	 * sampling rate at the input.
+	 * The sampling rate must not change during encoding.
+	 */
+	uint32_t     sample_rate;
+} __packed;
+
+/* FMT ID for apt-X Classic */
+#define ASM_MEDIA_FMT_APTX 0x000131ff
+
+/* FMT ID for apt-X HD */
+#define ASM_MEDIA_FMT_APTX_HD 0x00013200
+
+#define PCM_CHANNEL_L         1
+#define PCM_CHANNEL_R         2
+#define PCM_CHANNEL_C         3
+
+struct asm_custom_enc_cfg_aptx_t {
+	uint32_t    sample_rate;
+	/* Mono or stereo */
+	uint16_t    num_channels;
+	uint16_t    reserved;
+	/* num_ch == 1, then PCM_CHANNEL_C,
+	 * num_ch == 2, then {PCM_CHANNEL_L, PCM_CHANNEL_R}
+	 */
+	uint8_t     channel_mapping[8];
+	uint32_t    custom_size;
+} __packed;
+
+struct afe_enc_fmt_id_param_t {
+	/*
+	 * Supported values:
+	 *  #ASM_MEDIA_FMT_SBC
+	 *  #ASM_MEDIA_FMT_AAC_V2
+	 * Any OpenDSP supported values
+	 */
+	uint32_t    fmt_id;
+} __packed;
+
+struct afe_port_media_type_t {
+	/*
+	 * Minor version
+	 * @values #AFE_API_VERSION_PORT_MEDIA_TYPE.
+	 */
+	uint32_t    minor_version;
+
+	/*
+	 * Sampling rate of the port.
+	 * @values
+	 * #AFE_PORT_SAMPLE_RATE_8K
+	 * #AFE_PORT_SAMPLE_RATE_11_025K
+	 * #AFE_PORT_SAMPLE_RATE_12K
+	 * #AFE_PORT_SAMPLE_RATE_16K
+	 * #AFE_PORT_SAMPLE_RATE_22_05K
+	 * #AFE_PORT_SAMPLE_RATE_24K
+	 * #AFE_PORT_SAMPLE_RATE_32K
+	 * #AFE_PORT_SAMPLE_RATE_44_1K
+	 * #AFE_PORT_SAMPLE_RATE_48K
+	 * #AFE_PORT_SAMPLE_RATE_88_2K
+	 * #AFE_PORT_SAMPLE_RATE_96K
+	 * #AFE_PORT_SAMPLE_RATE_176_4K
+	 * #AFE_PORT_SAMPLE_RATE_192K
+	 * #AFE_PORT_SAMPLE_RATE_352_8K
+	 * #AFE_PORT_SAMPLE_RATE_384K
+	 */
+	uint32_t    sample_rate;
+
+	/*
+	 * Bit width of the sample.
+	 * @values 16, 24
+	 */
+	uint16_t    bit_width;
+
+	/*
+	 * Number of channels.
+	 * @values 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
+	 */
+	uint16_t    num_channels;
+
+	/*
+	 * Data format supported by this port.
+	 * If the port media type and device media type are different,
+	 * it signifies a encoding/decoding use case
+	 * @values
+	 * #AFE_PORT_DATA_FORMAT_PCM
+	 * #AFE_PORT_DATA_FORMAT_GENERIC_COMPRESSED
+	 */
+	uint16_t   data_format;
+
+	/*This field must be set to zero.*/
+	uint16_t   reserved;
+} __packed;
+
+union afe_enc_config_data {
+	struct asm_sbc_enc_cfg_t sbc_config;
+	struct asm_aac_enc_cfg_v2_t aac_config;
+	struct asm_custom_enc_cfg_aptx_t  aptx_config;
+};
+
+struct afe_enc_config {
+	u32 format;
+	union afe_enc_config_data data;
+};
+
+struct afe_enc_cfg_blk_param_t {
+	uint32_t enc_cfg_blk_size;
+	/*
+	 *Size of the encoder configuration block that follows this member
+	 */
+	union afe_enc_config_data enc_blk_config;
+};
+
+/*
+ * Payload of the AVS_ENCODER_PARAM_ID_PACKETIZER_ID parameter.
+ */
+struct avs_enc_packetizer_id_param_t {
+	/*
+	 * Supported values:
+	 * #AVS_MODULE_ID_PACKETIZER_COP
+	 * Any OpenDSP supported values
+	 */
+	uint32_t enc_packetizer_id;
+};
+
 union afe_port_config {
 	struct afe_param_id_pcm_cfg               pcm;
 	struct afe_param_id_i2s_cfg               i2s;
@@ -2751,6 +3078,10 @@ union afe_port_config {
 	struct afe_param_id_set_topology_cfg      topology;
 	struct afe_param_id_tdm_cfg               tdm;
 	struct afe_param_id_usb_audio_cfg         usb_audio;
+	struct afe_enc_fmt_id_param_t             enc_fmt;
+	struct afe_port_media_type_t              media_type;
+	struct afe_enc_cfg_blk_param_t            enc_blk_param;
+	struct avs_enc_packetizer_id_param_t      enc_pkt_id_param;
 } __packed;
 
 struct afe_audioif_config_command_no_payload {
diff --git a/include/sound/q6afe-v2.h b/include/sound/q6afe-v2.h
index 5cd65357dd95..6be903a4c8d0 100644
--- a/include/sound/q6afe-v2.h
+++ b/include/sound/q6afe-v2.h
@@ -274,6 +274,9 @@ int afe_rt_proxy_port_read(phys_addr_t buf_addr_p,
 void afe_set_cal_mode(u16 port_id, enum afe_cal_mode afe_cal_mode);
 int afe_port_start(u16 port_id, union afe_port_config *afe_config,
 	u32 rate);
+int afe_port_start_v2(u16 port_id, union afe_port_config *afe_config,
+		      u32 rate, u16 afe_in_channels,
+		      struct afe_enc_config *enc_config);
 int afe_spk_prot_feed_back_cfg(int src_port, int dst_port,
 	int l_ch, int r_ch, u32 enable);
 int afe_spk_prot_get_calib_data(struct afe_spkr_prot_get_vi_calib *calib);
diff --git a/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c
index 7684d27d60a0..c439c5cf2de5 100644
--- a/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c
+++ b/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c
@@ -37,6 +37,14 @@
 #define CHANNEL_STATUS_MASK 0x4
 #define AFE_API_VERSION_CLOCK_SET 1
 
+enum {
+	ENC_FMT_NONE,
+	ENC_FMT_SBC = ASM_MEDIA_FMT_SBC,
+	ENC_FMT_AAC_V2 = ASM_MEDIA_FMT_AAC_V2,
+	ENC_FMT_APTX = ASM_MEDIA_FMT_APTX,
+	ENC_FMT_APTX_HD = ASM_MEDIA_FMT_APTX_HD,
+};
+
 static const struct afe_clk_set lpass_clk_set_default = {
 	AFE_API_VERSION_CLOCK_SET,
 	Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT,
@@ -157,6 +165,8 @@ struct msm_dai_q6_dai_data {
 	u32 channels;
 	u32 bitwidth;
 	u32 cal_mode;
+	u32 afe_in_channels;
+	struct afe_enc_config enc_config;
 	union afe_port_config port_config;
 };
 
@@ -1330,9 +1340,20 @@ static int msm_dai_q6_prepare(struct snd_pcm_substream *substream,
 	int rc = 0;
 
 	if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
-		rc = afe_port_start(dai->id, &dai_data->port_config,
-					dai_data->rate);
-
+		if (dai_data->enc_config.format != ENC_FMT_NONE) {
+			pr_debug("%s: calling AFE_PORT_START_V2 with enc_format: %d\n",
+				 __func__, dai_data->enc_config.format);
+			rc = afe_port_start_v2(dai->id, &dai_data->port_config,
+					       dai_data->rate,
+					       dai_data->afe_in_channels,
+					       &dai_data->enc_config);
+			if (rc < 0)
+				pr_err("%s: afe_port_start_v2 failed error: %d\n",
+					__func__, rc);
+		} else {
+			rc = afe_port_start(dai->id, &dai_data->port_config,
+						dai_data->rate);
+		}
 		if (IS_ERR_VALUE(rc))
 			dev_err(dai->dev, "fail to open AFE port 0x%x\n",
 				dai->id);
@@ -1941,6 +1962,151 @@ static int msm_dai_q6_usb_audio_cfg_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
+static int  msm_dai_q6_afe_enc_cfg_info(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+	uinfo->count = sizeof(struct afe_enc_config);
+
+	return 0;
+}
+
+static int msm_dai_q6_afe_enc_cfg_get(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	int ret = 0;
+	struct msm_dai_q6_dai_data *dai_data = kcontrol->private_data;
+
+	if (dai_data) {
+		int format_size = sizeof(dai_data->enc_config.format);
+
+		pr_debug("%s:encoder config for %d format\n",
+			 __func__, dai_data->enc_config.format);
+		memcpy(ucontrol->value.bytes.data,
+			&dai_data->enc_config.format,
+			format_size);
+		switch (dai_data->enc_config.format) {
+		case ENC_FMT_SBC:
+			memcpy(ucontrol->value.bytes.data + format_size,
+				&dai_data->enc_config.data,
+				sizeof(struct asm_sbc_enc_cfg_t));
+			break;
+		case ENC_FMT_AAC_V2:
+			memcpy(ucontrol->value.bytes.data + format_size,
+				&dai_data->enc_config.data,
+				sizeof(struct asm_aac_enc_cfg_v2_t));
+			break;
+		case ENC_FMT_APTX:
+		case ENC_FMT_APTX_HD:
+			memcpy(ucontrol->value.bytes.data + format_size,
+				&dai_data->enc_config.data,
+				sizeof(struct asm_aac_enc_cfg_v2_t));
+			break;
+		default:
+			pr_debug("%s: unknown format = %d\n",
+				 __func__, dai_data->enc_config.format);
+			ret = -EINVAL;
+			break;
+		}
+	}
+
+	return ret;
+}
+
+static int msm_dai_q6_afe_enc_cfg_put(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	int ret = 0;
+	struct msm_dai_q6_dai_data *dai_data = kcontrol->private_data;
+
+	if (dai_data) {
+		int format_size = sizeof(dai_data->enc_config.format);
+
+		memset(&dai_data->enc_config, 0x0,
+			sizeof(struct afe_enc_config));
+		memcpy(&dai_data->enc_config.format,
+			ucontrol->value.bytes.data,
+			format_size);
+		pr_debug("%s: Received encoder config for %d format\n",
+			 __func__, dai_data->enc_config.format);
+		switch (dai_data->enc_config.format) {
+		case ENC_FMT_SBC:
+			memcpy(&dai_data->enc_config.data,
+				ucontrol->value.bytes.data + format_size,
+				sizeof(struct asm_sbc_enc_cfg_t));
+			break;
+		case ENC_FMT_AAC_V2:
+			memcpy(&dai_data->enc_config.data,
+				ucontrol->value.bytes.data + format_size,
+				sizeof(struct asm_aac_enc_cfg_v2_t));
+			break;
+		case ENC_FMT_APTX:
+		case ENC_FMT_APTX_HD:
+			memcpy(&dai_data->enc_config.data,
+				ucontrol->value.bytes.data + format_size,
+				sizeof(struct asm_custom_enc_cfg_aptx_t));
+			break;
+		default:
+			pr_debug("%s: Ignore enc config for unknown format = %d\n",
+				 __func__, dai_data->enc_config.format);
+			ret = -EINVAL;
+			break;
+		}
+	} else
+		ret = -EINVAL;
+
+	return ret;
+}
+
+static const char *const afe_input_chs_text[] = {"Zero", "One", "Two"};
+
+static const struct soc_enum afe_input_chs_enum[] = {
+	SOC_ENUM_SINGLE_EXT(3, afe_input_chs_text),
+};
+
+static int msm_dai_q6_afe_input_channel_get(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct msm_dai_q6_dai_data *dai_data = kcontrol->private_data;
+
+	if (dai_data) {
+		ucontrol->value.integer.value[0] = dai_data->afe_in_channels;
+		pr_debug("%s:afe input channel = %d\n",
+			  __func__, dai_data->afe_in_channels);
+	}
+
+	return 0;
+}
+
+static int msm_dai_q6_afe_input_channel_put(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct msm_dai_q6_dai_data *dai_data = kcontrol->private_data;
+
+	if (dai_data) {
+		dai_data->afe_in_channels = ucontrol->value.integer.value[0];
+		pr_debug("%s: updating afe input channel : %d\n",
+			__func__, dai_data->afe_in_channels);
+	}
+
+	return 0;
+}
+
+static const struct snd_kcontrol_new afe_enc_config_controls[] = {
+	{
+		.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+			   SNDRV_CTL_ELEM_ACCESS_INACTIVE),
+		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.name = "SLIM_7_RX Encoder Config",
+		.info = msm_dai_q6_afe_enc_cfg_info,
+		.get = msm_dai_q6_afe_enc_cfg_get,
+		.put = msm_dai_q6_afe_enc_cfg_put,
+	},
+	SOC_ENUM_EXT("AFE Input Channels", afe_input_chs_enum[0],
+		     msm_dai_q6_afe_input_channel_get,
+		     msm_dai_q6_afe_input_channel_put),
+};
+
 static const char * const afe_cal_mode_text[] = {
 	"CAL_MODE_DEFAULT", "CAL_MODE_NONE"
 };
@@ -2020,6 +2186,17 @@ static int msm_dai_q6_dai_probe(struct snd_soc_dai *dai)
 				 snd_ctl_new1(&sb_config_controls[1],
 				 dai_data));
 		break;
+	case SLIMBUS_7_RX:
+		rc = snd_ctl_add(dai->component->card->snd_card,
+				 snd_ctl_new1(&afe_enc_config_controls[0],
+				 dai_data));
+		rc = snd_ctl_add(dai->component->card->snd_card,
+				 snd_ctl_new1(&afe_enc_config_controls[1],
+				 dai_data));
+		rc = snd_ctl_add(dai->component->card->snd_card,
+				 snd_ctl_new1(&afe_enc_config_controls[2],
+				 dai_data));
+		break;
 	case RT_PROXY_DAI_001_RX:
 		rc = snd_ctl_add(dai->component->card->snd_card,
 				 snd_ctl_new1(&rt_proxy_config_controls[0],
diff --git a/sound/soc/msm/qdsp6v2/q6afe.c b/sound/soc/msm/qdsp6v2/q6afe.c
index b76cb7f4b210..d8efd587c3da 100644
--- a/sound/soc/msm/qdsp6v2/q6afe.c
+++ b/sound/soc/msm/qdsp6v2/q6afe.c
@@ -2632,8 +2632,113 @@ exit:
 	return ret;
 }
 
-int afe_port_start(u16 port_id, union afe_port_config *afe_config,
-	u32 rate) /* This function is no blocking */
+static int q6afe_send_enc_config(u16 port_id,
+			union afe_enc_config_data *cfg, u32 format,
+			union afe_port_config afe_config, u16 afe_in_channels)
+{
+	struct afe_audioif_config_command config;
+	int index;
+	int ret;
+	int payload_size = sizeof(config) - sizeof(struct apr_hdr) -
+				sizeof(config.param) - sizeof(config.port);
+
+	pr_debug("%s:update DSP for enc format = %d\n", __func__, format);
+	if (format != ASM_MEDIA_FMT_SBC && format != ASM_MEDIA_FMT_AAC_V2 &&
+	    format != ASM_MEDIA_FMT_APTX && format != ASM_MEDIA_FMT_APTX_HD) {
+		pr_err("%s:Unsuppported format Ignore AFE config\n", __func__);
+		return 0;
+	}
+	memset(&config, 0, sizeof(config));
+	index = q6audio_get_port_index(port_id);
+	config.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+				APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+	config.hdr.pkt_size = sizeof(config);
+	config.hdr.src_port = 0;
+	config.hdr.dest_port = 0;
+	config.hdr.token = index;
+
+	config.hdr.opcode = AFE_PORT_CMD_SET_PARAM_V2;
+	config.param.port_id = q6audio_get_port_id(port_id);
+	config.param.payload_size = payload_size + sizeof(config.port.enc_fmt);
+	config.param.payload_address_lsw = 0x00;
+	config.param.payload_address_msw = 0x00;
+	config.param.mem_map_handle = 0x00;
+	config.pdata.module_id = AFE_MODULE_ID_ENCODER;
+	config.pdata.param_id = AFE_ENCODER_PARAM_ID_ENC_FMT_ID;
+	config.pdata.param_size = sizeof(config.port.enc_fmt);
+	config.port.enc_fmt.fmt_id = format;
+	pr_debug("%s:sending AFE_ENCODER_PARAM_ID_ENC_FMT_ID payload: %d\n",
+			__func__, config.param.payload_size);
+	ret = afe_apr_send_pkt(&config, &this_afe.wait[index]);
+	if (ret) {
+		pr_err("%s:unable to send AFE_ENCODER_PARAM_ID_ENC_FMT_ID",
+				__func__);
+		goto exit;
+	}
+
+	config.param.payload_size = payload_size
+					+ sizeof(config.port.enc_blk_param);
+	pr_debug("%s:send AFE_ENCODER_PARAM_ID_ENC_CFG_BLK to DSP payload:%d\n",
+				__func__, config.param.payload_size);
+	config.pdata.param_id = AFE_ENCODER_PARAM_ID_ENC_CFG_BLK;
+	config.pdata.param_size = sizeof(config.port.enc_blk_param);
+	config.port.enc_blk_param.enc_cfg_blk_size =
+			sizeof(config.port.enc_blk_param.enc_blk_config);
+	config.port.enc_blk_param.enc_blk_config = *cfg;
+	ret = afe_apr_send_pkt(&config, &this_afe.wait[index]);
+	if (ret) {
+		pr_err("%s: AFE_ENCODER_PARAM_ID_ENC_CFG_BLK for port 0x%x failed %d\n",
+			__func__, port_id, ret);
+		goto exit;
+	}
+
+	config.param.payload_size =
+			payload_size + sizeof(config.port.enc_pkt_id_param);
+	pr_debug("%s:sending AFE_ENCODER_PARAM_ID_PACKETIZER to DSP payload = %d",
+					__func__, config.param.payload_size);
+	config.pdata.param_id = AFE_ENCODER_PARAM_ID_PACKETIZER_ID;
+	config.pdata.param_size = sizeof(config.port.enc_pkt_id_param);
+	config.port.enc_pkt_id_param.enc_packetizer_id =
+					AFE_MODULE_ID_PACKETIZER_COP;
+	ret = afe_apr_send_pkt(&config, &this_afe.wait[index]);
+	if (ret) {
+		pr_err("%s: AFE_ENCODER_PARAM_ID_PACKETIZER for port 0x%x failed %d\n",
+			__func__, port_id, ret);
+		goto exit;
+	}
+
+	config.param.payload_size =
+			payload_size + sizeof(config.port.media_type);
+	config.pdata.param_size = sizeof(config.port.media_type);
+
+	pr_debug("%s:Sending AFE_API_VERSION_PORT_MEDIA_TYPE to DSP", __func__);
+	config.pdata.module_id = AFE_MODULE_PORT;
+	config.pdata.param_id = AFE_PARAM_ID_PORT_MEDIA_TYPE;
+	config.port.media_type.minor_version = AFE_API_VERSION_PORT_MEDIA_TYPE;
+	config.port.media_type.sample_rate = afe_config.slim_sch.sample_rate;
+	config.port.media_type.bit_width = afe_config.slim_sch.bit_width;
+	if (afe_in_channels != 0)
+		config.port.media_type.num_channels = afe_in_channels;
+	else
+		config.port.media_type.num_channels =
+					afe_config.slim_sch.num_channels;
+	config.port.media_type.data_format = AFE_PORT_DATA_FORMAT_PCM;
+	config.port.media_type.reserved = 0;
+
+	ret = afe_apr_send_pkt(&config, &this_afe.wait[index]);
+	if (ret) {
+		pr_err("%s: AFE_API_VERSION_PORT_MEDIA_TYPE for port 0x%x failed %d\n",
+			__func__, port_id, ret);
+		goto exit;
+	}
+
+exit:
+	return ret;
+}
+
+static int __afe_port_start(u16 port_id, union afe_port_config *afe_config,
+			   u32 rate,  u16 afe_in_channels,
+			   union afe_enc_config_data *cfg, u32 enc_format)
 {
 	struct afe_audioif_config_command config;
 	int ret = 0;
@@ -2850,7 +2955,11 @@ int afe_port_start(u16 port_id, union afe_port_config *afe_config,
 	config.pdata.param_size = sizeof(config.port);
 
 	config.port = *afe_config;
-
+	if ((enc_format != ASM_MEDIA_FMT_NONE) &&
+	    (cfg_type == AFE_PARAM_ID_SLIMBUS_CONFIG)) {
+		config.port.slim_sch.data_format =
+				AFE_SB_DATA_FORMAT_GENERIC_COMPRESSED;
+	}
 	ret = afe_apr_send_pkt(&config, &this_afe.wait[index]);
 	if (ret) {
 		pr_err("%s: AFE enable for port 0x%x failed %d\n",
@@ -2858,6 +2967,19 @@ int afe_port_start(u16 port_id, union afe_port_config *afe_config,
 		goto fail_cmd;
 	}
 
+	if ((enc_format != ASM_MEDIA_FMT_NONE) &&
+	    (cfg_type == AFE_PARAM_ID_SLIMBUS_CONFIG)) {
+		pr_debug("%s: Found AFE encoder support for SLIMBUS enc_format = %d\n",
+					__func__, enc_format);
+		ret = q6afe_send_enc_config(port_id, cfg, enc_format,
+					*afe_config, afe_in_channels);
+		if (ret) {
+			pr_err("%s: AFE encoder config for port 0x%x failed %d\n",
+				__func__, port_id, ret);
+			goto fail_cmd;
+		}
+	}
+
 	port_index = afe_get_port_index(port_id);
 	if ((port_index >= 0) && (port_index < AFE_MAX_PORTS)) {
 		this_afe.afe_sample_rates[port_index] = rate;
@@ -2890,6 +3012,47 @@ fail_cmd:
 	return ret;
 }
 
+/**
+ * afe_port_start - to configure AFE session with
+ * specified port configuration
+ *
+ * @port_id: AFE port id number
+ * @afe_config: port configutation
+ * @rate: sampling rate of port
+ *
+ * Returns 0 on success or error value on port start failure.
+ */
+int afe_port_start(u16 port_id, union afe_port_config *afe_config,
+		   u32 rate)
+{
+	return __afe_port_start(port_id, afe_config, rate,
+				0, NULL, ASM_MEDIA_FMT_NONE);
+}
+EXPORT_SYMBOL(afe_port_start);
+
+/**
+ * afe_port_start_v2 - to configure AFE session with
+ * specified port configuration and encoder params
+ *
+ * @port_id: AFE port id number
+ * @afe_config: port configutation
+ * @rate: sampling rate of port
+ * @cfg: AFE encoder configuration information to setup encoder
+ * @afe_in_channels: AFE input channel configuration, this needs
+ *  update only if input channel is differ from AFE output
+ *
+ * Returns 0 on success or error value on port start failure.
+ */
+int afe_port_start_v2(u16 port_id, union afe_port_config *afe_config,
+		      u32 rate, u16 afe_in_channels,
+		      struct afe_enc_config *enc_cfg)
+{
+	return __afe_port_start(port_id, afe_config, rate,
+				afe_in_channels, &enc_cfg->data,
+				enc_cfg->format);
+}
+EXPORT_SYMBOL(afe_port_start_v2);
+
 int afe_get_port_index(u16 port_id)
 {
 	switch (port_id) {