diff --git a/bthost_ipc/bthost_ipc.c b/bthost_ipc/bthost_ipc.c
index bec3a56..865956f 100644
--- a/bthost_ipc/bthost_ipc.c
+++ b/bthost_ipc/bthost_ipc.c
@@ -103,6 +103,7 @@
 bool resp_received = false;
 uint8_t tws_channelmode = 0;
 static char a2dp_hal_imp[PROPERTY_VALUE_MAX] = "false";
+static char AAC_frame_ctrl_val[PROPERTY_VALUE_MAX] = "false";
 /*****************************************************************************
 **  Static functions
 ******************************************************************************/
@@ -112,6 +113,7 @@
 audio_aptx_adaptive_encoder_config_t aptx_adaptive_codec;
 audio_aptx_tws_encoder_config_t aptx_tws_codec;
 audio_aac_encoder_config_t aac_codec;
+audio_aac_encoder_config_v2_t aac_codec_v2;
 audio_ldac_encoder_config_t ldac_codec;
 audio_celt_encoder_config_t celt_codec;
 /*****************************************************************************
@@ -179,7 +181,6 @@
 {
     char byte,len;
     uint8_t *p_cfg = codec_cfg;
-    ALOGW("%s",__func__);
     ALOGW("%s: codec_type = %x",__func__, codec_cfg[CODEC_OFFSET]);
     if (codec_cfg[CODEC_OFFSET] == CODEC_TYPE_PCM)
     {
@@ -296,103 +297,228 @@
         return ((void *)(&sbc_codec));
     } else if (codec_cfg[CODEC_OFFSET] == CODEC_TYPE_AAC)
     {
-        uint16_t aac_samp_freq = 0;
-        uint32_t aac_bit_rate = 0;
-        memset(&aac_codec,0,sizeof(audio_aac_encoder_config_t));
-        p_cfg++;//skip dev idx
-        len = *p_cfg++;
-        p_cfg++;//skip media type
-        len--;
-        p_cfg++;//skip codec type
-        len--;
-        byte = *p_cfg++;
-        len--;
-        switch (byte & A2D_AAC_IE_OBJ_TYPE_MSK)
-        {
-            case A2D_AAC_IE_OBJ_TYPE_MPEG_2_AAC_LC:
-                aac_codec.enc_mode = AUDIO_FORMAT_AAC_SUB_LC;
-                break;
-            case A2D_AAC_IE_OBJ_TYPE_MPEG_4_AAC_LC:
-                aac_codec.enc_mode = AUDIO_FORMAT_AAC_SUB_LC;
-                break;
-            case A2D_AAC_IE_OBJ_TYPE_MPEG_4_AAC_LTP:
-                aac_codec.enc_mode = AUDIO_FORMAT_AAC_SUB_LTP;
-                break;
-            case A2D_AAC_IE_OBJ_TYPE_MPEG_4_AAC_SCA:
-                aac_codec.enc_mode = AUDIO_FORMAT_AAC_SUB_SCALABLE;
-                break;
-            default:
-                ALOGE("AAC:Unknown encoder mode");
+        bool is_AAC_frame_ctrl_enable = false;
+        property_get("persist.vendor.bt.aac_frm_ctl.enabled", AAC_frame_ctrl_val, "false");
+        if (!strcmp(AAC_frame_ctrl_val, "true"))
+          is_AAC_frame_ctrl_enable = true;
+        ALOGW("%s: AAC frame control enabled: %d", __func__, is_AAC_frame_ctrl_enable);
+        if (is_AAC_frame_ctrl_enable) {
+          uint16_t aac_samp_freq = 0;
+          uint32_t aac_bit_rate = 0;
+          memset(&aac_codec_v2,0,sizeof(audio_aac_encoder_config_v2_t));
+          p_cfg++;//skip dev idx
+          len = *p_cfg++;
+          p_cfg++;//skip media type
+          len--;
+          p_cfg++;//skip codec type
+          len--;
+          byte = *p_cfg++;
+          len--;
+          switch (byte & A2D_AAC_IE_OBJ_TYPE_MSK)
+          {
+              case A2D_AAC_IE_OBJ_TYPE_MPEG_2_AAC_LC:
+                  aac_codec_v2.enc_mode = AUDIO_FORMAT_AAC_SUB_LC;
+                  break;
+              case A2D_AAC_IE_OBJ_TYPE_MPEG_4_AAC_LC:
+                  aac_codec_v2.enc_mode = AUDIO_FORMAT_AAC_SUB_LC;
+                  break;
+              case A2D_AAC_IE_OBJ_TYPE_MPEG_4_AAC_LTP:
+                  aac_codec_v2.enc_mode = AUDIO_FORMAT_AAC_SUB_LTP;
+                  break;
+              case A2D_AAC_IE_OBJ_TYPE_MPEG_4_AAC_SCA:
+                  aac_codec_v2.enc_mode = AUDIO_FORMAT_AAC_SUB_SCALABLE;
+                  break;
+              default:
+                  ALOGE("AAC:Unknown encoder mode");
+          }
+          //USE 0 (AAC_LC) as hardcoded value till Audio
+          //define constants
+          aac_codec_v2.enc_mode = 0;
+          //USE LOAS(1) or LATM(4) hardcoded values till
+          //Audio define proper constants
+          aac_codec_v2.format_flag = 4;
+          byte = *p_cfg++;
+          len--;
+          aac_samp_freq = byte << 8; //1st byte of sample_freq
+          byte = *p_cfg++;
+          len--;
+          aac_samp_freq |= byte & 0x00F0; //1st nibble of second byte of samp_freq
+
+          switch (aac_samp_freq) {
+              case 0x8000: aac_codec_v2.sampling_rate = 8000; break;
+              case 0x4000: aac_codec_v2.sampling_rate = 11025; break;
+              case 0x2000: aac_codec_v2.sampling_rate = 12000; break;
+              case 0x1000: aac_codec_v2.sampling_rate = 16000; break;
+              case 0x0800: aac_codec_v2.sampling_rate = 22050; break;
+              case 0x0400: aac_codec_v2.sampling_rate = 24000; break;
+              case 0x0200: aac_codec_v2.sampling_rate = 32000; break;
+              case 0x0100: aac_codec_v2.sampling_rate = 44100; break;
+              case 0x0080: aac_codec_v2.sampling_rate = 48000; break;
+              case 0x0040: aac_codec_v2.sampling_rate = 64000; break;
+              case 0x0020: aac_codec_v2.sampling_rate = 88200; break;
+              case 0x0010: aac_codec_v2.sampling_rate = 96000; break;
+              default:
+                  ALOGE("Invalid sample_freq: %x", aac_samp_freq);
+          }
+
+          switch (byte & A2D_AAC_IE_CHANNELS_MSK)
+          {
+              case A2D_AAC_IE_CHANNELS_1:
+                   aac_codec_v2.channels = 1;
+                   break;
+              case A2D_AAC_IE_CHANNELS_2:
+                   aac_codec_v2.channels = 2;
+                   break;
+              default:
+                   ALOGE("AAC:Unknown channel mode");
+          }
+          byte = *p_cfg++; //Move to VBR byte
+          len--;
+          switch (byte & A2D_AAC_IE_VBR_MSK)
+          {
+              case A2D_AAC_IE_VBR:
+                  break;
+              default:
+                  ALOGE("AAC:VBR not supported");
+          }
+          aac_bit_rate = 0x7F&byte;
+          //Move it 2nd byte of 32 bit word. leaving the VBR bit
+          aac_bit_rate = aac_bit_rate << 16;
+          byte = *p_cfg++; //Move to 2nd byteof bitrate
+          len--;
+
+          //Move it to 3rd byte of 32bit word
+          aac_bit_rate |= 0x0000FF00 & (((uint32_t)byte)<<8);
+          byte = *p_cfg++; //Move to 3rd byte of bitrate
+          len--;
+
+          aac_bit_rate |= 0x000000FF & (((uint32_t)byte));
+          aac_codec_v2.bitrate = aac_bit_rate;
+          ALOGW("%s: Final AAC bitrate: %d",__func__, aac_bit_rate);
+
+          aac_codec_v2.frame_ctl.ctl_type = A2D_AAC_FRAME_PEAK_MTU;
+          aac_codec_v2.frame_ctl.ctl_value = *(uint16_t *)p_cfg;
+          //(p_cfg(+2 is because of mtu filled in stack is of 2bytes)
+          p_cfg = p_cfg + 2;
+
+          aac_codec_v2.bitrate = *(uint32_t *)p_cfg;
+          //p_cfg(+4 is because of bitrate filled in stack is occupying 4 bytes.
+          p_cfg = p_cfg + 4;
+          ALOGW("%s: AAC bitrate overwritten with actual value fetched from stack: %d",
+                      __func__, aac_codec_v2.bitrate);
+
+          aac_codec_v2.bits_per_sample = *(uint32_t *)p_cfg;
+          *codec_type = AUDIO_FORMAT_AAC;
+
+          if(sample_freq) *sample_freq = aac_codec_v2.sampling_rate;
+          ALOGW("%s: Copied full codec config bits_per_sample : %d, ctl_type : %d, ctl_value : %d",
+                 __func__, aac_codec_v2.bits_per_sample, aac_codec_v2.frame_ctl.ctl_type,
+                           aac_codec_v2.frame_ctl.ctl_value );
+          return ((void *)(&aac_codec_v2));
+        } else {
+          uint16_t aac_samp_freq = 0;
+          uint32_t aac_bit_rate = 0;
+          memset(&aac_codec,0,sizeof(audio_aac_encoder_config_t));
+          p_cfg++;//skip dev idx
+          len = *p_cfg++;
+          p_cfg++;//skip media type
+          len--;
+          p_cfg++;//skip codec type
+          len--;
+          byte = *p_cfg++;
+          len--;
+          switch (byte & A2D_AAC_IE_OBJ_TYPE_MSK)
+          {
+              case A2D_AAC_IE_OBJ_TYPE_MPEG_2_AAC_LC:
+                  aac_codec.enc_mode = AUDIO_FORMAT_AAC_SUB_LC;
+                  break;
+              case A2D_AAC_IE_OBJ_TYPE_MPEG_4_AAC_LC:
+                  aac_codec.enc_mode = AUDIO_FORMAT_AAC_SUB_LC;
+                  break;
+              case A2D_AAC_IE_OBJ_TYPE_MPEG_4_AAC_LTP:
+                  aac_codec.enc_mode = AUDIO_FORMAT_AAC_SUB_LTP;
+                  break;
+              case A2D_AAC_IE_OBJ_TYPE_MPEG_4_AAC_SCA:
+                  aac_codec.enc_mode = AUDIO_FORMAT_AAC_SUB_SCALABLE;
+                  break;
+              default:
+                  ALOGE("AAC:Unknown encoder mode");
+          }
+          //USE 0 (AAC_LC) as hardcoded value till Audio
+          //define constants
+          aac_codec.enc_mode = 0;
+          //USE LOAS(1) or LATM(4) hardcoded values till
+          //Audio define proper constants
+          aac_codec.format_flag = 4;
+          byte = *p_cfg++;
+          len--;
+          aac_samp_freq = byte << 8; //1st byte of sample_freq
+          byte = *p_cfg++;
+          len--;
+          aac_samp_freq |= byte & 0x00F0; //1st nibble of second byte of samp_freq
+
+          switch (aac_samp_freq) {
+              case 0x8000: aac_codec.sampling_rate = 8000; break;
+              case 0x4000: aac_codec.sampling_rate = 11025; break;
+              case 0x2000: aac_codec.sampling_rate = 12000; break;
+              case 0x1000: aac_codec.sampling_rate = 16000; break;
+              case 0x0800: aac_codec.sampling_rate = 22050; break;
+              case 0x0400: aac_codec.sampling_rate = 24000; break;
+              case 0x0200: aac_codec.sampling_rate = 32000; break;
+              case 0x0100: aac_codec.sampling_rate = 44100; break;
+              case 0x0080: aac_codec.sampling_rate = 48000; break;
+              case 0x0040: aac_codec.sampling_rate = 64000; break;
+              case 0x0020: aac_codec.sampling_rate = 88200; break;
+              case 0x0010: aac_codec.sampling_rate = 96000; break;
+              default:
+                  ALOGE("Invalid sample_freq: %x", aac_samp_freq);
+          }
+
+          switch (byte & A2D_AAC_IE_CHANNELS_MSK)
+          {
+              case A2D_AAC_IE_CHANNELS_1:
+                   aac_codec.channels = 1;
+                   break;
+              case A2D_AAC_IE_CHANNELS_2:
+                   aac_codec.channels = 2;
+                   break;
+              default:
+                   ALOGE("AAC:Unknown channel mode");
+          }
+          byte = *p_cfg++; //Move to VBR byte
+          len--;
+          switch (byte & A2D_AAC_IE_VBR_MSK)
+          {
+              case A2D_AAC_IE_VBR:
+                  break;
+              default:
+                  ALOGE("AAC:VBR not supported");
+          }
+          aac_bit_rate = 0x7F&byte;
+          //Move it 2nd byte of 32 bit word. leaving the VBR bit
+          aac_bit_rate = aac_bit_rate << 16;
+          byte = *p_cfg++; //Move to 2nd byteof bitrate
+          len--;
+
+          //Move it to 3rd byte of 32bit word
+          aac_bit_rate |= 0x0000FF00 & (((uint32_t)byte)<<8);
+          byte = *p_cfg++; //Move to 3rd byte of bitrate
+          len--;
+
+          aac_bit_rate |= 0x000000FF & (((uint32_t)byte));
+          aac_codec.bitrate = aac_bit_rate;
+          ALOGW("%s: Final AAC bitrate: %d",__func__, aac_bit_rate);
+          //+2 because 2 bytes occupying by MTU
+          //+4 because 4 bytes occupying by bitrate
+          p_cfg = p_cfg + 2 + 4;
+          aac_codec.bits_per_sample = *(uint32_t *)p_cfg;
+          *codec_type = AUDIO_FORMAT_AAC;
+
+          if(sample_freq) *sample_freq = aac_codec.sampling_rate;
+          ALOGW("%s: AAC: Done copying full codec config bits_per_sample : %d",
+                               __func__, aac_codec.bits_per_sample );
+          return ((void *)(&aac_codec));
         }
-        //USE 0 (AAC_LC) as hardcoded value till Audio
-        //define constants
-        aac_codec.enc_mode = 0;
-        //USE LOAS(1) or LATM(4) hardcoded values till
-        //Audio define proper constants
-        aac_codec.format_flag = 4;
-        byte = *p_cfg++;
-        len--;
-        aac_samp_freq = byte << 8; //1st byte of sample_freq
-        byte = *p_cfg++;
-        len--;
-        aac_samp_freq |= byte & 0x00F0; //1st nibble of second byte of samp_freq
-
-        switch (aac_samp_freq) {
-            case 0x8000: aac_codec.sampling_rate = 8000; break;
-            case 0x4000: aac_codec.sampling_rate = 11025; break;
-            case 0x2000: aac_codec.sampling_rate = 12000; break;
-            case 0x1000: aac_codec.sampling_rate = 16000; break;
-            case 0x0800: aac_codec.sampling_rate = 22050; break;
-            case 0x0400: aac_codec.sampling_rate = 24000; break;
-            case 0x0200: aac_codec.sampling_rate = 32000; break;
-            case 0x0100: aac_codec.sampling_rate = 44100; break;
-            case 0x0080: aac_codec.sampling_rate = 48000; break;
-            case 0x0040: aac_codec.sampling_rate = 64000; break;
-            case 0x0020: aac_codec.sampling_rate = 88200; break;
-            case 0x0010: aac_codec.sampling_rate = 96000; break;
-            default:
-                ALOGE("Invalid sample_freq: %x", aac_samp_freq);
-        }
-
-        switch (byte & A2D_AAC_IE_CHANNELS_MSK)
-        {
-            case A2D_AAC_IE_CHANNELS_1:
-                 aac_codec.channels = 1;
-                 break;
-            case A2D_AAC_IE_CHANNELS_2:
-                 aac_codec.channels = 2;
-                 break;
-            default:
-                 ALOGE("AAC:Unknown channel mode");
-        }
-        byte = *p_cfg++; //Move to VBR byte
-        len--;
-        switch (byte & A2D_AAC_IE_VBR_MSK)
-        {
-            case A2D_AAC_IE_VBR:
-                break;
-            default:
-                ALOGE("AAC:VBR not supported");
-        }
-        aac_bit_rate = 0x7F&byte;
-        //Move it 2nd byte of 32 bit word. leaving the VBR bit
-        aac_bit_rate = aac_bit_rate << 16;
-        byte = *p_cfg++; //Move to 2nd byteof bitrate
-        len--;
-
-        //Move it to 3rd byte of 32bit word
-        aac_bit_rate |= 0x0000FF00 & (((uint32_t)byte)<<8);
-        byte = *p_cfg++; //Move to 3rd byte of bitrate
-        len--;
-
-        aac_bit_rate |= 0x000000FF & (((uint32_t)byte));
-        aac_codec.bitrate = aac_bit_rate;
-        aac_codec.bits_per_sample = *(uint32_t *)p_cfg;
-        *codec_type = AUDIO_FORMAT_AAC;
-
-        if(sample_freq) *sample_freq = aac_codec.sampling_rate;
-        ALOGW("AAC: Done copying full codec config bits_per_sample : %d", aac_codec.bits_per_sample);
-        return ((void *)(&aac_codec));
     }
     else if (codec_cfg[CODEC_OFFSET] == NON_A2DP_CODEC_TYPE)
     {
diff --git a/bthost_ipc/bthost_ipc.h b/bthost_ipc/bthost_ipc.h
index 50e7abe..b72f0a3 100644
--- a/bthost_ipc/bthost_ipc.h
+++ b/bthost_ipc/bthost_ipc.h
@@ -244,6 +244,9 @@
 #define A2D_AAC_IE_VBR_MSK                     0x80
 #define A2D_AAC_IE_VBR                         0x80    /* supported */
 
+#define A2D_AAC_FRAME_PEAK_MTU       0  /* Configure peak MTU */
+#define A2D_AAC_FRAME_PEAK_BITRATE   1  /* Configure peak bitrate */
+
 #define A2DP_DEFAULT_SINK_LATENCY 0
 
 
@@ -355,6 +358,17 @@
     uint32_t bits_per_sample;
 } audio_ldac_encoder_config_t;
 
+/* Structure to control frame size of AAC encoded frames. */
+struct aac_frame_size_control_t {
+    /* Type of frame size control: MTU_SIZE / PEAK_BIT_RATE */
+    uint32_t ctl_type;
+    /* Control value
+     * MTU_SIZE: MTU size in bytes
+     * PEAK_BIT_RATE: Peak bitrate in bits per second.
+     */
+    uint32_t ctl_value;
+};
+
 /* Information about BT AAC encoder configuration
  * This data is used between audio HAL module and
  * BT IPC library to configure DSP encoder
@@ -368,6 +382,20 @@
     uint32_t bits_per_sample;
 } audio_aac_encoder_config_t;
 
+/* Information about BT AAC encoder configuration
+ * This data is used between audio HAL module and
+ * BT IPC library to configure DSP encoder with frame control
+ */
+typedef struct {
+    uint32_t enc_mode; /* LC, SBR, PS */
+    uint16_t format_flag; /* RAW, ADTS */
+    uint16_t channels; /* 1-Mono, 2-Stereo */
+    uint32_t sampling_rate;
+    uint32_t bitrate;
+    uint32_t bits_per_sample;
+    struct aac_frame_size_control_t frame_ctl;
+} audio_aac_encoder_config_v2_t;
+
 typedef struct {
     uint32_t sampling_rate; /* 32000 - 48000, 48000 */
     uint16_t channels; /* 1-Mono, 2-Stereo, 2*/