diff --git a/hal/Android.mk b/hal/Android.mk
index 1c3f946..32a9ee9 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -121,7 +121,8 @@
 	libtinycompress \
 	libaudioroute \
 	libdl \
-	libexpat
+	libexpat \
+        libmdmdetect
 
 LOCAL_C_INCLUDES += \
 	external/tinyalsa/include \
@@ -131,7 +132,8 @@
 	$(call include-path-for, audio-effects) \
 	$(LOCAL_PATH)/$(AUDIO_PLATFORM) \
 	$(LOCAL_PATH)/audio_extn \
-	$(LOCAL_PATH)/voice_extn
+	$(LOCAL_PATH)/voice_extn \
+        $(TARGET_OUT_HEADERS)/libmdmdetect/inc
 
 ifeq ($(strip $(AUDIO_FEATURE_ENABLED_LISTEN)),true)
     LOCAL_CFLAGS += -DAUDIO_LISTEN_ENABLED
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index c6a8e68..245f5ca 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -174,7 +174,7 @@
 #endif /* ANC_HEADSET_ENABLED */
 
 #ifndef AFE_PROXY_ENABLED
-#define audio_extn_set_afe_proxy_parameters(parms)        (0)
+#define audio_extn_set_afe_proxy_parameters(adev, parms)  (0)
 #define audio_extn_get_afe_proxy_parameters(query, reply) (0)
 #else
 /* Front left channel. */
@@ -290,7 +290,8 @@
     return ret;
 }
 
-void audio_extn_set_afe_proxy_parameters(struct str_parms *parms)
+void audio_extn_set_afe_proxy_parameters(struct audio_device *adev,
+                                         struct str_parms *parms)
 {
     int ret, val;
     char value[32]={0};
@@ -300,6 +301,7 @@
     if (ret >= 0) {
         val = atoi(value);
         aextnmod.proxy_channel_num = val;
+        adev->cur_wfd_channels = val;
         ALOGD("%s: channel capability set to: %d", __func__,
                aextnmod.proxy_channel_num);
     }
@@ -358,7 +360,7 @@
                                struct str_parms *parms)
 {
    audio_extn_set_anc_parameters(adev, parms);
-   audio_extn_set_afe_proxy_parameters(parms);
+   audio_extn_set_afe_proxy_parameters(adev, parms);
    audio_extn_fm_set_parameters(adev, parms);
    audio_extn_listen_set_parameters(adev, parms);
    audio_extn_hfp_set_parameters(adev, parms);
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index fb428db..d432670 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -184,8 +184,10 @@
 
 #ifndef HFP_ENABLED
 #define audio_extn_hfp_is_active(adev)                  (0)
+#define audio_extn_hfp_get_usecase()                    (0)
 #else
 bool audio_extn_hfp_is_active(struct audio_device *adev);
+audio_usecase_t audio_extn_hfp_get_usecase();
 #endif
 
 #endif /* AUDIO_EXTN_H */
diff --git a/hal/audio_extn/dolby.c b/hal/audio_extn/dolby.c
index bcc7381..99fa2b7 100644
--- a/hal/audio_extn/dolby.c
+++ b/hal/audio_extn/dolby.c
@@ -64,7 +64,7 @@
 
 /* DS1-DDP Endp Params */
 #define DDP_ENDP_NUM_PARAMS 17
-#define DDP_ENDP_NUM_DEVICES 22
+#define DDP_ENDP_NUM_DEVICES 23
 static int ddp_endp_params_id[DDP_ENDP_NUM_PARAMS] = {
     PARAM_ID_MAX_OUTPUT_CHANNELS, PARAM_ID_CTL_RUNNING_MODE,
     PARAM_ID_CTL_ERROR_CONCEAL, PARAM_ID_CTL_ERROR_MAX_RPTS,
@@ -147,7 +147,10 @@
               {8, 0, 0, 0, 0, 0, 0, 21, 1, 6, 0, 0, 0, 0, 0, 0, 0},
               {1, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0} },
           {AUDIO_DEVICE_OUT_PROXY, 2,
-              {8, 0, 0, 0, 0, 0, 0, 21, 1, 6, 0, 0, 0, 0, 0, 0, 0},
+              {8, 0, 0, 0, 0, 0, 0, 21, 1, 2, 0, 0, 0, 0, 0, 0, 0},
+              {1, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0} },
+          {AUDIO_DEVICE_OUT_PROXY, 6,
+              {8, 0, 0, 0, 0, 0, 0, 21, 1, 2, 0, 0, 0, 0, 0, 0, 0},
               {1, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0} },
 };
 
@@ -264,9 +267,16 @@
             (usecase->stream.out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
             ((usecase->stream.out->format == AUDIO_FORMAT_AC3) ||
              (usecase->stream.out->format == AUDIO_FORMAT_EAC3))) {
+            /*
+             * Use wfd /hdmi sink channel cap for dolby params if device is wfd
+             * or hdmi. Otherwise use stereo configuration
+             */
+            int channel_cap = usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ?
+                              adev->cur_hdmi_channels :
+                              usecase->devices & AUDIO_DEVICE_OUT_PROXY ?
+                              adev->cur_wfd_channels : 2;
             send_ddp_endp_params_stream(usecase->stream.out, usecase->devices,
-                           usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ?
-                           adev->cur_hdmi_channels : 2, false /* set cache */);
+                                        channel_cap, false /* set cache */);
         }
     }
 }
@@ -334,7 +344,9 @@
         update_ddp_endp_table(ddp_dev, dev_ch_cap,
                               PARAM_ID_OUT_CTL_STEREO_MODE, val);
     }
-
+    /* TODO: Do we need device channel caps here?
+     * We dont have that information as this is from dolby modules
+     */
     send_ddp_endp_params(adev, ddp_dev, dev_ch_cap);
 }
 
@@ -343,13 +355,20 @@
                                       audio_format_t format)
 {
     int id = 0;
+    /*
+     * Use wfd /hdmi sink channel cap for dolby params if device is wfd
+     * or hdmi. Otherwise use stereo configuration
+     */
+    int channel_cap = out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ?
+                      adev->cur_hdmi_channels :
+                      out->devices & AUDIO_DEVICE_OUT_PROXY ?
+                      adev->cur_wfd_channels : 2;
 
     switch (format) {
     case AUDIO_FORMAT_AC3:
         id = SND_AUDIOCODEC_AC3;
         send_ddp_endp_params_stream(out, out->devices,
-                            out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ?
-                            adev->cur_hdmi_channels : 2, true /* set_cache */);
+                            channel_cap, true /* set_cache */);
 #ifndef DS1_DOLBY_DAP_ENABLED
         audio_extn_dolby_set_dmid(adev);
 #endif
@@ -357,8 +376,7 @@
     case AUDIO_FORMAT_EAC3:
         id = SND_AUDIOCODEC_EAC3;
         send_ddp_endp_params_stream(out, out->devices,
-                            out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ?
-                            adev->cur_hdmi_channels : 2, true /* set_cache */);
+                            channel_cap, true /* set_cache */);
 #ifndef DS1_DOLBY_DAP_ENABLED
         audio_extn_dolby_set_dmid(adev);
 #endif
@@ -430,8 +448,7 @@
 
     list_for_each(node, &adev->usecase_list) {
         usecase = node_to_item(node, struct audio_usecase, list);
-        if ((usecase->type == PCM_PLAYBACK) &&
-            (usecase->id != USECASE_AUDIO_PLAYBACK_LOW_LATENCY))
+        if (usecase->type == PCM_PLAYBACK)
             send = true;
     }
     if (!send)
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index 2d6e1e0..c480490 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -220,7 +220,7 @@
 bool audio_extn_hfp_is_active(struct audio_device *adev)
 {
     struct audio_usecase *hfp_usecase = NULL;
-    hfp_usecase = get_usecase_from_list(adev, USECASE_AUDIO_HFP_SCO);
+    hfp_usecase = get_usecase_from_list(adev, hfpmod.ucid);
 
     if (hfp_usecase != NULL)
         return true;
@@ -228,6 +228,11 @@
         return false;
 }
 
+audio_usecase_t audio_extn_hfp_get_usecase()
+{
+    return hfpmod.ucid;
+}
+
 void audio_extn_hfp_set_parameters(struct audio_device *adev, struct str_parms *parms)
 {
     int ret;
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 5514007..69e9561 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -603,6 +603,7 @@
     struct audio_usecase *vc_usecase = NULL;
     struct audio_usecase *voip_usecase = NULL;
     struct audio_usecase *hfp_usecase = NULL;
+    audio_usecase_t hfp_ucid;
     struct listnode *node;
     int status = 0;
 
@@ -642,7 +643,8 @@
                     out_snd_device = voip_usecase->out_snd_device;
             }
         } else if (audio_extn_hfp_is_active(adev)) {
-            hfp_usecase = get_usecase_from_list(adev, USECASE_AUDIO_HFP_SCO);
+            hfp_ucid = audio_extn_hfp_get_usecase();
+            hfp_usecase = get_usecase_from_list(adev, hfp_ucid);
             if (hfp_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) {
                    in_snd_device = hfp_usecase->in_snd_device;
                    out_snd_device = hfp_usecase->out_snd_device;
@@ -921,24 +923,30 @@
         send_callback = false;
         switch(cmd->cmd) {
         case OFFLOAD_CMD_WAIT_FOR_BUFFER:
+            ALOGD("copl(%x):calling compress_wait", (unsigned int)out);
             compress_wait(out->compr, -1);
+            ALOGD("copl(%x):out of compress_wait", (unsigned int)out);
             send_callback = true;
             event = STREAM_CBK_EVENT_WRITE_READY;
             break;
         case OFFLOAD_CMD_PARTIAL_DRAIN:
             ret = compress_next_track(out->compr);
-            if(ret == 0)
+            if(ret == 0) {
+                ALOGD("copl(%x):calling compress_partial_drain", (unsigned int)out);
                 compress_partial_drain(out->compr);
+                ALOGD("copl(%x):out of compress_partial_drain", (unsigned int)out);
+            }
             else if(ret == -ETIMEDOUT)
                 compress_drain(out->compr);
             else
                 ALOGE("%s: Next track returned error %d",__func__, ret);
-
             send_callback = true;
             event = STREAM_CBK_EVENT_DRAIN_READY;
             break;
         case OFFLOAD_CMD_DRAIN:
+            ALOGD("copl(%x):calling compress_drain", (unsigned int)out);
             compress_drain(out->compr);
+            ALOGD("copl(%x):calling compress_drain", (unsigned int)out);
             send_callback = true;
             event = STREAM_CBK_EVENT_DRAIN_READY;
             break;
@@ -1013,6 +1021,12 @@
                       "no change in HDMI channels", __func__);
                 ret = false;
                 break;
+            } else if (usecase->id == USECASE_AUDIO_PLAYBACK_OFFLOAD &&
+                       popcount(usecase->stream.out->channel_mask) > 2) {
+                ALOGD("%s: multi-channel(%x) compress offload playback is active, "
+                      "no change in HDMI channels", __func__, usecase->stream.out->channel_mask);
+                ret = false;
+                break;
             }
         }
     }
@@ -1110,7 +1124,7 @@
     struct audio_usecase *uc_info;
     struct audio_device *adev = out->dev;
 
-    ALOGV("%s: enter: usecase(%d: %s) devices(%#x)",
+    ALOGD("%s: enter: usecase(%d: %s) devices(%#x)",
           __func__, out->usecase, use_case_table[out->usecase], out->devices);
     out->pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK);
     if (out->pcm_device_id < 0) {
@@ -1312,6 +1326,7 @@
                 out->pcm = NULL;
             }
         } else {
+            ALOGD("copl(%x):standby", (unsigned int)out);
             stop_compressed_output_l(out);
             out->gapless_mdata.encoder_delay = 0;
             out->gapless_mdata.encoder_padding = 0;
@@ -1595,9 +1610,9 @@
     }
 
     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
-        ALOGVV("%s: writing buffer (%d bytes) to compress device", __func__, bytes);
+        ALOGD("copl(%x): writing buffer (%d bytes) to compress device", (unsigned int)out, bytes);
         if (out->send_new_metadata) {
-            ALOGVV("send new gapless metadata");
+            ALOGD("copl(%x):send new gapless metadata", (unsigned int)out);
             compress_set_gapless_metadata(out->compr, &out->gapless_mdata);
             out->send_new_metadata = 0;
         }
@@ -1605,6 +1620,7 @@
         ret = compress_write(out->compr, buffer, bytes);
         ALOGVV("%s: writing buffer (%d bytes) to compress device returned %d", __func__, bytes, ret);
         if (ret >= 0 && ret < (ssize_t)bytes) {
+            ALOGD("No space available in compress driver, post msg to cb thread");
             send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
         }
         if (!out->playback_started) {
@@ -1737,6 +1753,7 @@
     int status = -ENOSYS;
     ALOGV("%s", __func__);
     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        ALOGD("copl(%x):pause compress driver", (unsigned int)out);
         pthread_mutex_lock(&out->lock);
         if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) {
             status = compress_pause(out->compr);
@@ -1753,6 +1770,7 @@
     int status = -ENOSYS;
     ALOGV("%s", __func__);
     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        ALOGD("copl(%x):resume compress driver", (unsigned int)out);
         status = 0;
         pthread_mutex_lock(&out->lock);
         if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) {
@@ -1785,9 +1803,11 @@
     struct stream_out *out = (struct stream_out *)stream;
     ALOGV("%s", __func__);
     if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        ALOGD("copl(%x):calling compress flush", (unsigned int)out);
         pthread_mutex_lock(&out->lock);
         stop_compressed_output_l(out);
         pthread_mutex_unlock(&out->lock);
+        ALOGD("copl(%x):out of compress flush", (unsigned int)out);
         return 0;
     }
     return -ENOSYS;
@@ -2126,6 +2146,9 @@
             goto error_open;
         }
     } else if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+        ALOGD("%s: copl(%x): sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)",
+              __func__, (unsigned int)out, config->sample_rate, config->channel_mask, devices, flags);
+
         if (config->offload_info.version != AUDIO_INFO_INITIALIZER.version ||
             config->offload_info.size != AUDIO_INFO_INITIALIZER.size) {
             ALOGE("%s: Unsupported Offload information", __func__);
@@ -2673,6 +2696,7 @@
     adev->snd_dev_ref_cnt = calloc(SND_DEVICE_MAX, sizeof(int));
     voice_init(adev);
     list_init(&adev->usecase_list);
+    adev->cur_wfd_channels = 2;
 
     /* Loads platform specific libraries dynamically */
     adev->platform = platform_init(adev);
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index a6824c1..2108a00 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -224,6 +224,7 @@
     bool speaker_lr_swap;
     struct voice voice;
     unsigned int cur_hdmi_channels;
+    unsigned int cur_wfd_channels;
 
     int snd_card;
     void *platform;
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index 128e4af..59bdb56 100644
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -125,7 +125,9 @@
 
 static void  update_hardware_info_8084(struct hardware_info *hw_info, const char *snd_card_name)
 {
-    if (!strcmp(snd_card_name, "apq8084-taiko-mtp-snd-card")) {
+    if (!strcmp(snd_card_name, "apq8084-taiko-mtp-snd-card") ||
+        !strncmp(snd_card_name, "apq8084-taiko-i2s-mtp-snd-card",
+                 sizeof("apq8084-taiko-i2s-mtp-snd-card"))) {
         strlcpy(hw_info->type, "mtp", sizeof(hw_info->type));
         strlcpy(hw_info->name, "apq8084", sizeof(hw_info->name));
         hw_info->snd_devices = NULL;
@@ -137,6 +139,13 @@
         hw_info->snd_devices = (snd_device_t *)taiko_apq8084_CDP_variant_devices;
         hw_info->num_snd_devices = ARRAY_SIZE(taiko_apq8084_CDP_variant_devices);
         strlcpy(hw_info->dev_extn, "-cdp", sizeof(hw_info->dev_extn));
+    } else if (!strncmp(snd_card_name, "apq8084-taiko-i2s-cdp-snd-card",
+                        sizeof("apq8084-taiko-i2s-cdp-snd-card"))) {
+        strlcpy(hw_info->type, " cdp", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "apq8084", sizeof(hw_info->name));
+        hw_info->snd_devices = NULL;
+        hw_info->num_snd_devices = 0;
+        strlcpy(hw_info->dev_extn, "", sizeof(hw_info->dev_extn));
     } else if (!strcmp(snd_card_name, "apq8084-taiko-liquid-snd-card")) {
         strlcpy(hw_info->type , " liquid", sizeof(hw_info->type));
         strlcpy(hw_info->name, "apq8084", sizeof(hw_info->type));
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index c727900..684436e 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -32,9 +32,15 @@
 #include "audio_extn.h"
 #include "voice_extn.h"
 #include "sound/compress_params.h"
+#include "mdm_detect.h"
 
 #define MIXER_XML_PATH "/system/etc/mixer_paths.xml"
 #define MIXER_XML_PATH_AUXPCM "/system/etc/mixer_paths_auxpcm.xml"
+#define MIXER_XML_PATH_I2S "/system/etc/mixer_paths_i2s.xml"
+
+#define PLATFORM_INFO_XML_PATH      "/system/etc/audio_platform_info.xml"
+#define PLATFORM_INFO_XML_PATH_I2S  "/system/etc/audio_platform_info_i2s.xml"
+
 #define LIB_ACDB_LOADER "libacdbloader.so"
 #define AUDIO_DATA_BLOCK_MIXER_CTL "HDMI EDID"
 
@@ -108,6 +114,7 @@
     int  fluence_type;
     int  btsco_sample_rate;
     bool slowtalk;
+    bool is_i2s_ext_modem;
     /* Audio calibration related functions */
     void                       *acdb_handle;
     int                        voice_feature_set;
@@ -410,7 +417,7 @@
     return 0;
 }
 
-static struct csd_data *open_csd_client()
+static struct csd_data *open_csd_client(bool i2s_ext_modem)
 {
     struct csd_data *csd = calloc(1, sizeof(struct csd_data));
 
@@ -512,6 +519,16 @@
                   __func__, dlerror());
             goto error;
         }
+
+        csd->get_sample_rate = (get_sample_rate_t)dlsym(csd->csd_client,
+                                             "csd_client_get_sample_rate");
+        if (csd->get_sample_rate == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_get_sample_rate",
+                  __func__, dlerror());
+
+            goto error;
+        }
+
         csd->init = (init_t)dlsym(csd->csd_client, "csd_client_init");
 
         if (csd->init == NULL) {
@@ -519,7 +536,7 @@
                   __func__, dlerror());
             goto error;
         } else {
-            csd->init();
+            csd->init(i2s_ext_modem);
         }
     }
     return csd;
@@ -540,10 +557,44 @@
     }
 }
 
+static void platform_csd_init(struct platform_data *plat_data)
+{
+    struct dev_info mdm_detect_info;
+    int ret = 0;
+
+    /* Call ESOC API to get the number of modems.
+     * If the number of modems is not zero, load CSD Client specific
+     * symbols. Voice call is handled by MDM and apps processor talks to
+     * MDM through CSD Client
+     */
+    ret = get_system_info(&mdm_detect_info);
+    if (ret > 0) {
+        ALOGE("%s: Failed to get system info, ret %d", __func__, ret);
+    }
+    ALOGD("%s: num_modems %d\n", __func__, mdm_detect_info.num_modems);
+
+    if (mdm_detect_info.num_modems > 0)
+        plat_data->csd = open_csd_client(plat_data->is_i2s_ext_modem);
+}
+
+static bool platform_is_i2s_ext_modem(const char *snd_card_name,
+                                      struct platform_data *plat_data)
+{
+    plat_data->is_i2s_ext_modem = false;
+
+    if (!strncmp(snd_card_name, "apq8084-taiko-i2s-mtp-snd-card",
+                 sizeof("apq8084-taiko-i2s-mtp-snd-card")) ||
+        !strncmp(snd_card_name, "apq8084-taiko-i2s-cdp-snd-card",
+                 sizeof("apq8084-taiko-i2s-cdp-snd-card"))) {
+        plat_data->is_i2s_ext_modem = true;
+    }
+    ALOGV("%s, is_i2s_ext_modem:%d",__func__, plat_data->is_i2s_ext_modem);
+
+    return plat_data->is_i2s_ext_modem;
+}
+
 void *platform_init(struct audio_device *adev)
 {
-    char platform[PROPERTY_VALUE_MAX];
-    char baseband[PROPERTY_VALUE_MAX];
     char value[PROPERTY_VALUE_MAX];
     struct platform_data *my_data = NULL;
     int retry_num = 0, snd_card_num = 0;
@@ -575,10 +626,16 @@
         if (!my_data->hw_info) {
             ALOGE("%s: Failed to init hardware info", __func__);
         } else {
-            if (audio_extn_read_xml(adev, snd_card_num, MIXER_XML_PATH,
-                                    MIXER_XML_PATH_AUXPCM) == -ENOSYS)
+            if (platform_is_i2s_ext_modem(snd_card_name, my_data)) {
+                ALOGD("%s: Call MIXER_XML_PATH_I2S", __func__);
+
+                adev->audio_route = audio_route_init(snd_card_num,
+                                                     MIXER_XML_PATH_I2S);
+            } else if (audio_extn_read_xml(adev, snd_card_num, MIXER_XML_PATH,
+                                    MIXER_XML_PATH_AUXPCM) == -ENOSYS) {
                 adev->audio_route = audio_route_init(snd_card_num,
                                                  MIXER_XML_PATH);
+            }
             if (!adev->audio_route) {
                 ALOGE("%s: Failed to init audio route controls, aborting.",
                        __func__);
@@ -677,18 +734,13 @@
     }
 
     /* Initialize ACDB ID's */
-    platform_info_init();
+    if (my_data->is_i2s_ext_modem)
+        platform_info_init(PLATFORM_INFO_XML_PATH_I2S);
+    else
+        platform_info_init(PLATFORM_INFO_XML_PATH);
 
-    /* If platform is apq8084 and baseband is MDM, load CSD Client specific
-     * symbols. Voice call is handled by MDM and apps processor talks to
-     * MDM through CSD Client
-     */
-    property_get("ro.board.platform", platform, "");
-    property_get("ro.baseband", baseband, "");
-    if (!strncmp("apq8084", platform, sizeof("apq8084")) &&
-        !strncmp("mdm", baseband, sizeof("mdm"))) {
-         my_data->csd = open_csd_client();
-    }
+    /* load csd client */
+    platform_csd_init(my_data);
 
     /* init usb */
     audio_extn_usb_init(adev);
@@ -961,6 +1013,20 @@
     return ret;
 }
 
+int platform_get_sample_rate(void *platform, uint32_t *rate)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    int ret = 0;
+
+    if ((my_data->csd != NULL) && my_data->is_i2s_ext_modem) {
+        ret = my_data->csd->get_sample_rate(rate);
+        if (ret < 0) {
+            ALOGE("%s: csd_get_sample_rate error %d\n", __func__, ret);
+        }
+    }
+    return ret;
+}
+
 int platform_set_voice_volume(void *platform, int volume)
 {
     struct platform_data *my_data = (struct platform_data *)platform;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index d9b4302..0f72a54 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -225,7 +225,7 @@
 
 #define LIB_CSD_CLIENT "libcsd-client.so"
 /* CSD-CLIENT related functions */
-typedef int (*init_t)();
+typedef int (*init_t)(bool);
 typedef int (*deinit_t)();
 typedef int (*disable_device_t)();
 typedef int (*enable_device_config_t)(int, int);
@@ -239,6 +239,7 @@
 typedef int (*stop_playback_t)(uint32_t);
 typedef int (*start_record_t)(uint32_t, int);
 typedef int (*stop_record_t)(uint32_t);
+typedef int (*get_sample_rate_t)(uint32_t *);
 /* CSD Client structure */
 struct csd_data {
     void *csd_client;
@@ -256,6 +257,7 @@
     stop_playback_t stop_playback;
     start_record_t start_record;
     stop_record_t stop_record;
+    get_sample_rate_t get_sample_rate;
 };
 
 #endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 81291a2..2ec6b50 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -44,6 +44,7 @@
 int platform_stop_voice_call(void *platform, uint32_t vsid);
 int platform_set_voice_volume(void *platform, int volume);
 int platform_set_mic_mute(void *platform, bool state);
+int platform_get_sample_rate(void *platform, uint32_t *rate);
 snd_device_t platform_get_output_snd_device(void *platform, audio_devices_t devices);
 snd_device_t platform_get_input_snd_device(void *platform, audio_devices_t out_device);
 int platform_set_hdmi_channels(void *platform, int channel_count);
@@ -63,7 +64,7 @@
 bool platform_listen_update_status(snd_device_t snd_device);
 
 /* From platform_info_parser.c */
-int platform_info_init(void);
+int platform_info_init(const char *filename);
 
 struct audio_offload_info_t;
 uint32_t platform_get_compress_offload_buffer_size(audio_offload_info_t* info);
diff --git a/hal/platform_info.c b/hal/platform_info.c
index 8f56107..85a05eb 100644
--- a/hal/platform_info.c
+++ b/hal/platform_info.c
@@ -38,7 +38,6 @@
 #include "platform_api.h"
 #include <platform.h>
 
-#define PLATFORM_INFO_XML_PATH      "/system/etc/audio_platform_info.xml"
 #define BUF_SIZE                    1024
 
 static void process_device(const XML_Char **attr)
@@ -52,20 +51,20 @@
 
     index = platform_get_snd_device_index((char *)attr[1]);
     if (index < 0) {
-        ALOGE("%s: Device %s in %s not found, no ACDB ID set!",
-              __func__, attr[1], PLATFORM_INFO_XML_PATH);
+        ALOGE("%s: Device %s in platform info xml not found, no ACDB ID set!",
+              __func__, attr[1]);
         goto done;
     }
 
     if (strcmp(attr[2], "acdb_id") != 0) {
-        ALOGE("%s: Device %s in %s has no acdb_id, no ACDB ID set!",
-              __func__, attr[1], PLATFORM_INFO_XML_PATH);
+        ALOGE("%s: Device %s in platform info xml has no acdb_id, no ACDB ID set!",
+              __func__, attr[1]);
         goto done;
     }
 
     if(platform_set_snd_device_acdb_id(index, atoi((char *)attr[3])) < 0) {
-        ALOGE("%s: Device %s in %s, ACDB ID %d was not set!",
-              __func__, attr[1], PLATFORM_INFO_XML_PATH, atoi((char *)attr[3]));
+        ALOGE("%s: Device %s in platform info xml ACDB ID %d was not set!",
+              __func__, attr[1], atoi((char *)attr[3]));
         goto done;
     }
 
@@ -91,7 +90,7 @@
 
 }
 
-int platform_info_init(void)
+int platform_info_init(const char *filename)
 {
     XML_Parser      parser;
     FILE            *file;
@@ -99,10 +98,10 @@
     int             bytes_read;
     void            *buf;
 
-    file = fopen(PLATFORM_INFO_XML_PATH, "r");
+    file = fopen(filename, "r");
     if (!file) {
         ALOGD("%s: Failed to open %s, using defaults.",
-            __func__, PLATFORM_INFO_XML_PATH);
+            __func__, filename);
         ret = -ENODEV;
         goto done;
     }
@@ -134,7 +133,7 @@
         if (XML_ParseBuffer(parser, bytes_read,
                             bytes_read == 0) == XML_STATUS_ERROR) {
             ALOGE("%s: XML_ParseBuffer failed, for %s",
-                __func__, PLATFORM_INFO_XML_PATH);
+                __func__, filename);
             ret = -EINVAL;
             goto err_free_parser;
         }
diff --git a/hal/voice.c b/hal/voice.c
index 28d44db..1c3ab38 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -106,6 +106,7 @@
     int i, ret = 0;
     struct audio_usecase *uc_info;
     int pcm_dev_rx_id, pcm_dev_tx_id;
+    uint32_t sample_rate = 8000;
     struct voice_session *session = NULL;
     struct pcm_config voice_config = pcm_config_voice_call;
 
@@ -133,6 +134,13 @@
         ret = -EIO;
         goto error_start_voice;
     }
+    ret = platform_get_sample_rate(adev->platform, &sample_rate);
+    if (ret < 0) {
+        ALOGE("platform_get_sample_rate error %d\n", ret);
+    } else {
+        voice_config.rate = sample_rate;
+    }
+    ALOGD("voice_config.rate %d\n", voice_config.rate);
 
     ALOGV("%s: Opening PCM playback device card_id(%d) device_id(%d)",
           __func__, adev->snd_card, pcm_dev_rx_id);
diff --git a/mm-audio/aenc-aac/qdsp6/inc/omx_aac_aenc.h b/mm-audio/aenc-aac/qdsp6/inc/omx_aac_aenc.h
index 276eaa3..623caa8 100644
--- a/mm-audio/aenc-aac/qdsp6/inc/omx_aac_aenc.h
+++ b/mm-audio/aenc-aac/qdsp6/inc/omx_aac_aenc.h
@@ -1,5 +1,5 @@
 /*--------------------------------------------------------------------------
-Copyright (c) 2010, The Linux Foundation. All rights reserved.
+Copyright (c) 2010-2014, The Linux Foundation. All rights reserved.
 
 Redistribution and use in source and binary forms, with or without
 modification, are permitted provided that the following conditions are met:
@@ -358,7 +358,10 @@
         OMX_COMPONENT_OUTPUT_DISABLE_PENDING  =0x7
     };
 
-
+    #define MIN_BITRATE 24000
+    #define MAX_BITRATE 192000
+    #define MAX_BITRATE_MULFACTOR 12
+    #define BITRATE_DIVFACTOR 2
     typedef Map<OMX_BUFFERHEADERTYPE*, OMX_BUFFERHEADERTYPE*>
     input_buffer_map;
 
@@ -619,6 +622,7 @@
                 OMX_U8 num_bits_reqd,
                 OMX_U32  value,
                 OMX_U16 *hdr_bit_index);
+    int get_updated_bit_rate(int bitrate);
 
 };
 #endif
diff --git a/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp b/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
index 52aa915..6521265 100644
--- a/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
+++ b/mm-audio/aenc-aac/qdsp6/src/omx_aac_aenc.cpp
@@ -1,5 +1,5 @@
 /*--------------------------------------------------------------------------
-Copyright (c) 2010, The Linux Foundation. All rights reserved.
+Copyright (c) 2010-2014, The Linux Foundation. All rights reserved.
 
 Redistribution and use in source and binary forms, with or without
 modification, are permitted provided that the following conditions are met:
@@ -1438,10 +1438,12 @@
                 }
                 drv_aac_enc_config.channels = m_aac_param.nChannels;
                 drv_aac_enc_config.sample_rate = m_aac_param.nSampleRate;
-                drv_aac_enc_config.bit_rate =  m_aac_param.nBitRate;
-                DEBUG_PRINT("aac config %lu,%lu,%lu %d\n",
+                drv_aac_enc_config.bit_rate =
+                get_updated_bit_rate(m_aac_param.nBitRate);
+                DEBUG_PRINT("aac config %lu,%lu,%lu %d updated bitrate %d\n",
                             m_aac_param.nChannels,m_aac_param.nSampleRate,
-			    m_aac_param.nBitRate,m_aac_param.eAACStreamFormat);
+			    m_aac_param.nBitRate,m_aac_param.eAACStreamFormat,
+                            drv_aac_enc_config.bit_rate);
                 switch(m_aac_param.eAACStreamFormat)
                 {
 
@@ -5014,3 +5016,44 @@
 
 }
 
+int omx_aac_aenc::get_updated_bit_rate(int bitrate)
+{
+	int updated_rate, min_bitrate, max_bitrate;
+
+        max_bitrate = m_aac_param.nSampleRate *
+        MAX_BITRATE_MULFACTOR;
+	switch(m_aac_param.eAACProfile)
+	{
+		case OMX_AUDIO_AACObjectLC:
+		    min_bitrate = m_aac_param.nSampleRate;
+		    if (m_aac_param.nChannels == 1) {
+		       min_bitrate = min_bitrate/BITRATE_DIVFACTOR;
+                       max_bitrate = max_bitrate/BITRATE_DIVFACTOR;
+                    }
+                break;
+		case OMX_AUDIO_AACObjectHE:
+		    min_bitrate = MIN_BITRATE;
+		    if (m_aac_param.nChannels == 1)
+                       max_bitrate = max_bitrate/BITRATE_DIVFACTOR;
+		break;
+		case OMX_AUDIO_AACObjectHE_PS:
+		    min_bitrate = MIN_BITRATE;
+		break;
+                default:
+                    return bitrate;
+                break;
+	}
+        /* Update MIN and MAX values*/
+        if (min_bitrate > MIN_BITRATE)
+              min_bitrate = MIN_BITRATE;
+        if (max_bitrate > MAX_BITRATE)
+              max_bitrate = MAX_BITRATE;
+        /* Update the bitrate in the range  */
+        if (bitrate < min_bitrate)
+            updated_rate = min_bitrate;
+        else if(bitrate > max_bitrate)
+            updated_rate = max_bitrate;
+        else
+             updated_rate = bitrate;
+	return updated_rate;
+}
diff --git a/post_proc/bass_boost.c b/post_proc/bass_boost.c
index a925e8e..9e6c1fc 100644
--- a/post_proc/bass_boost.c
+++ b/post_proc/bass_boost.c
@@ -211,9 +211,14 @@
     bassboost_context_t *bass_ctxt = (bassboost_context_t *)context;
 
     ALOGV("%s", __func__);
-
-    if (!offload_bassboost_get_enable_flag(&(bass_ctxt->offload_bass)))
+    if (!offload_bassboost_get_enable_flag(&(bass_ctxt->offload_bass))) {
         offload_bassboost_set_enable_flag(&(bass_ctxt->offload_bass), true);
+        if (bass_ctxt->ctl && bass_ctxt->strength)
+            offload_bassboost_send_params(bass_ctxt->ctl,
+                                          bass_ctxt->offload_bass,
+                                          OFFLOAD_SEND_BASSBOOST_ENABLE_FLAG |
+                                          OFFLOAD_SEND_BASSBOOST_STRENGTH);
+    }
     return 0;
 }
 
diff --git a/post_proc/virtualizer.c b/post_proc/virtualizer.c
index 9682b93..4190129 100644
--- a/post_proc/virtualizer.c
+++ b/post_proc/virtualizer.c
@@ -210,9 +210,14 @@
     virtualizer_context_t *virt_ctxt = (virtualizer_context_t *)context;
 
     ALOGV("%s", __func__);
-
-    if (!offload_virtualizer_get_enable_flag(&(virt_ctxt->offload_virt)))
+    if (!offload_virtualizer_get_enable_flag(&(virt_ctxt->offload_virt))) {
         offload_virtualizer_set_enable_flag(&(virt_ctxt->offload_virt), true);
+        if (virt_ctxt->ctl && virt_ctxt->strength)
+            offload_virtualizer_send_params(virt_ctxt->ctl,
+                                          virt_ctxt->offload_virt,
+                                          OFFLOAD_SEND_VIRTUALIZER_ENABLE_FLAG |
+                                          OFFLOAD_SEND_BASSBOOST_STRENGTH);
+    }
     return 0;
 }