diff --git a/Android.mk b/Android.mk
index 9bb3250..c63df05 100644
--- a/Android.mk
+++ b/Android.mk
@@ -1,4 +1,4 @@
-ifneq ($(filter mpq8092 msm8960 msm8226 msm8x26 msm8610 msm8974 msm8x74 apq8084,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter mpq8092 msm8960 msm8226 msm8x26 msm8610 msm8974 msm8x74 apq8084 msm8916,$(TARGET_BOARD_PLATFORM)),)
 
 MY_LOCAL_PATH := $(call my-dir)
 
diff --git a/hal/Android.mk b/hal/Android.mk
index a7e0a02..d78d13b 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -23,6 +23,12 @@
 endif
 endif
 
+ifneq ($(filter msm8916,$(TARGET_BOARD_PLATFORM)),)
+  AUDIO_PLATFORM = msm8916
+  MULTIPLE_HW_VARIANTS_ENABLED := true
+  LOCAL_CFLAGS := -DPLATFORM_MSM8916
+endif
+
 LOCAL_SRC_FILES := \
 	audio_hw.c \
 	voice.c \
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 89903ba..ad487b1 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -326,9 +326,12 @@
                               struct str_parms *query,
                               struct str_parms *reply)
 {
+    char *kv_pairs = NULL;
     audio_extn_get_afe_proxy_parameters(query, reply);
 
-    ALOGD("%s: returns %s", __func__, str_parms_to_str(reply));
+    kv_pairs = str_parms_to_str(reply);
+    ALOGD_IF(kv_pairs != NULL, "%s: returns %s", __func__, kv_pairs);
+    free(kv_pairs);
 }
 
 #ifdef AUXPCM_BT_ENABLED
diff --git a/hal/audio_extn/listen.c b/hal/audio_extn/listen.c
index 9166f8e..4a1980b 100644
--- a/hal/audio_extn/listen.c
+++ b/hal/audio_extn/listen.c
@@ -119,10 +119,13 @@
 void audio_extn_listen_set_parameters(struct audio_device *adev,
                                struct str_parms *parms)
 {
-    ALOGV("%s: enter: %s", __func__, str_parms_to_str(parms));
 
+    ALOGV("%s: enter", __func__);
     if (listen_dev) {
-        listen_dev->listen_set_parameters(&adev->device, str_parms_to_str(parms));
+         char *kv_pairs = str_parms_to_str(parms);
+         ALOGV_IF(kv_pairs != NULL, "%s: %s", __func__, kv_pairs);
+         listen_dev->listen_set_parameters(&adev->device, kv_pairs);
+         free(kv_pairs);
     }
 
     return;
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 1b008db..1521a7b 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -53,8 +53,10 @@
 #include "voice_extn.h"
 
 #include "sound/compress_params.h"
+
 #define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
-#define MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE (8 * 1024)
+#define MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE (2 * 1024)
+#define COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING (2 * 1024)
 #define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
 #define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
 /* ToDo: Check and update a proper value in msec */
@@ -157,7 +159,7 @@
  * If value is not power of 2  round it to
  * power of 2.
  */
-static uint32_t get_offload_buffer_size()
+static uint32_t get_offload_buffer_size(audio_offload_info_t* info)
 {
     char value[PROPERTY_VALUE_MAX] = {0};
     uint32_t fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE;
@@ -167,6 +169,13 @@
         //ring buffer size needs to be 4k aligned.
         CHECK(!(fragment_size * COMPRESS_OFFLOAD_NUM_FRAGMENTS % 4096));
     }
+
+    if (info != NULL && info->has_video && info->is_streaming) {
+        fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
+        ALOGV("%s: offload fragment size reduced for AV streaming to %d",
+               __func__, out->compr_config.fragment_size);
+    }
+
     if(fragment_size < MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE)
         fragment_size = MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
     else if(fragment_size > MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE)
@@ -720,6 +729,15 @@
         disable_snd_device(adev, usecase->in_snd_device, false);
     }
 
+    /* Applicable only on the targets that has external modem.
+     * New device information should be sent to modem before enabling
+     * the devices to reduce in-call device switch time.
+     */
+    if (usecase->type == VOICE_CALL)
+        status = platform_switch_voice_call_enable_device_config(adev->platform,
+                                                                 out_snd_device,
+                                                                 in_snd_device);
+
     /* Enable new sound devices */
     if (out_snd_device != SND_DEVICE_NONE) {
         if (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)
@@ -1347,11 +1365,21 @@
     struct compr_gapless_mdata tmp_mdata;
     tmp_mdata.encoder_delay = 0;
     tmp_mdata.encoder_padding = 0;
+
     if (!out || !parms) {
         ALOGE("%s: return invalid ",__func__);
         return -EINVAL;
     }
 
+    ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_FORMAT, value, sizeof(value));
+    if (ret >= 0) {
+        if (atoi(value) == SND_AUDIOSTREAMFORMAT_MP4ADTS) {
+            out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4ADTS;
+            ALOGV("ADTS format is set in offload mode");
+        }
+        out->send_new_metadata = 1;
+    }
+
     ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_SAMPLE_RATE, value, sizeof(value));
     if(ret >= 0)
         is_meta_data_params = true;
@@ -1512,7 +1540,8 @@
         voice_extn_out_get_parameters(out, query, reply);
         str = str_parms_to_str(reply);
         if (!strncmp(str, "", sizeof(""))) {
-           str = strdup(keys);
+            free(str);
+            str = strdup(keys);
         }
     }
     str_parms_destroy(query);
@@ -1889,6 +1918,17 @@
         /* no audio source uses val == 0 */
         if ((in->source != val) && (val != 0)) {
             in->source = val;
+            if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
+                (in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
+                (voice_extn_compress_voip_is_format_supported(in->format)) &&
+                (in->config.rate == 8000 || in->config.rate == 16000) &&
+                (popcount(in->channel_mask) == 1)) {
+                err = voice_extn_compress_voip_open_input_stream(in);
+                if (err != 0) {
+                    ALOGE("%s: Compress voip input cannot be opened, error:%d",
+                          __func__, err);
+                }
+            }
         }
     }
 
@@ -1903,6 +1943,7 @@
         }
     }
 
+done:
     pthread_mutex_unlock(&adev->lock);
     pthread_mutex_unlock(&in->lock);
 
@@ -2146,7 +2187,7 @@
         else
             out->compr_config.codec->id =
                 get_snd_codec_id(config->offload_info.format);
-        out->compr_config.fragment_size = get_offload_buffer_size();
+        out->compr_config.fragment_size = get_offload_buffer_size(&config->offload_info);
         out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
         out->compr_config.codec->sample_rate =
                     compress_get_alsa_rate(config->offload_info.sample_rate);
@@ -2155,6 +2196,7 @@
         out->compr_config.codec->ch_in =
                     popcount(config->channel_mask);
         out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
+        out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
 
         if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
             out->non_blocking = 1;
@@ -2502,16 +2544,7 @@
     in->config.rate = config->sample_rate;
     in->format = config->format;
 
-    if ((in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
-        (voice_extn_compress_voip_is_config_supported(config))) {
-        ret = voice_extn_compress_voip_open_input_stream(in);
-        if (ret != 0)
-        {
-            ALOGE("%s: Compress voip input cannot be opened, error:%d",
-                  __func__, ret);
-            goto err_open;
-        }
-    } else if (channel_count == 6) {
+    if (channel_count == 6) {
         if(audio_extn_ssr_get_enabled()) {
             if(audio_extn_ssr_init(adev, in)) {
                 ALOGE("%s: audio_extn_ssr_init failed", __func__);
@@ -2523,7 +2556,8 @@
             goto err_open;
         }
     } else if (audio_extn_compr_cap_enabled() &&
-            audio_extn_compr_cap_format_supported(config->format)) {
+            audio_extn_compr_cap_format_supported(config->format) &&
+            (in->dev->mode != AUDIO_MODE_IN_COMMUNICATION)) {
         audio_extn_compr_cap_init(adev, in);
     } else {
         in->config.channels = channel_count;
diff --git a/hal/msm8916/hw_info.c b/hal/msm8916/hw_info.c
new file mode 100644
index 0000000..63506f9
--- /dev/null
+++ b/hal/msm8916/hw_info.c
@@ -0,0 +1,198 @@
+/*
+ * Copyright (c) 2013-2014, 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:
+ *   * Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ *   * Redistributions in binary form must reproduce the above
+ *     copyright notice, this list of conditions and the following
+ *     disclaimer in the documentation and/or other materials provided
+ *     with the distribution.
+ *   * Neither the name of The Linux Foundation nor the names of its
+ *     contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#define LOG_TAG "hardware_info"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+
+#include <stdlib.h>
+#include <dlfcn.h>
+#include <cutils/log.h>
+#include <cutils/str_parms.h>
+#include "audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+
+
+struct hardware_info {
+    char name[HW_INFO_ARRAY_MAX_SIZE];
+    char type[HW_INFO_ARRAY_MAX_SIZE];
+    /* variables for handling target variants */
+    uint32_t num_snd_devices;
+    char dev_extn[HW_INFO_ARRAY_MAX_SIZE];
+    snd_device_t  *snd_devices;
+};
+
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+static const snd_device_t taiko_fluid_variant_devices[] = {
+    SND_DEVICE_OUT_SPEAKER,
+    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+    SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
+};
+
+static const snd_device_t taiko_CDP_variant_devices[] = {
+    SND_DEVICE_OUT_SPEAKER,
+    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+    SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
+    SND_DEVICE_IN_QUAD_MIC,
+};
+
+static const snd_device_t taiko_apq8084_CDP_variant_devices[] = {
+    SND_DEVICE_IN_HANDSET_MIC,
+};
+
+static const snd_device_t taiko_liquid_variant_devices[] = {
+    SND_DEVICE_OUT_SPEAKER,
+    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+    SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
+    SND_DEVICE_IN_SPEAKER_MIC,
+    SND_DEVICE_IN_HEADSET_MIC,
+    SND_DEVICE_IN_VOICE_DMIC,
+    SND_DEVICE_IN_VOICE_SPEAKER_DMIC,
+    SND_DEVICE_IN_VOICE_REC_DMIC_STEREO,
+    SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE,
+    SND_DEVICE_IN_QUAD_MIC,
+    SND_DEVICE_IN_HANDSET_STEREO_DMIC,
+    SND_DEVICE_IN_SPEAKER_STEREO_DMIC,
+};
+
+static const snd_device_t taiko_DB_variant_devices[] = {
+    SND_DEVICE_OUT_SPEAKER,
+    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+    SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
+    SND_DEVICE_IN_SPEAKER_MIC,
+    SND_DEVICE_IN_HEADSET_MIC,
+    SND_DEVICE_IN_QUAD_MIC,
+};
+
+static const snd_device_t tapan_lite_variant_devices[] = {
+    SND_DEVICE_OUT_SPEAKER,
+    SND_DEVICE_OUT_HEADPHONES,
+    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+    SND_DEVICE_OUT_VOICE_HEADPHONES,
+    SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
+    SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
+};
+
+static const snd_device_t tapan_skuf_variant_devices[] = {
+    SND_DEVICE_OUT_SPEAKER,
+    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+    SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
+    /*SND_DEVICE_OUT_SPEAKER_AND_ANC_FB_HEADSET,*/
+};
+
+static const snd_device_t tapan_lite_skuf_variant_devices[] = {
+    SND_DEVICE_OUT_SPEAKER,
+    SND_DEVICE_OUT_HEADPHONES,
+    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+    SND_DEVICE_OUT_VOICE_HEADPHONES,
+    SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
+    SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
+};
+
+static const snd_device_t helicon_skuab_variant_devices[] = {
+    SND_DEVICE_OUT_SPEAKER,
+    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+    SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
+};
+
+static void update_hardware_info_8916(struct hardware_info *hw_info, const char *snd_card_name)
+{
+    if (!strcmp(snd_card_name, "msm8916-snd-card")) {
+        strlcpy(hw_info->type, "", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8916", 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, "msm8916-skuab-snd-card")) {
+        strlcpy(hw_info->type, "skuab", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8916", sizeof(hw_info->name));
+        hw_info->snd_devices = (snd_device_t *)helicon_skuab_variant_devices;
+        hw_info->num_snd_devices = ARRAY_SIZE(helicon_skuab_variant_devices);
+        strlcpy(hw_info->dev_extn, "-skuab", sizeof(hw_info->dev_extn));
+    } else if (!strcmp(snd_card_name, "msm8916-skuaa-snd-card")) {
+        strlcpy(hw_info->type, " skuaa", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8916", 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 {
+        ALOGW("%s: Not an  8916 device", __func__);
+    }
+}
+
+void *hw_info_init(const char *snd_card_name)
+{
+    struct hardware_info *hw_info;
+
+    hw_info = malloc(sizeof(struct hardware_info));
+
+    if(strstr(snd_card_name, "msm8916")) {
+        ALOGV("8916 - variant soundcard");
+        update_hardware_info_8916(hw_info, snd_card_name);
+    } else {
+        ALOGE("%s: Unsupported target %s:",__func__, snd_card_name);
+        free(hw_info);
+        hw_info = NULL;
+    }
+
+    return hw_info;
+}
+
+void hw_info_deinit(void *hw_info)
+{
+    struct hardware_info *my_data = (struct hardware_info*) hw_info;
+
+    if(!my_data)
+        free(my_data);
+}
+
+void hw_info_append_hw_type(void *hw_info, snd_device_t snd_device,
+                            char *device_name)
+{
+    struct hardware_info *my_data = (struct hardware_info*) hw_info;
+    uint32_t i = 0;
+
+    snd_device_t *snd_devices =
+            (snd_device_t *) my_data->snd_devices;
+
+    if(snd_devices != NULL) {
+        for (i = 0; i <  my_data->num_snd_devices; i++) {
+            if (snd_device == (snd_device_t)snd_devices[i]) {
+                ALOGV("extract dev_extn device %d, extn = %s",
+                        (snd_device_t)snd_devices[i],  my_data->dev_extn);
+                CHECK(strlcat(device_name,  my_data->dev_extn,
+                        DEVICE_NAME_MAX_SIZE) < DEVICE_NAME_MAX_SIZE);
+                break;
+            }
+        }
+    }
+    ALOGD("%s : device_name = %s", __func__,device_name);
+}
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
new file mode 100644
index 0000000..c011f64
--- /dev/null
+++ b/hal/msm8916/platform.c
@@ -0,0 +1,1718 @@
+/*
+ * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "msm8916_platform"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+
+#include <stdlib.h>
+#include <dlfcn.h>
+#include <cutils/log.h>
+#include <cutils/properties.h>
+#include <cutils/str_parms.h>
+#include <audio_hw.h>
+#include <platform_api.h>
+#include "platform.h"
+#include "audio_extn.h"
+#include "voice_extn.h"
+
+#define MIXER_XML_PATH "/system/etc/mixer_paths.xml"
+#define MIXER_XML_PATH_AUXPCM "/system/etc/mixer_paths_auxpcm.xml"
+#define LIB_ACDB_LOADER "libacdbloader.so"
+#define AUDIO_DATA_BLOCK_MIXER_CTL "HDMI EDID"
+
+/*
+ * This file will have a maximum of 38 bytes:
+ *
+ * 4 bytes: number of audio blocks
+ * 4 bytes: total length of Short Audio Descriptor (SAD) blocks
+ * Maximum 10 * 3 bytes: SAD blocks
+ */
+#define MAX_SAD_BLOCKS      10
+#define SAD_BLOCK_SIZE      3
+
+/* EDID format ID for LPCM audio */
+#define EDID_FORMAT_LPCM    1
+
+/* Retry for delay in FW loading*/
+#define RETRY_NUMBER 10
+#define RETRY_US 500000
+#define MAX_SND_CARD 8
+
+#define SAMPLE_RATE_8KHZ  8000
+#define SAMPLE_RATE_16KHZ 16000
+
+#define AUDIO_PARAMETER_KEY_FLUENCE_TYPE  "fluence"
+#define AUDIO_PARAMETER_KEY_BTSCO         "bt_samplerate"
+#define AUDIO_PARAMETER_KEY_SLOWTALK      "st_enable"
+#define AUDIO_PARAMETER_KEY_VOLUME_BOOST  "volume_boost"
+
+enum {
+	VOICE_FEATURE_SET_DEFAULT,
+	VOICE_FEATURE_SET_VOLUME_BOOST
+};
+
+struct audio_block_header
+{
+    int reserved;
+    int length;
+};
+
+/* Audio calibration related functions */
+typedef void (*acdb_deallocate_t)();
+typedef int  (*acdb_init_t)(char *);
+typedef void (*acdb_send_audio_cal_t)(int, int);
+typedef void (*acdb_send_voice_cal_t)(int, int);
+typedef int (*acdb_reload_vocvoltable_t)(int);
+
+struct platform_data {
+    struct audio_device *adev;
+    bool fluence_in_spkr_mode;
+    bool fluence_in_voice_call;
+    bool fluence_in_voice_rec;
+    bool fluence_in_audio_rec;
+    int  fluence_type;
+    int  btsco_sample_rate;
+    bool slowtalk;
+    /* Audio calibration related functions */
+    void                       *acdb_handle;
+    int                        voice_feature_set;
+    acdb_init_t                acdb_init;
+    acdb_deallocate_t          acdb_deallocate;
+    acdb_send_audio_cal_t      acdb_send_audio_cal;
+    acdb_send_voice_cal_t      acdb_send_voice_cal;
+    acdb_reload_vocvoltable_t  acdb_reload_vocvoltable;
+
+    void *hw_info;
+    struct csd_data *csd;
+};
+
+static const int pcm_device_table[AUDIO_USECASE_MAX][2] = {
+    [USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = {DEEP_BUFFER_PCM_DEVICE,
+                                            DEEP_BUFFER_PCM_DEVICE},
+    [USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = {LOWLATENCY_PCM_DEVICE,
+                                           LOWLATENCY_PCM_DEVICE},
+    [USECASE_AUDIO_PLAYBACK_MULTI_CH] = {MULTIMEDIA2_PCM_DEVICE,
+                                        MULTIMEDIA2_PCM_DEVICE},
+    [USECASE_AUDIO_PLAYBACK_OFFLOAD] =
+                     {PLAYBACK_OFFLOAD_DEVICE, PLAYBACK_OFFLOAD_DEVICE},
+    [USECASE_AUDIO_RECORD] = {AUDIO_RECORD_PCM_DEVICE, AUDIO_RECORD_PCM_DEVICE},
+    [USECASE_AUDIO_RECORD_COMPRESS] = {COMPRESS_CAPTURE_DEVICE, COMPRESS_CAPTURE_DEVICE},
+    [USECASE_AUDIO_RECORD_LOW_LATENCY] = {LOWLATENCY_PCM_DEVICE,
+                                          LOWLATENCY_PCM_DEVICE},
+    [USECASE_AUDIO_RECORD_FM_VIRTUAL] = {MULTIMEDIA2_PCM_DEVICE,
+                                  MULTIMEDIA2_PCM_DEVICE},
+    [USECASE_AUDIO_PLAYBACK_FM] = {FM_PLAYBACK_PCM_DEVICE, FM_CAPTURE_PCM_DEVICE},
+    [USECASE_AUDIO_HFP_SCO] = {HFP_PCM_RX, HFP_SCO_RX},
+    [USECASE_AUDIO_HFP_SCO_WB] = {HFP_PCM_RX, HFP_SCO_RX},
+    [USECASE_VOICE_CALL] = {VOICE_CALL_PCM_DEVICE, VOICE_CALL_PCM_DEVICE},
+    [USECASE_VOICE2_CALL] = {VOICE2_CALL_PCM_DEVICE, VOICE2_CALL_PCM_DEVICE},
+    [USECASE_VOLTE_CALL] = {VOLTE_CALL_PCM_DEVICE, VOLTE_CALL_PCM_DEVICE},
+    [USECASE_QCHAT_CALL] = {QCHAT_CALL_PCM_DEVICE, QCHAT_CALL_PCM_DEVICE},
+    [USECASE_COMPRESS_VOIP_CALL] = {COMPRESS_VOIP_CALL_PCM_DEVICE, COMPRESS_VOIP_CALL_PCM_DEVICE},
+    [USECASE_INCALL_REC_UPLINK] = {AUDIO_RECORD_PCM_DEVICE,
+                                   AUDIO_RECORD_PCM_DEVICE},
+    [USECASE_INCALL_REC_DOWNLINK] = {AUDIO_RECORD_PCM_DEVICE,
+                                     AUDIO_RECORD_PCM_DEVICE},
+    [USECASE_INCALL_REC_UPLINK_AND_DOWNLINK] = {AUDIO_RECORD_PCM_DEVICE,
+                                                AUDIO_RECORD_PCM_DEVICE},
+    [USECASE_INCALL_REC_UPLINK_COMPRESS] = {COMPRESS_CAPTURE_DEVICE,
+                                            COMPRESS_CAPTURE_DEVICE},
+    [USECASE_INCALL_REC_DOWNLINK_COMPRESS] = {COMPRESS_CAPTURE_DEVICE,
+                                              COMPRESS_CAPTURE_DEVICE},
+    [USECASE_INCALL_REC_UPLINK_AND_DOWNLINK_COMPRESS] = {COMPRESS_CAPTURE_DEVICE,
+                                                         COMPRESS_CAPTURE_DEVICE},
+    [USECASE_INCALL_MUSIC_UPLINK] = {INCALL_MUSIC_UPLINK_PCM_DEVICE,
+                                     INCALL_MUSIC_UPLINK_PCM_DEVICE},
+    [USECASE_INCALL_MUSIC_UPLINK2] = {INCALL_MUSIC_UPLINK2_PCM_DEVICE,
+                                      INCALL_MUSIC_UPLINK2_PCM_DEVICE},
+    [USECASE_AUDIO_SPKR_CALIB_RX] = {SPKR_PROT_CALIB_RX_PCM_DEVICE, -1},
+    [USECASE_AUDIO_SPKR_CALIB_TX] = {-1, SPKR_PROT_CALIB_TX_PCM_DEVICE},
+};
+
+/* Array to store sound devices */
+static const char * const device_table[SND_DEVICE_MAX] = {
+    [SND_DEVICE_NONE] = "none",
+    /* Playback sound devices */
+    [SND_DEVICE_OUT_HANDSET] = "handset",
+    [SND_DEVICE_OUT_SPEAKER] = "speaker",
+    [SND_DEVICE_OUT_SPEAKER_REVERSE] = "speaker-reverse",
+    [SND_DEVICE_OUT_HEADPHONES] = "headphones",
+    [SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = "speaker-and-headphones",
+    [SND_DEVICE_OUT_VOICE_HANDSET] = "voice-handset",
+    [SND_DEVICE_OUT_VOICE_SPEAKER] = "voice-speaker",
+    [SND_DEVICE_OUT_VOICE_HEADPHONES] = "voice-headphones",
+    [SND_DEVICE_OUT_HDMI] = "hdmi",
+    [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = "speaker-and-hdmi",
+    [SND_DEVICE_OUT_BT_SCO] = "bt-sco-headset",
+    [SND_DEVICE_OUT_BT_SCO_WB] = "bt-sco-headset-wb",
+    [SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = "voice-tty-full-headphones",
+    [SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = "voice-tty-vco-headphones",
+    [SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = "voice-tty-hco-handset",
+    [SND_DEVICE_OUT_AFE_PROXY] = "afe-proxy",
+    [SND_DEVICE_OUT_USB_HEADSET] = "usb-headphones",
+    [SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = "speaker-and-usb-headphones",
+    [SND_DEVICE_OUT_TRANSMISSION_FM] = "transmission-fm",
+    [SND_DEVICE_OUT_ANC_HEADSET] = "anc-headphones",
+    [SND_DEVICE_OUT_ANC_FB_HEADSET] = "anc-fb-headphones",
+    [SND_DEVICE_OUT_VOICE_ANC_HEADSET] = "voice-anc-headphones",
+    [SND_DEVICE_OUT_VOICE_ANC_FB_HEADSET] = "voice-anc-fb-headphones",
+    [SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET] = "speaker-and-anc-headphones",
+    [SND_DEVICE_OUT_ANC_HANDSET] = "anc-handset",
+    [SND_DEVICE_OUT_SPEAKER_PROTECTED] = "speaker-protected",
+
+    /* Capture sound devices */
+    [SND_DEVICE_IN_HANDSET_MIC] = "handset-mic",
+    [SND_DEVICE_IN_HANDSET_MIC_AEC] = "handset-mic",
+    [SND_DEVICE_IN_HANDSET_MIC_NS] = "handset-mic",
+    [SND_DEVICE_IN_HANDSET_MIC_AEC_NS] = "handset-mic",
+    [SND_DEVICE_IN_HANDSET_DMIC] = "dmic-endfire",
+    [SND_DEVICE_IN_HANDSET_DMIC_AEC] = "dmic-endfire",
+    [SND_DEVICE_IN_HANDSET_DMIC_NS] = "dmic-endfire",
+    [SND_DEVICE_IN_HANDSET_DMIC_AEC_NS] = "dmic-endfire",
+    [SND_DEVICE_IN_SPEAKER_MIC] = "speaker-mic",
+    [SND_DEVICE_IN_SPEAKER_MIC_AEC] = "speaker-mic",
+    [SND_DEVICE_IN_SPEAKER_MIC_NS] = "speaker-mic",
+    [SND_DEVICE_IN_SPEAKER_MIC_AEC_NS] = "speaker-mic",
+    [SND_DEVICE_IN_SPEAKER_DMIC] = "speaker-dmic-endfire",
+    [SND_DEVICE_IN_SPEAKER_DMIC_AEC] = "speaker-dmic-endfire",
+    [SND_DEVICE_IN_SPEAKER_DMIC_NS] = "speaker-dmic-endfire",
+    [SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS] = "speaker-dmic-endfire",
+    [SND_DEVICE_IN_HEADSET_MIC] = "headset-mic",
+    [SND_DEVICE_IN_HEADSET_MIC_FLUENCE] = "headset-mic",
+    [SND_DEVICE_IN_VOICE_SPEAKER_MIC] = "voice-speaker-mic",
+    [SND_DEVICE_IN_VOICE_HEADSET_MIC] = "voice-headset-mic",
+    [SND_DEVICE_IN_HDMI_MIC] = "hdmi-mic",
+    [SND_DEVICE_IN_BT_SCO_MIC] = "bt-sco-mic",
+    [SND_DEVICE_IN_BT_SCO_MIC_WB] = "bt-sco-mic-wb",
+    [SND_DEVICE_IN_CAMCORDER_MIC] = "camcorder-mic",
+    [SND_DEVICE_IN_VOICE_DMIC] = "voice-dmic-ef",
+    [SND_DEVICE_IN_VOICE_SPEAKER_DMIC] = "voice-speaker-dmic-ef",
+    [SND_DEVICE_IN_VOICE_SPEAKER_QMIC] = "voice-speaker-qmic",
+    [SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC] = "voice-tty-full-headset-mic",
+    [SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC] = "voice-tty-vco-handset-mic",
+    [SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC] = "voice-tty-hco-headset-mic",
+    [SND_DEVICE_IN_VOICE_REC_MIC] = "voice-rec-mic",
+    [SND_DEVICE_IN_VOICE_REC_MIC_NS] = "voice-rec-mic",
+    [SND_DEVICE_IN_VOICE_REC_DMIC_STEREO] = "voice-rec-dmic-ef",
+    [SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE] = "voice-rec-dmic-ef-fluence",
+    [SND_DEVICE_IN_USB_HEADSET_MIC] = "usb-headset-mic",
+    [SND_DEVICE_IN_CAPTURE_FM] = "capture-fm",
+    [SND_DEVICE_IN_AANC_HANDSET_MIC] = "aanc-handset-mic",
+    [SND_DEVICE_IN_QUAD_MIC] = "quad-mic",
+    [SND_DEVICE_IN_HANDSET_STEREO_DMIC] = "handset-stereo-dmic-ef",
+    [SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = "speaker-stereo-dmic-ef",
+    [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = "vi-feedback",
+};
+
+/* ACDB IDs (audio DSP path configuration IDs) for each sound device */
+static int acdb_device_table[SND_DEVICE_MAX] = {
+    [SND_DEVICE_NONE] = -1,
+    [SND_DEVICE_OUT_HANDSET] = 7,
+    [SND_DEVICE_OUT_SPEAKER] = 14,
+    [SND_DEVICE_OUT_SPEAKER_REVERSE] = 14,
+    [SND_DEVICE_OUT_HEADPHONES] = 10,
+    [SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = 10,
+    [SND_DEVICE_OUT_VOICE_HANDSET] = 7,
+    [SND_DEVICE_OUT_VOICE_SPEAKER] = 14,
+    [SND_DEVICE_OUT_VOICE_HEADPHONES] = 10,
+    [SND_DEVICE_OUT_HDMI] = 18,
+    [SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 14,
+    [SND_DEVICE_OUT_BT_SCO] = 22,
+    [SND_DEVICE_OUT_BT_SCO_WB] = 39,
+    [SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES] = 17,
+    [SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES] = 17,
+    [SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = 37,
+    [SND_DEVICE_OUT_AFE_PROXY] = 0,
+    [SND_DEVICE_OUT_USB_HEADSET] = 45,
+    [SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = 14,
+    [SND_DEVICE_OUT_TRANSMISSION_FM] = 0,
+    [SND_DEVICE_OUT_ANC_HEADSET] = 26,
+    [SND_DEVICE_OUT_ANC_FB_HEADSET] = 27,
+    [SND_DEVICE_OUT_VOICE_ANC_HEADSET] = 26,
+    [SND_DEVICE_OUT_VOICE_ANC_FB_HEADSET] = 27,
+    [SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET] = 26,
+    [SND_DEVICE_OUT_ANC_HANDSET] = 103,
+    [SND_DEVICE_OUT_SPEAKER_PROTECTED] = 101,
+
+    [SND_DEVICE_IN_HANDSET_MIC] = 4,
+    [SND_DEVICE_IN_HANDSET_MIC_AEC] = 106,
+    [SND_DEVICE_IN_HANDSET_MIC_NS] = 107,
+    [SND_DEVICE_IN_HANDSET_MIC_AEC_NS] = 108,
+    [SND_DEVICE_IN_HANDSET_DMIC] = 41,
+    [SND_DEVICE_IN_HANDSET_DMIC_AEC] = 109,
+    [SND_DEVICE_IN_HANDSET_DMIC_NS] = 110,
+    [SND_DEVICE_IN_HANDSET_DMIC_AEC_NS] = 111,
+    [SND_DEVICE_IN_SPEAKER_MIC] = 11,
+    [SND_DEVICE_IN_SPEAKER_MIC_AEC] = 112,
+    [SND_DEVICE_IN_SPEAKER_MIC_NS] = 113,
+    [SND_DEVICE_IN_SPEAKER_MIC_AEC_NS] = 114,
+    [SND_DEVICE_IN_SPEAKER_DMIC] = 43,
+    [SND_DEVICE_IN_SPEAKER_DMIC_AEC] = 115,
+    [SND_DEVICE_IN_SPEAKER_DMIC_NS] = 116,
+    [SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS] = 117,
+    [SND_DEVICE_IN_HEADSET_MIC] = 8,
+    [SND_DEVICE_IN_HEADSET_MIC_FLUENCE] = 47,
+    [SND_DEVICE_IN_VOICE_SPEAKER_MIC] = 11,
+    [SND_DEVICE_IN_VOICE_HEADSET_MIC] = 8,
+    [SND_DEVICE_IN_HDMI_MIC] = 4,
+    [SND_DEVICE_IN_BT_SCO_MIC] = 21,
+    [SND_DEVICE_IN_BT_SCO_MIC_WB] = 38,
+    [SND_DEVICE_IN_CAMCORDER_MIC] = 4,
+    [SND_DEVICE_IN_VOICE_DMIC] = 41,
+    [SND_DEVICE_IN_VOICE_SPEAKER_DMIC] = 43,
+    [SND_DEVICE_IN_VOICE_SPEAKER_QMIC] = 19,
+    [SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC] = 16,
+    [SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC] = 36,
+    [SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC] = 16,
+    [SND_DEVICE_IN_VOICE_REC_MIC] = 4,
+    [SND_DEVICE_IN_VOICE_REC_MIC_NS] = 107,
+    [SND_DEVICE_IN_VOICE_REC_DMIC_STEREO] = 34,
+    [SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE] = 41,
+    [SND_DEVICE_IN_USB_HEADSET_MIC] = 44,
+    [SND_DEVICE_IN_CAPTURE_FM] = 0,
+    [SND_DEVICE_IN_AANC_HANDSET_MIC] = 104,
+    [SND_DEVICE_IN_QUAD_MIC] = 46,
+    [SND_DEVICE_IN_HANDSET_STEREO_DMIC] = 34,
+    [SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = 35,
+    [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = 102,
+};
+
+struct snd_device_index {
+    char name[100];
+    unsigned int index;
+};
+
+#define TO_NAME_INDEX(X)   #X, X
+
+/* Used to get index from parsed sting */
+struct snd_device_index snd_device_name_index[SND_DEVICE_MAX] = {
+    {TO_NAME_INDEX(SND_DEVICE_OUT_HANDSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_REVERSE)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HANDSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HEADPHONES)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_HDMI)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HDMI)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_BT_SCO_WB)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_AFE_PROXY)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_USB_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_TRANSMISSION_FM)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_ANC_FB_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_ANC_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_ANC_FB_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HANDSET)},
+    {TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_AEC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_AEC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_DMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_DMIC_AEC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_DMIC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_DMIC_AEC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_MIC_AEC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_MIC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_MIC_AEC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_AEC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HEADSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HEADSET_MIC_FLUENCE)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_HEADSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HDMI_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_BT_SCO_MIC_WB)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_CAMCORDER_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_DMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_DMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_QMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_MIC_NS)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_DMIC_STEREO)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_USB_HEADSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_FM)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_AANC_HANDSET_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_QUAD_MIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_STEREO_DMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_STEREO_DMIC)},
+    {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK)},
+};
+
+#define DEEP_BUFFER_PLATFORM_DELAY (29*1000LL)
+#define LOW_LATENCY_PLATFORM_DELAY (13*1000LL)
+
+static int set_echo_reference(struct mixer *mixer, const char* ec_ref)
+{
+    struct mixer_ctl *ctl;
+    const char *mixer_ctl_name = "EC_REF_RX";
+
+    ctl = mixer_get_ctl_by_name(mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer cmd - %s",
+              __func__, mixer_ctl_name);
+        return -EINVAL;
+    }
+    ALOGV("Setting EC Reference: %s", ec_ref);
+    mixer_ctl_set_enum_by_string(ctl, ec_ref);
+    return 0;
+}
+
+static struct csd_data *open_csd_client()
+{
+    struct csd_data *csd = calloc(1, sizeof(struct csd_data));
+
+    csd->csd_client = dlopen(LIB_CSD_CLIENT, RTLD_NOW);
+    if (csd->csd_client == NULL) {
+        ALOGE("%s: DLOPEN failed for %s", __func__, LIB_CSD_CLIENT);
+        goto error;
+    } else {
+        ALOGV("%s: DLOPEN successful for %s", __func__, LIB_CSD_CLIENT);
+
+        csd->deinit = (deinit_t)dlsym(csd->csd_client,
+                                             "csd_client_deinit");
+        if (csd->deinit == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_deinit", __func__,
+                  dlerror());
+            goto error;
+        }
+        csd->disable_device = (disable_device_t)dlsym(csd->csd_client,
+                                             "csd_client_disable_device");
+        if (csd->disable_device == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_disable_device",
+                  __func__, dlerror());
+            goto error;
+        }
+        csd->enable_device_config = (enable_device_config_t)dlsym(csd->csd_client,
+                                               "csd_client_enable_device_config");
+        if (csd->enable_device_config == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_enable_device_config",
+                  __func__, dlerror());
+            goto error;
+        }
+        csd->enable_device = (enable_device_t)dlsym(csd->csd_client,
+                                             "csd_client_enable_device");
+        if (csd->enable_device == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_enable_device",
+                  __func__, dlerror());
+            goto error;
+        }
+        csd->start_voice = (start_voice_t)dlsym(csd->csd_client,
+                                             "csd_client_start_voice");
+        if (csd->start_voice == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_start_voice",
+                  __func__, dlerror());
+            goto error;
+        }
+        csd->stop_voice = (stop_voice_t)dlsym(csd->csd_client,
+                                             "csd_client_stop_voice");
+        if (csd->stop_voice == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_stop_voice",
+                  __func__, dlerror());
+            goto error;
+        }
+        csd->volume = (volume_t)dlsym(csd->csd_client,
+                                             "csd_client_volume");
+        if (csd->volume == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_volume",
+                  __func__, dlerror());
+            goto error;
+        }
+        csd->mic_mute = (mic_mute_t)dlsym(csd->csd_client,
+                                             "csd_client_mic_mute");
+        if (csd->mic_mute == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_mic_mute",
+                  __func__, dlerror());
+            goto error;
+        }
+        csd->slow_talk = (slow_talk_t)dlsym(csd->csd_client,
+                                             "csd_client_slow_talk");
+        if (csd->slow_talk == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_slow_talk",
+                  __func__, dlerror());
+            goto error;
+        }
+        csd->start_playback = (start_playback_t)dlsym(csd->csd_client,
+                                             "csd_client_start_playback");
+        if (csd->start_playback == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_start_playback",
+                  __func__, dlerror());
+            goto error;
+        }
+        csd->stop_playback = (stop_playback_t)dlsym(csd->csd_client,
+                                             "csd_client_stop_playback");
+        if (csd->stop_playback == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_stop_playback",
+                  __func__, dlerror());
+            goto error;
+        }
+        csd->start_record = (start_record_t)dlsym(csd->csd_client,
+                                             "csd_client_start_record");
+        if (csd->start_record == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_start_record",
+                  __func__, dlerror());
+            goto error;
+        }
+        csd->stop_record = (stop_record_t)dlsym(csd->csd_client,
+                                             "csd_client_stop_record");
+        if (csd->stop_record == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_stop_record",
+                  __func__, dlerror());
+            goto error;
+        }
+        csd->init = (init_t)dlsym(csd->csd_client, "csd_client_init");
+
+        if (csd->init == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_init",
+                  __func__, dlerror());
+            goto error;
+        } else {
+            csd->init();
+        }
+    }
+    return csd;
+
+error:
+    free(csd);
+    csd = NULL;
+    return csd;
+}
+
+void close_csd_client(struct csd_data *csd)
+{
+    if (csd != NULL) {
+        csd->deinit();
+        dlclose(csd->csd_client);
+        free(csd);
+        csd = NULL;
+    }
+}
+
+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;
+    const char *snd_card_name;
+
+    my_data = calloc(1, sizeof(struct platform_data));
+
+    while (snd_card_num < MAX_SND_CARD) {
+        adev->mixer = mixer_open(snd_card_num);
+
+        while (!adev->mixer && retry_num < RETRY_NUMBER) {
+            usleep(RETRY_US);
+            adev->mixer = mixer_open(snd_card_num);
+            retry_num++;
+        }
+
+        if (!adev->mixer) {
+            ALOGE("%s: Unable to open the mixer card: %d", __func__,
+                   snd_card_num);
+            retry_num = 0;
+            snd_card_num++;
+            continue;
+        }
+
+        snd_card_name = mixer_get_name(adev->mixer);
+        ALOGV("%s: snd_card_name: %s", __func__, snd_card_name);
+
+        my_data->hw_info = hw_info_init(snd_card_name);
+        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)
+                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__);
+                free(my_data);
+                return NULL;
+            }
+            adev->snd_card = snd_card_num;
+            ALOGD("%s: Opened sound card:%d", __func__, snd_card_num);
+            break;
+        }
+        retry_num = 0;
+        snd_card_num++;
+    }
+
+    if (snd_card_num >= MAX_SND_CARD) {
+        ALOGE("%s: Unable to find correct sound card, aborting.", __func__);
+        free(my_data);
+        return NULL;
+    }
+
+    my_data->adev = adev;
+    my_data->btsco_sample_rate = SAMPLE_RATE_8KHZ;
+    my_data->fluence_in_spkr_mode = false;
+    my_data->fluence_in_voice_call = false;
+    my_data->fluence_in_voice_rec = false;
+    my_data->fluence_in_audio_rec = false;
+    my_data->fluence_type = FLUENCE_NONE;
+
+    property_get("ro.qc.sdk.audio.fluencetype", value, "");
+    if (!strncmp("fluencepro", value, sizeof("fluencepro"))) {
+        my_data->fluence_type = FLUENCE_QUAD_MIC | FLUENCE_DUAL_MIC;
+    } else if (!strncmp("fluence", value, sizeof("fluence"))) {
+        my_data->fluence_type = FLUENCE_DUAL_MIC;
+    } else {
+        my_data->fluence_type = FLUENCE_NONE;
+    }
+
+    if (my_data->fluence_type != FLUENCE_NONE) {
+        property_get("persist.audio.fluence.voicecall",value,"");
+        if (!strncmp("true", value, sizeof("true"))) {
+            my_data->fluence_in_voice_call = true;
+        }
+
+        property_get("persist.audio.fluence.voicerec",value,"");
+        if (!strncmp("true", value, sizeof("true"))) {
+            my_data->fluence_in_voice_rec = true;
+        }
+
+        property_get("persist.audio.fluence.audiorec",value,"");
+        if (!strncmp("true", value, sizeof("true"))) {
+            my_data->fluence_in_audio_rec = true;
+        }
+
+        property_get("persist.audio.fluence.speaker",value,"");
+        if (!strncmp("true", value, sizeof("true"))) {
+            my_data->fluence_in_spkr_mode = true;
+        }
+    }
+
+    my_data->voice_feature_set = VOICE_FEATURE_SET_DEFAULT;
+    my_data->acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
+    if (my_data->acdb_handle == NULL) {
+        ALOGE("%s: DLOPEN failed for %s", __func__, LIB_ACDB_LOADER);
+    } else {
+        ALOGV("%s: DLOPEN successful for %s", __func__, LIB_ACDB_LOADER);
+        my_data->acdb_deallocate = (acdb_deallocate_t)dlsym(my_data->acdb_handle,
+                                                    "acdb_loader_deallocate_ACDB");
+        if (!my_data->acdb_deallocate)
+            ALOGE("%s: Could not find the symbol acdb_loader_deallocate_ACDB from %s",
+                  __func__, LIB_ACDB_LOADER);
+
+        my_data->acdb_send_audio_cal = (acdb_send_audio_cal_t)dlsym(my_data->acdb_handle,
+                                                    "acdb_loader_send_audio_cal");
+        if (!my_data->acdb_send_audio_cal)
+            ALOGE("%s: Could not find the symbol acdb_send_audio_cal from %s",
+                  __func__, LIB_ACDB_LOADER);
+
+        my_data->acdb_send_voice_cal = (acdb_send_voice_cal_t)dlsym(my_data->acdb_handle,
+                                                    "acdb_loader_send_voice_cal");
+        if (!my_data->acdb_send_voice_cal)
+            ALOGE("%s: Could not find the symbol acdb_loader_send_voice_cal from %s",
+                  __func__, LIB_ACDB_LOADER);
+
+        my_data->acdb_reload_vocvoltable = (acdb_reload_vocvoltable_t)dlsym(my_data->acdb_handle,
+                                                    "acdb_loader_reload_vocvoltable");
+        if (!my_data->acdb_reload_vocvoltable)
+            ALOGE("%s: Could not find the symbol acdb_loader_reload_vocvoltable from %s",
+                  __func__, LIB_ACDB_LOADER);
+
+        my_data->acdb_init = (acdb_init_t)dlsym(my_data->acdb_handle,
+                                                    "acdb_loader_init_v2");
+        if (my_data->acdb_init == NULL)
+            ALOGE("%s: dlsym error %s for acdb_loader_init_v2", __func__, dlerror());
+        else
+            my_data->acdb_init(snd_card_name);
+    }
+
+    /* Initialize ACDB ID's */
+    platform_info_init();
+
+    /* init usb */
+    audio_extn_usb_init(adev);
+    /* update sound cards appropriately */
+    audio_extn_usb_set_proxy_sound_card(adev->snd_card);
+
+    /* Read one time ssr property */
+    audio_extn_ssr_update_enabled();
+    audio_extn_spkr_prot_init(adev);
+    return my_data;
+}
+
+void platform_deinit(void *platform)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+
+    hw_info_deinit(my_data->hw_info);
+    close_csd_client(my_data->csd);
+
+    free(platform);
+    /* deinit usb */
+    audio_extn_usb_deinit();
+}
+
+const char *platform_get_snd_device_name(snd_device_t snd_device)
+{
+    if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX)
+        return device_table[snd_device];
+    else
+        return "";
+}
+
+int platform_get_snd_device_name_extn(void *platform, snd_device_t snd_device,
+                                      char *device_name)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+
+    if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX) {
+        strlcpy(device_name, device_table[snd_device], DEVICE_NAME_MAX_SIZE);
+        hw_info_append_hw_type(my_data->hw_info, snd_device, device_name);
+    } else {
+        strlcpy(device_name, "", DEVICE_NAME_MAX_SIZE);
+        return -EINVAL;
+    }
+
+    return 0;
+}
+
+void platform_add_backend_name(char *mixer_path, snd_device_t snd_device)
+{
+    if (snd_device == SND_DEVICE_IN_BT_SCO_MIC)
+        strlcat(mixer_path, " bt-sco", MIXER_PATH_MAX_LENGTH);
+    else if (snd_device == SND_DEVICE_IN_BT_SCO_MIC_WB)
+        strlcat(mixer_path, " bt-sco-wb", MIXER_PATH_MAX_LENGTH);
+    else if(snd_device == SND_DEVICE_OUT_BT_SCO)
+        strlcat(mixer_path, " bt-sco", MIXER_PATH_MAX_LENGTH);
+    else if(snd_device == SND_DEVICE_OUT_BT_SCO_WB)
+        strlcat(mixer_path, " bt-sco-wb", MIXER_PATH_MAX_LENGTH);
+    else if (snd_device == SND_DEVICE_OUT_HDMI)
+        strlcat(mixer_path, " hdmi", MIXER_PATH_MAX_LENGTH);
+    else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_HDMI)
+        strlcat(mixer_path, " speaker-and-hdmi", MIXER_PATH_MAX_LENGTH);
+    else if (snd_device == SND_DEVICE_OUT_AFE_PROXY)
+        strlcat(mixer_path, " afe-proxy", MIXER_PATH_MAX_LENGTH);
+    else if (snd_device == SND_DEVICE_OUT_USB_HEADSET)
+        strlcat(mixer_path, " usb-headphones", MIXER_PATH_MAX_LENGTH);
+    else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET)
+        strlcat(mixer_path, " speaker-and-usb-headphones",
+                MIXER_PATH_MAX_LENGTH);
+    else if (snd_device == SND_DEVICE_IN_USB_HEADSET_MIC)
+        strlcat(mixer_path, " usb-headset-mic", MIXER_PATH_MAX_LENGTH);
+    else if (snd_device == SND_DEVICE_IN_CAPTURE_FM)
+        strlcat(mixer_path, " capture-fm", MIXER_PATH_MAX_LENGTH);
+    else if (snd_device == SND_DEVICE_OUT_TRANSMISSION_FM)
+        strlcat(mixer_path, " transmission-fm", MIXER_PATH_MAX_LENGTH);
+}
+
+int platform_get_pcm_device_id(audio_usecase_t usecase, int device_type)
+{
+    int device_id;
+    if (device_type == PCM_PLAYBACK)
+        device_id = pcm_device_table[usecase][0];
+    else
+        device_id = pcm_device_table[usecase][1];
+    return device_id;
+}
+
+int platform_get_snd_device_index(char *snd_device_index_name)
+{
+    int ret = 0;
+    int i;
+
+    if (snd_device_index_name == NULL) {
+        ALOGE("%s: snd_device_index_name is NULL", __func__);
+        ret = -ENODEV;
+        goto done;
+    }
+
+    for (i=0; i < SND_DEVICE_MAX; i++) {
+        if(strcmp(snd_device_name_index[i].name, snd_device_index_name) == 0) {
+            ret = snd_device_name_index[i].index;
+            goto done;
+        }
+    }
+    ALOGE("%s: Could not find index for snd_device_index_name = %s",
+            __func__, snd_device_index_name);
+    ret = -ENODEV;
+done:
+    return ret;
+}
+
+int platform_set_snd_device_acdb_id(snd_device_t snd_device, unsigned int acdb_id)
+{
+    int ret = 0;
+
+    if ((snd_device < SND_DEVICE_MIN) || (snd_device >= SND_DEVICE_MAX)) {
+        ALOGE("%s: Invalid snd_device = %d",
+            __func__, snd_device);
+        ret = -EINVAL;
+        goto done;
+    }
+
+    acdb_device_table[snd_device] = acdb_id;
+done:
+    return ret;
+}
+
+int platform_send_audio_calibration(void *platform, snd_device_t snd_device)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    int acdb_dev_id, acdb_dev_type;
+
+    acdb_dev_id = acdb_device_table[snd_device];
+    if (acdb_dev_id < 0) {
+        ALOGE("%s: Could not find acdb id for device(%d)",
+              __func__, snd_device);
+        return -EINVAL;
+    }
+    if (my_data->acdb_send_audio_cal) {
+        ("%s: sending audio calibration for snd_device(%d) acdb_id(%d)",
+              __func__, snd_device, acdb_dev_id);
+        if (snd_device >= SND_DEVICE_OUT_BEGIN &&
+                snd_device < SND_DEVICE_OUT_END)
+            acdb_dev_type = ACDB_DEV_TYPE_OUT;
+        else
+            acdb_dev_type = ACDB_DEV_TYPE_IN;
+        my_data->acdb_send_audio_cal(acdb_dev_id, acdb_dev_type);
+    }
+    return 0;
+}
+
+int platform_switch_voice_call_device_pre(void *platform)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    int ret = 0;
+
+    if (my_data->csd != NULL &&
+        my_data->adev->mode == AUDIO_MODE_IN_CALL) {
+        /* This must be called before disabling mixer controls on APQ side */
+        ret = my_data->csd->disable_device();
+        if (ret < 0) {
+            ALOGE("%s: csd_client_disable_device, failed, error %d",
+                  __func__, ret);
+        }
+    }
+    return ret;
+}
+int platform_switch_voice_call_enable_device_config(void *platform,
+                                                    snd_device_t out_snd_device,
+                                                    snd_device_t in_snd_device)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    int acdb_rx_id, acdb_tx_id;
+    int ret = 0;
+
+    acdb_rx_id = acdb_device_table[out_snd_device];
+    acdb_tx_id = acdb_device_table[in_snd_device];
+
+    if (my_data->csd != NULL) {
+        if (acdb_rx_id > 0 && acdb_tx_id > 0) {
+            ret = my_data->csd->enable_device_config(acdb_rx_id, acdb_tx_id);
+            if (ret < 0) {
+                ALOGE("%s: csd_enable_device_config, failed, error %d",
+                      __func__, ret);
+            }
+        } else {
+            ALOGE("%s: Incorrect ACDB IDs (rx: %d tx: %d)", __func__,
+                  acdb_rx_id, acdb_tx_id);
+        }
+    }
+    return ret;
+}
+
+
+int platform_switch_voice_call_device_post(void *platform,
+                                           snd_device_t out_snd_device,
+                                           snd_device_t in_snd_device)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    int acdb_rx_id, acdb_tx_id;
+
+    if (my_data->acdb_send_voice_cal == NULL) {
+        ALOGE("%s: dlsym error for acdb_send_voice_call", __func__);
+    } else {
+        acdb_rx_id = acdb_device_table[out_snd_device];
+        acdb_tx_id = acdb_device_table[in_snd_device];
+
+        if (acdb_rx_id > 0 && acdb_tx_id > 0)
+            my_data->acdb_send_voice_cal(acdb_rx_id, acdb_tx_id);
+        else
+            ALOGE("%s: Incorrect ACDB IDs (rx: %d tx: %d)", __func__,
+                  acdb_rx_id, acdb_tx_id);
+    }
+
+    return 0;
+}
+
+int platform_switch_voice_call_usecase_route_post(void *platform,
+                                                  snd_device_t out_snd_device,
+                                                  snd_device_t in_snd_device)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    int acdb_rx_id, acdb_tx_id;
+    int ret = 0;
+
+    acdb_rx_id = acdb_device_table[out_snd_device];
+    acdb_tx_id = acdb_device_table[in_snd_device];
+
+    if (my_data->csd != NULL) {
+        if (acdb_rx_id > 0 && acdb_tx_id > 0) {
+            ret = my_data->csd->enable_device(acdb_rx_id, acdb_tx_id,
+                                              my_data->adev->acdb_settings);
+            if (ret < 0) {
+                ALOGE("%s: csd_enable_device, failed, error %d",
+                      __func__, ret);
+            }
+        } else {
+            ALOGE("%s: Incorrect ACDB IDs (rx: %d tx: %d)", __func__,
+                  acdb_rx_id, acdb_tx_id);
+        }
+    }
+    return ret;
+}
+
+int platform_start_voice_call(void *platform, uint32_t vsid)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    int ret = 0;
+
+    if (my_data->csd != NULL) {
+        ret = my_data->csd->start_voice(vsid);
+        if (ret < 0) {
+            ALOGE("%s: csd_start_voice error %d\n", __func__, ret);
+        }
+    }
+    return ret;
+}
+
+int platform_stop_voice_call(void *platform, uint32_t vsid)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    int ret = 0;
+
+    if (my_data->csd != NULL) {
+        ret = my_data->csd->stop_voice(vsid);
+        if (ret < 0) {
+            ALOGE("%s: csd_stop_voice 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;
+    struct audio_device *adev = my_data->adev;
+    struct mixer_ctl *ctl;
+    const char *mixer_ctl_name = "Voice Rx Gain";
+    int vol_index = 0, ret = 0;
+    uint32_t set_values[ ] = {0,
+                              ALL_SESSION_VSID,
+                              DEFAULT_VOLUME_RAMP_DURATION_MS};
+
+    // Voice volume levels are mapped to adsp volume levels as follows.
+    // 100 -> 5, 80 -> 4, 60 -> 3, 40 -> 2, 20 -> 1  0 -> 0
+    // But this values don't changed in kernel. So, below change is need.
+    vol_index = (int)percent_to_index(volume, MIN_VOL_INDEX, MAX_VOL_INDEX);
+    set_values[0] = vol_index;
+
+    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer cmd - %s",
+              __func__, mixer_ctl_name);
+        return -EINVAL;
+    }
+    ALOGV("Setting voice volume index: %d", set_values[0]);
+    mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
+
+    if (my_data->csd != NULL) {
+        ret = my_data->csd->volume(ALL_SESSION_VSID, volume);
+        if (ret < 0) {
+            ALOGE("%s: csd_volume error %d", __func__, ret);
+        }
+    }
+    return ret;
+}
+
+int platform_set_mic_mute(void *platform, bool state)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    struct audio_device *adev = my_data->adev;
+    struct mixer_ctl *ctl;
+    const char *mixer_ctl_name = "Voice Tx Mute";
+    int ret = 0;
+    uint32_t set_values[ ] = {0,
+                              ALL_SESSION_VSID,
+                              DEFAULT_VOLUME_RAMP_DURATION_MS};
+
+    set_values[0] = state;
+    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer cmd - %s",
+              __func__, mixer_ctl_name);
+        return -EINVAL;
+    }
+    ALOGV("Setting voice mute state: %d", state);
+    mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
+
+    if (my_data->csd != NULL) {
+        ret = my_data->csd->mic_mute(ALL_SESSION_VSID, state);
+        if (ret < 0) {
+            ALOGE("%s: csd_mic_mute error %d", __func__, ret);
+        }
+    }
+    return ret;
+}
+
+snd_device_t platform_get_output_snd_device(void *platform, audio_devices_t devices)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    struct audio_device *adev = my_data->adev;
+    audio_mode_t mode = adev->mode;
+    snd_device_t snd_device = SND_DEVICE_NONE;
+
+    audio_channel_mask_t channel_mask = (adev->active_input == NULL) ?
+                                AUDIO_CHANNEL_IN_MONO : adev->active_input->channel_mask;
+    int channel_count = popcount(channel_mask);
+
+    ALOGV("%s: enter: output devices(%#x)", __func__, devices);
+    if (devices == AUDIO_DEVICE_NONE ||
+        devices & AUDIO_DEVICE_BIT_IN) {
+        ALOGV("%s: Invalid output devices (%#x)", __func__, devices);
+        goto exit;
+    }
+
+    if (popcount(devices) == 2) {
+        if (devices == (AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
+                        AUDIO_DEVICE_OUT_SPEAKER)) {
+            snd_device = SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES;
+        } else if (devices == (AUDIO_DEVICE_OUT_WIRED_HEADSET |
+                               AUDIO_DEVICE_OUT_SPEAKER)) {
+            if (audio_extn_get_anc_enabled())
+                snd_device = SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET;
+            else
+                snd_device = SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES;
+        } else if (devices == (AUDIO_DEVICE_OUT_AUX_DIGITAL |
+                               AUDIO_DEVICE_OUT_SPEAKER)) {
+            snd_device = SND_DEVICE_OUT_SPEAKER_AND_HDMI;
+        } else if (devices == (AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET |
+                               AUDIO_DEVICE_OUT_SPEAKER)) {
+            snd_device = SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET;
+        } else {
+            ALOGE("%s: Invalid combo device(%#x)", __func__, devices);
+            goto exit;
+        }
+        if (snd_device != SND_DEVICE_NONE) {
+            goto exit;
+        }
+    }
+
+    if (popcount(devices) != 1) {
+        ALOGE("%s: Invalid output devices(%#x)", __func__, devices);
+        goto exit;
+    }
+
+    if ((mode == AUDIO_MODE_IN_CALL) ||
+        voice_extn_compress_voip_is_active(adev)) {
+        if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+            devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+            if ((adev->voice.tty_mode != TTY_MODE_OFF) &&
+                !voice_extn_compress_voip_is_active(adev)) {
+                switch (adev->voice.tty_mode) {
+                case TTY_MODE_FULL:
+                    snd_device = SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES;
+                    break;
+                case TTY_MODE_VCO:
+                    snd_device = SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES;
+                    break;
+                case TTY_MODE_HCO:
+                    snd_device = SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET;
+                    break;
+                default:
+                    ALOGE("%s: Invalid TTY mode (%#x)",
+                          __func__, adev->voice.tty_mode);
+                }
+            } else if (audio_extn_get_anc_enabled()) {
+                if (audio_extn_should_use_fb_anc())
+                    snd_device = SND_DEVICE_OUT_VOICE_ANC_FB_HEADSET;
+                else
+                    snd_device = SND_DEVICE_OUT_VOICE_ANC_HEADSET;
+            } else {
+                snd_device = SND_DEVICE_OUT_VOICE_HEADPHONES;
+            }
+        } else if (devices & AUDIO_DEVICE_OUT_ALL_SCO) {
+            if (my_data->btsco_sample_rate == SAMPLE_RATE_16KHZ)
+                snd_device = SND_DEVICE_OUT_BT_SCO_WB;
+            else
+                snd_device = SND_DEVICE_OUT_BT_SCO;
+        } else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
+            snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+        } else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
+                   devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
+            snd_device = SND_DEVICE_OUT_USB_HEADSET;
+        } else if (devices & AUDIO_DEVICE_OUT_FM_TX) {
+            snd_device = SND_DEVICE_OUT_TRANSMISSION_FM;
+        } else if (devices & AUDIO_DEVICE_OUT_EARPIECE) {
+            if (audio_extn_should_use_handset_anc(channel_count))
+                snd_device = SND_DEVICE_OUT_ANC_HANDSET;
+            else
+                snd_device = SND_DEVICE_OUT_VOICE_HANDSET;
+        }
+        if (snd_device != SND_DEVICE_NONE) {
+            goto exit;
+        }
+    }
+
+    if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+        devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+        if (devices & AUDIO_DEVICE_OUT_WIRED_HEADSET
+            && audio_extn_get_anc_enabled()) {
+            if (audio_extn_should_use_fb_anc())
+                snd_device = SND_DEVICE_OUT_ANC_FB_HEADSET;
+            else
+                snd_device = SND_DEVICE_OUT_ANC_HEADSET;
+        }
+        else
+            snd_device = SND_DEVICE_OUT_HEADPHONES;
+    } else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
+        if (adev->speaker_lr_swap)
+            snd_device = SND_DEVICE_OUT_SPEAKER_REVERSE;
+        else
+            snd_device = SND_DEVICE_OUT_SPEAKER;
+    } else if (devices & AUDIO_DEVICE_OUT_ALL_SCO) {
+        if (my_data->btsco_sample_rate == SAMPLE_RATE_16KHZ)
+            snd_device = SND_DEVICE_OUT_BT_SCO_WB;
+        else
+            snd_device = SND_DEVICE_OUT_BT_SCO;
+    } else if (devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+        snd_device = SND_DEVICE_OUT_HDMI ;
+    } else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
+               devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
+        snd_device = SND_DEVICE_OUT_USB_HEADSET;
+    } else if (devices & AUDIO_DEVICE_OUT_FM_TX) {
+        snd_device = SND_DEVICE_OUT_TRANSMISSION_FM;
+    } else if (devices & AUDIO_DEVICE_OUT_EARPIECE) {
+        snd_device = SND_DEVICE_OUT_HANDSET;
+    } else if (devices & AUDIO_DEVICE_OUT_PROXY) {
+        ALOGD("%s: setting sink capability for Proxy", __func__);
+        audio_extn_set_afe_proxy_channel_mixer(adev);
+        snd_device = SND_DEVICE_OUT_AFE_PROXY;
+    } else {
+        ALOGE("%s: Unknown device(s) %#x", __func__, devices);
+    }
+exit:
+    ALOGV("%s: exit: snd_device(%s)", __func__, device_table[snd_device]);
+    return snd_device;
+}
+
+snd_device_t platform_get_input_snd_device(void *platform, audio_devices_t out_device)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    struct audio_device *adev = my_data->adev;
+    audio_source_t  source = (adev->active_input == NULL) ?
+                                AUDIO_SOURCE_DEFAULT : adev->active_input->source;
+
+    audio_mode_t    mode   = adev->mode;
+    audio_devices_t in_device = ((adev->active_input == NULL) ?
+                                    AUDIO_DEVICE_NONE : adev->active_input->device)
+                                & ~AUDIO_DEVICE_BIT_IN;
+    audio_channel_mask_t channel_mask = (adev->active_input == NULL) ?
+                                AUDIO_CHANNEL_IN_MONO : adev->active_input->channel_mask;
+    snd_device_t snd_device = SND_DEVICE_NONE;
+    int channel_count = popcount(channel_mask);
+
+    ALOGV("%s: enter: out_device(%#x) in_device(%#x)",
+          __func__, out_device, in_device);
+    if ((out_device != AUDIO_DEVICE_NONE) && ((mode == AUDIO_MODE_IN_CALL) ||
+        voice_extn_compress_voip_is_active(adev))) {
+        if ((adev->voice.tty_mode != TTY_MODE_OFF) &&
+            !voice_extn_compress_voip_is_active(adev)) {
+            if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+                out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+                switch (adev->voice.tty_mode) {
+                case TTY_MODE_FULL:
+                    snd_device = SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC;
+                    break;
+                case TTY_MODE_VCO:
+                    snd_device = SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC;
+                    break;
+                case TTY_MODE_HCO:
+                    snd_device = SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC;
+                    break;
+                default:
+                    ALOGE("%s: Invalid TTY mode (%#x)",
+                          __func__, adev->voice.tty_mode);
+                }
+                goto exit;
+            }
+        }
+        if (out_device & AUDIO_DEVICE_OUT_EARPIECE ||
+            out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE) {
+            if (out_device & AUDIO_DEVICE_OUT_EARPIECE &&
+                audio_extn_should_use_handset_anc(channel_count)) {
+                snd_device = SND_DEVICE_IN_AANC_HANDSET_MIC;
+            } else if (my_data->fluence_type == FLUENCE_NONE ||
+                my_data->fluence_in_voice_call == false) {
+                snd_device = SND_DEVICE_IN_HANDSET_MIC;
+                set_echo_reference(adev->mixer, EC_REF_RX);
+            } else {
+                snd_device = SND_DEVICE_IN_VOICE_DMIC;
+                adev->acdb_settings |= DMIC_FLAG;
+            }
+        } else if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+            snd_device = SND_DEVICE_IN_VOICE_HEADSET_MIC;
+        } else if (out_device & AUDIO_DEVICE_OUT_ALL_SCO) {
+            if (my_data->btsco_sample_rate == SAMPLE_RATE_16KHZ)
+                snd_device = SND_DEVICE_IN_BT_SCO_MIC_WB;
+            else
+                snd_device = SND_DEVICE_IN_BT_SCO_MIC;
+        } else if (out_device & AUDIO_DEVICE_OUT_SPEAKER) {
+            if (my_data->fluence_type != FLUENCE_NONE &&
+                my_data->fluence_in_voice_call &&
+                my_data->fluence_in_spkr_mode) {
+                if(my_data->fluence_type & FLUENCE_QUAD_MIC) {
+                    adev->acdb_settings |= QMIC_FLAG;
+                    snd_device = SND_DEVICE_IN_VOICE_SPEAKER_QMIC;
+                } else {
+                    adev->acdb_settings |= DMIC_FLAG;
+                    snd_device = SND_DEVICE_IN_VOICE_SPEAKER_DMIC;
+                }
+            } else {
+                snd_device = SND_DEVICE_IN_VOICE_SPEAKER_MIC;
+            }
+        }
+    } else if (source == AUDIO_SOURCE_CAMCORDER) {
+        if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC ||
+            in_device & AUDIO_DEVICE_IN_BACK_MIC) {
+            snd_device = SND_DEVICE_IN_CAMCORDER_MIC;
+        }
+    } else if (source == AUDIO_SOURCE_VOICE_RECOGNITION) {
+        if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+            if (channel_count == 2) {
+                snd_device = SND_DEVICE_IN_VOICE_REC_DMIC_STEREO;
+                adev->acdb_settings |= DMIC_FLAG;
+            } else if (adev->active_input->enable_ns)
+                snd_device = SND_DEVICE_IN_VOICE_REC_MIC_NS;
+            else if (my_data->fluence_type != FLUENCE_NONE &&
+                     my_data->fluence_in_voice_rec) {
+                snd_device = SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE;
+                adev->acdb_settings |= DMIC_FLAG;
+            } else {
+                snd_device = SND_DEVICE_IN_VOICE_REC_MIC;
+            }
+        }
+    } else if (source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
+        if (out_device & AUDIO_DEVICE_OUT_SPEAKER)
+            in_device = AUDIO_DEVICE_IN_BACK_MIC;
+        if (adev->active_input) {
+            if (adev->active_input->enable_aec &&
+                    adev->active_input->enable_ns) {
+                if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
+                    if (my_data->fluence_type & FLUENCE_DUAL_MIC &&
+                       my_data->fluence_in_spkr_mode) {
+                        snd_device = SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS;
+                        adev->acdb_settings |= DMIC_FLAG;
+                    } else
+                        snd_device = SND_DEVICE_IN_SPEAKER_MIC_AEC_NS;
+                } else if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+                    if (my_data->fluence_type & FLUENCE_DUAL_MIC) {
+                        snd_device = SND_DEVICE_IN_HANDSET_DMIC_AEC_NS;
+                        adev->acdb_settings |= DMIC_FLAG;
+                    } else
+                        snd_device = SND_DEVICE_IN_HANDSET_MIC_AEC_NS;
+                } else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+                    snd_device = SND_DEVICE_IN_HEADSET_MIC_FLUENCE;
+                }
+                set_echo_reference(adev->mixer, EC_REF_RX);
+            } else if (adev->active_input->enable_aec) {
+                if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
+                    if (my_data->fluence_type & FLUENCE_DUAL_MIC) {
+                        snd_device = SND_DEVICE_IN_SPEAKER_DMIC_AEC;
+                        adev->acdb_settings |= DMIC_FLAG;
+                    } else
+                        snd_device = SND_DEVICE_IN_SPEAKER_MIC_AEC;
+                } else if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+                    if (my_data->fluence_type & FLUENCE_DUAL_MIC) {
+                        snd_device = SND_DEVICE_IN_HANDSET_DMIC_AEC;
+                        adev->acdb_settings |= DMIC_FLAG;
+                    } else
+                        snd_device = SND_DEVICE_IN_HANDSET_MIC_AEC;
+                } else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+                    snd_device = SND_DEVICE_IN_HEADSET_MIC_FLUENCE;
+                }
+                set_echo_reference(adev->mixer, EC_REF_RX);
+            } else if (adev->active_input->enable_ns) {
+                if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
+                    if (my_data->fluence_type & FLUENCE_DUAL_MIC) {
+                        snd_device = SND_DEVICE_IN_SPEAKER_DMIC_NS;
+                        adev->acdb_settings |= DMIC_FLAG;
+                    } else
+                        snd_device = SND_DEVICE_IN_SPEAKER_MIC_NS;
+                } else if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+                    if (my_data->fluence_type & FLUENCE_DUAL_MIC) {
+                        snd_device = SND_DEVICE_IN_HANDSET_DMIC_NS;
+                        adev->acdb_settings |= DMIC_FLAG;
+                    } else
+                        snd_device = SND_DEVICE_IN_HANDSET_MIC_NS;
+                } else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+                    snd_device = SND_DEVICE_IN_HEADSET_MIC_FLUENCE;
+                }
+                set_echo_reference(adev->mixer, "NONE");
+            } else
+                set_echo_reference(adev->mixer, "NONE");
+        }
+    } else if (source == AUDIO_SOURCE_MIC) {
+        if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC &&
+                channel_count == 1 ) {
+            if(my_data->fluence_type & FLUENCE_DUAL_MIC &&
+                    my_data->fluence_in_audio_rec)
+                snd_device = SND_DEVICE_IN_HANDSET_DMIC;
+        }
+    } else if (source == AUDIO_SOURCE_FM_RX ||
+               source == AUDIO_SOURCE_FM_RX_A2DP) {
+        snd_device = SND_DEVICE_IN_CAPTURE_FM;
+    } else if (source == AUDIO_SOURCE_DEFAULT) {
+        goto exit;
+    }
+
+
+    if (snd_device != SND_DEVICE_NONE) {
+        goto exit;
+    }
+
+    if (in_device != AUDIO_DEVICE_NONE &&
+            !(in_device & AUDIO_DEVICE_IN_VOICE_CALL) &&
+            !(in_device & AUDIO_DEVICE_IN_COMMUNICATION)) {
+        if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+            if (audio_extn_ssr_get_enabled() && channel_count == 6)
+                snd_device = SND_DEVICE_IN_QUAD_MIC;
+            else if (channel_count == 2)
+                snd_device = SND_DEVICE_IN_HANDSET_STEREO_DMIC;
+            else
+                snd_device = SND_DEVICE_IN_HANDSET_MIC;
+        } else if (in_device & AUDIO_DEVICE_IN_BACK_MIC) {
+            snd_device = SND_DEVICE_IN_SPEAKER_MIC;
+        } else if (in_device & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+            snd_device = SND_DEVICE_IN_HEADSET_MIC;
+        } else if (in_device & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+            if (my_data->btsco_sample_rate == SAMPLE_RATE_16KHZ)
+                snd_device = SND_DEVICE_IN_BT_SCO_MIC_WB;
+            else
+                snd_device = SND_DEVICE_IN_BT_SCO_MIC;
+        } else if (in_device & AUDIO_DEVICE_IN_AUX_DIGITAL) {
+            snd_device = SND_DEVICE_IN_HDMI_MIC;
+        } else if (in_device & AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET ||
+                   in_device & AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET) {
+            snd_device = SND_DEVICE_IN_USB_HEADSET_MIC;
+        } else if (in_device & AUDIO_DEVICE_IN_FM_RX) {
+            snd_device = SND_DEVICE_IN_CAPTURE_FM;
+        } else {
+            ALOGE("%s: Unknown input device(s) %#x", __func__, in_device);
+            ALOGW("%s: Using default handset-mic", __func__);
+            snd_device = SND_DEVICE_IN_HANDSET_MIC;
+        }
+    } else {
+        if (out_device & AUDIO_DEVICE_OUT_EARPIECE) {
+            snd_device = SND_DEVICE_IN_HANDSET_MIC;
+        } else if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+            snd_device = SND_DEVICE_IN_HEADSET_MIC;
+        } else if (out_device & AUDIO_DEVICE_OUT_SPEAKER) {
+            if (channel_count > 1)
+                snd_device = SND_DEVICE_IN_SPEAKER_STEREO_DMIC;
+            else
+                snd_device = SND_DEVICE_IN_SPEAKER_MIC;
+        } else if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE) {
+            snd_device = SND_DEVICE_IN_HANDSET_MIC;
+        } else if (out_device & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET) {
+            if (my_data->btsco_sample_rate == SAMPLE_RATE_16KHZ)
+                snd_device = SND_DEVICE_IN_BT_SCO_MIC_WB;
+            else
+                snd_device = SND_DEVICE_IN_BT_SCO_MIC;
+        } else if (out_device & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+            snd_device = SND_DEVICE_IN_HDMI_MIC;
+        } else if (out_device & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
+                   out_device & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
+            snd_device = SND_DEVICE_IN_USB_HEADSET_MIC;
+        } else {
+            ALOGE("%s: Unknown output device(s) %#x", __func__, out_device);
+            ALOGW("%s: Using default handset-mic", __func__);
+            snd_device = SND_DEVICE_IN_HANDSET_MIC;
+        }
+    }
+exit:
+    ALOGV("%s: exit: in_snd_device(%s)", __func__, device_table[snd_device]);
+    return snd_device;
+}
+
+int platform_set_hdmi_channels(void *platform,  int channel_count)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    struct audio_device *adev = my_data->adev;
+    struct mixer_ctl *ctl;
+    const char *channel_cnt_str = NULL;
+    const char *mixer_ctl_name = "HDMI_RX Channels";
+    switch (channel_count) {
+    case 8:
+        channel_cnt_str = "Eight"; break;
+    case 7:
+        channel_cnt_str = "Seven"; break;
+    case 6:
+        channel_cnt_str = "Six"; break;
+    case 5:
+        channel_cnt_str = "Five"; break;
+    case 4:
+        channel_cnt_str = "Four"; break;
+    case 3:
+        channel_cnt_str = "Three"; break;
+    default:
+        channel_cnt_str = "Two"; break;
+    }
+    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer cmd - %s",
+              __func__, mixer_ctl_name);
+        return -EINVAL;
+    }
+    ALOGV("HDMI channel count: %s", channel_cnt_str);
+    mixer_ctl_set_enum_by_string(ctl, channel_cnt_str);
+    return 0;
+}
+
+int platform_edid_get_max_channels(void *platform)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    struct audio_device *adev = my_data->adev;
+    char block[MAX_SAD_BLOCKS * SAD_BLOCK_SIZE];
+    char *sad = block;
+    int num_audio_blocks;
+    int channel_count;
+    int max_channels = 0;
+    int i, ret, count;
+
+    struct mixer_ctl *ctl;
+
+    ctl = mixer_get_ctl_by_name(adev->mixer, AUDIO_DATA_BLOCK_MIXER_CTL);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer cmd - %s",
+              __func__, AUDIO_DATA_BLOCK_MIXER_CTL);
+        return 0;
+    }
+
+    mixer_ctl_update(ctl);
+
+    count = mixer_ctl_get_num_values(ctl);
+
+    /* Read SAD blocks, clamping the maximum size for safety */
+    if (count > (int)sizeof(block))
+        count = (int)sizeof(block);
+
+    ret = mixer_ctl_get_array(ctl, block, count);
+    if (ret != 0) {
+        ALOGE("%s: mixer_ctl_get_array() failed to get EDID info", __func__);
+        return 0;
+    }
+
+    /* Calculate the number of SAD blocks */
+    num_audio_blocks = count / SAD_BLOCK_SIZE;
+
+    for (i = 0; i < num_audio_blocks; i++) {
+        /* Only consider LPCM blocks */
+        if ((sad[0] >> 3) != EDID_FORMAT_LPCM) {
+            sad += 3;
+            continue;
+        }
+
+        channel_count = (sad[0] & 0x7) + 1;
+        if (channel_count > max_channels)
+            max_channels = channel_count;
+
+        /* Advance to next block */
+        sad += 3;
+    }
+
+    return max_channels;
+}
+
+static int platform_set_slowtalk(struct platform_data *my_data, bool state)
+{
+    int ret = 0;
+    struct audio_device *adev = my_data->adev;
+    struct mixer_ctl *ctl;
+    const char *mixer_ctl_name = "Slowtalk Enable";
+    uint32_t set_values[ ] = {0,
+                              ALL_SESSION_VSID};
+
+    set_values[0] = state;
+    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer cmd - %s",
+              __func__, mixer_ctl_name);
+        ret = -EINVAL;
+    } else {
+        ALOGV("Setting slowtalk state: %d", state);
+        ret = mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
+        my_data->slowtalk = state;
+    }
+
+    if (my_data->csd != NULL) {
+        ret = my_data->csd->slow_talk(ALL_SESSION_VSID, state);
+        if (ret < 0) {
+            ALOGE("%s: csd_client_disable_device, failed, error %d",
+                  __func__, ret);
+        }
+    }
+    return ret;
+}
+
+int platform_set_parameters(void *platform, struct str_parms *parms)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    char *str;
+    char value[256] = {0};
+    int val;
+    int ret = 0, err;
+
+    ALOGV("%s: enter: %s", __func__, str_parms_to_str(parms));
+
+    err = str_parms_get_int(parms, AUDIO_PARAMETER_KEY_BTSCO, &val);
+    if (err >= 0) {
+        str_parms_del(parms, AUDIO_PARAMETER_KEY_BTSCO);
+        my_data->btsco_sample_rate = val;
+        if (val == SAMPLE_RATE_16KHZ) {
+            audio_route_apply_path(my_data->adev->audio_route,
+                                   "bt-sco-wb-samplerate");
+            audio_route_update_mixer(my_data->adev->audio_route);
+        }
+    }
+
+    err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_SLOWTALK, value, sizeof(value));
+    if (err >= 0) {
+        bool state = false;
+        if (!strncmp("true", value, sizeof("true"))) {
+            state = true;
+        }
+
+        str_parms_del(parms, AUDIO_PARAMETER_KEY_SLOWTALK);
+        ret = platform_set_slowtalk(my_data, state);
+        if (ret)
+            ALOGE("%s: Failed to set slow talk err: %d", __func__, ret);
+    }
+
+    err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_VOLUME_BOOST,
+                            value, sizeof(value));
+    if (err >= 0) {
+        str_parms_del(parms, AUDIO_PARAMETER_KEY_VOLUME_BOOST);
+
+        if (my_data->acdb_reload_vocvoltable == NULL) {
+            ALOGE("%s: acdb_reload_vocvoltable is NULL", __func__);
+        } else if (!strcmp(value, "on")) {
+            if (!my_data->acdb_reload_vocvoltable(VOICE_FEATURE_SET_VOLUME_BOOST)) {
+                my_data->voice_feature_set = 1;
+            }
+        } else {
+            if (!my_data->acdb_reload_vocvoltable(VOICE_FEATURE_SET_DEFAULT)) {
+                my_data->voice_feature_set = 0;
+            }
+        }
+    }
+
+    ALOGV("%s: exit with code(%d)", __func__, ret);
+    return ret;
+}
+
+int platform_set_incall_recording_session_id(void *platform,
+                                             uint32_t session_id, int rec_mode)
+{
+    int ret = 0;
+    struct platform_data *my_data = (struct platform_data *)platform;
+    struct audio_device *adev = my_data->adev;
+    struct mixer_ctl *ctl;
+    const char *mixer_ctl_name = "Voc VSID";
+    int num_ctl_values;
+    int i;
+
+    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer cmd - %s",
+              __func__, mixer_ctl_name);
+        ret = -EINVAL;
+    } else {
+        num_ctl_values = mixer_ctl_get_num_values(ctl);
+        for (i = 0; i < num_ctl_values; i++) {
+            if (mixer_ctl_set_value(ctl, i, session_id)) {
+                ALOGV("Error: invalid session_id: %x", session_id);
+                ret = -EINVAL;
+                break;
+            }
+        }
+    }
+
+    if (my_data->csd != NULL) {
+        ret = my_data->csd->start_record(ALL_SESSION_VSID, rec_mode);
+        if (ret < 0) {
+            ALOGE("%s: csd_client_start_record failed, error %d",
+                  __func__, ret);
+        }
+    }
+
+    return ret;
+}
+
+int platform_stop_incall_recording_usecase(void *platform)
+{
+    int ret = 0;
+    struct platform_data *my_data = (struct platform_data *)platform;
+
+    if (my_data->csd != NULL) {
+        ret = my_data->csd->stop_record(ALL_SESSION_VSID);
+        if (ret < 0) {
+            ALOGE("%s: csd_client_stop_record failed, error %d",
+                  __func__, ret);
+        }
+    }
+
+    return ret;
+}
+
+int platform_start_incall_music_usecase(void *platform)
+{
+    int ret = 0;
+    struct platform_data *my_data = (struct platform_data *)platform;
+
+    if (my_data->csd != NULL) {
+        ret = my_data->csd->start_playback(ALL_SESSION_VSID);
+        if (ret < 0) {
+            ALOGE("%s: csd_client_start_playback failed, error %d",
+                  __func__, ret);
+        }
+    }
+
+    return ret;
+}
+
+int platform_stop_incall_music_usecase(void *platform)
+{
+    int ret = 0;
+    struct platform_data *my_data = (struct platform_data *)platform;
+
+    if (my_data->csd != NULL) {
+        ret = my_data->csd->stop_playback(ALL_SESSION_VSID);
+        if (ret < 0) {
+            ALOGE("%s: csd_client_stop_playback failed, error %d",
+                  __func__, ret);
+        }
+    }
+
+    return ret;
+}
+
+void platform_get_parameters(void *platform,
+                            struct str_parms *query,
+                            struct str_parms *reply)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    char *str = NULL;
+    char value[256] = {0};
+    int ret;
+    int fluence_type;
+
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_FLUENCE_TYPE,
+                            value, sizeof(value));
+    if (ret >= 0) {
+        if (my_data->fluence_type & FLUENCE_QUAD_MIC) {
+            strlcpy(value, "fluencepro", sizeof(value));
+        } else if (my_data->fluence_type & FLUENCE_DUAL_MIC) {
+            strlcpy(value, "fluence", sizeof(value));
+        } else {
+            strlcpy(value, "none", sizeof(value));
+        }
+
+        str_parms_add_str(reply, AUDIO_PARAMETER_KEY_FLUENCE_TYPE, value);
+    }
+
+    memset(value, 0, sizeof(value));
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_SLOWTALK,
+                            value, sizeof(value));
+    if (ret >= 0) {
+        str_parms_add_str(reply, AUDIO_PARAMETER_KEY_SLOWTALK,
+                          my_data->slowtalk?"true":"false");
+    }
+
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_VOLUME_BOOST,
+                            value, sizeof(value));
+    if (ret >= 0) {
+        if (my_data->voice_feature_set == VOICE_FEATURE_SET_VOLUME_BOOST) {
+            strlcpy(value, "on", sizeof(value));
+        } else {
+            strlcpy(value, "off", sizeof(value));
+        }
+
+        str_parms_add_str(reply, AUDIO_PARAMETER_KEY_VOLUME_BOOST, value);
+    }
+
+    ALOGV("%s: exit: returns - %s", __func__, str_parms_to_str(reply));
+}
+
+/* Delay in Us */
+int64_t platform_render_latency(audio_usecase_t usecase)
+{
+    switch (usecase) {
+        case USECASE_AUDIO_PLAYBACK_DEEP_BUFFER:
+            return DEEP_BUFFER_PLATFORM_DELAY;
+        case USECASE_AUDIO_PLAYBACK_LOW_LATENCY:
+            return LOW_LATENCY_PLATFORM_DELAY;
+        default:
+            return 0;
+    }
+}
+
+int platform_update_usecase_from_source(int source, int usecase)
+{
+    ALOGV("%s: input source :%d", __func__, source);
+    if(source == AUDIO_SOURCE_FM_RX_A2DP)
+        usecase = USECASE_AUDIO_RECORD_FM_VIRTUAL;
+    return usecase;
+}
+
+bool platform_listen_update_status(snd_device_t snd_device)
+{
+    if ((snd_device >= SND_DEVICE_IN_BEGIN) &&
+        (snd_device < SND_DEVICE_IN_END) &&
+        (snd_device != SND_DEVICE_IN_CAPTURE_FM) &&
+        (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK))
+        return true;
+    else
+        return false;
+}
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
new file mode 100644
index 0000000..cad5198
--- /dev/null
+++ b/hal/msm8916/platform.h
@@ -0,0 +1,226 @@
+/*
+ * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef QCOM_AUDIO_PLATFORM_H
+#define QCOM_AUDIO_PLATFORM_H
+
+enum {
+    FLUENCE_NONE,
+    FLUENCE_DUAL_MIC = 0x1,
+    FLUENCE_QUAD_MIC = 0x2,
+};
+
+/*
+ * Below are the devices for which is back end is same, SLIMBUS_0_RX.
+ * All these devices are handled by the internal HW codec. We can
+ * enable any one of these devices at any time
+ */
+#define AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND \
+    (AUDIO_DEVICE_OUT_EARPIECE | AUDIO_DEVICE_OUT_SPEAKER | \
+     AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE)
+
+/* Sound devices specific to the platform
+ * The DEVICE_OUT_* and DEVICE_IN_* should be mapped to these sound
+ * devices to enable corresponding mixer paths
+ */
+enum {
+    SND_DEVICE_NONE = 0,
+
+    /* Playback devices */
+    SND_DEVICE_MIN,
+    SND_DEVICE_OUT_BEGIN = SND_DEVICE_MIN,
+    SND_DEVICE_OUT_HANDSET = SND_DEVICE_OUT_BEGIN,
+    SND_DEVICE_OUT_SPEAKER,
+    SND_DEVICE_OUT_SPEAKER_REVERSE,
+    SND_DEVICE_OUT_HEADPHONES,
+    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+    SND_DEVICE_OUT_VOICE_HANDSET,
+    SND_DEVICE_OUT_VOICE_SPEAKER,
+    SND_DEVICE_OUT_VOICE_HEADPHONES,
+    SND_DEVICE_OUT_HDMI,
+    SND_DEVICE_OUT_SPEAKER_AND_HDMI,
+    SND_DEVICE_OUT_BT_SCO,
+    SND_DEVICE_OUT_BT_SCO_WB,
+    SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
+    SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
+    SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
+    SND_DEVICE_OUT_AFE_PROXY,
+    SND_DEVICE_OUT_USB_HEADSET,
+    SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET,
+    SND_DEVICE_OUT_TRANSMISSION_FM,
+    SND_DEVICE_OUT_ANC_HEADSET,
+    SND_DEVICE_OUT_ANC_FB_HEADSET,
+    SND_DEVICE_OUT_VOICE_ANC_HEADSET,
+    SND_DEVICE_OUT_VOICE_ANC_FB_HEADSET,
+    SND_DEVICE_OUT_SPEAKER_AND_ANC_HEADSET,
+    SND_DEVICE_OUT_ANC_HANDSET,
+    SND_DEVICE_OUT_SPEAKER_PROTECTED,
+    SND_DEVICE_OUT_END,
+
+    /*
+     * Note: IN_BEGIN should be same as OUT_END because total number of devices
+     * SND_DEVICES_MAX should not exceed MAX_RX + MAX_TX devices.
+     */
+    /* Capture devices */
+    SND_DEVICE_IN_BEGIN = SND_DEVICE_OUT_END,
+    SND_DEVICE_IN_HANDSET_MIC  = SND_DEVICE_IN_BEGIN,
+    SND_DEVICE_IN_HANDSET_MIC_AEC,
+    SND_DEVICE_IN_HANDSET_MIC_NS,
+    SND_DEVICE_IN_HANDSET_MIC_AEC_NS,
+    SND_DEVICE_IN_HANDSET_DMIC,
+    SND_DEVICE_IN_HANDSET_DMIC_AEC,
+    SND_DEVICE_IN_HANDSET_DMIC_NS,
+    SND_DEVICE_IN_HANDSET_DMIC_AEC_NS,
+    SND_DEVICE_IN_SPEAKER_MIC,
+    SND_DEVICE_IN_SPEAKER_MIC_AEC,
+    SND_DEVICE_IN_SPEAKER_MIC_NS,
+    SND_DEVICE_IN_SPEAKER_MIC_AEC_NS,
+    SND_DEVICE_IN_SPEAKER_DMIC,
+    SND_DEVICE_IN_SPEAKER_DMIC_AEC,
+    SND_DEVICE_IN_SPEAKER_DMIC_NS,
+    SND_DEVICE_IN_SPEAKER_DMIC_AEC_NS,
+    SND_DEVICE_IN_HEADSET_MIC,
+    SND_DEVICE_IN_HEADSET_MIC_FLUENCE,
+    SND_DEVICE_IN_VOICE_SPEAKER_MIC,
+    SND_DEVICE_IN_VOICE_HEADSET_MIC,
+    SND_DEVICE_IN_HDMI_MIC,
+    SND_DEVICE_IN_BT_SCO_MIC,
+    SND_DEVICE_IN_BT_SCO_MIC_WB,
+    SND_DEVICE_IN_CAMCORDER_MIC,
+    SND_DEVICE_IN_VOICE_DMIC,
+    SND_DEVICE_IN_VOICE_SPEAKER_DMIC,
+    SND_DEVICE_IN_VOICE_SPEAKER_QMIC,
+    SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC,
+    SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC,
+    SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC,
+    SND_DEVICE_IN_VOICE_REC_MIC,
+    SND_DEVICE_IN_VOICE_REC_MIC_NS,
+    SND_DEVICE_IN_VOICE_REC_DMIC_STEREO,
+    SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE,
+    SND_DEVICE_IN_USB_HEADSET_MIC,
+    SND_DEVICE_IN_CAPTURE_FM,
+    SND_DEVICE_IN_AANC_HANDSET_MIC,
+    SND_DEVICE_IN_QUAD_MIC,
+    SND_DEVICE_IN_HANDSET_STEREO_DMIC,
+    SND_DEVICE_IN_SPEAKER_STEREO_DMIC,
+    SND_DEVICE_IN_CAPTURE_VI_FEEDBACK,
+    SND_DEVICE_IN_END,
+
+    SND_DEVICE_MAX = SND_DEVICE_IN_END,
+
+};
+
+#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
+
+#define ALL_SESSION_VSID                0xFFFFFFFF
+#define DEFAULT_MUTE_RAMP_DURATION      500
+#define DEFAULT_VOLUME_RAMP_DURATION_MS 20
+#define MIXER_PATH_MAX_LENGTH 100
+
+#define MAX_VOL_INDEX 5
+#define MIN_VOL_INDEX 0
+#define percent_to_index(val, min, max) \
+            ((val) * ((max) - (min)) * 0.01 + (min) + .5)
+
+/*
+ * tinyAlsa library interprets period size as number of frames
+ * one frame = channel_count * sizeof (pcm sample)
+ * so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes
+ * DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes
+ * We should take care of returning proper size when AudioFlinger queries for
+ * the buffer size of an input/output stream
+ */
+#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 960
+#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 4
+#define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
+#define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
+
+#define HDMI_MULTI_PERIOD_SIZE  336
+#define HDMI_MULTI_PERIOD_COUNT 8
+#define HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6
+#define HDMI_MULTI_PERIOD_BYTES (HDMI_MULTI_PERIOD_SIZE * HDMI_MULTI_DEFAULT_CHANNEL_COUNT * 2)
+
+#define AUDIO_CAPTURE_PERIOD_DURATION_MSEC 20
+#define AUDIO_CAPTURE_PERIOD_COUNT 2
+
+#define DEVICE_NAME_MAX_SIZE 128
+#define HW_INFO_ARRAY_MAX_SIZE 32
+
+#define DEEP_BUFFER_PCM_DEVICE 0
+#define AUDIO_RECORD_PCM_DEVICE 0
+#define MULTIMEDIA2_PCM_DEVICE 1
+#define FM_PLAYBACK_PCM_DEVICE 5
+#define FM_CAPTURE_PCM_DEVICE  6
+#define HFP_PCM_RX 5
+#define HFP_SCO_RX 23
+#define HFP_ASM_RX_TX 24
+
+#define INCALL_MUSIC_UPLINK_PCM_DEVICE 1
+#define INCALL_MUSIC_UPLINK2_PCM_DEVICE 16
+#define SPKR_PROT_CALIB_RX_PCM_DEVICE 5
+#define SPKR_PROT_CALIB_TX_PCM_DEVICE 22
+#define PLAYBACK_OFFLOAD_DEVICE 9
+#define COMPRESS_VOIP_CALL_PCM_DEVICE 3
+
+
+#define LOWLATENCY_PCM_DEVICE 12
+#define EC_REF_RX "I2S_RX"
+#define COMPRESS_CAPTURE_DEVICE 19
+
+#define VOICE_CALL_PCM_DEVICE 2
+#define VOICE2_CALL_PCM_DEVICE 13
+#define VOLTE_CALL_PCM_DEVICE 15
+#define QCHAT_CALL_PCM_DEVICE 14
+
+#define LIB_CSD_CLIENT "libcsd-client.so"
+/* CSD-CLIENT related functions */
+typedef int (*init_t)();
+typedef int (*deinit_t)();
+typedef int (*disable_device_t)();
+typedef int (*enable_device_config_t)(int, int);
+typedef int (*enable_device_t)(int, int, uint32_t);
+typedef int (*volume_t)(uint32_t, int);
+typedef int (*mic_mute_t)(uint32_t, int);
+typedef int (*slow_talk_t)(uint32_t, uint8_t);
+typedef int (*start_voice_t)(uint32_t);
+typedef int (*stop_voice_t)(uint32_t);
+typedef int (*start_playback_t)(uint32_t);
+typedef int (*stop_playback_t)(uint32_t);
+typedef int (*start_record_t)(uint32_t, int);
+typedef int (*stop_record_t)(uint32_t);
+/* CSD Client structure */
+struct csd_data {
+    void *csd_client;
+    init_t init;
+    deinit_t deinit;
+    disable_device_t disable_device;
+    enable_device_config_t enable_device_config;
+    enable_device_t enable_device;
+    volume_t volume;
+    mic_mute_t mic_mute;
+    slow_talk_t slow_talk;
+    start_voice_t start_voice;
+    stop_voice_t stop_voice;
+    start_playback_t start_playback;
+    stop_playback_t stop_playback;
+    start_record_t start_record;
+    stop_record_t stop_record;
+};
+
+#endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index a25991f..fb14330 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -76,7 +76,7 @@
 
 /* Audio calibration related functions */
 typedef void (*acdb_deallocate_t)();
-typedef int  (*acdb_init_t)();
+typedef int  (*acdb_init_t)(char *);
 typedef void (*acdb_send_audio_cal_t)(int, int);
 typedef void (*acdb_send_voice_cal_t)(int, int);
 typedef int (*acdb_reload_vocvoltable_t)(int);
@@ -416,6 +416,13 @@
                   __func__, dlerror());
             goto error;
         }
+        csd->enable_device_config = (enable_device_config_t)dlsym(csd->csd_client,
+                                               "csd_client_enable_device_config");
+        if (csd->enable_device_config == NULL) {
+            ALOGE("%s: dlsym error %s for csd_client_enable_device_config",
+                  __func__, dlerror());
+            goto error;
+        }
         csd->enable_device = (enable_device_t)dlsym(csd->csd_client,
                                              "csd_client_enable_device");
         if (csd->enable_device == NULL) {
@@ -643,11 +650,11 @@
                   __func__, LIB_ACDB_LOADER);
 
         my_data->acdb_init = (acdb_init_t)dlsym(my_data->acdb_handle,
-                                                    "acdb_loader_init_ACDB");
+                                                    "acdb_loader_init_v2");
         if (my_data->acdb_init == NULL)
-            ALOGE("%s: dlsym error %s for acdb_loader_init_ACDB", __func__, dlerror());
+            ALOGE("%s: dlsym error %s for acdb_loader_init_v2", __func__, dlerror());
         else
-            my_data->acdb_init();
+            my_data->acdb_init(snd_card_name);
     }
 
     /* Initialize ACDB ID's */
@@ -831,6 +838,32 @@
     return ret;
 }
 
+int platform_switch_voice_call_enable_device_config(void *platform,
+                                                    snd_device_t out_snd_device,
+                                                    snd_device_t in_snd_device)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    int acdb_rx_id, acdb_tx_id;
+    int ret = 0;
+
+    acdb_rx_id = acdb_device_table[out_snd_device];
+    acdb_tx_id = acdb_device_table[in_snd_device];
+
+    if (my_data->csd != NULL) {
+        if (acdb_rx_id > 0 && acdb_tx_id > 0) {
+            ret = my_data->csd->enable_device_config(acdb_rx_id, acdb_tx_id);
+            if (ret < 0) {
+                ALOGE("%s: csd_enable_device_config, failed, error %d",
+                      __func__, ret);
+            }
+        } else {
+            ALOGE("%s: Incorrect ACDB IDs (rx: %d tx: %d)", __func__,
+                  acdb_rx_id, acdb_tx_id);
+        }
+    }
+    return ret;
+}
+
 int platform_switch_voice_call_device_post(void *platform,
                                            snd_device_t out_snd_device,
                                            snd_device_t in_snd_device)
@@ -1296,7 +1329,8 @@
         if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
             if (audio_extn_ssr_get_enabled() && channel_count == 6)
                 snd_device = SND_DEVICE_IN_QUAD_MIC;
-            else if (channel_count == 2)
+            else if (my_data->fluence_type & (FLUENCE_DUAL_MIC | FLUENCE_QUAD_MIC) &&
+                    channel_count == 2)
                 snd_device = SND_DEVICE_IN_HANDSET_STEREO_DMIC;
             else
                 snd_device = SND_DEVICE_IN_HANDSET_MIC;
@@ -1481,8 +1515,9 @@
     char value[256] = {0};
     int val;
     int ret = 0, err;
+    char *kv_pairs = str_parms_to_str(parms);
 
-    ALOGV("%s: enter: %s", __func__, str_parms_to_str(parms));
+    ALOGV_IF(kv_pairs != NULL, "%s: enter: %s", __func__, kv_pairs);
 
     err = str_parms_get_int(parms, AUDIO_PARAMETER_KEY_BTSCO, &val);
     if (err >= 0) {
@@ -1527,6 +1562,7 @@
     }
 
     ALOGV("%s: exit with code(%d)", __func__, ret);
+    free(kv_pairs);
     return ret;
 }
 
@@ -1625,6 +1661,7 @@
     char value[256] = {0};
     int ret;
     int fluence_type;
+    char *kv_pairs = NULL;
 
     ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_FLUENCE_TYPE,
                             value, sizeof(value));
@@ -1660,7 +1697,9 @@
         str_parms_add_str(reply, AUDIO_PARAMETER_KEY_VOLUME_BOOST, value);
     }
 
-    ALOGV("%s: exit: returns - %s", __func__, str_parms_to_str(reply));
+    kv_pairs = str_parms_to_str(reply);
+    ALOGV_IF(kv_pairs != NULL, "%s: exit: returns - %s", __func__, kv_pairs);
+    free(reply);
 }
 
 /* Delay in Us */
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index ca8469a..9749be4 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -174,7 +174,7 @@
 #define INCALL_MUSIC_UPLINK_PCM_DEVICE 1
 #define INCALL_MUSIC_UPLINK2_PCM_DEVICE 16
 #define SPKR_PROT_CALIB_RX_PCM_DEVICE 5
-#define SPKR_PROT_CALIB_TX_PCM_DEVICE 22
+#define SPKR_PROT_CALIB_TX_PCM_DEVICE 25
 #define PLAYBACK_OFFLOAD_DEVICE 9
 #define COMPRESS_VOIP_CALL_PCM_DEVICE 3
 
@@ -218,6 +218,7 @@
 typedef int (*init_t)();
 typedef int (*deinit_t)();
 typedef int (*disable_device_t)();
+typedef int (*enable_device_config_t)(int, int);
 typedef int (*enable_device_t)(int, int, uint32_t);
 typedef int (*volume_t)(uint32_t, int);
 typedef int (*mic_mute_t)(uint32_t, int);
@@ -234,6 +235,7 @@
     init_t init;
     deinit_t deinit;
     disable_device_t disable_device;
+    enable_device_config_t enable_device_config;
     enable_device_t enable_device;
     volume_t volume;
     mic_mute_t mic_mute;
diff --git a/hal/platform_api.h b/hal/platform_api.h
index a5f5074..2c12ea6 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -31,6 +31,9 @@
 int platform_set_snd_device_acdb_id(snd_device_t snd_device, unsigned int acdb_id);
 int platform_send_audio_calibration(void *platform, snd_device_t snd_device);
 int platform_switch_voice_call_device_pre(void *platform);
+int platform_switch_voice_call_enable_device_config(void *platform,
+                                                    snd_device_t out_snd_device,
+                                                    snd_device_t in_snd_device);
 int platform_switch_voice_call_device_post(void *platform,
                                            snd_device_t out_snd_device,
                                            snd_device_t in_snd_device);
diff --git a/hal/voice.c b/hal/voice.c
index 8783f01..28d44db 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -364,8 +364,9 @@
     char value[32];
     int val;
     int ret = 0, err;
+    char *kv_pairs = str_parms_to_str(parms);
 
-    ALOGV("%s: enter: %s", __func__, str_parms_to_str(parms));
+    ALOGV_IF(kv_pairs != NULL, "%s: enter: %s", __func__, kv_pairs);
 
     ret = voice_extn_set_parameters(adev, parms);
     if (ret != 0)
@@ -413,6 +414,7 @@
 
 done:
     ALOGV("%s: exit with code(%d)", __func__, ret);
+    free(kv_pairs);
     return ret;
 }
 
diff --git a/hal/voice_extn/compress_voip.c b/hal/voice_extn/compress_voip.c
index 5c87c9c..47ac2c8 100644
--- a/hal/voice_extn/compress_voip.c
+++ b/hal/voice_extn/compress_voip.c
@@ -59,8 +59,9 @@
     struct pcm *pcm_rx;
     struct pcm *pcm_tx;
     struct stream_out *out_stream;
-    int ref_count;
-    int out_stream_count;
+    uint32_t out_stream_count;
+    uint32_t in_stream_count;
+    uint32_t sample_rate;
 };
 
 #define MODE_IS127              0x2
@@ -78,13 +79,15 @@
 #define AUDIO_PARAMETER_VALUE_VOIP_TRUE             "true"
 #define AUDIO_PARAMETER_KEY_VOIP_CHECK              "voip_flag"
 #define AUDIO_PARAMETER_KEY_VOIP_OUT_STREAM_COUNT   "voip_out_stream_count"
+#define AUDIO_PARAMETER_KEY_VOIP_SAMPLE_RATE        "voip_sample_rate"
 
 static struct voip_data voip_data = {
   .pcm_rx = NULL,
   .pcm_tx = NULL,
   .out_stream = NULL,
-  .ref_count = 0,
-  .out_stream_count = 0
+  .out_stream_count = 0,
+  .in_stream_count = 0,
+  .sample_rate = 0
 };
 
 static int voip_set_volume(struct audio_device *adev, int volume);
@@ -280,10 +283,10 @@
     int i, ret = 0;
     struct audio_usecase *uc_info;
 
-    ALOGD("%s: enter, ref_count=%d", __func__, voip_data.ref_count);
-    voip_data.ref_count--;
+    ALOGD("%s: enter, out_stream_count=%d, in_stream_count=%d",
+           __func__, voip_data.out_stream_count, voip_data.in_stream_count);
 
-    if (!voip_data.ref_count) {
+    if (!voip_data.out_stream_count && !voip_data.in_stream_count) {
         uc_info = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
         if (uc_info == NULL) {
             ALOGE("%s: Could not find the usecase (%d) in the list",
@@ -310,8 +313,10 @@
 
         list_remove(&uc_info->list);
         free(uc_info);
+        voip_data.sample_rate = 0;
     } else
-        ALOGV("%s: NO-OP because ref_count=%d", __func__, voip_data.ref_count);
+        ALOGV("%s: NO-OP because out_stream_count=%d, in_stream_count=%d",
+               __func__, voip_data.out_stream_count, voip_data.in_stream_count);
 
     ALOGV("%s: exit: status(%d)", __func__, ret);
     return ret;
@@ -327,12 +332,15 @@
     ALOGD("%s: enter", __func__);
 
     uc_info = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
-    if ((uc_info == NULL) && (voip_data.out_stream)) {
+    if (uc_info == NULL) {
         ALOGV("%s: voip usecase is added to the list", __func__);
         uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
         uc_info->id = USECASE_COMPRESS_VOIP_CALL;
         uc_info->type = VOIP_CALL;
-        uc_info->stream.out = voip_data.out_stream;
+        if (voip_data.out_stream)
+            uc_info->stream.out = voip_data.out_stream;
+        else
+            uc_info->stream.out = adev->primary_output;
         uc_info->in_snd_device = SND_DEVICE_NONE;
         uc_info->out_snd_device = SND_DEVICE_NONE;
 
@@ -388,12 +396,15 @@
             ALOGE("%s: error %d\n", __func__, ret);
             goto error_start_voip;
         }
-        voip_data.ref_count = 0;
-    }
-    else
+    } else {
         ALOGV("%s: voip usecase is already enabled", __func__);
+        if (voip_data.out_stream)
+            uc_info->stream.out = voip_data.out_stream;
+        else
+            uc_info->stream.out = adev->primary_output;
+        select_devices(adev, USECASE_COMPRESS_VOIP_CALL);
+    }
 
-    voip_data.ref_count++;
     return 0;
 
 error_start_voip:
@@ -411,8 +422,9 @@
     int ret = 0, err, rate;
     int min_rate, max_rate;
     bool flag;
+    char *kv_pairs = str_parms_to_str(parms);
 
-    ALOGV("%s: enter: %s", __func__, str_parms_to_str(parms));
+    ALOGV_IF(kv_pairs != NULL, "%s: enter: %s", __func__, kv_pairs);
 
     err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_VOIP_RATE,
                             value, sizeof(value));
@@ -453,6 +465,7 @@
 
 done:
     ALOGV("%s: exit", __func__);
+    free(kv_pairs);
     return ret;
 }
 
@@ -470,6 +483,13 @@
         str_parms_add_int(reply, AUDIO_PARAMETER_KEY_VOIP_OUT_STREAM_COUNT,
                           voip_data.out_stream_count);
     }
+
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_VOIP_SAMPLE_RATE,
+                            value, sizeof(value));
+    if (ret >= 0) {
+        str_parms_add_int(reply, AUDIO_PARAMETER_KEY_VOIP_SAMPLE_RATE,
+                          voip_data.sample_rate);
+    }
 }
 
 void voice_extn_compress_voip_out_get_parameters(struct stream_out *out,
@@ -499,6 +519,7 @@
 {
     int ret, val;
     char value[32]={0};
+    char *kv_pairs = NULL;
 
     ALOGV("%s: enter", __func__);
 
@@ -511,7 +532,9 @@
             str_parms_add_int(reply, AUDIO_PARAMETER_KEY_VOIP_CHECK, false);
     }
 
-    ALOGD("%s: exit: return - %s", __func__, str_parms_to_str(reply));
+    kv_pairs = str_parms_to_str(reply);
+    ALOGD_IF(kv_pairs != NULL, "%s: exit: return - %s", __func__, kv_pairs);
+    free(kv_pairs);
 }
 
 int voice_extn_compress_voip_out_get_buffer_size(struct stream_out *out)
@@ -571,9 +594,9 @@
 
     ALOGD("%s: enter", __func__);
 
+    voip_data.out_stream_count--;
     ret = voip_stop_call(adev);
     voip_data.out_stream = NULL;
-    voip_data.out_stream_count--;
 
     ALOGV("%s: exit: status(%d)", __func__, ret);
     return ret;
@@ -595,7 +618,7 @@
 
     voip_data.out_stream = out;
     voip_data.out_stream_count++;
-
+    voip_data.sample_rate = out->sample_rate;
     ret = voip_set_mode(out->dev, out->format);
 
     ALOGV("%s: exit", __func__);
@@ -610,6 +633,7 @@
 
     ALOGD("%s: enter", __func__);
 
+    voip_data.in_stream_count--;
     status = voip_stop_call(adev);
 
     ALOGV("%s: exit: status(%d)", __func__, status);
@@ -625,15 +649,25 @@
 
     ALOGD("%s: enter", __func__);
 
+    if ((voip_data.sample_rate != 0) &&
+        (voip_data.sample_rate != in->config.rate)) {
+        ret = -ENOTSUP;
+        goto done;
+    } else {
+        voip_data.sample_rate = in->config.rate;
+    }
+
     in->usecase = USECASE_COMPRESS_VOIP_CALL;
     if (in->config.rate == 16000)
         in->config = pcm_config_voip_wb;
     else
         in->config = pcm_config_voip_nb;
 
+    voip_data.in_stream_count++;
     ret = voip_set_mode(in->dev, in->format);
 
-    ALOGV("%s: exit", __func__);
+done:
+    ALOGV("%s: exit, ret=%d", __func__, ret);
     return ret;
 }
 
@@ -729,7 +763,8 @@
     if (ret) {
         if ((popcount(config->channel_mask) == 1) &&
             (config->sample_rate == 8000 || config->sample_rate == 16000))
-            ret = true;
+            ret = ((voip_data.sample_rate == 0) ? true:
+                    (voip_data.sample_rate == config->sample_rate));
         else
             ret = false;
     }
diff --git a/hal/voice_extn/voice_extn.c b/hal/voice_extn/voice_extn.c
index b8bc2df..5612e0c 100644
--- a/hal/voice_extn/voice_extn.c
+++ b/hal/voice_extn/voice_extn.c
@@ -427,8 +427,9 @@
     char *str;
     int value;
     int ret = 0, err;
+    char *kv_pairs = str_parms_to_str(parms);
 
-    ALOGV("%s: enter: %s", __func__, str_parms_to_str(parms));
+    ALOGV_IF(kv_pairs != NULL, "%s: enter: %s", __func__, kv_pairs);
 
     err = str_parms_get_int(parms, AUDIO_PARAMETER_KEY_VSID, &value);
     if (err >= 0) {
@@ -458,6 +459,7 @@
 
 done:
     ALOGV("%s: exit with code(%d)", __func__, ret);
+    free(kv_pairs);
     return ret;
 }
 
@@ -485,9 +487,10 @@
 {
     int ret;
     char value[VOICE_EXTN_PARAMETER_VALUE_MAX_LEN] = {0};
-    char *str = NULL;
+    char *str = str_parms_to_str(query);
 
-    ALOGV("%s: enter %s", __func__, str_parms_to_str(query));
+    ALOGV_IF(str != NULL, "%s: enter %s", __func__, str);
+    free(str);
 
     ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_AUDIO_MODE, value,
                             sizeof(value));
@@ -507,7 +510,9 @@
     }
     voice_extn_compress_voip_get_parameters(adev, query, reply);
 
-    ALOGV("%s: exit: returns \"%s\"", __func__, str_parms_to_str(reply));
+    str = str_parms_to_str(reply);
+    ALOGV_IF(str != NULL, "%s: exit: returns \"%s\"", __func__, str);
+    free(str);
 }
 
 void voice_extn_out_get_parameters(struct stream_out *out,
diff --git a/mm-audio/aenc-aac/Android.mk b/mm-audio/aenc-aac/Android.mk
index 8ab45b3..7cd804a 100644
--- a/mm-audio/aenc-aac/Android.mk
+++ b/mm-audio/aenc-aac/Android.mk
@@ -24,6 +24,8 @@
 ifeq ($(call is-board-platform,mpq8092),true)
 include $(AENC_AAC_PATH)/qdsp6/Android.mk
 endif
-
+ifeq ($(call is-board-platform,msm8916),true)
+include $(AENC_AAC_PATH)/qdsp6/Android.mk
+endif
 
 endif
diff --git a/mm-audio/aenc-amrnb/Android.mk b/mm-audio/aenc-amrnb/Android.mk
index 2601ede..79474c6 100644
--- a/mm-audio/aenc-amrnb/Android.mk
+++ b/mm-audio/aenc-amrnb/Android.mk
@@ -24,6 +24,8 @@
 ifeq ($(call is-board-platform,mpq8092),true)
 include $(AENC_AMR_PATH)/qdsp6/Android.mk
 endif
-
+ifeq ($(call is-board-platform,msm8916),true)
+include $(AENC_AMR_PATH)/qdsp6/Android.mk
+endif
 
 endif
diff --git a/mm-audio/aenc-evrc/Android.mk b/mm-audio/aenc-evrc/Android.mk
index 2f42d6b..649f860 100644
--- a/mm-audio/aenc-evrc/Android.mk
+++ b/mm-audio/aenc-evrc/Android.mk
@@ -24,6 +24,8 @@
 ifeq ($(call is-board-platform,mpq8092),true)
 include $(AENC_EVRC_PATH)/qdsp6/Android.mk
 endif
-
+ifeq ($(call is-board-platform,msm8916),true)
+include $(AENC_EVRC_PATH)/qdsp6/Android.mk
+endif
 
 endif
diff --git a/mm-audio/aenc-qcelp13/Android.mk b/mm-audio/aenc-qcelp13/Android.mk
index fe18efc..c919f19 100644
--- a/mm-audio/aenc-qcelp13/Android.mk
+++ b/mm-audio/aenc-qcelp13/Android.mk
@@ -24,5 +24,8 @@
 ifeq ($(call is-board-platform,mpq8092),true)
 include $(AENC_QCELP13_PATH)/qdsp6/Android.mk
 endif
+ifeq ($(call is-board-platform,msm8916),true)
+include $(AENC_QCELP13_PATH)/qdsp6/Android.mk
+endif
 
 endif
diff --git a/policy_hal/Android.mk b/policy_hal/Android.mk
index b6a06e4..c68ab6e 100644
--- a/policy_hal/Android.mk
+++ b/policy_hal/Android.mk
@@ -30,6 +30,14 @@
 LOCAL_CFLAGS += -DAUDIO_EXTN_INCALL_MUSIC_ENABLED
 endif
 
+
+ifeq ($(strip $(TARGET_BOARD_PLATFORM)),msm8916)
+LOCAL_CFLAGS += -DVOICE_CONCURRENCY
+LOCAL_CFLAGS += -DWFD_CONCURRENCY
+endif
+
+
+
 include $(BUILD_SHARED_LIBRARY)
 
 endif
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index 69587dc..5142353 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -418,12 +418,36 @@
 
 AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy(AudioSystem::stream_type stream)
 {
-#ifdef QCOM_INCALL_MUSIC_ENABLED
-    if (stream == AudioSystem::INCALL_MUSIC)
-        return STRATEGY_MEDIA;
+       // stream to strategy mapping
+        switch (stream) {
+        case AudioSystem::VOICE_CALL:
+        case AudioSystem::BLUETOOTH_SCO:
+            return STRATEGY_PHONE;
+        case AudioSystem::RING:
+        case AudioSystem::ALARM:
+            return STRATEGY_SONIFICATION;
+        case AudioSystem::NOTIFICATION:
+            return STRATEGY_SONIFICATION_RESPECTFUL;
+        case AudioSystem::DTMF:
+            return STRATEGY_DTMF;
+        default:
+            ALOGE("unknown stream type");
+        case AudioSystem::SYSTEM:
+            // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
+            // while key clicks are played produces a poor result
+        case AudioSystem::TTS:
+        case AudioSystem::MUSIC:
+#ifdef AUDIO_EXTN_INCALL_MUSIC_ENABLED
+        case AudioSystem::INCALL_MUSIC:
 #endif
+#ifdef QCOM_INCALL_MUSIC_ENABLED
+        case AudioSystem::INCALL_MUSIC:
+#endif
+             return STRATEGY_MEDIA;
+        case AudioSystem::ENFORCED_AUDIBLE:
+            return STRATEGY_ENFORCED_AUDIBLE;
+    }
 
-    return getStrategy(stream);
 }
 
 audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
@@ -881,6 +905,422 @@
 #endif
     return AudioPolicyManagerBase::computeVolume(stream, index, output, device);
 }
+
+
+audio_io_handle_t AudioPolicyManager::getOutput(AudioSystem::stream_type stream,
+                                    uint32_t samplingRate,
+                                    uint32_t format,
+                                    uint32_t channelMask,
+                                    AudioSystem::output_flags flags,
+                                    const audio_offload_info_t *offloadInfo)
+{
+    audio_io_handle_t output = 0;
+    uint32_t latency = 0;
+    routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+    audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+    IOProfile *profile = NULL;
+
+#ifdef VOICE_CONCURRENCY
+    if (isInCall()) {
+        ALOGV(" IN call mode adding ULL flags .. flags: %x ", flags );
+        //For voip paths
+        if(flags & AudioSystem::OUTPUT_FLAG_DIRECT)
+            flags = AudioSystem::OUTPUT_FLAG_DIRECT;
+        else //route every thing else to ULL path
+            flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+    }
+#endif
+
+#ifdef WFD_CONCURRENCY
+    if ((mAvailableOutputDevices & AUDIO_DEVICE_OUT_PROXY)
+          && (stream != AudioSystem::MUSIC)) {
+        ALOGV(" WFD mode adding ULL flags for non music stream.. flags: %x ", flags );
+        //For voip paths
+        if(flags & AudioSystem::OUTPUT_FLAG_DIRECT)
+            flags = AudioSystem::OUTPUT_FLAG_DIRECT;
+        else //route every thing else to ULL path
+            flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+    }
+#endif
+
+    ALOGV(" getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x ",
+          device, stream, samplingRate, format, channelMask, flags);
+
+
+
+#ifdef AUDIO_POLICY_TEST
+    if (mCurOutput != 0) {
+        ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
+                mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+        if (mTestOutputs[mCurOutput] == 0) {
+            ALOGV("getOutput() opening test output");
+            AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
+            outputDesc->mDevice = mTestDevice;
+            outputDesc->mSamplingRate = mTestSamplingRate;
+            outputDesc->mFormat = mTestFormat;
+            outputDesc->mChannelMask = mTestChannels;
+            outputDesc->mLatency = mTestLatencyMs;
+            outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
+            outputDesc->mRefCount[stream] = 0;
+            mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice,
+                                            &outputDesc->mSamplingRate,
+                                            &outputDesc->mFormat,
+                                            &outputDesc->mChannelMask,
+                                            &outputDesc->mLatency,
+                                            outputDesc->mFlags,
+                                            offloadInfo);
+            if (mTestOutputs[mCurOutput]) {
+                AudioParameter outputCmd = AudioParameter();
+                outputCmd.addInt(String8("set_id"),mCurOutput);
+                mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
+                addOutput(mTestOutputs[mCurOutput], outputDesc);
+            }
+        }
+        return mTestOutputs[mCurOutput];
+    }
+#endif //AUDIO_POLICY_TEST
+
+    // open a direct output if required by specified parameters
+    //force direct flag if offload flag is set: offloading implies a direct output stream
+    // and all common behaviors are driven by checking only the direct flag
+    // this should normally be set appropriately in the policy configuration file
+    if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+        flags = (AudioSystem::output_flags)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+    }
+
+    if ((format == AudioSystem::PCM_16_BIT) &&(AudioSystem::popCount(channelMask) > 2)) {
+        ALOGV("owerwrite flag(%x) for PCM16 multi-channel(CM:%x) playback", flags ,channelMask);
+        flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_DIRECT;
+    }
+
+    // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+    // creating an offloaded track and tearing it down immediately after start when audioflinger
+    // detects there is an active non offloadable effect.
+    // FIXME: We should check the audio session here but we do not have it in this context.
+    // This may prevent offloading in rare situations where effects are left active by apps
+    // in the background.
+    if ((((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
+            !isNonOffloadableEffectEnabled()) &&
+            flags & AUDIO_OUTPUT_FLAG_DIRECT) {
+        profile = getProfileForDirectOutput(device,
+                                           samplingRate,
+                                           format,
+                                           channelMask,
+                                           (audio_output_flags_t)flags);
+    }
+
+    if (profile != NULL) {
+        AudioOutputDescriptor *outputDesc = NULL;
+
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+            if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+                outputDesc = desc;
+                // reuse direct output if currently open and configured with same parameters
+                if ((samplingRate == outputDesc->mSamplingRate) &&
+                        (format == outputDesc->mFormat) &&
+                        (channelMask == outputDesc->mChannelMask)) {
+                    outputDesc->mDirectOpenCount++;
+                    ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
+                    return mOutputs.keyAt(i);
+                }
+            }
+        }
+        // close direct output if currently open and configured with different parameters
+        if (outputDesc != NULL) {
+            closeOutput(outputDesc->mId);
+        }
+        outputDesc = new AudioOutputDescriptor(profile);
+        outputDesc->mDevice = device;
+        outputDesc->mSamplingRate = samplingRate;
+        outputDesc->mFormat = (audio_format_t)format;
+        outputDesc->mChannelMask = (audio_channel_mask_t)channelMask;
+        outputDesc->mLatency = 0;
+        outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
+        outputDesc->mRefCount[stream] = 0;
+        outputDesc->mStopTime[stream] = 0;
+        outputDesc->mDirectOpenCount = 1;
+        output = mpClientInterface->openOutput(profile->mModule->mHandle,
+                                        &outputDesc->mDevice,
+                                        &outputDesc->mSamplingRate,
+                                        &outputDesc->mFormat,
+                                        &outputDesc->mChannelMask,
+                                        &outputDesc->mLatency,
+                                        outputDesc->mFlags,
+                                        offloadInfo);
+
+        // only accept an output with the requested parameters
+        if (output == 0 ||
+            (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
+            (format != 0 && format != outputDesc->mFormat) ||
+            (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
+            ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
+                    "format %d %d, channelMask %04x %04x", output, samplingRate,
+                    outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
+                    outputDesc->mChannelMask);
+            if (output != 0) {
+                mpClientInterface->closeOutput(output);
+            }
+            delete outputDesc;
+            return 0;
+        }
+        audio_io_handle_t srcOutput = getOutputForEffect();
+        addOutput(output, outputDesc);
+        audio_io_handle_t dstOutput = getOutputForEffect();
+        if (dstOutput == output) {
+            mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
+        }
+        mPreviousOutputs = mOutputs;
+        ALOGV("getOutput() returns new direct output %d", output);
+        return output;
+    }
+
+    // ignoring channel mask due to downmix capability in mixer
+
+    // open a non direct output
+
+    // for non direct outputs, only PCM is supported
+    if (audio_is_linear_pcm((audio_format_t)format)) {
+        // get which output is suitable for the specified stream. The actual
+        // routing change will happen when startOutput() will be called
+        SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+
+        output = selectOutput(outputs, flags);
+    }
+    ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
+            "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
+
+    ALOGV("getOutput() returns output %d", output);
+
+    return output;
+}
+
+
+// This function checks for the parameters which can be offloaded.
+// This can be enhanced depending on the capability of the DSP and policy
+// of the system.
+bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo)
+{
+    ALOGV(" isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
+     " BitRate=%u, duration=%lld us, has_video=%d",
+     offloadInfo.sample_rate, offloadInfo.channel_mask,
+     offloadInfo.format,
+     offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
+     offloadInfo.has_video);
+
+#ifdef VOICE_CONCURRENCY
+    if(isInCall())
+    {
+        ALOGD("\n  blocking  compress offload on call mode\n");
+        return false;
+    }
+#endif
+
+    // Check if offload has been disabled
+    char propValue[PROPERTY_VALUE_MAX];
+    if (property_get("audio.offload.disable", propValue, "0")) {
+        if (atoi(propValue) != 0) {
+            ALOGV("offload disabled by audio.offload.disable=%s", propValue );
+            return false;
+        }
+    }
+
+    // Check if stream type is music, then only allow offload as of now.
+    if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
+    {
+        ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
+        return false;
+    }
+
+    //TODO: enable audio offloading with video when ready
+    if (offloadInfo.has_video)
+    {
+        if(property_get("av.offload.enable", propValue, NULL)) {
+            bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+            if (!prop_enabled) {
+               ALOGW("offload disabled by av.offload.enable = %s ", propValue );
+               return false;
+            }
+        }
+        if(offloadInfo.is_streaming &&
+           property_get("av.streaming.offload.enable", propValue, NULL)) {
+            bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+            if (!prop_enabled) {
+               ALOGW("offload disabled by av.streaming.offload.enable = %s ", propValue );
+               return false;
+            }
+        }
+        ALOGV("isOffloadSupported: has_video == true, property\
+                set to enable offload");
+    }
+
+    //If duration is less than minimum value defined in property, return false
+    if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
+        if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
+            ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
+            return false;
+        }
+    } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
+        ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
+        //duration checks only valid for MP3/AAC formats,
+        //do not check duration for other audio formats, e.g. dolby AAC/AC3 and amrwb+ formats
+        if (offloadInfo.format == AUDIO_FORMAT_MP3 || offloadInfo.format == AUDIO_FORMAT_AAC)
+            return false;
+    }
+
+    // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+    // creating an offloaded track and tearing it down immediately after start when audioflinger
+    // detects there is an active non offloadable effect.
+    // FIXME: We should check the audio session here but we do not have it in this context.
+    // This may prevent offloading in rare situations where effects are left active by apps
+    // in the background.
+    if (isNonOffloadableEffectEnabled()) {
+        return false;
+    }
+
+    // See if there is a profile to support this.
+    // AUDIO_DEVICE_NONE
+    IOProfile *profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
+                                            offloadInfo.sample_rate,
+                                            offloadInfo.format,
+                                            offloadInfo.channel_mask,
+                                            AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+    ALOGV("isOffloadSupported() profile %sfound", profile != NULL ? "" : "NOT ");
+    return (profile != NULL);
+}
+
+void AudioPolicyManager::setPhoneState(int state)
+
+{
+    ALOGV("setPhoneState() state %d", state);
+    audio_devices_t newDevice = AUDIO_DEVICE_NONE;
+    if (state < 0 || state >= AudioSystem::NUM_MODES) {
+        ALOGW("setPhoneState() invalid state %d", state);
+        return;
+    }
+
+    if (state == mPhoneState ) {
+        ALOGW("setPhoneState() setting same state %d", state);
+        return;
+    }
+
+    // if leaving call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (isInCall()) {
+        ALOGV("setPhoneState() in call state management: new state is %d", state);
+        for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+            handleIncallSonification(stream, false, true);
+        }
+    }
+
+    // store previous phone state for management of sonification strategy below
+    int oldState = mPhoneState;
+    mPhoneState = state;
+    bool force = false;
+
+    // are we entering or starting a call
+    if (!isStateInCall(oldState) && isStateInCall(state)) {
+        ALOGV("  Entering call in setPhoneState()");
+        // force routing command to audio hardware when starting a call
+        // even if no device change is needed
+        force = true;
+        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+            mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+                    sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
+        }
+    } else if (isStateInCall(oldState) && !isStateInCall(state)) {
+        ALOGV("  Exiting call in setPhoneState()");
+        // force routing command to audio hardware when exiting a call
+        // even if no device change is needed
+        force = true;
+        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+            mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+                    sVolumeProfiles[AUDIO_STREAM_DTMF][j];
+        }
+    } else if (isStateInCall(state) && (state != oldState)) {
+        ALOGV("  Switching between telephony and VoIP in setPhoneState()");
+        // force routing command to audio hardware when switching between telephony and VoIP
+        // even if no device change is needed
+        force = true;
+    }
+
+    // check for device and output changes triggered by new phone state
+    newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/);
+    checkA2dpSuspend();
+    checkOutputForAllStrategies();
+    updateDevicesAndOutputs();
+
+    AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+
+    // force routing command to audio hardware when ending call
+    // even if no device change is needed
+    if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) {
+        newDevice = hwOutputDesc->device();
+    }
+
+    int delayMs = 0;
+    if (isStateInCall(state)) {
+        nsecs_t sysTime = systemTime();
+        for (size_t i = 0; i < mOutputs.size(); i++) {
+            AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+            // mute media and sonification strategies and delay device switch by the largest
+            // latency of any output where either strategy is active.
+            // This avoid sending the ring tone or music tail into the earpiece or headset.
+            if ((desc->isStrategyActive(STRATEGY_MEDIA,
+                                     SONIFICATION_HEADSET_MUSIC_DELAY,
+                                     sysTime) ||
+                    desc->isStrategyActive(STRATEGY_SONIFICATION,
+                                         SONIFICATION_HEADSET_MUSIC_DELAY,
+                                         sysTime)) &&
+                    (delayMs < (int)desc->mLatency*2)) {
+                delayMs = desc->mLatency*2;
+            }
+            setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
+            setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+                getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
+            setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
+            setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+                getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
+        }
+    }
+
+    // change routing is necessary
+    setOutputDevice(mPrimaryOutput, newDevice, force, delayMs);
+
+    // if entering in call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (isStateInCall(state)) {
+        ALOGV("setPhoneState() in call state management: new state is %d", state);
+        for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+            handleIncallSonification(stream, true, true);
+        }
+    }
+
+    // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
+    if (state == AudioSystem::MODE_RINGTONE &&
+        isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
+        mLimitRingtoneVolume = true;
+    } else {
+        mLimitRingtoneVolume = false;
+    }
+
+#ifdef VOICE_CONCURRENCY
+    //Call invalidate to reset all opened non ULL audio tracks
+    if(isInCall())
+    {
+        // Move tracks associated to this strategy from previous output to new output
+        for (int i = AudioSystem::SYSTEM; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+                ALOGV("\n Invalidate on call mode for stream :: %d  \n", i);
+                //FIXME see fixme on name change
+                mpClientInterface->setStreamOutput((AudioSystem::stream_type)i,
+                                                  0 /* ignored */);
+        }
+    }
+#endif
+
+}
+
 extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
 {
     return new AudioPolicyManager(clientInterface);
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
index 7a8cfa9..34ca701 100644
--- a/policy_hal/AudioPolicyManager.h
+++ b/policy_hal/AudioPolicyManager.h
@@ -48,6 +48,17 @@
                                             uint32_t format,
                                             uint32_t channels,
                                             AudioSystem::audio_in_acoustics acoustics);
+        virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+                                            uint32_t samplingRate = 0,
+                                            uint32_t format = AudioSystem::FORMAT_DEFAULT,
+                                            uint32_t channels = 0,
+                                            AudioSystem::output_flags flags =
+                                                    AudioSystem::OUTPUT_FLAG_INDIRECT,
+                                            const audio_offload_info_t *offloadInfo = NULL);
+
+        virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
+
+        virtual void setPhoneState(int state);
 protected:
         // return the strategy corresponding to a given stream type
         static routing_strategy getStrategy(AudioSystem::stream_type stream);