diff --git a/Android.mk b/Android.mk
index 0e94762..e293fd6 100644
--- a/Android.mk
+++ b/Android.mk
@@ -1,11 +1,15 @@
-ifneq ($(filter msm8960 msm8226 msm8x26 msm8610 msm8974 msm8x74 apq8084,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter mpq8092 msm8960 msm8226 msm8x26 msm8610 msm8974 msm8x74 apq8084,$(TARGET_BOARD_PLATFORM)),)
 
 MY_LOCAL_PATH := $(call my-dir)
 
 ifeq ($(BOARD_USES_LEGACY_ALSA_AUDIO),true)
 include $(MY_LOCAL_PATH)/legacy/Android.mk
 else
+ifneq ($(filter mpq8092,$(TARGET_BOARD_PLATFORM)),)
+include $(MY_LOCAL_PATH)/hal_mpq/Android.mk
+else
 include $(MY_LOCAL_PATH)/hal/Android.mk
+endif
 include $(MY_LOCAL_PATH)/voice_processing/Android.mk
 include $(MY_LOCAL_PATH)/mm-audio/Android.mk
 include $(MY_LOCAL_PATH)/policy_hal/Android.mk
diff --git a/hal_mpq/Android.mk b/hal_mpq/Android.mk
new file mode 100644
index 0000000..338ee40
--- /dev/null
+++ b/hal_mpq/Android.mk
@@ -0,0 +1,57 @@
+ifeq ($(strip $(BOARD_USES_ALSA_AUDIO)),true)
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_ARM_MODE := arm
+
+AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM)
+
+LOCAL_SRC_FILES := \
+	audio_hw.c \
+	$(AUDIO_PLATFORM)/hw_info.c \
+	$(AUDIO_PLATFORM)/platform.c
+
+ifneq ($(strip $(AUDIO_FEATURE_DISABLED_ANC_HEADSET)),true)
+    LOCAL_CFLAGS += -DANC_HEADSET_ENABLED
+endif
+
+ifneq ($(strip $(AUDIO_FEATURE_DISABLED_PROXY_DEVICE)),true)
+    LOCAL_CFLAGS += -DAFE_PROXY_ENABLED
+endif
+
+
+ifdef MULTIPLE_HW_VARIANTS_ENABLED
+  LOCAL_CFLAGS += -DHW_VARIANTS_ENABLED
+  LOCAL_SRC_FILES += $(AUDIO_PLATFORM)/hw_info.c
+endif
+
+LOCAL_SHARED_LIBRARIES := \
+	liblog \
+	libcutils \
+	libtinyalsa \
+	libtinycompress \
+	libaudioroute \
+	libdl
+
+LOCAL_C_INCLUDES := \
+	external/tinyalsa/include \
+	external/tinycompress/include \
+	$(call include-path-for, audio-route) \
+	$(call include-path-for, audio-effects) \
+	$(LOCAL_PATH)/$(AUDIO_PLATFORM)
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_AUXPCM_BT)),true)
+    LOCAL_CFLAGS += -DAUXPCM_BT_ENABLED
+endif
+
+LOCAL_MODULE := audio.primary.$(TARGET_BOARD_PLATFORM)
+
+LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
+
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
+
+endif
diff --git a/hal_mpq/audio_hw.c b/hal_mpq/audio_hw.c
new file mode 100644
index 0000000..8df46be
--- /dev/null
+++ b/hal_mpq/audio_hw.c
@@ -0,0 +1,2400 @@
+/*
+ * Copyright (c) 2013, 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 "audio_hw_primary"
+/*#define LOG_NDEBUG 0*/
+/*#define VERY_VERY_VERBOSE_LOGGING*/
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include <errno.h>
+#include <pthread.h>
+#include <stdint.h>
+#include <sys/time.h>
+#include <stdlib.h>
+#include <math.h>
+#include <dlfcn.h>
+#include <sys/resource.h>
+#include <sys/prctl.h>
+
+#include <cutils/log.h>
+#include <cutils/str_parms.h>
+#include <cutils/properties.h>
+#include <cutils/atomic.h>
+#include <cutils/sched_policy.h>
+
+#include <hardware/audio_effect.h>
+#include <system/thread_defs.h>
+#include <audio_effects/effect_aec.h>
+#include <audio_effects/effect_ns.h>
+#include "audio_hw.h"
+#include "platform_api.h"
+#include <platform.h>
+
+#include "sound/compress_params.h"
+
+#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
+#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
+/* ToDo: Check and update a proper value in msec */
+#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 96
+#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
+
+struct pcm_config pcm_config_deep_buffer = {
+    .channels = 2,
+    .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
+    .period_size = DEEP_BUFFER_OUTPUT_PERIOD_SIZE,
+    .period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT,
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4,
+    .stop_threshold = INT_MAX,
+    .avail_min = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4,
+};
+
+struct pcm_config pcm_config_low_latency = {
+    .channels = 2,
+    .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
+    .period_size = LOW_LATENCY_OUTPUT_PERIOD_SIZE,
+    .period_count = LOW_LATENCY_OUTPUT_PERIOD_COUNT,
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
+    .stop_threshold = INT_MAX,
+    .avail_min = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
+};
+
+struct pcm_config pcm_config_hdmi_multi = {
+    .channels = HDMI_MULTI_DEFAULT_CHANNEL_COUNT, /* changed when the stream is opened */
+    .rate = DEFAULT_OUTPUT_SAMPLING_RATE, /* changed when the stream is opened */
+    .period_size = HDMI_MULTI_PERIOD_SIZE,
+    .period_count = HDMI_MULTI_PERIOD_COUNT,
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = 0,
+    .stop_threshold = INT_MAX,
+    .avail_min = 0,
+};
+
+struct pcm_config pcm_config_audio_capture = {
+    .channels = 2,
+    .period_count = AUDIO_CAPTURE_PERIOD_COUNT,
+    .format = PCM_FORMAT_S16_LE,
+};
+
+const char * const use_case_table[AUDIO_USECASE_MAX] = {
+    [USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = "deep-buffer-playback",
+    [USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = "low-latency-playback",
+    [USECASE_AUDIO_PLAYBACK_MULTI_CH] = "multi-channel-playback",
+    [USECASE_AUDIO_PLAYBACK_OFFLOAD] = "compress-offload-playback",
+    [USECASE_AUDIO_RECORD] = "audio-record",
+    [USECASE_AUDIO_RECORD_COMPRESS] = "audio-record-compress",
+    [USECASE_AUDIO_RECORD_LOW_LATENCY] = "low-latency-record",
+    [USECASE_AUDIO_RECORD_FM_VIRTUAL] = "fm-virtual-record",
+    [USECASE_AUDIO_PLAYBACK_FM] = "play-fm",
+    [USECASE_VOICE_CALL] = "voice-call",
+
+    [USECASE_VOICE2_CALL] = "voice2-call",
+    [USECASE_VOLTE_CALL] = "volte-call",
+    [USECASE_QCHAT_CALL] = "qchat-call",
+    [USECASE_COMPRESS_VOIP_CALL] = "compress-voip-call",
+    [USECASE_INCALL_REC_UPLINK] = "incall-rec-uplink",
+    [USECASE_INCALL_REC_DOWNLINK] = "incall-rec-downlink",
+    [USECASE_INCALL_REC_UPLINK_AND_DOWNLINK] = "incall-rec-uplink-and-downlink",
+    [USECASE_INCALL_MUSIC_UPLINK] = "incall_music_uplink",
+    [USECASE_INCALL_MUSIC_UPLINK2] = "incall_music_uplink2",
+    [USECASE_AUDIO_SPKR_CALIB_RX] = "spkr-rx-calib",
+    [USECASE_AUDIO_SPKR_CALIB_TX] = "spkr-vi-record",
+};
+
+
+#define STRING_TO_ENUM(string) { #string, string }
+
+struct string_to_enum {
+    const char *name;
+    uint32_t value;
+};
+
+static const struct string_to_enum out_channels_name_to_enum_table[] = {
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+};
+
+static struct audio_device *adev = NULL;
+static pthread_mutex_t adev_init_lock;
+static unsigned int audio_device_ref_count;
+
+static int set_voice_volume_l(struct audio_device *adev, float volume);
+
+static bool is_supported_format(audio_format_t format)
+{
+    if (format == AUDIO_FORMAT_MP3 ||
+            format == AUDIO_FORMAT_AAC)
+        return true;
+
+    return false;
+}
+
+static int get_snd_codec_id(audio_format_t format)
+{
+    int id = 0;
+
+    switch (format) {
+    case AUDIO_FORMAT_MP3:
+        id = SND_AUDIOCODEC_MP3;
+        break;
+    case AUDIO_FORMAT_AAC:
+        id = SND_AUDIOCODEC_AAC;
+        break;
+    default:
+        ALOGE("%s: Unsupported audio format", __func__);
+    }
+
+    return id;
+}
+
+int enable_audio_route(struct audio_device *adev,
+                              struct audio_usecase *usecase,
+                              bool update_mixer)
+{
+    snd_device_t snd_device;
+    char mixer_path[MIXER_PATH_MAX_LENGTH];
+
+    if (usecase == NULL)
+        return -EINVAL;
+
+    ALOGV("%s: enter: usecase(%d)", __func__, usecase->id);
+
+    if (usecase->type == PCM_CAPTURE)
+        snd_device = usecase->in_snd_device;
+    else
+        snd_device = usecase->out_snd_device;
+
+    strlcpy(mixer_path, use_case_table[usecase->id], sizeof(mixer_path));
+    platform_add_backend_name(mixer_path, snd_device);
+    ALOGV("%s: apply mixer path: %s", __func__, mixer_path);
+    audio_route_apply_path(adev->audio_route, mixer_path);
+    if (update_mixer)
+        audio_route_update_mixer(adev->audio_route);
+
+    ALOGV("%s: exit", __func__);
+    return 0;
+}
+
+int disable_audio_route(struct audio_device *adev,
+                        struct audio_usecase *usecase,
+                        bool update_mixer)
+{
+    snd_device_t snd_device;
+    char mixer_path[MIXER_PATH_MAX_LENGTH];
+
+    if (usecase == NULL)
+        return -EINVAL;
+
+    ALOGV("%s: enter: usecase(%d)", __func__, usecase->id);
+    if (usecase->type == PCM_CAPTURE)
+        snd_device = usecase->in_snd_device;
+    else
+        snd_device = usecase->out_snd_device;
+    strlcpy(mixer_path, use_case_table[usecase->id], sizeof(mixer_path));
+    platform_add_backend_name(mixer_path, snd_device);
+    ALOGV("%s: reset mixer path: %s", __func__, mixer_path);
+    audio_route_reset_path(adev->audio_route, mixer_path);
+    if (update_mixer)
+        audio_route_update_mixer(adev->audio_route);
+
+    ALOGV("%s: exit", __func__);
+    return 0;
+}
+
+int enable_snd_device(struct audio_device *adev,
+                             snd_device_t snd_device,
+                             bool update_mixer)
+{
+    char device_name[DEVICE_NAME_MAX_SIZE] = {0};
+
+    if (snd_device < SND_DEVICE_MIN ||
+        snd_device >= SND_DEVICE_MAX) {
+        ALOGE("%s: Invalid sound device %d", __func__, snd_device);
+        return -EINVAL;
+    }
+
+    adev->snd_dev_ref_cnt[snd_device]++;
+
+    if(platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0 ) {
+        ALOGE("%s: Invalid sound device returned", __func__);
+        return -EINVAL;
+    }
+    if (adev->snd_dev_ref_cnt[snd_device] > 1) {
+        ALOGV("%s: snd_device(%d: %s) is already active",
+              __func__, snd_device, device_name);
+        return 0;
+    }
+
+    {
+        ALOGV("%s: snd_device(%d: %s)", __func__,
+        snd_device, device_name);
+        if (platform_send_audio_calibration(adev->platform, snd_device) < 0) {
+            adev->snd_dev_ref_cnt[snd_device]--;
+            return -EINVAL;
+        }
+        audio_route_apply_path(adev->audio_route, device_name);
+    }
+    if (update_mixer)
+        audio_route_update_mixer(adev->audio_route);
+
+    return 0;
+}
+
+int disable_snd_device(struct audio_device *adev,
+                       snd_device_t snd_device,
+                       bool update_mixer)
+{
+    char device_name[DEVICE_NAME_MAX_SIZE] = {0};
+
+    if (snd_device < SND_DEVICE_MIN ||
+        snd_device >= SND_DEVICE_MAX) {
+        ALOGE("%s: Invalid sound device %d", __func__, snd_device);
+        return -EINVAL;
+    }
+    if (adev->snd_dev_ref_cnt[snd_device] <= 0) {
+        ALOGE("%s: device ref cnt is already 0", __func__);
+        return -EINVAL;
+    }
+
+    adev->snd_dev_ref_cnt[snd_device]--;
+
+    if(platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0) {
+        ALOGE("%s: Invalid sound device returned", __func__);
+        return -EINVAL;
+    }
+
+    if (adev->snd_dev_ref_cnt[snd_device] == 0) {
+        ALOGV("%s: snd_device(%d: %s)", __func__,
+              snd_device, device_name);
+        audio_route_reset_path(adev->audio_route, device_name);
+
+        if (update_mixer)
+            audio_route_update_mixer(adev->audio_route);
+    }
+
+    return 0;
+}
+
+static void check_usecases_codec_backend(struct audio_device *adev,
+                                          struct audio_usecase *uc_info,
+                                          snd_device_t snd_device)
+{
+    struct listnode *node;
+    struct audio_usecase *usecase;
+    bool switch_device[AUDIO_USECASE_MAX];
+    int i, num_uc_to_switch = 0;
+
+    /*
+     * This function is to make sure that all the usecases that are active on
+     * the hardware codec backend are always routed to any one device that is
+     * handled by the hardware codec.
+     * For example, if low-latency and deep-buffer usecases are currently active
+     * on speaker and out_set_parameters(headset) is received on low-latency
+     * output, then we have to make sure deep-buffer is also switched to headset,
+     * because of the limitation that both the devices cannot be enabled
+     * at the same time as they share the same backend.
+     */
+    /* Disable all the usecases on the shared backend other than the
+       specified usecase */
+    for (i = 0; i < AUDIO_USECASE_MAX; i++)
+        switch_device[i] = false;
+
+    list_for_each(node, &adev->usecase_list) {
+        usecase = node_to_item(node, struct audio_usecase, list);
+        if (usecase->type == PCM_PLAYBACK &&
+                usecase != uc_info &&
+                usecase->out_snd_device != snd_device &&
+                usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) {
+            ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
+                  __func__, use_case_table[usecase->id],
+                  platform_get_snd_device_name(usecase->out_snd_device));
+            disable_audio_route(adev, usecase, false);
+            switch_device[usecase->id] = true;
+            num_uc_to_switch++;
+        }
+    }
+
+    if (num_uc_to_switch) {
+        /* Make sure all the streams are de-routed before disabling the device */
+        audio_route_update_mixer(adev->audio_route);
+
+        list_for_each(node, &adev->usecase_list) {
+            usecase = node_to_item(node, struct audio_usecase, list);
+            if (switch_device[usecase->id]) {
+                disable_snd_device(adev, usecase->out_snd_device, false);
+            }
+        }
+
+        list_for_each(node, &adev->usecase_list) {
+            usecase = node_to_item(node, struct audio_usecase, list);
+            if (switch_device[usecase->id]) {
+                enable_snd_device(adev, snd_device, false);
+            }
+        }
+        /* Make sure new snd device is enabled before re-routing the streams */
+        audio_route_update_mixer(adev->audio_route);
+
+        /* Re-route all the usecases on the shared backend other than the
+           specified usecase to new snd devices */
+        list_for_each(node, &adev->usecase_list) {
+            usecase = node_to_item(node, struct audio_usecase, list);
+            /* Update the out_snd_device only before enabling the audio route */
+            if (switch_device[usecase->id] ) {
+                usecase->out_snd_device = snd_device;
+                enable_audio_route(adev, usecase, false);
+            }
+        }
+
+        audio_route_update_mixer(adev->audio_route);
+    }
+}
+
+static void check_and_route_capture_usecases(struct audio_device *adev,
+                                             struct audio_usecase *uc_info,
+                                             snd_device_t snd_device)
+{
+    struct listnode *node;
+    struct audio_usecase *usecase;
+    bool switch_device[AUDIO_USECASE_MAX];
+    int i, num_uc_to_switch = 0;
+
+    /*
+     * This function is to make sure that all the active capture usecases
+     * are always routed to the same input sound device.
+     * For example, if audio-record and voice-call usecases are currently
+     * active on speaker(rx) and speaker-mic (tx) and out_set_parameters(earpiece)
+     * is received for voice call then we have to make sure that audio-record
+     * usecase is also switched to earpiece i.e. voice-dmic-ef,
+     * because of the limitation that two devices cannot be enabled
+     * at the same time if they share the same backend.
+     */
+    for (i = 0; i < AUDIO_USECASE_MAX; i++)
+        switch_device[i] = false;
+
+    list_for_each(node, &adev->usecase_list) {
+        usecase = node_to_item(node, struct audio_usecase, list);
+        if (usecase->type == PCM_CAPTURE &&
+                usecase != uc_info &&
+                usecase->in_snd_device != snd_device) {
+            ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
+                  __func__, use_case_table[usecase->id],
+                  platform_get_snd_device_name(usecase->in_snd_device));
+            disable_audio_route(adev, usecase, false);
+            switch_device[usecase->id] = true;
+            num_uc_to_switch++;
+        }
+    }
+
+    if (num_uc_to_switch) {
+        /* Make sure all the streams are de-routed before disabling the device */
+        audio_route_update_mixer(adev->audio_route);
+
+        list_for_each(node, &adev->usecase_list) {
+            usecase = node_to_item(node, struct audio_usecase, list);
+            if (switch_device[usecase->id]) {
+                disable_snd_device(adev, usecase->in_snd_device, false);
+                enable_snd_device(adev, snd_device, false);
+            }
+        }
+
+        /* Make sure new snd device is enabled before re-routing the streams */
+        audio_route_update_mixer(adev->audio_route);
+
+        /* Re-route all the usecases on the shared backend other than the
+           specified usecase to new snd devices */
+        list_for_each(node, &adev->usecase_list) {
+            usecase = node_to_item(node, struct audio_usecase, list);
+            /* Update the in_snd_device only before enabling the audio route */
+            if (switch_device[usecase->id] ) {
+                usecase->in_snd_device = snd_device;
+                enable_audio_route(adev, usecase, false);
+            }
+        }
+
+        audio_route_update_mixer(adev->audio_route);
+    }
+}
+
+static int disable_all_usecases_of_type(struct audio_device *adev,
+                                        usecase_type_t usecase_type,
+                                        bool update_mixer)
+{
+    struct audio_usecase *usecase;
+    struct listnode *node;
+    int ret = 0;
+
+    list_for_each(node, &adev->usecase_list) {
+        usecase = node_to_item(node, struct audio_usecase, list);
+        if (usecase->type == usecase_type) {
+            ALOGV("%s: usecase id %d", __func__, usecase->id);
+            ret = disable_audio_route(adev, usecase, update_mixer);
+            if (ret) {
+                ALOGE("%s: Failed to disable usecase id %d",
+                      __func__, usecase->id);
+            }
+        }
+    }
+
+    return ret;
+}
+
+static int enable_all_usecases_of_type(struct audio_device *adev,
+                                       usecase_type_t usecase_type,
+                                       bool update_mixer)
+{
+    struct audio_usecase *usecase;
+    struct listnode *node;
+    int ret = 0;
+
+    list_for_each(node, &adev->usecase_list) {
+        usecase = node_to_item(node, struct audio_usecase, list);
+        if (usecase->type == usecase_type) {
+            ALOGV("%s: usecase id %d", __func__, usecase->id);
+            ret = enable_audio_route(adev, usecase, update_mixer);
+            if (ret) {
+                ALOGE("%s: Failed to enable usecase id %d",
+                      __func__, usecase->id);
+            }
+        }
+    }
+
+    return ret;
+}
+
+/* must be called with hw device mutex locked */
+static int read_hdmi_channel_masks(struct stream_out *out)
+{
+    int ret = 0;
+    int channels = platform_edid_get_max_channels(out->dev->platform);
+
+    switch (channels) {
+        /*
+         * Do not handle stereo output in Multi-channel cases
+         * Stereo case is handled in normal playback path
+         */
+    case 6:
+        ALOGV("%s: HDMI supports 5.1", __func__);
+        out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
+        break;
+    case 8:
+        ALOGV("%s: HDMI supports 5.1 and 7.1 channels", __func__);
+        out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
+        out->supported_channel_masks[1] = AUDIO_CHANNEL_OUT_7POINT1;
+        break;
+    default:
+        ALOGE("HDMI does not support multi channel playback");
+        ret = -ENOSYS;
+        break;
+    }
+    return ret;
+}
+
+static audio_usecase_t get_voice_usecase_id_from_list(struct audio_device *adev)
+{
+    struct audio_usecase *usecase;
+    struct listnode *node;
+
+    list_for_each(node, &adev->usecase_list) {
+        usecase = node_to_item(node, struct audio_usecase, list);
+        if (usecase->type == VOICE_CALL) {
+            ALOGV("%s: usecase id %d", __func__, usecase->id);
+            return usecase->id;
+        }
+    }
+    return USECASE_INVALID;
+}
+
+struct audio_usecase *get_usecase_from_list(struct audio_device *adev,
+                                            audio_usecase_t uc_id)
+{
+    struct audio_usecase *usecase;
+    struct listnode *node;
+
+    list_for_each(node, &adev->usecase_list) {
+        usecase = node_to_item(node, struct audio_usecase, list);
+        if (usecase->id == uc_id)
+            return usecase;
+    }
+    return NULL;
+}
+
+int select_devices(struct audio_device *adev, audio_usecase_t uc_id)
+{
+    snd_device_t out_snd_device = SND_DEVICE_NONE;
+    snd_device_t in_snd_device = SND_DEVICE_NONE;
+    struct audio_usecase *usecase = NULL;
+    struct audio_usecase *vc_usecase = NULL;
+    struct audio_usecase *voip_usecase = NULL;
+    struct listnode *node;
+    int status = 0;
+
+    usecase = get_usecase_from_list(adev, uc_id);
+    if (usecase == NULL) {
+        ALOGE("%s: Could not find the usecase(%d)", __func__, uc_id);
+        return -EINVAL;
+    }
+
+    if ((usecase->type == VOICE_CALL) ||
+        (usecase->type == VOIP_CALL)) {
+        out_snd_device = platform_get_output_snd_device(adev->platform,
+                                                        usecase->stream.out->devices);
+        in_snd_device = platform_get_input_snd_device(adev->platform, usecase->stream.out->devices);
+        usecase->devices = usecase->stream.out->devices;
+    } else {
+        /*
+         * If the voice call is active, use the sound devices of voice call usecase
+         * so that it would not result any device switch. All the usecases will
+         * be switched to new device when select_devices() is called for voice call
+         * usecase. This is to avoid switching devices for voice call when
+         * check_usecases_codec_backend() is called below.
+         */
+        if (usecase->type == PCM_PLAYBACK) {
+            usecase->devices = usecase->stream.out->devices;
+            in_snd_device = SND_DEVICE_NONE;
+            if (out_snd_device == SND_DEVICE_NONE) {
+                out_snd_device = platform_get_output_snd_device(adev->platform,
+                                            usecase->stream.out->devices);
+                if (usecase->stream.out == adev->primary_output &&
+                        adev->active_input &&
+                        adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
+                    select_devices(adev, adev->active_input->usecase);
+                }
+            }
+        } else if (usecase->type == PCM_CAPTURE) {
+            usecase->devices = usecase->stream.in->device;
+            out_snd_device = SND_DEVICE_NONE;
+            if (in_snd_device == SND_DEVICE_NONE) {
+                if (adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION &&
+                        adev->primary_output && !adev->primary_output->standby) {
+                    in_snd_device = platform_get_input_snd_device(adev->platform,
+                                        adev->primary_output->devices);
+                } else {
+                    in_snd_device = platform_get_input_snd_device(adev->platform,
+                                                                  AUDIO_DEVICE_NONE);
+                }
+            }
+        }
+    }
+
+    if (out_snd_device == usecase->out_snd_device &&
+        in_snd_device == usecase->in_snd_device) {
+        return 0;
+    }
+
+    ALOGD("%s: out_snd_device(%d: %s) in_snd_device(%d: %s)", __func__,
+          out_snd_device, platform_get_snd_device_name(out_snd_device),
+          in_snd_device,  platform_get_snd_device_name(in_snd_device));
+
+    /*
+     * Limitation: While in call, to do a device switch we need to disable
+     * and enable both RX and TX devices though one of them is same as current
+     * device.
+     */
+    if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) {
+        status = platform_switch_voice_call_device_pre(adev->platform);
+        disable_all_usecases_of_type(adev, VOICE_CALL, true);
+    }
+
+    /* Disable current sound devices */
+    if (usecase->out_snd_device != SND_DEVICE_NONE) {
+        disable_audio_route(adev, usecase, true);
+        disable_snd_device(adev, usecase->out_snd_device, false);
+    }
+
+    if (usecase->in_snd_device != SND_DEVICE_NONE) {
+        disable_audio_route(adev, usecase, true);
+        disable_snd_device(adev, usecase->in_snd_device, false);
+    }
+
+    /* Enable new sound devices */
+    if (out_snd_device != SND_DEVICE_NONE) {
+        if (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)
+            check_usecases_codec_backend(adev, usecase, out_snd_device);
+        enable_snd_device(adev, out_snd_device, false);
+    }
+
+    if (in_snd_device != SND_DEVICE_NONE) {
+        check_and_route_capture_usecases(adev, usecase, in_snd_device);
+        enable_snd_device(adev, in_snd_device, false);
+    }
+
+    if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL)
+        status = platform_switch_voice_call_device_post(adev->platform,
+                                                        out_snd_device,
+                                                        in_snd_device);
+
+    audio_route_update_mixer(adev->audio_route);
+
+    usecase->in_snd_device = in_snd_device;
+    usecase->out_snd_device = out_snd_device;
+
+    if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL)
+        enable_all_usecases_of_type(adev, usecase->type, true);
+    else
+        enable_audio_route(adev, usecase, true);
+
+    /* Applicable only on the targets that has external modem.
+     * Enable device command should be sent to modem only after
+     * enabling voice call mixer controls
+     */
+    if (usecase->type == VOICE_CALL)
+        status = platform_switch_voice_call_usecase_route_post(adev->platform,
+                                                               out_snd_device,
+                                                               in_snd_device);
+
+    return status;
+}
+
+static int stop_input_stream(struct stream_in *in)
+{
+    int i, ret = 0;
+    struct audio_usecase *uc_info;
+    struct audio_device *adev = in->dev;
+
+    adev->active_input = NULL;
+
+    ALOGV("%s: enter: usecase(%d: %s)", __func__,
+          in->usecase, use_case_table[in->usecase]);
+    uc_info = get_usecase_from_list(adev, in->usecase);
+    if (uc_info == NULL) {
+        ALOGE("%s: Could not find the usecase (%d) in the list",
+              __func__, in->usecase);
+        return -EINVAL;
+    }
+
+    /* 1. Disable stream specific mixer controls */
+    disable_audio_route(adev, uc_info, true);
+
+    /* 2. Disable the tx device */
+    disable_snd_device(adev, uc_info->in_snd_device, true);
+
+    list_remove(&uc_info->list);
+    free(uc_info);
+
+    ALOGV("%s: exit: status(%d)", __func__, ret);
+    return ret;
+}
+
+int start_input_stream(struct stream_in *in)
+{
+    /* 1. Enable output device and stream routing controls */
+    int ret = 0;
+    struct audio_usecase *uc_info;
+    struct audio_device *adev = in->dev;
+
+    in->usecase = platform_update_usecase_from_source(in->source,in->usecase);
+    ALOGV("%s: enter: usecase(%d)", __func__, in->usecase);
+
+    in->pcm_device_id = platform_get_pcm_device_id(in->usecase, PCM_CAPTURE);
+    if (in->pcm_device_id < 0) {
+        ALOGE("%s: Could not find PCM device id for the usecase(%d)",
+              __func__, in->usecase);
+        ret = -EINVAL;
+        goto error_config;
+    }
+
+    adev->active_input = in;
+    uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
+    uc_info->id = in->usecase;
+    uc_info->type = PCM_CAPTURE;
+    uc_info->stream.in = in;
+    uc_info->devices = in->device;
+    uc_info->in_snd_device = SND_DEVICE_NONE;
+    uc_info->out_snd_device = SND_DEVICE_NONE;
+
+    list_add_tail(&adev->usecase_list, &uc_info->list);
+    select_devices(adev, in->usecase);
+
+    ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d",
+          __func__, SOUND_CARD, in->pcm_device_id, in->config.channels);
+    in->pcm = pcm_open(SOUND_CARD, in->pcm_device_id,
+                           PCM_IN, &in->config);
+    if (in->pcm && !pcm_is_ready(in->pcm)) {
+        ALOGE("%s: %s", __func__, pcm_get_error(in->pcm));
+        pcm_close(in->pcm);
+        in->pcm = NULL;
+        ret = -EIO;
+        goto error_open;
+    }
+    ALOGV("%s: exit", __func__);
+    return ret;
+
+error_open:
+    stop_input_stream(in);
+
+error_config:
+    adev->active_input = NULL;
+    ALOGD("%s: exit: status(%d)", __func__, ret);
+
+    return ret;
+}
+
+/* must be called with out->lock locked */
+static int send_offload_cmd_l(struct stream_out* out, int command)
+{
+    struct offload_cmd *cmd = (struct offload_cmd *)calloc(1, sizeof(struct offload_cmd));
+
+    ALOGVV("%s %d", __func__, command);
+
+    cmd->cmd = command;
+    list_add_tail(&out->offload_cmd_list, &cmd->node);
+    pthread_cond_signal(&out->offload_cond);
+    return 0;
+}
+
+/* must be called iwth out->lock locked */
+static void stop_compressed_output_l(struct stream_out *out)
+{
+    out->offload_state = OFFLOAD_STATE_IDLE;
+    out->playback_started = 0;
+    out->send_new_metadata = 1;
+    if (out->compr != NULL) {
+        compress_stop(out->compr);
+        while (out->offload_thread_blocked) {
+            pthread_cond_wait(&out->cond, &out->lock);
+        }
+    }
+}
+
+static void *offload_thread_loop(void *context)
+{
+    struct stream_out *out = (struct stream_out *) context;
+    struct listnode *item;
+
+    out->offload_state = OFFLOAD_STATE_IDLE;
+    out->playback_started = 0;
+
+    setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
+    set_sched_policy(0, SP_FOREGROUND);
+    prctl(PR_SET_NAME, (unsigned long)"Offload Callback", 0, 0, 0);
+
+    ALOGV("%s", __func__);
+    pthread_mutex_lock(&out->lock);
+    for (;;) {
+        struct offload_cmd *cmd = NULL;
+        stream_callback_event_t event;
+        bool send_callback = false;
+
+        ALOGVV("%s offload_cmd_list %d out->offload_state %d",
+              __func__, list_empty(&out->offload_cmd_list),
+              out->offload_state);
+        if (list_empty(&out->offload_cmd_list)) {
+            ALOGV("%s SLEEPING", __func__);
+            pthread_cond_wait(&out->offload_cond, &out->lock);
+            ALOGV("%s RUNNING", __func__);
+            continue;
+        }
+
+        item = list_head(&out->offload_cmd_list);
+        cmd = node_to_item(item, struct offload_cmd, node);
+        list_remove(item);
+
+        ALOGVV("%s STATE %d CMD %d out->compr %p",
+               __func__, out->offload_state, cmd->cmd, out->compr);
+
+        if (cmd->cmd == OFFLOAD_CMD_EXIT) {
+            free(cmd);
+            break;
+        }
+
+        if (out->compr == NULL) {
+            ALOGE("%s: Compress handle is NULL", __func__);
+            pthread_cond_signal(&out->cond);
+            continue;
+        }
+        out->offload_thread_blocked = true;
+        pthread_mutex_unlock(&out->lock);
+        send_callback = false;
+        switch(cmd->cmd) {
+        case OFFLOAD_CMD_WAIT_FOR_BUFFER:
+            compress_wait(out->compr, -1);
+            send_callback = true;
+            event = STREAM_CBK_EVENT_WRITE_READY;
+            break;
+        case OFFLOAD_CMD_PARTIAL_DRAIN:
+            compress_next_track(out->compr);
+            compress_partial_drain(out->compr);
+            send_callback = true;
+            event = STREAM_CBK_EVENT_DRAIN_READY;
+            break;
+        case OFFLOAD_CMD_DRAIN:
+            compress_drain(out->compr);
+            send_callback = true;
+            event = STREAM_CBK_EVENT_DRAIN_READY;
+            break;
+        default:
+            ALOGE("%s unknown command received: %d", __func__, cmd->cmd);
+            break;
+        }
+        pthread_mutex_lock(&out->lock);
+        out->offload_thread_blocked = false;
+        pthread_cond_signal(&out->cond);
+        if (send_callback) {
+            out->offload_callback(event, NULL, out->offload_cookie);
+        }
+        free(cmd);
+    }
+
+    pthread_cond_signal(&out->cond);
+    while (!list_empty(&out->offload_cmd_list)) {
+        item = list_head(&out->offload_cmd_list);
+        list_remove(item);
+        free(node_to_item(item, struct offload_cmd, node));
+    }
+    pthread_mutex_unlock(&out->lock);
+
+    return NULL;
+}
+
+static int create_offload_callback_thread(struct stream_out *out)
+{
+    pthread_cond_init(&out->offload_cond, (const pthread_condattr_t *) NULL);
+    list_init(&out->offload_cmd_list);
+    pthread_create(&out->offload_thread, (const pthread_attr_t *) NULL,
+                    offload_thread_loop, out);
+    return 0;
+}
+
+static int destroy_offload_callback_thread(struct stream_out *out)
+{
+    pthread_mutex_lock(&out->lock);
+    stop_compressed_output_l(out);
+    send_offload_cmd_l(out, OFFLOAD_CMD_EXIT);
+
+    pthread_mutex_unlock(&out->lock);
+    pthread_join(out->offload_thread, (void **) NULL);
+    pthread_cond_destroy(&out->offload_cond);
+
+    return 0;
+}
+
+static bool allow_hdmi_channel_config(struct audio_device *adev)
+{
+    struct listnode *node;
+    struct audio_usecase *usecase;
+    bool ret = true;
+
+    list_for_each(node, &adev->usecase_list) {
+        usecase = node_to_item(node, struct audio_usecase, list);
+        if (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+            /*
+             * If voice call is already existing, do not proceed further to avoid
+             * disabling/enabling both RX and TX devices, CSD calls, etc.
+             * Once the voice call done, the HDMI channels can be configured to
+             * max channels of remaining use cases.
+             */
+            if (usecase->id == USECASE_VOICE_CALL) {
+                ALOGD("%s: voice call is active, no change in HDMI channels",
+                      __func__);
+                ret = false;
+                break;
+            } else if (usecase->id == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
+                ALOGD("%s: multi channel playback is active, "
+                      "no change in HDMI channels", __func__);
+                ret = false;
+                break;
+            }
+        }
+    }
+    return ret;
+}
+
+static int check_and_set_hdmi_channels(struct audio_device *adev,
+                                       unsigned int channels)
+{
+    struct listnode *node;
+    struct audio_usecase *usecase;
+
+    /* Check if change in HDMI channel config is allowed */
+    if (!allow_hdmi_channel_config(adev))
+        return 0;
+
+    if (channels == adev->cur_hdmi_channels) {
+        ALOGD("%s: Requested channels are same as current", __func__);
+        return 0;
+    }
+
+    platform_set_hdmi_channels(adev->platform, channels);
+    adev->cur_hdmi_channels = channels;
+
+    /*
+     * Deroute all the playback streams routed to HDMI so that
+     * the back end is deactivated. Note that backend will not
+     * be deactivated if any one stream is connected to it.
+     */
+    list_for_each(node, &adev->usecase_list) {
+        usecase = node_to_item(node, struct audio_usecase, list);
+        if (usecase->type == PCM_PLAYBACK &&
+                usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+            disable_audio_route(adev, usecase, true);
+        }
+    }
+
+    /*
+     * Enable all the streams disabled above. Now the HDMI backend
+     * will be activated with new channel configuration
+     */
+    list_for_each(node, &adev->usecase_list) {
+        usecase = node_to_item(node, struct audio_usecase, list);
+        if (usecase->type == PCM_PLAYBACK &&
+                usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+            enable_audio_route(adev, usecase, true);
+        }
+    }
+
+    return 0;
+}
+
+static int stop_output_stream(struct stream_out *out)
+{
+    int i, ret = 0;
+    struct audio_usecase *uc_info;
+    struct audio_device *adev = out->dev;
+
+    ALOGV("%s: enter: usecase(%d: %s)", __func__,
+          out->usecase, use_case_table[out->usecase]);
+    uc_info = get_usecase_from_list(adev, out->usecase);
+    if (uc_info == NULL) {
+        ALOGE("%s: Could not find the usecase (%d) in the list",
+              __func__, out->usecase);
+        return -EINVAL;
+    }
+
+    if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD &&
+            adev->visualizer_stop_output != NULL)
+        adev->visualizer_stop_output(out->handle);
+
+    /* 1. Get and set stream specific mixer controls */
+    disable_audio_route(adev, uc_info, true);
+
+    /* 2. Disable the rx device */
+    disable_snd_device(adev, uc_info->out_snd_device, true);
+
+    list_remove(&uc_info->list);
+    free(uc_info);
+
+    /* Must be called after removing the usecase from list */
+    if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
+        check_and_set_hdmi_channels(adev, DEFAULT_HDMI_OUT_CHANNELS);
+
+    ALOGV("%s: exit: status(%d)", __func__, ret);
+    return ret;
+}
+
+int start_output_stream(struct stream_out *out)
+{
+    int ret = 0;
+    struct audio_usecase *uc_info;
+    struct audio_device *adev = out->dev;
+
+    ALOGV("%s: enter: usecase(%d: %s) devices(%#x)",
+          __func__, out->usecase, use_case_table[out->usecase], out->devices);
+    out->pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK);
+    if (out->pcm_device_id < 0) {
+        ALOGE("%s: Invalid PCM device id(%d) for the usecase(%d)",
+              __func__, out->pcm_device_id, out->usecase);
+        ret = -EINVAL;
+        goto error_config;
+    }
+
+    uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
+    uc_info->id = out->usecase;
+    uc_info->type = PCM_PLAYBACK;
+    uc_info->stream.out = out;
+    uc_info->devices = out->devices;
+    uc_info->in_snd_device = SND_DEVICE_NONE;
+    uc_info->out_snd_device = SND_DEVICE_NONE;
+
+    /* This must be called before adding this usecase to the list */
+    if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
+        check_and_set_hdmi_channels(adev, out->config.channels);
+
+    list_add_tail(&adev->usecase_list, &uc_info->list);
+
+    select_devices(adev, out->usecase);
+
+    ALOGV("%s: Opening PCM device card_id(%d) device_id(%d)",
+          __func__, 0, out->pcm_device_id);
+    if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        out->pcm = pcm_open(SOUND_CARD, out->pcm_device_id,
+                               PCM_OUT | PCM_MONOTONIC, &out->config);
+        if (out->pcm && !pcm_is_ready(out->pcm)) {
+            ALOGE("%s: %s", __func__, pcm_get_error(out->pcm));
+            pcm_close(out->pcm);
+            out->pcm = NULL;
+            ret = -EIO;
+            goto error_open;
+        }
+    } else {
+        out->pcm = NULL;
+        out->compr = compress_open(SOUND_CARD, out->pcm_device_id,
+                                   COMPRESS_IN, &out->compr_config);
+        if (out->compr && !is_compress_ready(out->compr)) {
+            ALOGE("%s: %s", __func__, compress_get_error(out->compr));
+            compress_close(out->compr);
+            out->compr = NULL;
+            ret = -EIO;
+            goto error_open;
+        }
+        if (out->offload_callback)
+            compress_nonblock(out->compr, out->non_blocking);
+
+        if (adev->visualizer_start_output != NULL)
+            adev->visualizer_start_output(out->handle);
+    }
+    ALOGV("%s: exit", __func__);
+    return 0;
+error_open:
+    stop_output_stream(out);
+error_config:
+    return ret;
+}
+
+static int check_input_parameters(uint32_t sample_rate,
+                                  audio_format_t format,
+                                  int channel_count)
+{
+    int ret = 0;
+
+    if ((format != AUDIO_FORMAT_PCM_16_BIT))  ret = -EINVAL;
+
+    switch (channel_count) {
+    case 1:
+    case 2:
+    case 6:
+        break;
+    default:
+        ret = -EINVAL;
+    }
+
+    switch (sample_rate) {
+    case 8000:
+    case 11025:
+    case 12000:
+    case 16000:
+    case 22050:
+    case 24000:
+    case 32000:
+    case 44100:
+    case 48000:
+        break;
+    default:
+        ret = -EINVAL;
+    }
+
+    return ret;
+}
+
+static size_t get_input_buffer_size(uint32_t sample_rate,
+                                    audio_format_t format,
+                                    int channel_count)
+{
+    size_t size = 0;
+
+    if (check_input_parameters(sample_rate, format, channel_count) != 0)
+        return 0;
+
+    size = (sample_rate * AUDIO_CAPTURE_PERIOD_DURATION_MSEC) / 1000;
+    /* ToDo: should use frame_size computed based on the format and
+       channel_count here. */
+    size *= sizeof(short) * channel_count;
+
+    /* make sure the size is multiple of 64 */
+    size += 0x3f;
+    size &= ~0x3f;
+
+    return size;
+}
+
+static uint32_t out_get_sample_rate(const struct audio_stream *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+
+    return out->sample_rate;
+}
+
+static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    return -ENOSYS;
+}
+
+static size_t out_get_buffer_size(const struct audio_stream *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+
+    if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD)
+        return out->compr_config.fragment_size;
+
+    return out->config.period_size * audio_stream_frame_size(stream);
+}
+
+static uint32_t out_get_channels(const struct audio_stream *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+
+    return out->channel_mask;
+}
+
+static audio_format_t out_get_format(const struct audio_stream *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+
+    return out->format;
+}
+
+static int out_set_format(struct audio_stream *stream, audio_format_t format)
+{
+    return -ENOSYS;
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    struct audio_device *adev = out->dev;
+
+    ALOGV("%s: enter: usecase(%d: %s)", __func__,
+          out->usecase, use_case_table[out->usecase]);
+    if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
+        /* Ignore standby in case of voip call because the voip output
+         * stream is closed in adev_close_output_stream()
+         */
+        ALOGV("%s: Ignore Standby in VOIP call", __func__);
+        return 0;
+    }
+
+    pthread_mutex_lock(&out->lock);
+    pthread_mutex_lock(&adev->lock);
+    if (!out->standby) {
+        out->standby = true;
+        if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+            if (out->pcm) {
+                pcm_close(out->pcm);
+                out->pcm = NULL;
+            }
+        } else {
+            stop_compressed_output_l(out);
+            out->gapless_mdata.encoder_delay = 0;
+            out->gapless_mdata.encoder_padding = 0;
+            if (out->compr != NULL) {
+                compress_close(out->compr);
+                out->compr = NULL;
+            }
+        }
+        stop_output_stream(out);
+    }
+    pthread_mutex_unlock(&adev->lock);
+    pthread_mutex_unlock(&out->lock);
+    ALOGV("%s: exit", __func__);
+    return 0;
+}
+
+static int out_dump(const struct audio_stream *stream, int fd)
+{
+    return 0;
+}
+
+static int parse_compress_metadata(struct stream_out *out, struct str_parms *parms)
+{
+    int ret = 0;
+    char value[32];
+    struct compr_gapless_mdata tmp_mdata;
+
+    if (!out || !parms) {
+        return -EINVAL;
+    }
+
+    ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value));
+    if (ret >= 0) {
+        tmp_mdata.encoder_delay = atoi(value); //whats a good limit check?
+    } else {
+        return -EINVAL;
+    }
+
+    ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES, value, sizeof(value));
+    if (ret >= 0) {
+        tmp_mdata.encoder_padding = atoi(value);
+    } else {
+        return -EINVAL;
+    }
+
+    out->gapless_mdata = tmp_mdata;
+    out->send_new_metadata = 1;
+    ALOGV("%s new encoder delay %u and padding %u", __func__,
+          out->gapless_mdata.encoder_delay, out->gapless_mdata.encoder_padding);
+
+    return 0;
+}
+
+
+static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    struct audio_device *adev = out->dev;
+    struct audio_usecase *usecase;
+    struct listnode *node;
+    struct str_parms *parms;
+    char value[32];
+    int ret, val = 0;
+    bool select_new_device = false;
+
+    ALOGD("%s: enter: usecase(%d: %s) kvpairs: %s",
+          __func__, out->usecase, use_case_table[out->usecase], kvpairs);
+    parms = str_parms_create_str(kvpairs);
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+    if (ret >= 0) {
+        val = atoi(value);
+        pthread_mutex_lock(&out->lock);
+        pthread_mutex_lock(&adev->lock);
+
+        /*
+         * When HDMI cable is unplugged the music playback is paused and
+         * the policy manager sends routing=0. But the audioflinger
+         * continues to write data until standby time (3sec).
+         * As the HDMI core is turned off, the write gets blocked.
+         * Avoid this by routing audio to speaker until standby.
+         */
+        if (out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL &&
+                val == AUDIO_DEVICE_NONE) {
+            val = AUDIO_DEVICE_OUT_SPEAKER;
+        }
+
+        /*
+         * select_devices() call below switches all the usecases on the same
+         * backend to the new device. Refer to check_usecases_codec_backend() in
+         * the select_devices(). But how do we undo this?
+         *
+         * For example, music playback is active on headset (deep-buffer usecase)
+         * and if we go to ringtones and select a ringtone, low-latency usecase
+         * will be started on headset+speaker. As we can't enable headset+speaker
+         * and headset devices at the same time, select_devices() switches the music
+         * playback to headset+speaker while starting low-lateny usecase for ringtone.
+         * So when the ringtone playback is completed, how do we undo the same?
+         *
+         * We are relying on the out_set_parameters() call on deep-buffer output,
+         * once the ringtone playback is ended.
+         * NOTE: We should not check if the current devices are same as new devices.
+         *       Because select_devices() must be called to switch back the music
+         *       playback to headset.
+         */
+        if (val != 0) {
+            out->devices = val;
+
+            if (!out->standby)
+                select_devices(adev, out->usecase);
+        }
+
+        pthread_mutex_unlock(&adev->lock);
+        pthread_mutex_unlock(&out->lock);
+    }
+
+    if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        parse_compress_metadata(out, parms);
+    }
+
+    str_parms_destroy(parms);
+    ALOGV("%s: exit: code(%d)", __func__, ret);
+    return ret;
+}
+
+static char* out_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    struct str_parms *query = str_parms_create_str(keys);
+    char *str;
+    char value[256];
+    struct str_parms *reply = str_parms_create();
+    size_t i, j;
+    int ret;
+    bool first = true;
+    ALOGV("%s: enter: keys - %s", __func__, keys);
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value));
+    if (ret >= 0) {
+        value[0] = '\0';
+        i = 0;
+        while (out->supported_channel_masks[i] != 0) {
+            for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) {
+                if (out_channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) {
+                    if (!first) {
+                        strlcat(value, "|", sizeof(value));
+                    }
+                    strlcat(value, out_channels_name_to_enum_table[j].name, sizeof(value));
+                    first = false;
+                    break;
+                }
+            }
+            i++;
+        }
+        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value);
+        str = str_parms_to_str(reply);
+    }
+    str_parms_destroy(query);
+    str_parms_destroy(reply);
+    ALOGV("%s: exit: returns - %s", __func__, str);
+    return str;
+}
+
+static uint32_t out_get_latency(const struct audio_stream_out *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+
+    if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD)
+        return COMPRESS_OFFLOAD_PLAYBACK_LATENCY;
+
+    return (out->config.period_count * out->config.period_size * 1000) /
+           (out->config.rate);
+}
+
+static int out_set_volume(struct audio_stream_out *stream, float left,
+                          float right)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    int volume[2];
+
+    if (out->usecase == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
+        /* only take left channel into account: the API is for stereo anyway */
+        out->muted = (left == 0.0f);
+        return 0;
+    } else if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        const char *mixer_ctl_name = "Compress Playback Volume";
+        struct audio_device *adev = out->dev;
+        struct mixer_ctl *ctl;
+
+        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;
+        }
+        volume[0] = (int)(left * COMPRESS_PLAYBACK_VOLUME_MAX);
+        volume[1] = (int)(right * COMPRESS_PLAYBACK_VOLUME_MAX);
+        mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
+        return 0;
+    }
+
+    return -ENOSYS;
+}
+
+static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
+                         size_t bytes)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    struct audio_device *adev = out->dev;
+    ssize_t ret = 0;
+
+    pthread_mutex_lock(&out->lock);
+    if (out->standby) {
+        out->standby = false;
+        pthread_mutex_lock(&adev->lock);
+            ret = start_output_stream(out);
+        pthread_mutex_unlock(&adev->lock);
+        /* ToDo: If use case is compress offload should return 0 */
+        if (ret != 0) {
+            out->standby = true;
+            goto exit;
+        }
+    }
+
+    if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        ALOGVV("%s: writing buffer (%d bytes) to compress device", __func__, bytes);
+        if (out->send_new_metadata) {
+            ALOGVV("send new gapless metadata");
+            compress_set_gapless_metadata(out->compr, &out->gapless_mdata);
+            out->send_new_metadata = 0;
+        }
+
+        ret = compress_write(out->compr, buffer, bytes);
+        ALOGVV("%s: writing buffer (%d bytes) to compress device returned %d", __func__, bytes, ret);
+        if (ret >= 0 && ret < (ssize_t)bytes) {
+            send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
+        }
+        if (!out->playback_started) {
+            compress_start(out->compr);
+            out->playback_started = 1;
+            out->offload_state = OFFLOAD_STATE_PLAYING;
+        }
+        pthread_mutex_unlock(&out->lock);
+        return ret;
+    } else {
+        if (out->pcm) {
+            if (out->muted)
+                memset((void *)buffer, 0, bytes);
+            ALOGVV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes);
+            ret = pcm_write(out->pcm, (void *)buffer, bytes);
+            if (ret == 0)
+                out->written += bytes / (out->config.channels * sizeof(short));
+        }
+    }
+
+exit:
+    pthread_mutex_unlock(&out->lock);
+
+    if (ret != 0) {
+        if (out->pcm)
+            ALOGE("%s: error %d - %s", __func__, ret, pcm_get_error(out->pcm));
+        out_standby(&out->stream.common);
+        usleep(bytes * 1000000 / audio_stream_frame_size(&out->stream.common) /
+               out_get_sample_rate(&out->stream.common));
+    }
+    return bytes;
+}
+
+static int out_get_render_position(const struct audio_stream_out *stream,
+                                   uint32_t *dsp_frames)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    *dsp_frames = 0;
+    if ((out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) && (dsp_frames != NULL)) {
+        pthread_mutex_lock(&out->lock);
+        if (out->compr != NULL) {
+            compress_get_tstamp(out->compr, (unsigned long *)dsp_frames,
+                    &out->sample_rate);
+            ALOGVV("%s rendered frames %d sample_rate %d",
+                   __func__, *dsp_frames, out->sample_rate);
+        }
+        pthread_mutex_unlock(&out->lock);
+        return 0;
+    } else
+        return -EINVAL;
+}
+
+static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    return 0;
+}
+
+static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    return 0;
+}
+
+static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
+                                        int64_t *timestamp)
+{
+    return -EINVAL;
+}
+
+static int out_get_presentation_position(const struct audio_stream_out *stream,
+                                   uint64_t *frames, struct timespec *timestamp)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    int ret = -1;
+    unsigned long dsp_frames;
+
+    pthread_mutex_lock(&out->lock);
+
+    if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        if (out->compr != NULL) {
+            compress_get_tstamp(out->compr, &dsp_frames,
+                    &out->sample_rate);
+            ALOGVV("%s rendered frames %ld sample_rate %d",
+                   __func__, dsp_frames, out->sample_rate);
+            *frames = dsp_frames;
+            ret = 0;
+            /* this is the best we can do */
+            clock_gettime(CLOCK_MONOTONIC, timestamp);
+        }
+    } else {
+        if (out->pcm) {
+            size_t avail;
+            if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) {
+                size_t kernel_buffer_size = out->config.period_size * out->config.period_count;
+                int64_t signed_frames = out->written - kernel_buffer_size + avail;
+                // This adjustment accounts for buffering after app processor.
+                // It is based on estimated DSP latency per use case, rather than exact.
+                signed_frames -=
+                    (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL);
+
+                // It would be unusual for this value to be negative, but check just in case ...
+                if (signed_frames >= 0) {
+                    *frames = signed_frames;
+                    ret = 0;
+                }
+            }
+        }
+    }
+
+    pthread_mutex_unlock(&out->lock);
+
+    return ret;
+}
+
+static int out_set_callback(struct audio_stream_out *stream,
+            stream_callback_t callback, void *cookie)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+
+    ALOGV("%s", __func__);
+    pthread_mutex_lock(&out->lock);
+    out->offload_callback = callback;
+    out->offload_cookie = cookie;
+    pthread_mutex_unlock(&out->lock);
+    return 0;
+}
+
+static int out_pause(struct audio_stream_out* stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    int status = -ENOSYS;
+    ALOGV("%s", __func__);
+    if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        pthread_mutex_lock(&out->lock);
+        if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) {
+            status = compress_pause(out->compr);
+            out->offload_state = OFFLOAD_STATE_PAUSED;
+        }
+        pthread_mutex_unlock(&out->lock);
+    }
+    return status;
+}
+
+static int out_resume(struct audio_stream_out* stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    int status = -ENOSYS;
+    ALOGV("%s", __func__);
+    if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        status = 0;
+        pthread_mutex_lock(&out->lock);
+        if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) {
+            status = compress_resume(out->compr);
+            out->offload_state = OFFLOAD_STATE_PLAYING;
+        }
+        pthread_mutex_unlock(&out->lock);
+    }
+    return status;
+}
+
+static int out_drain(struct audio_stream_out* stream, audio_drain_type_t type )
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    int status = -ENOSYS;
+    ALOGV("%s", __func__);
+    if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        pthread_mutex_lock(&out->lock);
+        if (type == AUDIO_DRAIN_EARLY_NOTIFY)
+            status = send_offload_cmd_l(out, OFFLOAD_CMD_PARTIAL_DRAIN);
+        else
+            status = send_offload_cmd_l(out, OFFLOAD_CMD_DRAIN);
+        pthread_mutex_unlock(&out->lock);
+    }
+    return status;
+}
+
+static int out_flush(struct audio_stream_out* stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    ALOGV("%s", __func__);
+    if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        pthread_mutex_lock(&out->lock);
+        stop_compressed_output_l(out);
+        pthread_mutex_unlock(&out->lock);
+        return 0;
+    }
+    return -ENOSYS;
+}
+
+/** audio_stream_in implementation **/
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+
+    return in->config.rate;
+}
+
+static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    return -ENOSYS;
+}
+
+static size_t in_get_buffer_size(const struct audio_stream *stream)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+
+    return in->config.period_size * audio_stream_frame_size(stream);
+}
+
+static uint32_t in_get_channels(const struct audio_stream *stream)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+
+    return in->channel_mask;
+}
+
+static audio_format_t in_get_format(const struct audio_stream *stream)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+
+    return in->format;
+}
+
+static int in_set_format(struct audio_stream *stream, audio_format_t format)
+{
+    return -ENOSYS;
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+    struct audio_device *adev = in->dev;
+    int status = 0;
+    ALOGV("%s: enter", __func__);
+
+    if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
+        /* Ignore standby in case of voip call because the voip input
+         * stream is closed in adev_close_input_stream()
+         */
+        ALOGV("%s: Ignore Standby in VOIP call", __func__);
+        return status;
+    }
+
+    pthread_mutex_lock(&in->lock);
+    if (!in->standby) {
+        in->standby = true;
+        if (in->pcm) {
+            pcm_close(in->pcm);
+            in->pcm = NULL;
+        }
+        pthread_mutex_lock(&adev->lock);
+        status = stop_input_stream(in);
+        pthread_mutex_unlock(&adev->lock);
+    }
+    pthread_mutex_unlock(&in->lock);
+    ALOGV("%s: exit:  status(%d)", __func__, status);
+    return status;
+}
+
+static int in_dump(const struct audio_stream *stream, int fd)
+{
+    return 0;
+}
+
+static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+    struct audio_device *adev = in->dev;
+    struct str_parms *parms;
+    char *str;
+    char value[32];
+    int ret, val = 0;
+
+    ALOGV("%s: enter: kvpairs=%s", __func__, kvpairs);
+    parms = str_parms_create_str(kvpairs);
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value));
+
+    pthread_mutex_lock(&in->lock);
+    pthread_mutex_lock(&adev->lock);
+    if (ret >= 0) {
+        val = atoi(value);
+        /* no audio source uses val == 0 */
+        if ((in->source != val) && (val != 0)) {
+            in->source = val;
+        }
+    }
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+    if (ret >= 0) {
+        val = atoi(value);
+        if ((in->device != val) && (val != 0)) {
+            in->device = val;
+            /* If recording is in progress, change the tx device to new device */
+            if (!in->standby)
+                ret = select_devices(adev, in->usecase);
+        }
+    }
+
+    pthread_mutex_unlock(&adev->lock);
+    pthread_mutex_unlock(&in->lock);
+
+    str_parms_destroy(parms);
+    ALOGV("%s: exit: status(%d)", __func__, ret);
+    return ret;
+}
+
+static char* in_get_parameters(const struct audio_stream *stream,
+                               const char *keys)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+    struct str_parms *query = str_parms_create_str(keys);
+    char *str;
+    char value[256];
+    struct str_parms *reply = str_parms_create();
+    ALOGV("%s: enter: keys - %s", __func__, keys);
+
+    str = str_parms_to_str(reply);
+    str_parms_destroy(query);
+    str_parms_destroy(reply);
+
+    ALOGV("%s: exit: returns - %s", __func__, str);
+    return str;
+}
+
+static int in_set_gain(struct audio_stream_in *stream, float gain)
+{
+    return 0;
+}
+
+static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
+                       size_t bytes)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+    struct audio_device *adev = in->dev;
+    int i, ret = -1;
+
+    pthread_mutex_lock(&in->lock);
+    if (in->standby) {
+        pthread_mutex_lock(&adev->lock);
+            ret = start_input_stream(in);
+        pthread_mutex_unlock(&adev->lock);
+        if (ret != 0) {
+            goto exit;
+        }
+        in->standby = 0;
+    }
+
+    if (in->pcm) {
+         ret = pcm_read(in->pcm, buffer, bytes);
+    }
+
+exit:
+    pthread_mutex_unlock(&in->lock);
+
+    if (ret != 0) {
+        in_standby(&in->stream.common);
+        ALOGV("%s: read failed - sleeping for buffer duration", __func__);
+        usleep(bytes * 1000000 / audio_stream_frame_size(&in->stream.common) /
+               in_get_sample_rate(&in->stream.common));
+    }
+    return bytes;
+}
+
+static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
+{
+    return 0;
+}
+
+static int add_remove_audio_effect(const struct audio_stream *stream,
+                                   effect_handle_t effect,
+                                   bool enable)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+    int status = 0;
+    effect_descriptor_t desc;
+
+    status = (*effect)->get_descriptor(effect, &desc);
+    if (status != 0)
+        return status;
+
+    pthread_mutex_lock(&in->lock);
+    pthread_mutex_lock(&in->dev->lock);
+    if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
+            in->enable_aec != enable &&
+            (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0)) {
+        in->enable_aec = enable;
+        if (!in->standby)
+            select_devices(in->dev, in->usecase);
+    }
+    if (in->enable_ns != enable &&
+            (memcmp(&desc.type, FX_IID_NS, sizeof(effect_uuid_t)) == 0)) {
+        in->enable_ns = enable;
+        if (!in->standby)
+            select_devices(in->dev, in->usecase);
+    }
+    pthread_mutex_unlock(&in->dev->lock);
+    pthread_mutex_unlock(&in->lock);
+
+    return 0;
+}
+
+static int in_add_audio_effect(const struct audio_stream *stream,
+                               effect_handle_t effect)
+{
+    ALOGV("%s: effect %p", __func__, effect);
+    return add_remove_audio_effect(stream, effect, true);
+}
+
+static int in_remove_audio_effect(const struct audio_stream *stream,
+                                  effect_handle_t effect)
+{
+    ALOGV("%s: effect %p", __func__, effect);
+    return add_remove_audio_effect(stream, effect, false);
+}
+
+static int adev_open_output_stream(struct audio_hw_device *dev,
+                                   audio_io_handle_t handle,
+                                   audio_devices_t devices,
+                                   audio_output_flags_t flags,
+                                   struct audio_config *config,
+                                   struct audio_stream_out **stream_out)
+{
+    struct audio_device *adev = (struct audio_device *)dev;
+    struct stream_out *out;
+    int i, ret;
+
+    ALOGV("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)",
+          __func__, config->sample_rate, config->channel_mask, devices, flags);
+    *stream_out = NULL;
+    out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
+
+    if (devices == AUDIO_DEVICE_NONE)
+        devices = AUDIO_DEVICE_OUT_SPEAKER;
+
+    out->flags = flags;
+    out->devices = devices;
+    out->dev = adev;
+    out->format = config->format;
+    out->sample_rate = config->sample_rate;
+    out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+    out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;
+    out->handle = handle;
+
+    /* Init use case and pcm_config */
+    if (out->flags == AUDIO_OUTPUT_FLAG_DIRECT &&
+        out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+        pthread_mutex_lock(&adev->lock);
+        ret = read_hdmi_channel_masks(out);
+        pthread_mutex_unlock(&adev->lock);
+        if (ret != 0)
+            goto error_open;
+
+        if (config->sample_rate == 0)
+            config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+        if (config->channel_mask == 0)
+            config->channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
+
+        out->channel_mask = config->channel_mask;
+        out->sample_rate = config->sample_rate;
+        out->usecase = USECASE_AUDIO_PLAYBACK_MULTI_CH;
+        out->config = pcm_config_hdmi_multi;
+        out->config.rate = config->sample_rate;
+        out->config.channels = popcount(out->channel_mask);
+        out->config.period_size = HDMI_MULTI_PERIOD_BYTES / (out->config.channels * 2);
+    } else if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+        if (config->offload_info.version != AUDIO_INFO_INITIALIZER.version ||
+            config->offload_info.size != AUDIO_INFO_INITIALIZER.size) {
+            ALOGE("%s: Unsupported Offload information", __func__);
+            ret = -EINVAL;
+            goto error_open;
+        }
+        if (!is_supported_format(config->offload_info.format)) {
+            ALOGE("%s: Unsupported audio format", __func__);
+            ret = -EINVAL;
+            goto error_open;
+        }
+
+        out->compr_config.codec = (struct snd_codec *)
+                                    calloc(1, sizeof(struct snd_codec));
+
+        out->usecase = USECASE_AUDIO_PLAYBACK_OFFLOAD;
+        if (config->offload_info.channel_mask)
+            out->channel_mask = config->offload_info.channel_mask;
+        else if (config->channel_mask)
+            out->channel_mask = config->channel_mask;
+        out->format = config->offload_info.format;
+        out->sample_rate = config->offload_info.sample_rate;
+
+        out->stream.set_callback = out_set_callback;
+        out->stream.pause = out_pause;
+        out->stream.resume = out_resume;
+        out->stream.drain = out_drain;
+        out->stream.flush = out_flush;
+
+        out->compr_config.codec->id =
+                get_snd_codec_id(config->offload_info.format);
+        out->compr_config.fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+        out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
+        out->compr_config.codec->sample_rate =
+                    compress_get_alsa_rate(config->offload_info.sample_rate);
+        out->compr_config.codec->bit_rate =
+                    config->offload_info.bit_rate;
+        out->compr_config.codec->ch_in =
+                    popcount(config->channel_mask);
+        out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
+
+        if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
+            out->non_blocking = 1;
+
+        out->send_new_metadata = 1;
+        create_offload_callback_thread(out);
+        ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
+                __func__, config->offload_info.version,
+                config->offload_info.bit_rate);
+    } else if (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
+        out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
+        out->config = pcm_config_low_latency;
+        out->sample_rate = out->config.rate;
+    } else {
+        out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
+        out->config = pcm_config_deep_buffer;
+        out->sample_rate = out->config.rate;
+    }
+
+    if (flags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+        if(adev->primary_output == NULL)
+            adev->primary_output = out;
+        else {
+            ALOGE("%s: Primary output is already opened", __func__);
+            ret = -EEXIST;
+            goto error_open;
+        }
+    }
+
+    /* Check if this usecase is already existing */
+    pthread_mutex_lock(&adev->lock);
+    if (get_usecase_from_list(adev, out->usecase) != NULL) {
+        ALOGE("%s: Usecase (%d) is already present", __func__, out->usecase);
+        pthread_mutex_unlock(&adev->lock);
+        ret = -EEXIST;
+        goto error_open;
+    }
+    pthread_mutex_unlock(&adev->lock);
+
+    out->stream.common.get_sample_rate = out_get_sample_rate;
+    out->stream.common.set_sample_rate = out_set_sample_rate;
+    out->stream.common.get_buffer_size = out_get_buffer_size;
+    out->stream.common.get_channels = out_get_channels;
+    out->stream.common.get_format = out_get_format;
+    out->stream.common.set_format = out_set_format;
+    out->stream.common.standby = out_standby;
+    out->stream.common.dump = out_dump;
+    out->stream.common.set_parameters = out_set_parameters;
+    out->stream.common.get_parameters = out_get_parameters;
+    out->stream.common.add_audio_effect = out_add_audio_effect;
+    out->stream.common.remove_audio_effect = out_remove_audio_effect;
+    out->stream.get_latency = out_get_latency;
+    out->stream.set_volume = out_set_volume;
+    out->stream.write = out_write;
+    out->stream.get_render_position = out_get_render_position;
+    out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
+    out->stream.get_presentation_position = out_get_presentation_position;
+
+    out->standby = 1;
+    /* out->muted = false; by calloc() */
+    /* out->written = 0; by calloc() */
+
+    pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL);
+    pthread_cond_init(&out->cond, (const pthread_condattr_t *) NULL);
+
+    config->format = out->stream.common.get_format(&out->stream.common);
+    config->channel_mask = out->stream.common.get_channels(&out->stream.common);
+    config->sample_rate = out->stream.common.get_sample_rate(&out->stream.common);
+
+    *stream_out = &out->stream;
+    ALOGV("%s: exit", __func__);
+    return 0;
+
+error_open:
+    free(out);
+    *stream_out = NULL;
+    ALOGD("%s: exit: ret %d", __func__, ret);
+    return ret;
+}
+
+static void adev_close_output_stream(struct audio_hw_device *dev,
+                                     struct audio_stream_out *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    struct audio_device *adev = out->dev;
+    int ret = 0;
+
+    ALOGV("%s: enter", __func__);
+    out_standby(&stream->common);
+
+    if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        destroy_offload_callback_thread(out);
+
+        if (out->compr_config.codec != NULL)
+            free(out->compr_config.codec);
+    }
+    pthread_cond_destroy(&out->cond);
+    pthread_mutex_destroy(&out->lock);
+    free(stream);
+    ALOGV("%s: exit", __func__);
+}
+
+static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
+{
+    struct audio_device *adev = (struct audio_device *)dev;
+    struct str_parms *parms;
+    char *str;
+    char value[32];
+    int val;
+    int ret;
+
+    ALOGD("%s: enter: %s", __func__, kvpairs);
+
+    pthread_mutex_lock(&adev->lock);
+    parms = str_parms_create_str(kvpairs);
+
+    platform_set_parameters(adev->platform, parms);
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_NREC, value, sizeof(value));
+    if (ret >= 0) {
+        /* When set to false, HAL should disable EC and NS
+         * But it is currently not supported.
+         */
+        if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
+            adev->bluetooth_nrec = true;
+        else
+            adev->bluetooth_nrec = false;
+    }
+
+    ret = str_parms_get_str(parms, "screen_state", value, sizeof(value));
+    if (ret >= 0) {
+        if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
+            adev->screen_off = false;
+        else
+            adev->screen_off = true;
+    }
+
+    ret = str_parms_get_int(parms, "rotation", &val);
+    if (ret >= 0) {
+        bool reverse_speakers = false;
+        switch(val) {
+        // FIXME: note that the code below assumes that the speakers are in the correct placement
+        //   relative to the user when the device is rotated 90deg from its default rotation. This
+        //   assumption is device-specific, not platform-specific like this code.
+        case 270:
+            reverse_speakers = true;
+            break;
+        case 0:
+        case 90:
+        case 180:
+            break;
+        default:
+            ALOGE("%s: unexpected rotation of %d", __func__, val);
+        }
+        if (adev->speaker_lr_swap != reverse_speakers) {
+            adev->speaker_lr_swap = reverse_speakers;
+            // only update the selected device if there is active pcm playback
+            struct audio_usecase *usecase;
+            struct listnode *node;
+            list_for_each(node, &adev->usecase_list) {
+                usecase = node_to_item(node, struct audio_usecase, list);
+                if (usecase->type == PCM_PLAYBACK) {
+                    select_devices(adev, usecase->id);
+                    break;
+                }
+            }
+        }
+    }
+
+    str_parms_destroy(parms);
+
+    pthread_mutex_unlock(&adev->lock);
+    ALOGV("%s: exit with code(%d)", __func__, ret);
+    return ret;
+}
+
+static char* adev_get_parameters(const struct audio_hw_device *dev,
+                                 const char *keys)
+{
+    struct audio_device *adev = (struct audio_device *)dev;
+    struct str_parms *reply = str_parms_create();
+    struct str_parms *query = str_parms_create_str(keys);
+    char *str;
+
+    pthread_mutex_lock(&adev->lock);
+
+    platform_get_parameters(adev->platform, query, reply);
+    str = str_parms_to_str(reply);
+    str_parms_destroy(query);
+    str_parms_destroy(reply);
+
+    pthread_mutex_unlock(&adev->lock);
+    ALOGV("%s: exit: returns - %s", __func__, str);
+    return str;
+}
+
+static int adev_init_check(const struct audio_hw_device *dev)
+{
+    return 0;
+}
+
+static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
+{
+    int ret = 0;
+    return ret;
+}
+
+static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
+{
+    return -ENOSYS;
+}
+
+static int adev_get_master_volume(struct audio_hw_device *dev,
+                                  float *volume)
+{
+    return -ENOSYS;
+}
+
+static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
+{
+    return -ENOSYS;
+}
+
+static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
+{
+    return -ENOSYS;
+}
+
+static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
+{
+    struct audio_device *adev = (struct audio_device *)dev;
+    pthread_mutex_lock(&adev->lock);
+    if (adev->mode != mode) {
+        ALOGD("%s mode %d\n", __func__, mode);
+        adev->mode = mode;
+    }
+    pthread_mutex_unlock(&adev->lock);
+    return 0;
+}
+
+static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
+{
+    int ret = 0;
+
+    return ret;
+}
+
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
+{
+    return 0;
+}
+
+static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
+                                         const struct audio_config *config)
+{
+    int channel_count = popcount(config->channel_mask);
+
+    return get_input_buffer_size(config->sample_rate, config->format, channel_count);
+}
+
+static int adev_open_input_stream(struct audio_hw_device *dev,
+                                  audio_io_handle_t handle,
+                                  audio_devices_t devices,
+                                  struct audio_config *config,
+                                  struct audio_stream_in **stream_in)
+{
+    struct audio_device *adev = (struct audio_device *)dev;
+    struct stream_in *in;
+    int ret = 0, buffer_size, frame_size;
+    int channel_count = popcount(config->channel_mask);
+
+    ALOGV("%s: enter", __func__);
+    *stream_in = NULL;
+    if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
+        return -EINVAL;
+
+    in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
+
+    in->stream.common.get_sample_rate = in_get_sample_rate;
+    in->stream.common.set_sample_rate = in_set_sample_rate;
+    in->stream.common.get_buffer_size = in_get_buffer_size;
+    in->stream.common.get_channels = in_get_channels;
+    in->stream.common.get_format = in_get_format;
+    in->stream.common.set_format = in_set_format;
+    in->stream.common.standby = in_standby;
+    in->stream.common.dump = in_dump;
+    in->stream.common.set_parameters = in_set_parameters;
+    in->stream.common.get_parameters = in_get_parameters;
+    in->stream.common.add_audio_effect = in_add_audio_effect;
+    in->stream.common.remove_audio_effect = in_remove_audio_effect;
+    in->stream.set_gain = in_set_gain;
+    in->stream.read = in_read;
+    in->stream.get_input_frames_lost = in_get_input_frames_lost;
+
+    in->device = devices;
+    in->source = AUDIO_SOURCE_DEFAULT;
+    in->dev = adev;
+    in->standby = 1;
+    in->channel_mask = config->channel_mask;
+
+    /* Update config params with the requested sample rate and channels */
+    in->usecase = USECASE_AUDIO_RECORD;
+    in->config = pcm_config_audio_capture;
+    in->config.rate = config->sample_rate;
+    in->format = config->format;
+
+    {
+        in->config.channels = channel_count;
+        frame_size = audio_stream_frame_size((struct audio_stream *)in);
+        buffer_size = get_input_buffer_size(config->sample_rate,
+                                            config->format,
+                                            channel_count);
+        in->config.period_size = buffer_size / frame_size;
+    }
+
+    *stream_in = &in->stream;
+    ALOGV("%s: exit", __func__);
+    return ret;
+
+err_open:
+    free(in);
+    *stream_in = NULL;
+    return ret;
+}
+
+static void adev_close_input_stream(struct audio_hw_device *dev,
+                                    struct audio_stream_in *stream)
+{
+    int ret;
+    struct stream_in *in = (struct stream_in *)stream;
+    ALOGV("%s", __func__);
+
+    in_standby(&stream->common);
+
+    free(stream);
+
+    return;
+}
+
+static int adev_dump(const audio_hw_device_t *device, int fd)
+{
+    return 0;
+}
+
+static int adev_close(hw_device_t *device)
+{
+    struct audio_device *adev = (struct audio_device *)device;
+
+    if (!adev)
+        return 0;
+
+    pthread_mutex_lock(&adev_init_lock);
+
+    if ((--audio_device_ref_count) == 0) {
+        audio_route_free(adev->audio_route);
+        free(adev->snd_dev_ref_cnt);
+        platform_deinit(adev->platform);
+        free(device);
+        adev = NULL;
+    }
+    pthread_mutex_unlock(&adev_init_lock);
+    return 0;
+}
+
+static int adev_open(const hw_module_t *module, const char *name,
+                     hw_device_t **device)
+{
+    int i, ret;
+
+    ALOGD("%s: enter", __func__);
+    if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL;
+
+    pthread_mutex_lock(&adev_init_lock);
+    if (audio_device_ref_count != 0){
+            *device = &adev->device.common;
+            audio_device_ref_count++;
+            ALOGD("%s: returning existing instance of adev", __func__);
+            ALOGD("%s: exit", __func__);
+            pthread_mutex_unlock(&adev_init_lock);
+            return 0;
+    }
+
+    adev = calloc(1, sizeof(struct audio_device));
+
+    adev->device.common.tag = HARDWARE_DEVICE_TAG;
+    adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
+    adev->device.common.module = (struct hw_module_t *)module;
+    adev->device.common.close = adev_close;
+
+    adev->device.init_check = adev_init_check;
+    adev->device.set_voice_volume = adev_set_voice_volume;
+    adev->device.set_master_volume = adev_set_master_volume;
+    adev->device.get_master_volume = adev_get_master_volume;
+    adev->device.set_master_mute = adev_set_master_mute;
+    adev->device.get_master_mute = adev_get_master_mute;
+    adev->device.set_mode = adev_set_mode;
+    adev->device.set_mic_mute = adev_set_mic_mute;
+    adev->device.get_mic_mute = adev_get_mic_mute;
+    adev->device.set_parameters = adev_set_parameters;
+    adev->device.get_parameters = adev_get_parameters;
+    adev->device.get_input_buffer_size = adev_get_input_buffer_size;
+    adev->device.open_output_stream = adev_open_output_stream;
+    adev->device.close_output_stream = adev_close_output_stream;
+    adev->device.open_input_stream = adev_open_input_stream;
+    adev->device.close_input_stream = adev_close_input_stream;
+    adev->device.dump = adev_dump;
+
+    /* Set the default route before the PCM stream is opened */
+    adev->mode = AUDIO_MODE_NORMAL;
+    adev->active_input = NULL;
+    adev->primary_output = NULL;
+    adev->out_device = AUDIO_DEVICE_NONE;
+    adev->bluetooth_nrec = true;
+    adev->acdb_settings = TTY_MODE_OFF;
+    /* adev->cur_hdmi_channels = 0;  by calloc() */
+    adev->snd_dev_ref_cnt = calloc(SND_DEVICE_MAX, sizeof(int));
+    list_init(&adev->usecase_list);
+
+    /* Loads platform specific libraries dynamically */
+    adev->platform = platform_init(adev);
+    if (!adev->platform) {
+        free(adev->snd_dev_ref_cnt);
+        free(adev);
+        ALOGE("%s: Failed to init platform data, aborting.", __func__);
+        *device = NULL;
+        return -EINVAL;
+    }
+
+    if (access(VISUALIZER_LIBRARY_PATH, R_OK) == 0) {
+        adev->visualizer_lib = dlopen(VISUALIZER_LIBRARY_PATH, RTLD_NOW);
+        if (adev->visualizer_lib == NULL) {
+            ALOGE("%s: DLOPEN failed for %s", __func__, VISUALIZER_LIBRARY_PATH);
+        } else {
+            ALOGV("%s: DLOPEN successful for %s", __func__, VISUALIZER_LIBRARY_PATH);
+            adev->visualizer_start_output =
+                        (int (*)(audio_io_handle_t))dlsym(adev->visualizer_lib,
+                                                        "visualizer_hal_start_output");
+            adev->visualizer_stop_output =
+                        (int (*)(audio_io_handle_t))dlsym(adev->visualizer_lib,
+                                                        "visualizer_hal_stop_output");
+        }
+    }
+    *device = &adev->device.common;
+
+    audio_device_ref_count++;
+    pthread_mutex_unlock(&adev_init_lock);
+
+    ALOGV("%s: exit", __func__);
+    return 0;
+}
+
+static struct hw_module_methods_t hal_module_methods = {
+    .open = adev_open,
+};
+
+struct audio_module HAL_MODULE_INFO_SYM = {
+    .common = {
+        .tag = HARDWARE_MODULE_TAG,
+        .module_api_version = AUDIO_MODULE_API_VERSION_0_1,
+        .hal_api_version = HARDWARE_HAL_API_VERSION,
+        .id = AUDIO_HARDWARE_MODULE_ID,
+        .name = "MPQ Audio HAL",
+        .author = "The Linux Foundation",
+        .methods = &hal_module_methods,
+    },
+};
diff --git a/hal_mpq/audio_hw.h b/hal_mpq/audio_hw.h
new file mode 100644
index 0000000..262fda8
--- /dev/null
+++ b/hal_mpq/audio_hw.h
@@ -0,0 +1,244 @@
+/*
+ * Copyright (c) 2013, 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_HW_H
+#define QCOM_AUDIO_HW_H
+
+#include <cutils/list.h>
+#include <hardware/audio.h>
+#include <tinyalsa/asoundlib.h>
+#include <tinycompress/tinycompress.h>
+
+#include <audio_route/audio_route.h>
+
+#define VISUALIZER_LIBRARY_PATH "/system/lib/soundfx/libqcomvisualizer.so"
+
+/* Flags used to initialize acdb_settings variable that goes to ACDB library */
+#define DMIC_FLAG       0x00000002
+#define QMIC_FLAG       0x00000004
+#define TTY_MODE_OFF    0x00000010
+#define TTY_MODE_FULL   0x00000020
+#define TTY_MODE_VCO    0x00000040
+#define TTY_MODE_HCO    0x00000080
+#define TTY_MODE_CLEAR  0xFFFFFF0F
+
+#define ACDB_DEV_TYPE_OUT 1
+#define ACDB_DEV_TYPE_IN 2
+
+#define MAX_SUPPORTED_CHANNEL_MASKS 2
+#define DEFAULT_HDMI_OUT_CHANNELS   2
+
+typedef int snd_device_t;
+
+/* These are the supported use cases by the hardware.
+ * Each usecase is mapped to a specific PCM device.
+ * Refer to pcm_device_table[].
+ */
+typedef enum {
+    USECASE_INVALID = -1,
+    /* Playback usecases */
+    USECASE_AUDIO_PLAYBACK_DEEP_BUFFER = 0,
+    USECASE_AUDIO_PLAYBACK_LOW_LATENCY,
+    USECASE_AUDIO_PLAYBACK_MULTI_CH,
+    USECASE_AUDIO_PLAYBACK_OFFLOAD,
+
+    /* FM usecase */
+    USECASE_AUDIO_PLAYBACK_FM,
+
+    /* Capture usecases */
+    USECASE_AUDIO_RECORD,
+    USECASE_AUDIO_RECORD_COMPRESS,
+    USECASE_AUDIO_RECORD_LOW_LATENCY,
+    USECASE_AUDIO_RECORD_FM_VIRTUAL,
+
+    /* Voice usecase */
+    USECASE_VOICE_CALL,
+
+    /* Voice extension usecases */
+    USECASE_VOICE2_CALL,
+    USECASE_VOLTE_CALL,
+    USECASE_QCHAT_CALL,
+    USECASE_COMPRESS_VOIP_CALL,
+
+    USECASE_INCALL_REC_UPLINK,
+    USECASE_INCALL_REC_DOWNLINK,
+    USECASE_INCALL_REC_UPLINK_AND_DOWNLINK,
+
+    USECASE_INCALL_MUSIC_UPLINK,
+    USECASE_INCALL_MUSIC_UPLINK2,
+
+    USECASE_AUDIO_SPKR_CALIB_RX,
+    USECASE_AUDIO_SPKR_CALIB_TX,
+    AUDIO_USECASE_MAX
+} audio_usecase_t;
+
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+/*
+ * 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
+ */
+
+enum {
+    OFFLOAD_CMD_EXIT,               /* exit compress offload thread loop*/
+    OFFLOAD_CMD_DRAIN,              /* send a full drain request to DSP */
+    OFFLOAD_CMD_PARTIAL_DRAIN,      /* send a partial drain request to DSP */
+    OFFLOAD_CMD_WAIT_FOR_BUFFER,    /* wait for buffer released by DSP */
+};
+
+enum {
+    OFFLOAD_STATE_IDLE,
+    OFFLOAD_STATE_PLAYING,
+    OFFLOAD_STATE_PAUSED,
+};
+
+struct offload_cmd {
+    struct listnode node;
+    int cmd;
+    int data[];
+};
+
+struct stream_out {
+    struct audio_stream_out stream;
+    pthread_mutex_t lock; /* see note below on mutex acquisition order */
+    pthread_cond_t  cond;
+    struct pcm_config config;
+    struct compr_config compr_config;
+    struct pcm *pcm;
+    struct compress *compr;
+    int standby;
+    int pcm_device_id;
+    unsigned int sample_rate;
+    audio_channel_mask_t channel_mask;
+    audio_format_t format;
+    audio_devices_t devices;
+    audio_output_flags_t flags;
+    audio_usecase_t usecase;
+    /* Array of supported channel mask configurations. +1 so that the last entry is always 0 */
+    audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
+    bool muted;
+    uint64_t written; /* total frames written, not cleared when entering standby */
+    audio_io_handle_t handle;
+
+    int non_blocking;
+    int playback_started;
+    int offload_state;
+    pthread_cond_t offload_cond;
+    pthread_t offload_thread;
+    struct listnode offload_cmd_list;
+    bool offload_thread_blocked;
+
+    stream_callback_t offload_callback;
+    void *offload_cookie;
+    struct compr_gapless_mdata gapless_mdata;
+    int send_new_metadata;
+
+    struct audio_device *dev;
+};
+
+struct stream_in {
+    struct audio_stream_in stream;
+    pthread_mutex_t lock; /* see note below on mutex acquisition order */
+    struct pcm_config config;
+    struct pcm *pcm;
+    int standby;
+    int source;
+    int pcm_device_id;
+    int device;
+    audio_channel_mask_t channel_mask;
+    audio_usecase_t usecase;
+    bool enable_aec;
+    bool enable_ns;
+    audio_format_t format;
+
+    struct audio_device *dev;
+};
+
+typedef enum {
+    PCM_PLAYBACK,
+    PCM_CAPTURE,
+    VOICE_CALL,
+    VOIP_CALL
+} usecase_type_t;
+
+union stream_ptr {
+    struct stream_in *in;
+    struct stream_out *out;
+};
+
+struct audio_usecase {
+    struct listnode list;
+    audio_usecase_t id;
+    usecase_type_t  type;
+    audio_devices_t devices;
+    snd_device_t out_snd_device;
+    snd_device_t in_snd_device;
+    union stream_ptr stream;
+};
+
+struct audio_device {
+    struct audio_hw_device device;
+    pthread_mutex_t lock; /* see note below on mutex acquisition order */
+    struct mixer *mixer;
+    audio_mode_t mode;
+    audio_devices_t out_device;
+    struct stream_in *active_input;
+    struct stream_out *primary_output;
+    bool bluetooth_nrec;
+    bool screen_off;
+    int *snd_dev_ref_cnt;
+    struct listnode usecase_list;
+    struct audio_route *audio_route;
+    int acdb_settings;
+    bool speaker_lr_swap;
+    unsigned int cur_hdmi_channels;
+
+    void *platform;
+
+    void *visualizer_lib;
+    int (*visualizer_start_output)(audio_io_handle_t);
+    int (*visualizer_stop_output)(audio_io_handle_t);
+};
+
+int select_devices(struct audio_device *adev,
+                          audio_usecase_t uc_id);
+int disable_audio_route(struct audio_device *adev,
+                               struct audio_usecase *usecase,
+                               bool update_mixer);
+int disable_snd_device(struct audio_device *adev,
+                              snd_device_t snd_device,
+                              bool update_mixer);
+int enable_snd_device(struct audio_device *adev,
+                             snd_device_t snd_device,
+                             bool update_mixer);
+int enable_audio_route(struct audio_device *adev,
+                              struct audio_usecase *usecase,
+                              bool update_mixer);
+struct audio_usecase *get_usecase_from_list(struct audio_device *adev,
+                                                   audio_usecase_t uc_id);
+/*
+ * NOTE: when multiple mutexes have to be acquired, always take the
+ * stream_in or stream_out mutex first, followed by the audio_device mutex.
+ */
+
+#endif // QCOM_AUDIO_HW_H
diff --git a/hal_mpq/mpq8092/hw_info.c b/hal_mpq/mpq8092/hw_info.c
new file mode 100644
index 0000000..97b7804
--- /dev/null
+++ b/hal_mpq/mpq8092/hw_info.c
@@ -0,0 +1,326 @@
+/*
+ * Copyright (c) 2013, 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]))
+
+#define LITERAL_TO_STRING(x) #x
+#define CHECK(condition) LOG_ALWAYS_FATAL_IF(!(condition), "%s",\
+            __FILE__ ":" LITERAL_TO_STRING(__LINE__)\
+            " ASSERT_FATAL(" #condition ") failed.")
+
+static const snd_device_t tabla_cdp_variant_devices[] = {
+    SND_DEVICE_OUT_SPEAKER,
+    SND_DEVICE_OUT_HEADPHONES,
+    SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+};
+
+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_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_8092(struct hardware_info *hw_info, const char *snd_card_name)
+{
+    if (!strcmp(snd_card_name, "mpq8092-tabla-cdp-snd-card")) {
+        strlcpy(hw_info->type, "cdp", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "mpq8092", 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 8084 device", __func__);
+    }
+}
+
+static void  update_hardware_info_8084(struct hardware_info *hw_info, const char *snd_card_name)
+{
+    if (!strcmp(snd_card_name, "apq8084-taiko-mtp-snd-card")) {
+        strlcpy(hw_info->type, "mtp", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "apq8084", sizeof(hw_info->name));
+        hw_info->snd_devices = NULL;
+        hw_info->num_snd_devices = 0;
+        strlcpy(hw_info->dev_extn, "", sizeof(hw_info->dev_extn));
+    } else if (!strcmp(snd_card_name, "apq8084-taiko-cdp-snd-card")) {
+        strlcpy(hw_info->type, " cdp", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "apq8084", sizeof(hw_info->name));
+        hw_info->snd_devices = (snd_device_t *)taiko_CDP_variant_devices;
+        hw_info->num_snd_devices = ARRAY_SIZE(taiko_CDP_variant_devices);
+        strlcpy(hw_info->dev_extn, "", sizeof(hw_info->dev_extn));
+    } else if (!strcmp(snd_card_name, "apq8084-taiko-liquid-snd-card")) {
+        strlcpy(hw_info->type , " liquid", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "apq8084", sizeof(hw_info->type));
+        hw_info->snd_devices = (snd_device_t *)taiko_liquid_variant_devices;
+        hw_info->num_snd_devices = ARRAY_SIZE(taiko_liquid_variant_devices);
+        strlcpy(hw_info->dev_extn, "", sizeof(hw_info->dev_extn));
+    } else {
+        ALOGW("%s: Not an 8084 device", __func__);
+    }
+}
+
+static void  update_hardware_info_8974(struct hardware_info *hw_info, const char *snd_card_name)
+{
+    if (!strcmp(snd_card_name, "msm8974-taiko-mtp-snd-card")) {
+        strlcpy(hw_info->type, " mtp", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8974", 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, "msm8974-taiko-cdp-snd-card")) {
+        strlcpy(hw_info->type, " cdp", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8974", sizeof(hw_info->name));
+        hw_info->snd_devices = (snd_device_t *)taiko_CDP_variant_devices;
+        hw_info->num_snd_devices = ARRAY_SIZE(taiko_CDP_variant_devices);
+        strlcpy(hw_info->dev_extn, "-cdp", sizeof(hw_info->dev_extn));
+    } else if (!strcmp(snd_card_name, "msm8974-taiko-fluid-snd-card")) {
+        strlcpy(hw_info->type, " fluid", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8974", sizeof(hw_info->name));
+        hw_info->snd_devices = (snd_device_t *) taiko_fluid_variant_devices;
+        hw_info->num_snd_devices = ARRAY_SIZE(taiko_fluid_variant_devices);
+        strlcpy(hw_info->dev_extn, "-fluid", sizeof(hw_info->dev_extn));
+    } else if (!strcmp(snd_card_name, "msm8974-taiko-liquid-snd-card")) {
+        strlcpy(hw_info->type, " liquid", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8974", sizeof(hw_info->name));
+        hw_info->snd_devices = (snd_device_t *)taiko_liquid_variant_devices;
+        hw_info->num_snd_devices = ARRAY_SIZE(taiko_liquid_variant_devices);
+        strlcpy(hw_info->dev_extn, "-liquid", sizeof(hw_info->dev_extn));
+    } else if (!strcmp(snd_card_name, "apq8074-taiko-db-snd-card")) {
+        strlcpy(hw_info->type, " dragon-board", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8974", sizeof(hw_info->name));
+        hw_info->snd_devices = (snd_device_t *)taiko_DB_variant_devices;
+        hw_info->num_snd_devices = ARRAY_SIZE(taiko_DB_variant_devices);
+        strlcpy(hw_info->dev_extn, "-DB", sizeof(hw_info->dev_extn));
+    } else {
+        ALOGW("%s: Not an 8974 device", __func__);
+    }
+}
+
+static void update_hardware_info_8610(struct hardware_info *hw_info, const char *snd_card_name)
+{
+    if (!strcmp(snd_card_name, "msm8x10-snd-card")) {
+        strlcpy(hw_info->type, "", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8x10", 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, "msm8x10-skuab-snd-card")) {
+        strlcpy(hw_info->type, "skuab", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8x10", 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, "msm8x10-skuaa-snd-card")) {
+        strlcpy(hw_info->type, " skuaa", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8x10", 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  8x10 device", __func__);
+    }
+}
+
+static void update_hardware_info_8226(struct hardware_info *hw_info, const char *snd_card_name)
+{
+    if (!strcmp(snd_card_name, "msm8226-tapan-snd-card")) {
+        strlcpy(hw_info->type, "", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8226", 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, "msm8226-tapan9302-snd-card")) {
+        strlcpy(hw_info->type, "tapan_lite", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8226", sizeof(hw_info->name));
+        hw_info->snd_devices = (snd_device_t *)tapan_lite_variant_devices;
+        hw_info->num_snd_devices = ARRAY_SIZE(tapan_lite_variant_devices);
+        strlcpy(hw_info->dev_extn, "-lite", sizeof(hw_info->dev_extn));
+    } else if (!strcmp(snd_card_name, "msm8226-tapan-skuf-snd-card")) {
+        strlcpy(hw_info->type, " skuf", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8226", sizeof(hw_info->name));
+        hw_info->snd_devices = (snd_device_t *) tapan_skuf_variant_devices;
+        hw_info->num_snd_devices = ARRAY_SIZE(tapan_skuf_variant_devices);
+        strlcpy(hw_info->dev_extn, "-skuf", sizeof(hw_info->dev_extn));
+    } else if (!strcmp(snd_card_name, "msm8226-tapan9302-skuf-snd-card")) {
+        strlcpy(hw_info->type, " tapan9302-skuf", sizeof(hw_info->type));
+        strlcpy(hw_info->name, "msm8226", sizeof(hw_info->name));
+        hw_info->snd_devices = (snd_device_t *)tapan_lite_skuf_variant_devices;
+        hw_info->num_snd_devices = ARRAY_SIZE(tapan_lite_skuf_variant_devices);
+        strlcpy(hw_info->dev_extn, "-skuf-lite", sizeof(hw_info->dev_extn));
+    } else {
+        ALOGW("%s: Not an  8x26 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, "mpq8092") ||
+              strstr(snd_card_name, "mpq8092")) {
+        ALOGV("8092 - variant soundcard");
+        update_hardware_info_8092(hw_info, snd_card_name);
+    } else if(strstr(snd_card_name, "msm8974") ||
+              strstr(snd_card_name, "apq8074")) {
+        ALOGV("8974 - variant soundcard");
+        update_hardware_info_8974(hw_info, snd_card_name);
+    } else if(strstr(snd_card_name, "msm8226")) {
+        ALOGV("8x26 - variant soundcard");
+        update_hardware_info_8226(hw_info, snd_card_name);
+    } else if(strstr(snd_card_name, "msm8x10")) {
+        ALOGV("8x10 - variant soundcard");
+        update_hardware_info_8610(hw_info, snd_card_name);
+    } else if(strstr(snd_card_name, "apq8084")) {
+        ALOGV("8084 - variant soundcard");
+        update_hardware_info_8084(hw_info, snd_card_name);
+    } else {
+        ALOGE("%s: Unupported target %s:",__func__, snd_card_name);
+        CHECK(0);
+        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_mpq/mpq8092/platform.c b/hal_mpq/mpq8092/platform.c
new file mode 100644
index 0000000..d7d67d5
--- /dev/null
+++ b/hal_mpq/mpq8092/platform.c
@@ -0,0 +1,1272 @@
+/*
+ * Copyright (c) 2013, 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 "msm8974_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"
+
+#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 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"
+
+struct audio_block_header
+{
+    int reserved;
+    int length;
+};
+
+/* Audio calibration related functions */
+typedef void (*acdb_deallocate_t)();
+typedef int  (*acdb_init_t)();
+typedef void (*acdb_send_audio_cal_t)(int, int);
+typedef void (*acdb_send_voice_cal_t)(int, 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;
+    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;
+
+    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_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_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_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 const 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] = 0,
+    [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_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,
+};
+
+#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 = (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;
+    int retry_num = 0;
+    const char *snd_card_name;
+
+    adev->mixer = mixer_open(MIXER_CARD);
+
+    while (!adev->mixer && retry_num < RETRY_NUMBER) {
+        usleep(RETRY_US);
+        adev->mixer = mixer_open(MIXER_CARD);
+        retry_num++;
+    }
+
+    if (!adev->mixer) {
+        ALOGE("Unable to open the mixer, aborting.");
+        return NULL;
+    }
+
+    adev->audio_route = audio_route_init(MIXER_CARD, MIXER_XML_PATH);
+
+    if (!adev->audio_route) {
+        ALOGE("%s: Failed to init audio route controls, aborting.", __func__);
+        return NULL;
+    }
+
+    my_data = calloc(1, sizeof(struct platform_data));
+
+    snd_card_name = mixer_get_name(adev->mixer);
+    my_data->hw_info = hw_info_init(snd_card_name);
+    if (!my_data->hw_info) {
+        ALOGE("%s: Failed to init hardware info", __func__);
+    }
+
+    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->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");
+        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)
+            ALOGW("%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");
+        my_data->acdb_init = (acdb_init_t)dlsym(my_data->acdb_handle,
+                                                    "acdb_loader_init_ACDB");
+        if (my_data->acdb_init == NULL)
+            ALOGE("%s: dlsym error %s for acdb_loader_init_ACDB", __func__, dlerror());
+        else
+            my_data->acdb_init();
+    }
+
+    /* If platform is apq8084 and baseband is MDM, load CSD Client specific
+     * symbols. Voice call is handled by MDM and apps processor talks to
+     * MDM through CSD Client
+     */
+    property_get("ro.board.platform", platform, "");
+    property_get("ro.baseband", baseband, "");
+    if (!strncmp("apq8084", platform, sizeof("apq8084")) &&
+        !strncmp("mdm", baseband, sizeof("mdm"))) {
+         my_data->csd = open_csd_client();
+    }
+
+    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);
+}
+
+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_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_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)) {
+            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 (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
+        devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+        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 {
+        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 (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, "SLIM_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, "SLIM_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 (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[32];
+    int val;
+    int ret = 0;
+
+    ALOGV("%s: enter: %s", __func__, str_parms_to_str(parms));
+
+    ret = str_parms_get_int(parms, AUDIO_PARAMETER_KEY_BTSCO, &val);
+    if (ret >= 0) {
+        str_parms_del(parms, AUDIO_PARAMETER_KEY_BTSCO);
+        my_data->btsco_sample_rate = val;
+    }
+
+    ret = str_parms_get_int(parms, AUDIO_PARAMETER_KEY_SLOWTALK, &val);
+    if (ret >= 0) {
+        str_parms_del(parms, AUDIO_PARAMETER_KEY_SLOWTALK);
+        ret = platform_set_slowtalk(my_data, val);
+        if (ret)
+            ALOGE("%s: Failed to set slow talk err: %d", __func__, ret);
+    }
+
+    ALOGV("%s: exit with code(%d)", __func__, ret);
+    return ret;
+}
+
+int platform_set_incall_recoding_session_id(void *platform,
+                                            uint32_t session_id)
+{
+    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;
+            }
+        }
+    }
+
+    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_int(reply, AUDIO_PARAMETER_KEY_SLOWTALK,
+                          my_data->slowtalk);
+    }
+
+    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;
+}
diff --git a/hal_mpq/mpq8092/platform.h b/hal_mpq/mpq8092/platform.h
new file mode 100644
index 0000000..3cd56cf
--- /dev/null
+++ b/hal_mpq/mpq8092/platform.h
@@ -0,0 +1,242 @@
+/*
+ * Copyright (c) 2013, 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 MIXER_CARD 0
+#define SOUND_CARD 0
+
+#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 8
+#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 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
+
+#ifdef PLATFORM_MSM8610
+#define LOWLATENCY_PCM_DEVICE 12
+#else
+#define LOWLATENCY_PCM_DEVICE 15
+#endif
+#define COMPRESS_CAPTURE_DEVICE 19
+
+#ifdef PLATFORM_MSM8x26
+#define VOICE_CALL_PCM_DEVICE 2
+#define VOICE2_CALL_PCM_DEVICE 14
+#define VOLTE_CALL_PCM_DEVICE 17
+#define QCHAT_CALL_PCM_DEVICE 18
+#elif PLATFORM_APQ8084
+#define VOICE_CALL_PCM_DEVICE 20
+#define VOICE2_CALL_PCM_DEVICE 13
+#define VOLTE_CALL_PCM_DEVICE 21
+#define QCHAT_CALL_PCM_DEVICE 06
+#else
+#define VOICE_CALL_PCM_DEVICE 2
+#define VOICE2_CALL_PCM_DEVICE 13
+#define VOLTE_CALL_PCM_DEVICE 14
+#define QCHAT_CALL_PCM_DEVICE 20
+#endif
+
+#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_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, int);
+/* CSD Client structure */
+struct csd_data {
+    void *csd_client;
+    init_t init;
+    deinit_t deinit;
+    disable_device_t disable_device;
+    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;
+};
+
+void *hw_info_init(const char *snd_card_name);
+void hw_info_deinit(void *hw_info);
+void hw_info_append_hw_type(void *hw_info, snd_device_t snd_device,
+                             char *device_name);
+
+#endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal_mpq/platform_api.h b/hal_mpq/platform_api.h
new file mode 100644
index 0000000..44ad790
--- /dev/null
+++ b/hal_mpq/platform_api.h
@@ -0,0 +1,55 @@
+/*
+ * Copyright (c) 2013, 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_API_H
+#define QCOM_AUDIO_PLATFORM_API_H
+
+void *platform_init(struct audio_device *adev);
+void platform_deinit(void *platform);
+const char *platform_get_snd_device_name(snd_device_t snd_device);
+int platform_get_snd_device_name_extn(void *platform, snd_device_t snd_device,
+                                      char *device_name);
+void platform_add_backend_name(char *mixer_path, snd_device_t snd_device);
+int platform_get_pcm_device_id(audio_usecase_t usecase, int device_type);
+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_device_post(void *platform,
+                                           snd_device_t out_snd_device,
+                                           snd_device_t in_snd_device);
+int platform_switch_voice_call_usecase_route_post(void *platform,
+                                                  snd_device_t out_snd_device,
+                                                  snd_device_t in_snd_device);
+int platform_start_voice_call(void *platform, uint32_t vsid);
+int platform_stop_voice_call(void *platform, uint32_t vsid);
+int platform_set_voice_volume(void *platform, int volume);
+int platform_set_mic_mute(void *platform, bool state);
+snd_device_t platform_get_output_snd_device(void *platform, audio_devices_t devices);
+snd_device_t platform_get_input_snd_device(void *platform, audio_devices_t out_device);
+int platform_set_hdmi_channels(void *platform, int channel_count);
+int platform_edid_get_max_channels(void *platform);
+void platform_get_parameters(void *platform, struct str_parms *query,
+                             struct str_parms *reply);
+int platform_set_parameters(void *platform, struct str_parms *parms);
+int platform_set_incall_recoding_session_id(void *platform, uint32_t session_id);
+
+/* returns the latency for a usecase in Us */
+int64_t platform_render_latency(audio_usecase_t usecase);
+int platform_update_usecase_from_source(int source, audio_usecase_t usecase);
+
+#endif // QCOM_AUDIO_PLATFORM_API_H