diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index 4c1416d..dd6063f 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -309,6 +309,7 @@
                                   struct listnode *streams_output_cfg_list);
 void audio_extn_utils_update_stream_app_type_cfg(void *platform,
                                   struct listnode *streams_output_cfg_list,
+                                  audio_devices_t devices,
                                   audio_output_flags_t flags,
                                   audio_format_t format,
                                   uint32_t sample_rate,
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index e148ead..5f4c6ba 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -139,12 +139,12 @@
 
     pthread_mutex_lock(&st_dev->lock);
     st_ses_info = get_sound_trigger_info(in->capture_handle);
+    pthread_mutex_unlock(&st_dev->lock);
     if (st_ses_info) {
         event.u.ses_info = st_ses_info->st_ses;
         ALOGV("%s: AUDIO_EVENT_STOP_LAB pcm %p", __func__, st_ses_info->st_ses.pcm);
         st_dev->st_callback(AUDIO_EVENT_STOP_LAB, &event);
     }
-    pthread_mutex_unlock(&st_dev->lock);
 }
 void audio_extn_sound_trigger_check_and_get_session(struct stream_in *in)
 {
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 5287ae8..6818161 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -422,6 +422,7 @@
 
 void audio_extn_utils_update_stream_app_type_cfg(void *platform,
                                   struct listnode *streams_output_cfg_list,
+                                  audio_devices_t devices,
                                   audio_output_flags_t flags,
                                   audio_format_t format,
                                   uint32_t sample_rate,
@@ -433,7 +434,14 @@
     struct stream_format *sf_info;
     struct stream_sample_rate *ss_info;
 
-    ALOGV("%s: flags: %x, format: %x", __func__, flags, format);
+    if ((24 == bit_width) &&
+        (devices & AUDIO_DEVICE_OUT_SPEAKER)) {
+        sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+        ALOGI("%s Allowing 24-bit playback on speaker ONLY at default sampling rate", __func__);
+    }
+
+    ALOGV("%s: flags: %x, format: %x sample_rate %d",
+           __func__, flags, format, sample_rate);
     list_for_each(node_i, streams_output_cfg_list) {
         so_info = node_to_item(node_i, struct streams_output_cfg, list);
         if (so_info->flags == flags) {
@@ -516,7 +524,7 @@
     }
 
     if ((24 == usecase->stream.out->bit_width) &&
-            (AUDIO_DEVICE_OUT_SPEAKER == snd_device)) {
+        (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
         sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
     } else {
         sample_rate = out->app_type_cfg.sample_rate;
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 627900b..390fb46 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -60,9 +60,14 @@
 #define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 96
 #define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
 
+#define PROXY_OPEN_RETRY_COUNT           100
+#define PROXY_OPEN_WAIT_TIME             20
 
 #define USECASE_AUDIO_PLAYBACK_PRIMARY USECASE_AUDIO_PLAYBACK_DEEP_BUFFER
 
+static unsigned int configured_low_latency_capture_period_size =
+        LOW_LATENCY_CAPTURE_PERIOD_SIZE;
+
 struct pcm_config pcm_config_deep_buffer = {
     .channels = 2,
     .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
@@ -102,6 +107,37 @@
     .format = PCM_FORMAT_S16_LE,
 };
 
+#define AFE_PROXY_CHANNEL_COUNT 2
+#define AFE_PROXY_SAMPLING_RATE 48000
+
+#define AFE_PROXY_PLAYBACK_PERIOD_SIZE  768
+#define AFE_PROXY_PLAYBACK_PERIOD_COUNT 4
+
+struct pcm_config pcm_config_afe_proxy_playback = {
+    .channels = AFE_PROXY_CHANNEL_COUNT,
+    .rate = AFE_PROXY_SAMPLING_RATE,
+    .period_size = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
+    .period_count = AFE_PROXY_PLAYBACK_PERIOD_COUNT,
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
+    .stop_threshold = INT_MAX,
+    .avail_min = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
+};
+
+#define AFE_PROXY_RECORD_PERIOD_SIZE  768
+#define AFE_PROXY_RECORD_PERIOD_COUNT 4
+
+struct pcm_config pcm_config_afe_proxy_record = {
+    .channels = AFE_PROXY_CHANNEL_COUNT,
+    .rate = AFE_PROXY_SAMPLING_RATE,
+    .period_size = AFE_PROXY_RECORD_PERIOD_SIZE,
+    .period_count = AFE_PROXY_RECORD_PERIOD_COUNT,
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = AFE_PROXY_RECORD_PERIOD_SIZE,
+    .stop_threshold = INT_MAX,
+    .avail_min = AFE_PROXY_RECORD_PERIOD_SIZE,
+};
+
 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",
@@ -142,6 +178,9 @@
     [USECASE_INCALL_MUSIC_UPLINK2] = "incall_music_uplink2",
     [USECASE_AUDIO_SPKR_CALIB_RX] = "spkr-rx-calib",
     [USECASE_AUDIO_SPKR_CALIB_TX] = "spkr-vi-record",
+
+    [USECASE_AUDIO_PLAYBACK_AFE_PROXY] = "afe-proxy-playback",
+    [USECASE_AUDIO_RECORD_AFE_PROXY] = "afe-proxy-record",
 };
 
 static const audio_usecase_t offload_usecases[] = {
@@ -288,6 +327,19 @@
     return 0;
 }
 
+int pcm_ioctl(struct pcm *pcm, int request, ...)
+{
+    va_list ap;
+    void * arg;
+    int pcm_fd = *(int*)pcm;
+
+    va_start(ap, request);
+    arg = va_arg(ap, void *);
+    va_end(ap);
+
+    return ioctl(pcm_fd, request, arg);
+}
+
 int enable_audio_route(struct audio_device *adev,
                        struct audio_usecase *usecase)
 {
@@ -704,7 +756,7 @@
          * usecase. This is to avoid switching devices for voice call when
          * check_usecases_codec_backend() is called below.
          */
-        if (adev->voice.in_call && adev->mode == AUDIO_MODE_IN_CALL) {
+        if (voice_is_in_call(adev) && adev->mode == AUDIO_MODE_IN_CALL) {
             vc_usecase = get_usecase_from_list(adev,
                                                get_voice_usecase_id_from_list(adev));
             if ((vc_usecase) && ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
@@ -744,14 +796,14 @@
             usecase->devices = usecase->stream.in->device;
             out_snd_device = SND_DEVICE_NONE;
             if (in_snd_device == SND_DEVICE_NONE) {
+                audio_devices_t out_device = AUDIO_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);
+                    out_device = adev->primary_output->devices;
+                } else if (usecase->id == USECASE_AUDIO_RECORD_AFE_PROXY) {
+                    out_device = AUDIO_DEVICE_OUT_TELEPHONY_TX;
                 }
+                in_snd_device = platform_get_input_snd_device(adev->platform, out_device);
             }
         }
     }
@@ -822,26 +874,17 @@
     usecase->out_snd_device = out_snd_device;
 
     if (usecase->type == PCM_PLAYBACK) {
-        if ((24 == usecase->stream.out->bit_width) &&
-            (AUDIO_DEVICE_OUT_SPEAKER == usecase->stream.out->devices)) {
-            audio_extn_utils_update_stream_app_type_cfg(adev->platform,
+        audio_extn_utils_update_stream_app_type_cfg(adev->platform,
                                                 &adev->streams_output_cfg_list,
-                                                usecase->stream.out->flags,
-                                                usecase->stream.out->format,
-                                                DEFAULT_OUTPUT_SAMPLING_RATE,
-                                                usecase->stream.out->bit_width,
-                                                &usecase->stream.out->app_type_cfg);
-        } else {
-            audio_extn_utils_update_stream_app_type_cfg(adev->platform,
-                                                &adev->streams_output_cfg_list,
+                                                usecase->stream.out->devices,
                                                 usecase->stream.out->flags,
                                                 usecase->stream.out->format,
                                                 usecase->stream.out->sample_rate,
                                                 usecase->stream.out->bit_width,
                                                 &usecase->stream.out->app_type_cfg);
-        }
-    ALOGI("%s Selected apptype: %d", __func__, usecase->stream.out->app_type_cfg.app_type);
+        ALOGI("%s Selected apptype: %d", __func__, usecase->stream.out->app_type_cfg.app_type);
     }
+
     enable_audio_route(adev, usecase);
 
     /* Applicable only on the targets that has external modem.
@@ -943,16 +986,33 @@
     select_devices(adev, in->usecase);
 
     ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d",
-          __func__, adev->snd_card,
-          in->pcm_device_id, in->config.channels);
-    in->pcm = pcm_open(adev->snd_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;
+          __func__, adev->snd_card, in->pcm_device_id, in->config.channels);
+
+    unsigned int flags = PCM_IN;
+    unsigned int pcm_open_retry_count = 0;
+
+    if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
+        flags |= PCM_MMAP | PCM_NOIRQ;
+        pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
+    }
+
+    while (1) {
+        in->pcm = pcm_open(adev->snd_card, in->pcm_device_id,
+                           flags, &in->config);
+        if (in->pcm == NULL || !pcm_is_ready(in->pcm)) {
+            ALOGE("%s: %s", __func__, pcm_get_error(in->pcm));
+            if (in->pcm != NULL) {
+                pcm_close(in->pcm);
+                in->pcm = NULL;
+            }
+            if (pcm_open_retry_count-- == 0) {
+                ret = -EIO;
+                goto error_open;
+            }
+            usleep(PROXY_OPEN_WAIT_TIME * 1000);
+            continue;
+        }
+        break;
     }
 
     ALOGV("%s: exit", __func__);
@@ -1354,18 +1414,34 @@
 
     select_devices(adev, out->usecase);
 
-    ALOGV("%s: Opening PCM device card_id(%d) device_id(%d)",
-          __func__, 0, out->pcm_device_id);
+    ALOGV("%s: Opening PCM device card_id(%d) device_id(%d) format(%#x)",
+          __func__, adev->snd_card, out->pcm_device_id, out->config.format);
     if (!is_offload_usecase(out->usecase)) {
-        out->pcm = pcm_open(adev->snd_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;
+        unsigned int flags = PCM_OUT;
+        unsigned int pcm_open_retry_count = 0;
+        if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
+            flags |= PCM_MMAP | PCM_NOIRQ;
+            pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
+        } else
+            flags |= PCM_MONOTONIC;
+
+        while (1) {
+            out->pcm = pcm_open(adev->snd_card, out->pcm_device_id,
+                               flags, &out->config);
+            if (out->pcm == NULL || !pcm_is_ready(out->pcm)) {
+                ALOGE("%s: %s", __func__, pcm_get_error(out->pcm));
+                if (out->pcm != NULL) {
+                    pcm_close(out->pcm);
+                    out->pcm = NULL;
+                }
+                if (pcm_open_retry_count-- == 0) {
+                    ret = -EIO;
+                    goto error_open;
+                }
+                usleep(PROXY_OPEN_WAIT_TIME * 1000);
+                continue;
+            }
+            break;
         }
     } else {
         out->pcm = NULL;
@@ -1439,7 +1515,8 @@
 
 static size_t get_input_buffer_size(uint32_t sample_rate,
                                     audio_format_t format,
-                                    int channel_count)
+                                    int channel_count,
+                                    bool is_low_latency)
 {
     size_t size = 0;
 
@@ -1447,13 +1524,19 @@
         return 0;
 
     size = (sample_rate * AUDIO_CAPTURE_PERIOD_DURATION_MSEC) / 1000;
+    if (is_low_latency)
+        size = configured_low_latency_capture_period_size;
     /* 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;
+    /* make sure the size is multiple of 32 bytes
+     * At 48 kHz mono 16-bit PCM:
+     *  5.000 ms = 240 frames = 15*16*1*2 = 480, a whole multiple of 32 (15)
+     *  3.333 ms = 160 frames = 10*16*1*2 = 320, a whole multiple of 32 (10)
+     */
+    size += 0x1f;
+    size &= ~0x1f;
 
     return size;
 }
@@ -1630,6 +1713,10 @@
     return 0;
 }
 
+static bool output_drives_call(struct audio_device *adev, struct stream_out *out)
+{
+    return out == adev->primary_output || out == adev->voice_tx_output;
+}
 
 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
 {
@@ -1691,22 +1778,15 @@
                 select_devices(adev, out->usecase);
 
             if ((adev->mode == AUDIO_MODE_IN_CALL) &&
-                !adev->voice.in_call &&
-                (out == adev->primary_output)) {
-                ret = voice_start_call(adev);
-            } else if ((adev->mode == AUDIO_MODE_IN_CALL) &&
-                       adev->voice.in_call &&
-                       (out == adev->primary_output)) {
-                voice_update_devices_for_all_voice_usecases(adev);
+                    output_drives_call(adev, out)) {
+                adev->current_call_output = out;
+                if (!voice_is_in_call(adev))
+                    ret = voice_start_call(adev);
+                else
+                    voice_update_devices_for_all_voice_usecases(adev);
             }
         }
 
-        if ((adev->mode == AUDIO_MODE_NORMAL) &&
-                adev->voice.in_call &&
-                (out == adev->primary_output)) {
-            ret = voice_stop_call(adev);
-        }
-
         pthread_mutex_unlock(&adev->lock);
         pthread_mutex_unlock(&out->lock);
     }
@@ -1895,7 +1975,10 @@
             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 (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY)
+                ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
+            else
+                ret = pcm_write(out->pcm, (void *)buffer, bytes);
             if (ret < 0)
                 ret = -errno;
             else if (ret == 0)
@@ -2243,7 +2326,7 @@
     err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
     if (err >= 0) {
         val = atoi(value);
-        if ((in->device != val) && (val != 0)) {
+        if (((int)in->device != val) && (val != 0)) {
             in->device = val;
             /* If recording is in progress, change the tx device to new device */
             if (!in->standby && !in->is_st_session)
@@ -2332,6 +2415,8 @@
             ret = audio_extn_ssr_read(stream, buffer, bytes);
         else if (audio_extn_compr_cap_usecase_supported(in->usecase))
             ret = audio_extn_compr_cap_read(in, buffer, bytes);
+        else if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY)
+            ret = pcm_mmap_read(in->pcm, buffer, bytes);
         else
             ret = pcm_read(in->pcm, buffer, bytes);
         if (ret < 0)
@@ -2433,7 +2518,6 @@
     struct stream_out *out;
     int i, ret = 0;
     audio_format_t format;
-    int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
 
     *stream_out = NULL;
 
@@ -2577,11 +2661,6 @@
             out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
         }
 
-        if (out->bit_width == 24 && !platform_check_24_bit_support()) {
-            ALOGW("24 bit support is not enabled, using 16 bit backend");
-            out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
-        }
-
         if (config->offload_info.format == AUDIO_FORMAT_FLAC)
             out->compr_config.codec->options.flac_dec.sample_size = PCM_OUTPUT_BIT_WIDTH;
 
@@ -2606,6 +2685,28 @@
                   __func__, ret);
             goto error_open;
         }
+    } else  if (out->devices == AUDIO_DEVICE_OUT_TELEPHONY_TX) {
+        if (config->sample_rate == 0)
+            config->sample_rate = AFE_PROXY_SAMPLING_RATE;
+        if (config->sample_rate != 48000 && config->sample_rate != 16000 &&
+                config->sample_rate != 8000) {
+            config->sample_rate = AFE_PROXY_SAMPLING_RATE;
+            ret = -EINVAL;
+            goto error_open;
+        }
+        out->sample_rate = config->sample_rate;
+        out->config.rate = config->sample_rate;
+        if (config->format == AUDIO_FORMAT_DEFAULT)
+            config->format = AUDIO_FORMAT_PCM_16_BIT;
+        if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
+            config->format = AUDIO_FORMAT_PCM_16_BIT;
+            ret = -EINVAL;
+            goto error_open;
+        }
+        out->format = config->format;
+        out->usecase = USECASE_AUDIO_PLAYBACK_AFE_PROXY;
+        out->config = pcm_config_afe_proxy_playback;
+        adev->voice_tx_output = out;
     } else if (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
         format = AUDIO_FORMAT_PCM_16_BIT;
         out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
@@ -2619,20 +2720,11 @@
         out->sample_rate = out->config.rate;
     }
 
-    if ((24 == out->bit_width) &&
-        (devices == AUDIO_DEVICE_OUT_SPEAKER)) {
-        sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
-        ALOGI("%s 24-bit playback on Speaker is allowed ONLY at 48khz. Hence changing sample rate to: %d",
-               __func__, sample_rate);
-    } else {
-        sample_rate = out->sample_rate;
-    }
-
-    ALOGV("%s flags %x, format %x, sample_rate %d, out->bit_width %d",
-           __func__, flags, format, sample_rate, out->bit_width);
+    ALOGV("%s devices %d,flags %x, format %x, out->sample_rate %d, out->bit_width %d",
+           __func__, devices, flags, format, out->sample_rate, out->bit_width);
     audio_extn_utils_update_stream_app_type_cfg(adev->platform,
                                                 &adev->streams_output_cfg_list,
-                                                flags, format, sample_rate,
+                                                devices, flags, format, out->sample_rate,
                                                 out->bit_width, &out->app_type_cfg);
     if ((out->usecase == USECASE_AUDIO_PLAYBACK_PRIMARY) ||
         (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
@@ -2725,6 +2817,10 @@
         if (out->compr_config.codec != NULL)
             free(out->compr_config.codec);
     }
+
+    if (adev->voice_tx_output == out)
+        adev->voice_tx_output = NULL;
+
     pthread_cond_destroy(&out->cond);
     pthread_mutex_destroy(&out->lock);
     free(stream);
@@ -2938,10 +3034,16 @@
 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);
+        ALOGD("%s: mode %d\n", __func__, mode);
         adev->mode = mode;
+        if ((mode == AUDIO_MODE_NORMAL || mode == AUDIO_MODE_IN_COMMUNICATION) &&
+                voice_is_in_call(adev)) {
+            voice_stop_call(adev);
+            adev->current_call_output = NULL;
+        }
     }
     pthread_mutex_unlock(&adev->lock);
     return 0;
@@ -2970,7 +3072,8 @@
 {
     int channel_count = audio_channel_count_from_in_mask(config->channel_mask);
 
-    return get_input_buffer_size(config->sample_rate, config->format, channel_count);
+    return get_input_buffer_size(config->sample_rate, config->format, channel_count,
+            false /* is_low_latency: since we don't know, be conservative */);
 }
 
 static int adev_open_input_stream(struct audio_hw_device *dev,
@@ -2986,7 +3089,7 @@
     struct stream_in *in;
     int ret = 0, buffer_size, frame_size;
     int channel_count = audio_channel_count_from_in_mask(config->channel_mask);
-
+    bool is_low_latency = false;
 
     *stream_in = NULL;
     if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
@@ -3030,11 +3133,39 @@
 
     /* Update config params with the requested sample rate and channels */
     in->usecase = USECASE_AUDIO_RECORD;
+    if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE &&
+            (flags & AUDIO_INPUT_FLAG_FAST) != 0) {
+        is_low_latency = true;
+#if LOW_LATENCY_CAPTURE_USE_CASE
+        in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY;
+#endif
+    }
     in->config = pcm_config_audio_capture;
     in->config.rate = config->sample_rate;
     in->format = config->format;
 
-    if (channel_count == 6) {
+    if (in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) {
+        if (config->sample_rate == 0)
+            config->sample_rate = AFE_PROXY_SAMPLING_RATE;
+        if (config->sample_rate != 48000 && config->sample_rate != 16000 &&
+                config->sample_rate != 8000) {
+            config->sample_rate = AFE_PROXY_SAMPLING_RATE;
+            ret = -EINVAL;
+            goto err_open;
+        }
+        if (config->format == AUDIO_FORMAT_DEFAULT)
+            config->format = AUDIO_FORMAT_PCM_16_BIT;
+        if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
+            config->format = AUDIO_FORMAT_PCM_16_BIT;
+            ret = -EINVAL;
+            goto err_open;
+        }
+
+        in->usecase = USECASE_AUDIO_RECORD_AFE_PROXY;
+        in->config = pcm_config_afe_proxy_record;
+        in->config.channels = channel_count;
+        in->config.rate = config->sample_rate;
+    } else if (channel_count == 6) {
         if(audio_extn_ssr_get_enabled()) {
             if(audio_extn_ssr_init(in)) {
                 ALOGE("%s: audio_extn_ssr_init failed", __func__);
@@ -3053,7 +3184,8 @@
         frame_size = audio_stream_in_frame_size(&in->stream);
         buffer_size = get_input_buffer_size(config->sample_rate,
                                             config->format,
-                                            channel_count);
+                                            channel_count,
+                                            is_low_latency);
         in->config.period_size = buffer_size / frame_size;
     }
 
@@ -3135,6 +3267,23 @@
     return 0;
 }
 
+/* This returns 1 if the input parameter looks at all plausible as a low latency period size,
+ * or 0 otherwise.  A return value of 1 doesn't mean the value is guaranteed to work,
+ * just that it _might_ work.
+ */
+static int period_size_is_plausible_for_low_latency(int period_size)
+{
+    switch (period_size) {
+    case 160:
+    case 240:
+    case 320:
+    case 480:
+        return 1;
+    default:
+        return 0;
+    }
+}
+
 static int adev_open(const hw_module_t *module, const char *name,
                      hw_device_t **device)
 {
@@ -3258,25 +3407,31 @@
                                                     &adev->streams_output_cfg_list);
 
     audio_device_ref_count++;
+
+    char value[PROPERTY_VALUE_MAX];
+    int trial;
+    if (property_get("audio_hal.period_size", value, NULL) > 0) {
+        trial = atoi(value);
+        if (period_size_is_plausible_for_low_latency(trial)) {
+            pcm_config_low_latency.period_size = trial;
+            pcm_config_low_latency.start_threshold = trial / 4;
+            pcm_config_low_latency.avail_min = trial / 4;
+            configured_low_latency_capture_period_size = trial;
+        }
+    }
+    if (property_get("audio_hal.in_period_size", value, NULL) > 0) {
+        trial = atoi(value);
+        if (period_size_is_plausible_for_low_latency(trial)) {
+            configured_low_latency_capture_period_size = trial;
+        }
+    }
+
     pthread_mutex_unlock(&adev_init_lock);
 
     ALOGV("%s: exit", __func__);
     return 0;
 }
 
-int pcm_ioctl(struct pcm *pcm, int request, ...)
-{
-    va_list ap;
-    void * arg;
-    int pcm_fd = *(int*)pcm;
-
-    va_start(ap, request);
-    arg = va_arg(ap, void *);
-    va_end(ap);
-
-    return ioctl(pcm_fd, request, arg);
-}
-
 static struct hw_module_methods_t hal_module_methods = {
     .open = adev_open,
 };
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index bdb9aae..d05f743 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -32,9 +32,6 @@
 #define VISUALIZER_LIBRARY_PATH "/system/lib/soundfx/libqcomvisualizer.so"
 #define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/system/lib/soundfx/libqcompostprocbundle.so"
 
-#define BT_SCO_SAMPLE_RATE "bt-sco-samplerate"
-#define BT_SCO_WB_SAMPLE_RATE "bt-sco-wb-samplerate"
-
 /* Flags used to initialize acdb_settings variable that goes to ACDB library */
 #define NONE_FLAG            0x00000000
 #define ANC_FLAG	     0x00000001
@@ -114,6 +111,10 @@
 
     USECASE_AUDIO_SPKR_CALIB_RX,
     USECASE_AUDIO_SPKR_CALIB_TX,
+
+    USECASE_AUDIO_PLAYBACK_AFE_PROXY,
+    USECASE_AUDIO_RECORD_AFE_PROXY,
+
     AUDIO_USECASE_MAX
 };
 
@@ -203,7 +204,7 @@
     int standby;
     int source;
     int pcm_device_id;
-    int device;
+    audio_devices_t device;
     audio_channel_mask_t channel_mask;
     audio_usecase_t usecase;
     bool enable_aec;
@@ -269,6 +270,8 @@
     audio_devices_t out_device;
     struct stream_in *active_input;
     struct stream_out *primary_output;
+    struct stream_out *voice_tx_output;
+    struct stream_out *current_call_output;
     bool bluetooth_nrec;
     bool screen_off;
     int *snd_dev_ref_cnt;
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 005c0b8..b978ae3 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -282,15 +282,15 @@
 static int acdb_device_table[SND_DEVICE_MAX] = {
     [SND_DEVICE_NONE] = -1,
     [SND_DEVICE_OUT_HANDSET] = 7,
-    [SND_DEVICE_OUT_SPEAKER] = 131,
-    [SND_DEVICE_OUT_SPEAKER_REVERSE] = 131,
+    [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] = 131,
+    [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,
@@ -298,7 +298,7 @@
     [SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET] = 37,
     [SND_DEVICE_OUT_AFE_PROXY] = 0,
     [SND_DEVICE_OUT_USB_HEADSET] = 45,
-    [SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = 131,
+    [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,
@@ -765,7 +765,7 @@
     struct platform_data *my_data = NULL;
     int retry_num = 0, snd_card_num = 0;
     const char *snd_card_name;
-    char mixer_xml_path[100];
+    char mixer_xml_path[100],ffspEnable[PROPERTY_VALUE_MAX];
     char *cvd_version = NULL;
 
     my_data = calloc(1, sizeof(struct platform_data));
@@ -867,7 +867,13 @@
             my_data->fluence_mode = FLUENCE_BROADSIDE;
         }
     }
-
+    property_get("persist.audio.FFSP.enable", ffspEnable, "");
+    if (!strncmp("true", ffspEnable, sizeof("true"))) {
+        acdb_device_table[SND_DEVICE_OUT_SPEAKER] = 131;
+        acdb_device_table[SND_DEVICE_OUT_SPEAKER_REVERSE] = 131;
+        acdb_device_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 131;
+        acdb_device_table[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = 131;
+    }
     my_data->voice_feature_set = VOICE_FEATURE_SET_DEFAULT;
     my_data->acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
     if (my_data->acdb_handle == NULL) {
@@ -2375,10 +2381,6 @@
     return fragment_size;
 }
 
-bool platform_check_24_bit_support() {
-    return false;
-}
-
 bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev __unused,
                                               struct audio_usecase *usecase __unused)
 {
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index dec9eda..e38503f 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -164,6 +164,10 @@
 #define LOW_LATENCY_OUTPUT_PERIOD_SIZE 240
 #define LOW_LATENCY_OUTPUT_PERIOD_COUNT 2
 
+#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
+#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
+#define LOW_LATENCY_CAPTURE_USE_CASE 0
+
 #define HDMI_MULTI_PERIOD_SIZE  336
 #define HDMI_MULTI_PERIOD_COUNT 8
 #define HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index e2c30f4..eea10da 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1030,10 +1030,6 @@
     return false;
 }
 
-bool platform_check_24_bit_support() {
-    return false;
-}
-
 bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev __unused,
                                               struct audio_usecase *usecase __unused)
 {
diff --git a/hal/msm8960/platform.h b/hal/msm8960/platform.h
index 20984d8..950ea84 100644
--- a/hal/msm8960/platform.h
+++ b/hal/msm8960/platform.h
@@ -138,4 +138,11 @@
 /* Define macro for Internal FM volume mixer */
 #define FM_RX_VOLUME "Internal FM RX Volume"
 
+#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
+#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
+#define LOW_LATENCY_CAPTURE_USE_CASE 0
+
+#define AFE_PROXY_PLAYBACK_PCM_DEVICE 7
+#define AFE_PROXY_RECORD_PCM_DEVICE 8
+
 #endif // QCOM_AUDIO_PLATFORM_H
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index de1c04d..f034ee1 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -220,6 +220,12 @@
                                       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},
+
+    [USECASE_AUDIO_PLAYBACK_AFE_PROXY] = {AFE_PROXY_PLAYBACK_PCM_DEVICE,
+                                          AFE_PROXY_RECORD_PCM_DEVICE},
+    [USECASE_AUDIO_RECORD_AFE_PROXY] = {AFE_PROXY_PLAYBACK_PCM_DEVICE,
+                                        AFE_PROXY_RECORD_PCM_DEVICE},
+
 };
 
 /* Array to store sound devices */
@@ -245,6 +251,7 @@
     [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_VOICE_TX] = "voice-tx",
     [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",
@@ -292,6 +299,8 @@
     [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_RX] = "voice-rx",
+
     [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",
@@ -332,6 +341,7 @@
     [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_VOICE_TX] = 45,
     [SND_DEVICE_OUT_AFE_PROXY] = 0,
     [SND_DEVICE_OUT_USB_HEADSET] = 45,
     [SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] = 14,
@@ -378,6 +388,8 @@
     [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_RX] = 44,
+
     [SND_DEVICE_IN_VOICE_REC_MIC] = 4,
     [SND_DEVICE_IN_VOICE_REC_MIC_NS] = 107,
     [SND_DEVICE_IN_VOICE_REC_DMIC_STEREO] = 34,
@@ -716,6 +728,9 @@
     backend_table[SND_DEVICE_OUT_BT_SCO_WB] = strdup("bt-sco-wb");
     backend_table[SND_DEVICE_OUT_HDMI] = strdup("hdmi");
     backend_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("speaker-and-hdmi");
+    backend_table[SND_DEVICE_OUT_VOICE_TX] = strdup("afe-proxy");
+    backend_table[SND_DEVICE_IN_VOICE_RX] = strdup("afe-proxy");
+
     backend_table[SND_DEVICE_OUT_AFE_PROXY] = strdup("afe-proxy");
     backend_table[SND_DEVICE_OUT_USB_HEADSET] = strdup("usb-headphones");
     backend_table[SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET] =
@@ -1648,7 +1663,9 @@
                 snd_device = SND_DEVICE_OUT_ANC_HANDSET;
             else
                 snd_device = SND_DEVICE_OUT_VOICE_HANDSET;
-        }
+        } else if (devices & AUDIO_DEVICE_OUT_TELEPHONY_TX)
+            snd_device = SND_DEVICE_OUT_VOICE_TX;
+
         if (snd_device != SND_DEVICE_NONE) {
             goto exit;
         }
@@ -1805,7 +1822,8 @@
                 if (audio_extn_hfp_is_active(adev))
                     platform_set_echo_reference(adev->platform, true);
             }
-        }
+        } else if (out_device & AUDIO_DEVICE_OUT_TELEPHONY_TX)
+            snd_device = SND_DEVICE_IN_VOICE_RX;
     } else if (source == AUDIO_SOURCE_CAMCORDER) {
         if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC ||
             in_device & AUDIO_DEVICE_IN_BACK_MIC) {
@@ -2520,17 +2538,6 @@
     return fragment_size;
 }
 
-bool platform_check_24_bit_support() {
-
-    char value[PROPERTY_VALUE_MAX] = {0};
-    property_get("audio.offload.24bit.enable", value, "0");
-    if (atoi(value)) {
-        ALOGW("Property audio.offload.24bit.enable is set");
-        return true;
-    }
-    return false;
-}
-
 int platform_set_codec_backend_cfg(struct audio_device* adev,
                          unsigned int bit_width, unsigned int sample_rate)
 {
@@ -2659,9 +2666,11 @@
         }
     }
 
-    // 16 bit playbacks is allowed through 16 bit/48 khz backend only
+    // 24 bit playback on speakers and all 16 bit playbacks is allowed through
+    // 16 bit/48 khz backend only
     if ((16 == bit_width) ||
-        ((24 == bit_width) && (SND_DEVICE_OUT_SPEAKER == usecase->devices))) {
+        ((24 == bit_width) &&
+         (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER))) {
         sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
     }
     // Force routing if the expected bitwdith or samplerate
@@ -2680,12 +2689,6 @@
 
 bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev, struct audio_usecase *usecase)
 {
-    // check if 24bit configuration is enabled first
-    if (!platform_check_24_bit_support()) {
-        ALOGW("24bit not enable, no need to check for backend change");
-        return false;
-    }
-
     ALOGV("platform_check_and_set_codec_backend_cfg usecase = %d",usecase->id );
 
     unsigned int new_bit_width, old_bit_width;
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 62aa40a..831ee58 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -70,6 +70,7 @@
     SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
     SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
+    SND_DEVICE_OUT_VOICE_TX,
     SND_DEVICE_OUT_AFE_PROXY,
     SND_DEVICE_OUT_USB_HEADSET,
     SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET,
@@ -127,6 +128,7 @@
     SND_DEVICE_IN_VOICE_REC_MIC_NS,
     SND_DEVICE_IN_VOICE_REC_DMIC_STEREO,
     SND_DEVICE_IN_VOICE_REC_DMIC_FLUENCE,
+    SND_DEVICE_IN_VOICE_RX,
     SND_DEVICE_IN_USB_HEADSET_MIC,
     SND_DEVICE_IN_CAPTURE_FM,
     SND_DEVICE_IN_AANC_HANDSET_MIC,
@@ -178,6 +180,10 @@
 #define AUDIO_CAPTURE_PERIOD_DURATION_MSEC 20
 #define AUDIO_CAPTURE_PERIOD_COUNT 2
 
+#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000
+#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240
+#define LOW_LATENCY_CAPTURE_USE_CASE 1
+
 #define DEVICE_NAME_MAX_SIZE 128
 #define HW_INFO_ARRAY_MAX_SIZE 32
 
@@ -278,6 +284,9 @@
 #define VOWLAN_CALL_PCM_DEVICE 36
 #endif
 
+#define AFE_PROXY_PLAYBACK_PCM_DEVICE 7
+#define AFE_PROXY_RECORD_PCM_DEVICE 8
+
 #ifdef PLATFORM_MSM8x26
 #define HFP_SCO_RX 28
 #define HFP_ASM_RX_TX 29
diff --git a/hal/platform_api.h b/hal/platform_api.h
index b11a0c7..520b27b 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -89,7 +89,6 @@
 uint32_t platform_get_pcm_offload_buffer_size(audio_offload_info_t* info);
 
 bool platform_check_and_set_codec_backend_cfg(struct audio_device* adev, struct audio_usecase *usecase);
-bool platform_check_24_bit_support();
 int platform_get_usecase_index(const char * usecase);
 int platform_set_usecase_pcm_id(audio_usecase_t usecase, int32_t type, int32_t pcm_id);
 void platform_set_echo_reference(void *platform, bool enable);
diff --git a/hal/voice.c b/hal/voice.c
index ecad7a7..c4fa163 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -131,8 +131,8 @@
 
     uc_info->id = usecase_id;
     uc_info->type = VOICE_CALL;
-    uc_info->stream.out = adev->primary_output;
-    uc_info->devices = adev->primary_output->devices;
+    uc_info->stream.out = adev->current_call_output ;
+    uc_info->devices = adev->current_call_output ->devices;
     uc_info->in_snd_device = SND_DEVICE_NONE;
     uc_info->out_snd_device = SND_DEVICE_NONE;
 
@@ -213,6 +213,11 @@
     return call_state;
 }
 
+bool voice_is_in_call(struct audio_device *adev)
+{
+    return adev->voice.in_call;
+}
+
 bool voice_is_in_call_rec_stream(struct stream_in *in)
 {
     bool in_call_rec = false;
@@ -494,6 +499,7 @@
         if (usecase->type == VOICE_CALL) {
             ALOGV("%s: updating device for usecase:%s", __func__,
                   use_case_table[usecase->id]);
+            usecase->stream.out = adev->current_call_output;
             select_devices(adev, usecase->id);
         }
     }
diff --git a/hal/voice_extn/voice_extn.c b/hal/voice_extn/voice_extn.c
index 8952522..e5b979f 100644
--- a/hal/voice_extn/voice_extn.c
+++ b/hal/voice_extn/voice_extn.c
@@ -299,7 +299,8 @@
          * occured, otherwise voice calls will be started unintendedly on
          * speaker.
          */
-        if (is_call_active || adev->voice.in_call) {
+        if (is_call_active ||
+                (adev->voice.in_call && adev->mode == AUDIO_MODE_IN_CALL)) {
             /* Device routing is not triggered for voice calls on the subsequent
              * subs, Hence update the call states if voice call is already
              * active on other sub.