diff --git a/Android.mk b/Android.mk
index 9c5e7ff..f7bdd70 100644
--- a/Android.mk
+++ b/Android.mk
@@ -1,6 +1,6 @@
 # TODO:  Find a better way to separate build configs for ADP vs non-ADP devices
 ifneq ($(TARGET_BOARD_AUTO),true)
-  ifneq ($(filter msm8960 msm8226 msm8x26 msm8x84 msm8084 msm8992 msm8994 msm8996 msm8909 msm8952,$(TARGET_BOARD_PLATFORM)),)
+  ifneq ($(filter msm8960 msm8226 msm8x26 msm8x84 msm8084 msm8992 msm8994 msm8996 msm8909 msm8952 msmcobalt,$(TARGET_BOARD_PLATFORM)),)
 
     MY_LOCAL_PATH := $(call my-dir)
 
diff --git a/hal/Android.mk b/hal/Android.mk
index fc2be7d..d02524c 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -10,7 +10,7 @@
 ifneq ($(filter msm8960,$(TARGET_BOARD_PLATFORM)),)
   LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="2"
 endif
-ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994 msm8996,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8974 msm8226 msm8084 msm8992 msm8994 msm8996 msmcobalt,$(TARGET_BOARD_PLATFORM)),)
   # B-family platform uses msm8974 code base
   AUDIO_PLATFORM = msm8974
 ifneq ($(filter msm8974,$(TARGET_BOARD_PLATFORM)),)
@@ -41,6 +41,12 @@
   LOCAL_CFLAGS += -DKPI_OPTIMIZE_ENABLED
   MULTIPLE_HW_VARIANTS_ENABLED := true
 endif
+ifneq ($(filter msmcobalt,$(TARGET_BOARD_PLATFORM)),)
+  LOCAL_CFLAGS := -DPLATFORM_MSMCOBALT
+  LOCAL_CFLAGS += -DMAX_TARGET_SPECIFIC_CHANNEL_CNT="4"
+  LOCAL_CFLAGS += -DKPI_OPTIMIZE_ENABLED
+  MULTIPLE_HW_VARIANTS_ENABLED := true
+endif
 endif
 
 ifneq ($(filter msm8916 msm8909 msm8952,$(TARGET_BOARD_PLATFORM)),)
@@ -121,7 +127,7 @@
     LOCAL_SRC_FILES += audio_extn/dsm_feedback.c
 endif
 
-ifneq ($(filter msm8992 msm8994 msm8996,$(TARGET_BOARD_PLATFORM)),)
+ifneq ($(filter msm8992 msm8994 msm8996 msmcobalt,$(TARGET_BOARD_PLATFORM)),)
   # push codec/mad calibration to HW dep node
   # applicable to msm8992/8994 or newer platforms
   LOCAL_CFLAGS += -DHWDEP_CAL_ENABLED
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 35b726e..21ce868 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -350,6 +350,79 @@
         adev->adm_abandon_focus(adev->adm_data, in->capture_handle);
 }
 
+// Time string format similar to logcat, buffer_length must be >= 19 chars.
+static void ns2string(int64_t ns, char *buffer, int buffer_length)
+{
+    const int one_second = 1000000000;
+    const time_t sec = ns / one_second;
+    struct tm tm;
+    localtime_r(&sec, &tm);
+    snprintf(buffer, buffer_length, "%02d-%02d %02d:%02d:%02d.%03d",
+        tm.tm_mon + 1, // localtime_r uses months in 0 - 11 range
+        tm.tm_mday, tm.tm_hour, tm.tm_min, tm.tm_sec,
+        (int)(ns % one_second / 1000000));
+}
+
+// Convert timespec to nsec.
+static int64_t ts2ns(const struct timespec *ts)
+{
+    return ts->tv_sec * 1000000000LL + ts->tv_nsec;
+}
+
+// Log errors: consecutive errors with the same code will
+// be aggregated if they occur within one second.
+// A mutual exclusion lock must be held before calling.
+static void log_error_l(struct error_log *log, int code) {
+    ++log->errors;
+
+    struct timespec now_ts = { 0, 0 };
+    (void)clock_gettime(CLOCK_REALTIME, &now_ts);
+    const int64_t now = ts2ns(&now_ts);
+
+    // Within 1 second, cluster the same error codes together.
+    const int one_second = 1000000000;
+    if (code == log->entries[log->idx].code &&
+            now - log->entries[log->idx].last_time < one_second) {
+        log->entries[log->idx].count++;
+        log->entries[log->idx].last_time = now;
+        return;
+    }
+
+    // Add new error entry.
+    if (++log->idx >= ARRAY_SIZE(log->entries)) {
+        log->idx = 0;
+    }
+    log->entries[log->idx].count = 1;
+    log->entries[log->idx].code = code;
+    log->entries[log->idx].first_time = now;
+    log->entries[log->idx].last_time = now;
+}
+
+// Dump information in the error log. A mutual exclusion lock
+// should be held, but if that cannot be obtained, one should
+// make a copy of the error log before calling -- the call is
+// still safe, but there might be some misinterpreted data.
+static void log_dump_l(const struct error_log *log, int fd)
+{
+    dprintf(fd, "      Errors: %u\n", log->errors);
+    if (log->errors == 0)
+        return;
+
+    dprintf(fd, "      Index Code  Freq          First time           Last time\n");
+    for (size_t i = 0; i < ARRAY_SIZE(log->entries); ++i) {
+        if (log->entries[i].count != 0) {
+            char first_time[32];
+            char last_time[32];
+            ns2string(log->entries[i].first_time, first_time, sizeof(first_time));
+            ns2string(log->entries[i].last_time, last_time, sizeof(last_time));
+            dprintf(fd, "      %c%4zu %4d %5d  %s  %s\n",
+                    i == log->idx ? '*' : ' ', // mark head position
+                    i, log->entries[i].code, log->entries[i].count,
+                    first_time, last_time);
+        }
+    }
+}
+
 static int parse_snd_card_status(struct str_parms * parms, int * card,
                                  card_status_t * status)
 {
@@ -546,7 +619,9 @@
             ALOGE("%s: spkr_start_processing failed", __func__);
             goto on_error;
         }
-    } else if (platform_can_split_snd_device(snd_device, &num_devices, new_snd_devices)) {
+    } else if (platform_can_split_snd_device(snd_device,
+                                             &num_devices,
+                                             new_snd_devices) == 0) {
         for (i = 0; i < num_devices; i++) {
             enable_snd_device(adev, new_snd_devices[i]);
         }
@@ -590,7 +665,9 @@
             snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
             audio_extn_spkr_prot_is_enabled()) {
             audio_extn_spkr_prot_stop_processing(snd_device);
-        } else if (platform_can_split_snd_device(snd_device, &num_devices, new_snd_devices)) {
+        } else if (platform_can_split_snd_device(snd_device,
+                                                 &num_devices,
+                                                 new_snd_devices) == 0) {
             for (i = 0; i < num_devices; i++) {
                 disable_snd_device(adev, new_snd_devices[i]);
             }
@@ -612,6 +689,114 @@
     return 0;
 }
 
+/*
+  legend:
+  uc - existing usecase
+  new_uc - new usecase
+  d1, d11, d2 - SND_DEVICE enums
+  a1, a2 - corresponding ANDROID device enums
+  B, B1, B2 - backend strings
+
+case 1
+  uc->dev  d1 (a1)               B1
+  new_uc->dev d1 (a1), d2 (a2)   B1, B2
+
+  resolution: disable and enable uc->dev on d1
+
+case 2
+  uc->dev d1 (a1)        B1
+  new_uc->dev d11 (a1)   B1
+
+  resolution: need to switch uc since d1 and d11 are related
+  (e.g. speaker and voice-speaker)
+  use ANDROID_DEVICE_OUT enums to match devices since SND_DEVICE enums may vary
+
+case 3
+  uc->dev d1 (a1)        B1
+  new_uc->dev d2 (a2)    B2
+
+  resolution: no need to switch uc
+
+case 4
+  uc->dev d1 (a1)      B
+  new_uc->dev d2 (a2)  B
+
+  resolution: disable enable uc-dev on d2 since backends match
+  we cannot enable two streams on two different devices if they
+  share the same backend. e.g. if offload is on speaker device using
+  QUAD_MI2S backend and a low-latency stream is started on voice-handset
+  using the same backend, offload must also be switched to voice-handset.
+
+case 5
+  uc->dev  d1 (a1)                  B
+  new_uc->dev d1 (a1), d2 (a2)      B
+
+  resolution: disable enable uc-dev on d2 since backends match
+  we cannot enable two streams on two different devices if they
+  share the same backend.
+
+case 6
+  uc->dev  d1 a1    B1
+  new_uc->dev d2 a1 B2
+
+  resolution: no need to switch
+
+case 7
+
+  uc->dev d1 (a1), d2 (a2)       B1, B2
+  new_uc->dev d1                 B1
+
+  resolution: no need to switch
+
+*/
+static snd_device_t derive_playback_snd_device(struct audio_usecase *uc,
+                                               struct audio_usecase *new_uc,
+                                               snd_device_t new_snd_device)
+{
+    audio_devices_t a1 = uc->stream.out->devices;
+    audio_devices_t a2 = new_uc->stream.out->devices;
+
+    snd_device_t d1 = uc->out_snd_device;
+    snd_device_t d2 = new_snd_device;
+
+    // Treat as a special case when a1 and a2 are not disjoint
+    if ((a1 != a2) && (a1 & a2)) {
+        snd_device_t d3[2];
+        int num_devices = 0;
+        int ret = platform_can_split_snd_device(popcount(a1) > 1 ? d1 : d2,
+                                                &num_devices,
+                                                d3);
+        if (ret < 0) {
+            if (ret != -ENOSYS) {
+                ALOGW("%s failed to split snd_device %d",
+                      __func__,
+                      popcount(a1) > 1 ? d1 : d2);
+            }
+            goto end;
+        }
+
+        // NB: case 7 is hypothetical and isn't a practical usecase yet.
+        // But if it does happen, we need to give priority to d2 if
+        // the combo devices active on the existing usecase share a backend.
+        // This is because we cannot have a usecase active on a combo device
+        // and a new usecase requests one device in this combo pair.
+        if (platform_check_backends_match(d3[0], d3[1])) {
+            return d2; // case 5
+        } else {
+            return d1; // case 1
+        }
+    } else {
+        if (platform_check_backends_match(d1, d2)) {
+            return d2; // case 2, 4
+        } else {
+            return d1; // case 6, 3
+        }
+    }
+
+end:
+    return d2; // return whatever was calculated before.
+}
+
 static void check_and_route_playback_usecases(struct audio_device *adev,
                                               struct audio_usecase *uc_info,
                                               snd_device_t snd_device)
@@ -660,10 +845,15 @@
             }
         }
 
+        snd_device_t d_device;
         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);
+                d_device = derive_playback_snd_device(usecase, uc_info,
+                                                      snd_device);
+                enable_snd_device(adev, d_device);
+                /* Update the out_snd_device before enabling the audio route */
+                usecase->out_snd_device = d_device;
             }
         }
 
@@ -671,9 +861,7 @@
            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);
             }
         }
@@ -1700,8 +1888,25 @@
     return 0;
 }
 
-static int out_dump(const struct audio_stream *stream __unused, int fd __unused)
+static int out_dump(const struct audio_stream *stream, int fd)
 {
+    struct stream_out *out = (struct stream_out *)stream;
+
+    // We try to get the lock for consistency,
+    // but it isn't necessary for these variables.
+    // If we're not in standby, we may be blocked on a write.
+    const bool locked = (pthread_mutex_trylock(&out->lock) == 0);
+    dprintf(fd, "      Standby: %s\n", out->standby ? "yes" : "no");
+    dprintf(fd, "      Frames written: %lld\n", (long long)out->written);
+
+    if (locked) {
+        log_dump_l(&out->error_log, fd);
+        pthread_mutex_unlock(&out->lock);
+    } else {
+        // We don't have the lock here, copy for safety.
+        struct error_log log = out->error_log;
+        log_dump_l(&log, fd);
+    }
     return 0;
 }
 
@@ -1992,6 +2197,7 @@
     struct stream_out *out = (struct stream_out *)stream;
     struct audio_device *adev = out->dev;
     ssize_t ret = 0;
+    int error_code = ERROR_CODE_STANDBY;
 
     lock_output_stream(out);
     if (out->standby) {
@@ -2044,9 +2250,13 @@
             out->playback_started = 1;
             out->offload_state = OFFLOAD_STATE_PLAYING;
         }
+        if (ret < 0) {
+            log_error_l(&out->error_log, ERROR_CODE_WRITE);
+        }
         pthread_mutex_unlock(&out->lock);
         return ret;
     } else {
+        error_code = ERROR_CODE_WRITE;
         if (out->pcm) {
             if (out->muted)
                 memset((void *)buffer, 0, bytes);
@@ -2063,23 +2273,35 @@
             else
                 ret = pcm_write(out->pcm, (void *)buffer, bytes);
 
-            if (ret == 0)
-                out->written += bytes / (out->config.channels * sizeof(short));
-
             release_out_focus(out, ns);
+        } else {
+            LOG_ALWAYS_FATAL("out->pcm is NULL after starting output stream");
         }
     }
 
 exit:
+    // For PCM we always consume the buffer and return #bytes regardless of ret.
+    if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+        out->written += bytes / (out->config.channels * sizeof(short));
+    }
+    long long sleeptime_us = 0;
+    if (ret != 0) {
+        log_error_l(&out->error_log, error_code);
+        if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+            ALOGE_IF(out->pcm != NULL,
+                    "%s: error %zd - %s", __func__, ret, pcm_get_error(out->pcm));
+            sleeptime_us = bytes * 1000000LL / audio_stream_out_frame_size(stream) /
+                out_get_sample_rate(&out->stream.common);
+            // usleep not guaranteed for values over 1 second but we don't limit here.
+        }
+    }
+
     pthread_mutex_unlock(&out->lock);
 
     if (ret != 0) {
         out_on_error(&out->stream.common);
-        if (out->pcm)
-            ALOGE("%s: error %zu - %s", __func__, ret, pcm_get_error(out->pcm));
-        if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD)
-            usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) /
-                   out_get_sample_rate(&out->stream.common));
+        if (sleeptime_us != 0)
+            usleep(sleeptime_us);
     }
     return bytes;
 }
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 5ce3018..4963316 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -51,6 +51,8 @@
 #define MAX_SUPPORTED_CHANNEL_MASKS 2
 #define DEFAULT_HDMI_OUT_CHANNELS   2
 
+#define ERROR_LOG_ENTRIES 16
+
 typedef enum card_status_t {
     CARD_STATUS_OFFLINE,
     CARD_STATUS_ONLINE
@@ -148,6 +150,24 @@
     int data[];
 };
 
+enum {
+    ERROR_CODE_STANDBY,
+    ERROR_CODE_WRITE,
+};
+
+struct error_log_entry {
+    int32_t code;
+    int32_t count;
+    int64_t first_time;
+    int64_t last_time;
+};
+
+struct error_log {
+    uint32_t errors;
+    uint32_t idx;
+    struct error_log_entry entries[ERROR_LOG_ENTRIES];
+};
+
 struct stream_out {
     struct audio_stream_out stream;
     pthread_mutex_t lock; /* see note below on mutex acquisition order */
@@ -188,6 +208,8 @@
     bool routing_change;
     struct audio_device *dev;
     card_status_t card_status;
+
+    struct error_log error_log;
 };
 
 struct stream_in {
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 52e9fd8..72d9591 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -1424,35 +1424,35 @@
     return ret;
 }
 
-bool platform_can_split_snd_device(snd_device_t snd_device,
+int platform_can_split_snd_device(snd_device_t snd_device,
                                    int *num_devices,
                                    snd_device_t *new_snd_devices)
 {
-    bool status = false;
+    int ret = -EINVAL;
 
     if (NULL == num_devices || NULL == new_snd_devices) {
         ALOGE("%s: NULL pointer ..", __func__);
-        return false;
+        return -EINVAL;
     }
 
     /*
      * If wired headset/headphones/line devices share the same backend
-     * with speaker/earpiece this routine returns false.
+     * with speaker/earpiece this routine -EINVAL.
      */
     if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES &&
         !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_HEADPHONES)) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_HEADPHONES;
-        status = true;
+        ret = 0;
     } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_LINE &&
                !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_LINE)) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_LINE;
-        status = true;
+        ret = 0;
     }
-    return status;
+    return ret;
 }
 
 snd_device_t platform_get_output_snd_device(void *platform, audio_devices_t devices)
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index d7fc84c..94f4621 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1075,11 +1075,11 @@
     return true;
 }
 
-bool platform_can_split_snd_device(snd_device_t in_snd_device __unused,
+int platform_can_split_snd_device(snd_device_t in_snd_device __unused,
                                    int *num_devices __unused,
                                    snd_device_t *out_snd_devices __unused)
 {
-    return false;
+    return -ENOSYS;
 }
 
 bool platform_check_backends_match(snd_device_t snd_device1 __unused,
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 441e2a1..a5cfc18 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -1010,24 +1010,37 @@
     list_init(&operator_info_list);
 
     set_platform_defaults(my_data);
+    bool card_verifed[MAX_SND_CARD] = {0};
+    const int retry_limit = property_get_int32("audio.snd_card.open.retries", RETRY_NUMBER);
 
-    while (snd_card_num < MAX_SND_CARD) {
-        adev->mixer = mixer_open(snd_card_num);
+    for (;;) {
+        if (snd_card_num >= MAX_SND_CARD) {
+            if (retry_num++ >= retry_limit) {
+                ALOGE("%s: Unable to find correct sound card, aborting.", __func__);
+                goto init_failed;
+            }
 
-        while (!adev->mixer && retry_num < RETRY_NUMBER) {
+            snd_card_num = 0;
             usleep(RETRY_US);
-            adev->mixer = mixer_open(snd_card_num);
-            retry_num++;
+            continue;
         }
 
+        if (card_verifed[snd_card_num]) {
+            ++snd_card_num;
+            continue;
+        }
+
+        adev->mixer = mixer_open(snd_card_num);
+
         if (!adev->mixer) {
             ALOGE("%s: Unable to open the mixer card: %d", __func__,
-                   snd_card_num);
-            retry_num = 0;
-            snd_card_num++;
+               snd_card_num);
+            ++snd_card_num;
             continue;
         }
 
+        card_verifed[snd_card_num] = true;
+
         snd_card_name = mixer_get_name(adev->mixer);
         my_data->hw_info = hw_info_init(snd_card_name);
 
@@ -1106,8 +1119,9 @@
                         min(strlen(snd_card_name), strlen(my_data->snd_card_name))) != 0) {
             ALOGI("%s: found valid sound card %s, but not primary sound card %s",
                    __func__, snd_card_name, my_data->snd_card_name);
-            retry_num = 0;
-            snd_card_num++;
+            ++snd_card_num;
+            mixer_close(adev->mixer);
+            adev->mixer = NULL;
             hw_info_deinit(my_data->hw_info);
             my_data->hw_info = NULL;
             continue;
@@ -1120,6 +1134,10 @@
 
         if (!adev->audio_route) {
             ALOGE("%s: Failed to init audio route controls, aborting.", __func__);
+            mixer_close(adev->mixer);
+            adev->mixer = NULL;
+            hw_info_deinit(my_data->hw_info);
+            my_data->hw_info = NULL;
             goto init_failed;
         }
         adev->snd_card = snd_card_num;
@@ -1127,11 +1145,6 @@
         break;
     }
 
-    if (snd_card_num >= MAX_SND_CARD) {
-        ALOGE("%s: Unable to find correct sound card, aborting.", __func__);
-        goto init_failed;
-    }
-
     //set max volume step for voice call
     property_get("ro.config.vc_call_vol_steps", value, TOSTRING(MAX_VOL_INDEX));
     my_data->max_vol_index = atoi(value);
@@ -1850,47 +1863,46 @@
     return ret;
 }
 
-bool platform_can_split_snd_device(snd_device_t snd_device,
-                                   int *num_devices,
-                                   snd_device_t *new_snd_devices)
+int platform_can_split_snd_device(snd_device_t snd_device,
+                                  int *num_devices,
+                                  snd_device_t *new_snd_devices)
 {
-    bool status = false;
-
+    int ret = -EINVAL;
     if (NULL == num_devices || NULL == new_snd_devices) {
         ALOGE("%s: NULL pointer ..", __func__);
-        return false;
+        return -EINVAL;
     }
 
     /*
      * If wired headset/headphones/line devices share the same backend
-     * with speaker/earpiece this routine returns false.
+     * with speaker/earpiece this routine returns -EINVAL.
      */
     if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES &&
         !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_HEADPHONES)) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_HEADPHONES;
-        status = true;
+        ret = 0;
     } else if (snd_device == SND_DEVICE_OUT_SPEAKER_AND_LINE &&
                !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER, SND_DEVICE_OUT_LINE)) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
         new_snd_devices[1] = SND_DEVICE_OUT_LINE;
-        status = true;
+        ret = 0;
     } else if (snd_device == SND_DEVICE_OUT_SPEAKER_SAFE_AND_HEADPHONES &&
                !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER_SAFE, SND_DEVICE_OUT_HEADPHONES)) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER_SAFE;
         new_snd_devices[1] = SND_DEVICE_OUT_HEADPHONES;
-        status = true;
+        ret = 0;
     } else if (snd_device == SND_DEVICE_OUT_SPEAKER_SAFE_AND_LINE &&
                !platform_check_backends_match(SND_DEVICE_OUT_SPEAKER_SAFE, SND_DEVICE_OUT_LINE)) {
         *num_devices = 2;
         new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER_SAFE;
         new_snd_devices[1] = SND_DEVICE_OUT_LINE;
-        status = true;
+        ret = 0;
     }
-    return status;
+    return ret;
 }
 
 snd_device_t platform_get_output_snd_device(void *platform, audio_devices_t devices)
@@ -2742,6 +2754,14 @@
     struct platform_data *my_data = (struct platform_data *)adev->platform;
 
     if (my_data->speaker_lr_swap != swap_channels) {
+
+        // do not swap channels in audio modes with concurrent capture and playback
+        // as this may break the echo reference
+        if ((adev->mode == AUDIO_MODE_IN_COMMUNICATION) || (adev->mode == AUDIO_MODE_IN_CALL)) {
+            ALOGV("%s: will not swap due to audio mode %d", __func__, adev->mode);
+            return 0;
+        }
+
         my_data->speaker_lr_swap = swap_channels;
 
         list_for_each(node, &adev->usecase_list) {
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 41e600e..fbc84c1 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -100,9 +100,9 @@
 void platform_set_echo_reference(struct audio_device *adev, bool enable, audio_devices_t out_device);
 int platform_swap_lr_channels(struct audio_device *adev, bool swap_channels);
 
-bool platform_can_split_snd_device(snd_device_t in_snd_device,
-                                   int *num_devices,
-                                   snd_device_t *out_snd_devices);
+int platform_can_split_snd_device(snd_device_t in_snd_device,
+                                  int *num_devices,
+                                  snd_device_t *out_snd_devices);
 
 bool platform_check_backends_match(snd_device_t snd_device1, snd_device_t snd_device2);
 
diff --git a/post_proc/volume_listener.c b/post_proc/volume_listener.c
index 3367b76..4a48a25 100644
--- a/post_proc/volume_listener.c
+++ b/post_proc/volume_listener.c
@@ -32,7 +32,8 @@
 #define VOL_FLAG ( EFFECT_FLAG_TYPE_INSERT | \
                    EFFECT_FLAG_VOLUME_IND | \
                    EFFECT_FLAG_DEVICE_IND | \
-                   EFFECT_FLAG_OFFLOAD_SUPPORTED)
+                   EFFECT_FLAG_OFFLOAD_SUPPORTED | \
+                   EFFECT_FLAG_NO_PROCESS)
 
 #define PRINT_STREAM_TYPE(i) ALOGV("descriptor found and is of stream type %s ",\
                                                             i == MUSIC?"MUSIC": \
@@ -332,41 +333,6 @@
     return sample;
 }
 
-static int vol_effect_process(effect_handle_t self,
-                              audio_buffer_t *in_buffer,
-                              audio_buffer_t *out_buffer)
-{
-    int status = 0;
-    ALOGV("%s Called ", __func__);
-
-    vol_listener_context_t *context = (vol_listener_context_t *)self;
-    pthread_mutex_lock(&vol_listner_init_lock);
-
-    if (context->state != VOL_LISTENER_STATE_ACTIVE) {
-        ALOGE("%s: state is not active .. return error", __func__);
-        status = -EINVAL;
-        goto exit;
-    }
-
-    // calculation based on channel count 2
-    if (in_buffer->raw != out_buffer->raw) {
-        if (context->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
-            size_t i;
-            for (i = 0; i < out_buffer->frameCount*2; i++) {
-                out_buffer->s16[i] = clamp16(out_buffer->s16[i] + in_buffer->s16[i]);
-            }
-        } else {
-            memcpy(out_buffer->raw, in_buffer->raw, out_buffer->frameCount * 2 * sizeof(int16_t));
-        }
-
-    }
-
-exit:
-    pthread_mutex_unlock(&vol_listner_init_lock);
-    return status;
-}
-
-
 static int vol_effect_command(effect_handle_t self,
                               uint32_t cmd_code, uint32_t cmd_size,
                               void *p_cmd_data, uint32_t *reply_size,
@@ -818,7 +784,7 @@
 
 /* effect_handle_t interface implementation for volume listener effect */
 static const struct effect_interface_s effect_interface = {
-    vol_effect_process,
+    NULL,
     vol_effect_command,
     vol_effect_get_descriptor,
     NULL,