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/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..05bf132 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -822,26 +822,17 @@
     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.
@@ -2433,7 +2424,6 @@
     struct stream_out *out;
     int i, ret = 0;
     audio_format_t format;
-    int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
 
     *stream_out = NULL;
 
@@ -2619,20 +2609,11 @@
         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)) {
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index ff9076e..ff1f9b8 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -2559,9 +2559,11 @@
         }
     }
 
-    // 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