Merge "hal: set default sample rate during voice/voip call"
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index d8deae8..ca98ccc 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -5792,7 +5792,10 @@
    if (voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
        list_for_each(node, &adev->usecase_list) {
            uc =  node_to_item(node, struct audio_usecase, list);
-           if (uc && (uc->type == VOICE_CALL || uc->type == VOIP_CALL) && uc->stream.out) {
+           if (uc && uc->stream.out &&
+               (uc->type == VOICE_CALL ||
+                uc->type == VOIP_CALL ||
+                uc->id == USECASE_AUDIO_PLAYBACK_VOIP)) {
                out_snd_device = platform_get_output_snd_device(adev->platform, uc->stream.out);
                backend_idx = platform_get_backend_index(out_snd_device);
                break;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index a31b048..d2af736 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -5971,7 +5971,10 @@
    if (voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
        list_for_each(node, &adev->usecase_list) {
            uc =  node_to_item(node, struct audio_usecase, list);
-           if (uc && (uc->type == VOICE_CALL || uc->type == VOIP_CALL) && uc->stream.out) {
+           if (uc && uc->stream.out &&
+               (uc->type == VOICE_CALL ||
+                uc->type == VOIP_CALL ||
+                uc->id == USECASE_AUDIO_PLAYBACK_VOIP)) {
                out_snd_device = platform_get_output_snd_device(adev->platform, uc->stream.out);
                backend_idx = platform_get_backend_index(out_snd_device);
                break;