commit 8827cf62246285f6b553c119265b1a2234be9628
author kunleiz <kunleiz@codeaurora.org> Thu Feb 28 16:53:25 2019 +0800
committer Gerrit - the friendly Code Review server <code-review@localhost> Thu Mar 07 03:21:29 2019 -0800
tree 4021bcf4410a573f2eee4b61c0bb4accad692dbe
parent ad630027a34343d2b658ef1a99e27623174a5f5d

hal: set default sample rate during voice/voip call

During voip call, backend sample rate is changed from
48Khz to 44.1Khz when starting 44.1Khz playback. This
is due to voip call using audio path is failed to check
to cause backend sample rate is changed.

Add voip call using audio path usecase condition check to
ensure default sample rate is set.

CRs-Fixed: 2405457
Change-Id: I9001abc059b554be6dda132a3a6b2c54227effd2

hal/msm8916/platform.c

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;

hal/msm8974/platform.c

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;