Chromium Code Reviews| Index: webrtc/modules/audio_device/android/opensles_player.cc |
| diff --git a/webrtc/modules/audio_device/android/opensles_player.cc b/webrtc/modules/audio_device/android/opensles_player.cc |
| index b9ccfd594d36dad0d7b85fbe1f8dbd4f16249f09..113dc70463047c8930b3788930ea18e8bd33a51a 100644 |
| --- a/webrtc/modules/audio_device/android/opensles_player.cc |
| +++ b/webrtc/modules/audio_device/android/opensles_player.cc |
| @@ -38,6 +38,7 @@ namespace webrtc { |
| OpenSLESPlayer::OpenSLESPlayer(AudioManager* audio_manager) |
| : audio_parameters_(audio_manager->GetPlayoutAudioParameters()), |
| + stream_type_(audio_manager->OutputStreamType()), |
| audio_device_buffer_(NULL), |
| initialized_(false), |
| playing_(false), |
| @@ -48,6 +49,9 @@ OpenSLESPlayer::OpenSLESPlayer(AudioManager* audio_manager) |
| simple_buffer_queue_(nullptr), |
| volume_(nullptr) { |
| ALOGD("ctor%s", GetThreadInfo().c_str()); |
| + RTC_DCHECK(stream_type_ == SL_ANDROID_STREAM_VOICE || |
|
magjed_webrtc
2015/10/28 09:08:33
Maybe add ' << stream_type_' at end of DCHECK so y
henrika_webrtc
2015/10/28 10:48:01
Done.
|
| + stream_type_ == SL_ANDROID_STREAM_RING || |
| + stream_type_ == SL_ANDROID_STREAM_MEDIA); |
| // Use native audio output parameters provided by the audio manager and |
| // define the PCM format structure. |
| pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(), |
| @@ -347,7 +351,7 @@ bool OpenSLESPlayer::CreateAudioPlayer() { |
| false); |
| // Set audio player configuration to SL_ANDROID_STREAM_VOICE which |
| // corresponds to android.media.AudioManager.STREAM_VOICE_CALL. |
| - SLint32 stream_type = SL_ANDROID_STREAM_VOICE; |
| + SLint32 stream_type = stream_type_; |
| RETURN_ON_ERROR( |
| (*player_config) |
| ->SetConfiguration(player_config, SL_ANDROID_KEY_STREAM_TYPE, |