Index: content/renderer/media/webrtc_audio_device_impl.cc |
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc |
index 2d5bd365a245b2fe2a311ea908dee7c24f34c106..1039fd25f63a883443bead6b3aa4100138c398cd 100644 |
--- a/content/renderer/media/webrtc_audio_device_impl.cc |
+++ b/content/renderer/media/webrtc_audio_device_impl.cc |
@@ -597,8 +597,9 @@ int32_t WebRtcAudioDeviceImpl::StopPlayout() { |
// webrtc::VoiceEngine assumes that it is OK to call Stop() just in case. |
return 0; |
} |
- playing_ = !audio_output_device_->Stop(); |
- return (!playing_ ? 0 : -1); |
+ audio_output_device_->Stop(); |
+ playing_ = false; |
+ return 0; |
} |
bool WebRtcAudioDeviceImpl::Playing() const { |
@@ -646,8 +647,9 @@ int32_t WebRtcAudioDeviceImpl::StopRecording() { |
// webrtc::VoiceEngine assumes that it is OK to call Stop() just in case. |
return 0; |
} |
- recording_ = !audio_input_device_->Stop(); |
- return (!recording_ ? 0 : -1); |
+ audio_input_device_->Stop(); |
+ recording_ = false; |
+ return 0; |
} |
bool WebRtcAudioDeviceImpl::Recording() const { |