Index: webrtc/audio/audio_send_stream.cc |
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc |
index c18331cea6c21dc83766ee869585ec0e9a42437a..160a818323b139381c06ca16164b3d8b667e8491 100644 |
--- a/webrtc/audio/audio_send_stream.cc |
+++ b/webrtc/audio/audio_send_stream.cc |
@@ -97,10 +97,20 @@ AudioSendStream::~AudioSendStream() { |
void AudioSendStream::Start() { |
RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
+ ScopedVoEInterface<VoEBase> base(voice_engine()); |
+ int error = base->StartSend(config_.voe_channel_id); |
+ if (error != 0) { |
+ LOG(LS_ERROR) << "AudioSendStream::Start failed with error: " << error; |
+ } |
} |
void AudioSendStream::Stop() { |
RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
+ ScopedVoEInterface<VoEBase> base(voice_engine()); |
+ int error = base->StopSend(config_.voe_channel_id); |
+ if (error != 0) { |
+ LOG(LS_ERROR) << "AudioSendStream::Stop failed with error: " << error; |
+ } |
} |
void AudioSendStream::SignalNetworkState(NetworkState state) { |