Index: webrtc/pc/channel.cc |
diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc |
index b05c0bc5024919d3b69e0955e58b22f0a7573958..c6553a003a73d2ef783a9bb9db3938dbd0848b7b 100644 |
--- a/webrtc/pc/channel.cc |
+++ b/webrtc/pc/channel.cc |
@@ -1334,9 +1334,9 @@ bool VoiceChannel::Init() { |
bool VoiceChannel::SetAudioSend(uint32_t ssrc, |
bool enable, |
const AudioOptions* options, |
- AudioRenderer* renderer) { |
+ AudioSource* source) { |
return InvokeOnWorker(Bind(&VoiceMediaChannel::SetAudioSend, media_channel(), |
- ssrc, enable, options, renderer)); |
+ ssrc, enable, options, source)); |
} |
// TODO(juberti): Handle early media the right way. We should get an explicit |
@@ -1454,10 +1454,7 @@ void VoiceChannel::ChangeState() { |
// Send outgoing data if we're the active call, we have the remote content, |
// and we have had some form of connectivity. |
bool send = IsReadyToSend(); |
- SendFlags send_flag = send ? SEND_MICROPHONE : SEND_NOTHING; |
- if (!media_channel()->SetSend(send_flag)) { |
- LOG(LS_ERROR) << "Failed to SetSend " << send_flag << " on voice channel"; |
- } |
+ media_channel()->SetSend(send); |
LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send; |
} |