Prevent a voice channel from sending data before a renderer is set.

webrtc/pc/channel.cc

Index: webrtc/pc/channel.cc
diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc
index 1d5d159d05fd786eafc6fb909fb96298d64e3c52..82affc1059d99977cdefc4f7aa018a4085522a35 100644
--- a/webrtc/pc/channel.cc
+++ b/webrtc/pc/channel.cc
@@ -1342,9 +1342,9 @@ 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
@@ -1462,9 +1462,8 @@ 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";
+  if (!media_channel()->SetSend(send)) {
+    LOG(LS_ERROR) << "Failed to SetSend " << send << " on voice channel";

[the sun, 2016/03/07 21:24:54]

SetSend() always returns true now. This will never happen. Log line should be
moved into AudioSendStream::Start()/Stop()

[Taylor Brandstetter, 2016/03/08 00:00:31]

Done.

   }
 
   LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send;