Remove MediaChannel::SetRemoteRenderer().

talk/app/webrtc/webrtcsession.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 1215 matching lines @@
     return false;
   return webrtc::GetTrackIdBySsrc(base_remote_description(), ssrc, track_id);
 }
 
 std::string WebRtcSession::BadStateErrMsg(State state) {
   std::ostringstream desc;
   desc << "Called in wrong state: " << GetStateString(state);
   return desc.str();
 }
 
-void WebRtcSession::SetAudioPlayout(uint32_t ssrc,
-                                    bool enable,
-                                    cricket::AudioRenderer* renderer) {
+void WebRtcSession::SetAudioPlayout(uint32_t ssrc, bool enable) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_) {
     LOG(LS_ERROR) << "SetAudioPlayout: No audio channel exists.";
     return;
   }
-  if (!voice_channel_->SetRemoteRenderer(ssrc, renderer)) {
-    // SetRenderer() can fail if the ssrc does not match any playout channel.
-    LOG(LS_ERROR) << "SetAudioPlayout: ssrc is incorrect: " << ssrc;
-    return;
-  }
   if (!voice_channel_->SetOutputVolume(ssrc, enable ? 1 : 0)) {
     // Allow that SetOutputVolume fail if |enable| is false but assert
     // otherwise. This in the normal case when the underlying media channel has
     // already been deleted.
     ASSERT(enable == false);
   }
 }
 
 void WebRtcSession::SetAudioSend(uint32_t ssrc,
                                  bool enable,
@@ skipping 920 matching lines @@
   if (!srtp_cipher.empty()) {
     metrics_observer_->IncrementSparseEnumCounter(
         srtp_counter_type, rtc::GetSrtpCryptoSuiteFromName(srtp_cipher));
   }
   if (ssl_cipher) {
     metrics_observer_->IncrementSparseEnumCounter(ssl_counter_type, ssl_cipher);
   }
 }
 
 }  // namespace webrtc