Remove GetOutputScaling from VoiceMediaChannel.

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 1188 matching lines @@
 }
 
 void WebRtcSession::SetAudioPlayoutVolume(uint32 ssrc, double volume) {
   ASSERT(signaling_thread()->IsCurrent());
   ASSERT(volume >= 0 && volume <= 10);
   if (!voice_channel_) {
     LOG(LS_ERROR) << "SetAudioPlayoutVolume: No audio channel exists.";
     return;
   }
 
-  if (!voice_channel_->SetOutputScaling(ssrc, volume, volume))
+  if (!voice_channel_->SetOutputScaling(ssrc, volume, volume)) {
     ASSERT(false);
+  }
 }
 
 bool WebRtcSession::SetCaptureDevice(uint32 ssrc,
                                      cricket::VideoCapturer* camera) {
   ASSERT(signaling_thread()->IsCurrent());
 
   if (!video_channel_) {
     // |video_channel_| doesnt't exist. Probably because the remote end doesnt't
     // support video.
     LOG(LS_WARNING) << "Video not used in this call.";
@@ skipping 875 matching lines @@
 
   if (!srtp_cipher.empty()) {
     metrics_observer_->AddHistogramSample(srtp_name, srtp_cipher);
   }
   if (!ssl_cipher.empty()) {
     metrics_observer_->AddHistogramSample(ssl_name, ssl_cipher);
   }
 }
 
 }  // namespace webrtc