Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine.

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 890 matching lines @@
   virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
   virtual bool SetRtpSendParameters(
       uint32_t ssrc,
       const webrtc::RtpParameters& parameters) = 0;
   virtual webrtc::RtpParameters GetRtpReceiveParameters(
       uint32_t ssrc) const = 0;
   virtual bool SetRtpReceiveParameters(
       uint32_t ssrc,
       const webrtc::RtpParameters& parameters) = 0;
   // Starts or stops playout of received audio.
-  virtual bool SetPlayout(bool playout) = 0;
+  virtual void SetPlayout(bool playout) = 0;
   // Starts or stops sending (and potentially capture) of local audio.
   virtual void SetSend(bool send) = 0;
   // Configure stream for sending.
   virtual bool SetAudioSend(uint32_t ssrc,
                             bool enable,
                             const AudioOptions* options,
                             AudioSource* source) = 0;
   // Gets current energy levels for all incoming streams.
   virtual bool GetActiveStreams(AudioInfo::StreamList* actives) = 0;
   // Get the current energy level of the stream sent to the speaker.
@@ skipping 208 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_