Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().

webrtc/media/engine/fakewebrtccall.cc

 /*
  *  Copyright (c) 2015 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 29 matching lines @@
 }
 
 bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, int event,
                                              int duration_ms) {
   latest_telephone_event_.payload_type = payload_type;
   latest_telephone_event_.event_code = event;
   latest_telephone_event_.duration_ms = duration_ms;
   return true;
 }
 
+void FakeAudioSendStream::SetMuted(bool muted) {
+  muted_ = muted;
+}
+
 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {
   return stats_;
 }
 
 FakeAudioReceiveStream::FakeAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config)
     : config_(config), received_packets_(0) {
   RTC_DCHECK(config.voe_channel_id != -1);
 }
 
@@ skipping 400 matching lines @@
     case webrtc::MediaType::ANY:
       ADD_FAILURE()
           << "SignalChannelNetworkState called with unknown parameter.";
   }
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
 }
 }  // namespace cricket