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

webrtc/audio/audio_send_stream.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 109 matching lines @@
   }
 }
 
 bool AudioSendStream::SendTelephoneEvent(int payload_type, int event,
                                          int duration_ms) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) &&
          channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
 }
 
+void AudioSendStream::SetMuted(bool muted) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel_proxy_->SetInputMute(muted);
+}
+
 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   webrtc::AudioSendStream::Stats stats;
   stats.local_ssrc = config_.rtp.ssrc;
   ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine());
   ScopedVoEInterface<VoECodec> codec(voice_engine());
   ScopedVoEInterface<VoEVolumeControl> volume(voice_engine());
 
   webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
   stats.bytes_sent = call_stats.bytesSent;
@@ skipping 89 matching lines @@
 
 VoiceEngine* AudioSendStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc