Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().

webrtc/audio/audio_receive_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 192 matching lines @@
   stats.decoding_plc_cng = ds.decoded_plc_cng;
 
   return stats;
 }
 
 void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   channel_proxy_->SetSink(std::move(sink));
 }
 
+void AudioReceiveStream::SetGain(float gain) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel_proxy_->SetChannelOutputVolumeScaling(gain);
+}
+
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_;
 }
 
 void AudioReceiveStream::SignalNetworkState(NetworkState state) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
@@ skipping 34 matching lines @@
 
 VoiceEngine* AudioReceiveStream::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