Replace AudioConferenceMixer with AudioMixer.

webrtc/audio/audio_receive_stream.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
 
 #include <memory>
 
 #include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/api/call/audio_receive_stream.h"
 #include "webrtc/api/call/audio_state.h"
+#include "webrtc/audio/audio_state.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
 namespace webrtc {
 class CongestionController;
 class RemoteBitrateEstimator;
 class RtcEventLog;
 
 namespace voe {
@@ skipping 28 matching lines @@
   const webrtc::AudioReceiveStream::Config& config() const;
 
   // AudioMixer::Source
   AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
                                        AudioFrame* audio_frame) override;
   int PreferredSampleRate() const override;
   int Ssrc() const override;
 
  private:
   VoiceEngine* voice_engine() const;
+  AudioState* audio_state() const;
+  int SetVoiceEnginePlayout(bool playout);

[aleloi, 2016/11/17 18:12:26]

Added method for VoE->StartPlayout() / StopPlayout() to use in Start() and
Stop().

 
   rtc::ThreadChecker thread_checker_;
   RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr;
   const webrtc::AudioReceiveStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
+  bool playing_ ACCESS_ON(thread_checker_) = false;
+
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_