Replace AudioConferenceMixer with AudioMixer.

webrtc/audio/audio_receive_stream.h

Index: webrtc/audio/audio_receive_stream.h
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index aeaadc6d686cb7da6edab932c4c09ffdbad4cb76..6e7da09d1ddf42d3b3916f312175af7f5db60bf3 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -16,6 +16,7 @@
 #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"
@@ -64,6 +65,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
 
  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;
@@ -72,6 +75,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   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