Support for remote audio into tracks

webrtc/voice_engine/channel.h

Index: webrtc/voice_engine/channel.h
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index d3b1b93645bf79a93a0d9769e6a6499010266eee..9184b93e099bc3df8338212ec668fae55a8006ab 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_
 
+#include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
@@ -192,6 +193,8 @@ public:
         CriticalSectionWrapper* callbackCritSect);
     int32_t UpdateLocalTimeStamp();
 
+    void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
+
     // API methods
 
     // VoEBase
@@ -508,6 +511,7 @@ private:
     TelephoneEventHandler* telephone_event_handler_;
     rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule;
     rtc::scoped_ptr<AudioCodingModule> audio_coding_;
+    rtc::scoped_ptr<AudioSinkInterface> audio_sink_;
     AudioLevel _outputAudioLevel;
     bool _externalTransport;
     AudioFrame _audioFrame;