| Index: webrtc/voice_engine/channel.h
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| diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
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| index fb9d61ddc84b744c7c1cd3d787e20b30519c86d8..13d8e49877a3d74f924dcded2105013babfb10ae 100644
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| --- a/webrtc/voice_engine/channel.h
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| +++ b/webrtc/voice_engine/channel.h
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| @@ -28,7 +28,6 @@
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|  #include "webrtc/modules/audio_processing/rms_level.h"
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|  #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
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|  #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
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| -#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
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|  #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
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|  #include "webrtc/voice_engine/audio_level.h"
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|  #include "webrtc/voice_engine/file_player.h"
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| @@ -53,6 +52,7 @@
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|  class RemoteNtpTimeEstimator;
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|  class RtcEventLog;
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|  class RTPPayloadRegistry;
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| +class RtpReceiver;
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|  class RTPReceiverAudio;
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|  class RtpPacketReceived;
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|  class RtpRtcp;
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| @@ -399,10 +399,6 @@
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|  
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|    void OnRecoverableUplinkPacketLossRate(float recoverable_packet_loss_rate);
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|  
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| -  std::vector<RtpSource> GetSources() const {
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| -    return rtp_receiver_->GetSources();
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| -  }
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| -
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|   private:
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|    class ProcessAndEncodeAudioTask;
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|  
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| 
 |