Define RtpTransportControllerSendInterface.

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 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 36 matching lines @@
 class ProcessThread;
 class RateLimiter;
 class ReceiveStatistics;
 class RemoteNtpTimeEstimator;
 class RtcEventLog;
 class RTPPayloadRegistry;
 class RtpReceiver;
 class RTPReceiverAudio;
 class RtpPacketReceived;
 class RtpRtcp;
+class RtpTransportControllerSendInterface;
 class TelephoneEventHandler;
 class VoERTPObserver;
 class VoiceEngineObserver;
 
 struct CallStatistics;
 struct ReportBlock;
 struct SenderInfo;
 
 namespace voe {
 
@@ skipping 199 matching lines @@
   // VoERTP_RTCP
   int SetLocalSSRC(unsigned int ssrc);
   int GetLocalSSRC(unsigned int& ssrc);
   int GetRemoteSSRC(unsigned int& ssrc);
   int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
   int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
   void EnableSendTransportSequenceNumber(int id);
   void EnableReceiveTransportSequenceNumber(int id);
 
   void RegisterSenderCongestionControlObjects(
-      RtpPacketSender* rtp_packet_sender,
-      TransportFeedbackObserver* transport_feedback_observer,
-      PacketRouter* packet_router,
+      RtpTransportControllerSendInterface* transport,
       RtcpBandwidthObserver* bandwidth_observer);
   void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
   void ResetCongestionControlObjects();
 
   void SetRTCPStatus(bool enable);
   int GetRTCPStatus(bool& enabled);
   int SetRTCP_CNAME(const char cName[256]);
   int GetRemoteRTCP_CNAME(char cName[256]);
   int SendApplicationDefinedRTCPPacket(unsigned char subType,
                                        unsigned int name,
@@ skipping 214 matching lines @@
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_