Define RtpTransportControllerSendInterface.

webrtc/audio/audio_send_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_SEND_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/bitrate_allocator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h"
 
 namespace webrtc {
-class SendSideCongestionController;
 class VoiceEngine;
 class RtcEventLog;
 class RtcpBandwidthObserver;
 class RtcpRttStats;
-class PacketRouter;
+class RtpTransportControllerSendInterface;
 
 namespace voe {
 class ChannelProxy;
 }  // namespace voe
 
 namespace internal {
 class AudioSendStream final : public webrtc::AudioSendStream,
                               public webrtc::BitrateAllocatorObserver,
                               public webrtc::PacketFeedbackObserver {
  public:
   AudioSendStream(const webrtc::AudioSendStream::Config& config,
                   const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
                   rtc::TaskQueue* worker_queue,
-                  PacketRouter* packet_router,
-                  SendSideCongestionController* send_side_cc,
+                  RtpTransportControllerSendInterface* transport,
                   BitrateAllocator* bitrate_allocator,
                   RtcEventLog* event_log,
                   RtcpRttStats* rtcp_rtt_stats);
   ~AudioSendStream() override;
 
   // webrtc::AudioSendStream implementation.
   void Start() override;
   void Stop() override;
   bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
                           int duration_ms) override;
@@ skipping 23 matching lines @@
   bool SetupSendCodec();
 
   rtc::ThreadChecker worker_thread_checker_;
   rtc::ThreadChecker pacer_thread_checker_;
   rtc::TaskQueue* worker_queue_;
   const webrtc::AudioSendStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   BitrateAllocator* const bitrate_allocator_;
-  SendSideCongestionController* const send_side_cc_;
+  RtpTransportControllerSendInterface* const transport_;
   std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
 
   rtc::CriticalSection packet_loss_tracker_cs_;
   TransportFeedbackPacketLossTracker packet_loss_tracker_
       GUARDED_BY(&packet_loss_tracker_cs_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_