Prepare the AudioSendStream to be hooked up to send-side BWE.

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.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
-#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/file_player.h"
 #include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/voice_engine/dtmf_inband.h"
 #include "webrtc/voice_engine/dtmf_inband_queue.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/level_indicator.h"
@@ skipping 10 matching lines @@
 
 class TimestampWrapAroundHandler;
 }
 
 namespace webrtc {
 
 class AudioDeviceModule;
 class Config;
 class CriticalSectionWrapper;
 class FileWrapper;
+class PacketRouter;
 class ProcessThread;
 class ReceiveStatistics;
 class RemoteNtpTimeEstimator;
 class RtcEventLog;
 class RTPPayloadRegistry;
 class RtpReceiver;
 class RTPReceiverAudio;
 class RtpRtcp;
 class TelephoneEventHandler;
 class VoEMediaProcess;
 class VoERTPObserver;
 class VoiceEngineObserver;
 
 struct CallStatistics;
 struct ReportBlock;
 struct SenderInfo;
 
 namespace voe {
 
 class OutputMixer;
 class Statistics;
 class StatisticsProxy;
 class TransmitMixer;
 class VoERtcpObserver;
 
+namespace {
+class TransportFeedbackProxy;
+class TransportSequenceNumberProxy;
+class PacketSenderProxy;
+}  // namespace
+
 // Helper class to simplify locking scheme for members that are accessed from
 // multiple threads.
 // Example: a member can be set on thread T1 and read by an internal audio
 // thread T2. Accessing the member via this class ensures that we are
 // safe and also avoid TSan v2 warnings.
 class ChannelState {
  public:
     struct State {
         State() : rx_apm_is_enabled(false),
                   input_external_media(false),
@@ skipping 228 matching lines @@
 #endif
 
     // 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);
     int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
     int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
+    void EnableSendTransportSequenceNumber(int id);
+
+    void SetCongestionControlObjects(
+        RtpPacketSender* rtp_packet_sender,
+        TransportFeedbackObserver* transport_feedback_observer,
+        PacketRouter* packet_router);
+
     void SetRTCPStatus(bool enable);
     int GetRTCPStatus(bool& enabled);
     int SetRTCP_CNAME(const char cName[256]);
     int GetRemoteRTCP_CNAME(char cName[256]);
     int GetRemoteRTCPData(unsigned int& NTPHigh, unsigned int& NTPLow,
                           unsigned int& timestamp,
                           unsigned int& playoutTimestamp, unsigned int* jitter,
                           unsigned short* fractionLost);
     int SendApplicationDefinedRTCPPacket(unsigned char subType,
                                          unsigned int name, const char* data,
@@ skipping 243 matching lines @@
     bool _RxVadDetection;
     bool _rxAgcIsEnabled;
     bool _rxNsIsEnabled;
     bool restored_packet_in_use_;
     // RtcpBandwidthObserver
     rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_;
     rtc::scoped_ptr<NetworkPredictor> network_predictor_;
     // An associated send channel.
     rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
     ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
+
+    bool pacing_enabled_;
+    PacketRouter* packet_router_ = nullptr;
+    rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
+    rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
+    rtc::scoped_ptr<PacketSenderProxy> packet_sender_proxy_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_