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/base/thread_checker.h"
+#include "webrtc/call/congestion_controller.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/pacing/paced_sender.h"
+#include "webrtc/modules/pacing/packet_router.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 283 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 SetSendTransportSequenceNumber(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 115 matching lines @@
       associate_send_channel_ = channel;
     }
 
     // Disassociate a send channel if it was associated.
     void DisassociateSendChannel(int channel_id);
 
 protected:
     void OnIncomingFractionLoss(int fraction_lost);
 
 private:
+ RtpRtcp* CreateRtpRtcp(
+     RtpPacketSender* packet_sender,
+     TransportSequenceNumberAllocator* sequence_number_allocator,
+     TransportFeedbackObserver* transport_feedback_callback);
     bool ReceivePacket(const uint8_t* packet, size_t packet_length,
                        const RTPHeader& header, bool in_order);
     bool HandleRtxPacket(const uint8_t* packet,
                          size_t packet_length,
                          const RTPHeader& header);
     bool IsPacketInOrder(const RTPHeader& header) const;
     bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
     int ResendPackets(const uint16_t* sequence_numbers, int length);
     int InsertInbandDtmfTone();
     int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
@@ skipping 108 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_);
+
+    PacketRouter* packet_router_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_