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

webrtc/audio/audio_send_stream.cc

 /*
  *  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.
  */
 
 #include "webrtc/audio/audio_send_stream.h"
 
 #include <string>
 
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/call/congestion_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_volume_control.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 std::string AudioSendStream::Config::Rtp::ToString() const {
   std::stringstream ss;
@@ skipping 18 matching lines @@
   // TODO(solenberg): Encoder config.
   ss << ", cng_payload_type: " << cng_payload_type;
   ss << ", red_payload_type: " << red_payload_type;
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
-    const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
+    const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+    CongestionController* congestion_controller)
     : config_(config), audio_state_(audio_state) {
   LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
+  RTC_DCHECK(congestion_controller);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
+  channel_proxy_->SetCongestionControlObjects(
+      congestion_controller->pacer(),
+      congestion_controller->GetTransportFeedbackObserver(),
+      congestion_controller->packet_router());
   channel_proxy_->SetRTCPStatus(true);
   channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
   channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
+
   for (const auto& extension : config.rtp.extensions) {
     if (extension.name == RtpExtension::kAbsSendTime) {
       channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
     } else if (extension.name == RtpExtension::kAudioLevel) {
       channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
+    } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
+      channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
 }
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
+  channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
 }
 
 void AudioSendStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 void AudioSendStream::Stop() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
@@ skipping 106 matching lines @@
 
 VoiceEngine* AudioSendStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc