Define RtpTransportControllerSendInterface.

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index c43d0da573f2c92963c31e83bc27605fb2d1e80d..5eb1f6762276430a90dd6193d434866994718272 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -19,6 +19,7 @@
 #include "webrtc/base/event.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/task_queue.h"
+#include "webrtc/call/rtp_transport_controller.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -43,8 +44,7 @@ AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     rtc::TaskQueue* worker_queue,
-    PacketRouter* packet_router,
-    CongestionController* congestion_controller,
+    RtpTransportControllerSenderInterface* transport,
     BitrateAllocator* bitrate_allocator,
     RtcEventLog* event_log,
     RtcpRttStats* rtcp_rtt_stats)
@@ -52,19 +52,18 @@ AudioSendStream::AudioSendStream(
       config_(config),
       audio_state_(audio_state),
       bitrate_allocator_(bitrate_allocator),
-      congestion_controller_(congestion_controller) {
+      transport_(transport) {
   LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
-  RTC_DCHECK(congestion_controller);
+  RTC_DCHECK(transport);
+  RTC_DCHECK(transport->congestion_controller());
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetRtcEventLog(event_log);
   channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
-  channel_proxy_->RegisterSenderCongestionControlObjects(
-      congestion_controller->pacer(),
-      congestion_controller->GetTransportFeedbackObserver(), packet_router);
+  channel_proxy_->RegisterSenderCongestionControlObjects(transport);
   channel_proxy_->SetRTCPStatus(true);
   channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
   channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
@@ -80,7 +79,7 @@ AudioSendStream::AudioSendStream(
       channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
     } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
-      congestion_controller->EnablePeriodicAlrProbing(true);
+      transport->congestion_controller()->EnablePeriodicAlrProbing(true);
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
@@ -256,7 +255,8 @@ const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
 
 void AudioSendStream::SetTransportOverhead(int transport_overhead_per_packet) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  congestion_controller_->SetTransportOverhead(transport_overhead_per_packet);
+  transport_->congestion_controller()->SetTransportOverhead(
+      transport_overhead_per_packet);
   channel_proxy_->SetTransportOverhead(transport_overhead_per_packet);
 }