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 438d1cc78a5aca5d7657b6368bfbac03fa5aed8e..cb6594aed8f5461a5b95a74bc549293499bd2b39 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/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
@@ -44,8 +45,7 @@ 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,
+    RtpTransportControllerSendInterface* transport,
     BitrateAllocator* bitrate_allocator,
     RtcEventLog* event_log,
     RtcpRttStats* rtcp_rtt_stats)
@@ -53,11 +53,12 @@ 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);
@@ -78,16 +79,16 @@ AudioSendStream::AudioSendStream(
       channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
     } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
-      congestion_controller->EnablePeriodicAlrProbing(true);
-      bandwidth_observer_.reset(congestion_controller->GetBitrateController()
+      transport->congestion_controller()->EnablePeriodicAlrProbing(true);
+      bandwidth_observer_.reset(transport->congestion_controller()
+                                    ->GetBitrateController()
                                     ->CreateRtcpBandwidthObserver());
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
   channel_proxy_->RegisterSenderCongestionControlObjects(
-      congestion_controller->pacer(), congestion_controller, packet_router,
-      bandwidth_observer_.get());
+      transport, bandwidth_observer_.get());
   if (!SetupSendCodec()) {
     LOG(LS_ERROR) << "Failed to set up send codec state.";
   }
@@ -254,7 +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);
 }