commit bba9dec4d57d8f2b7a3b67a61e63cc33667a2b54
author stefan <stefan@webrtc.org> Mon Feb 01 12:39:55 2016
committer Commit bot <commit-bot@chromium.org> Mon Feb 01 12:40:04 2016
tree 2e99c0e61305498ecf13c65eff60c0b4dcf96348
parent 1f611fa58ba3860dab2494c4fd9de778935bd376

Use separate rtp module lists for send and receive in PacketRouter.

This makes it possible to handle send and receive streams with the same SSRC, which is currently the case in some peer connection tests.

Also moves sending transport feedback to the pacer thread.

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1628683002

Cr-Commit-Position: refs/heads/master@{#11443}

webrtc/audio/audio_receive_stream.cc

diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index ebe2713..b731126 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -117,8 +117,8 @@
     }
   }
   // Configure bandwidth estimation.
-  channel_proxy_->SetCongestionControlObjects(
-      nullptr, nullptr, congestion_controller->packet_router());
+  channel_proxy_->RegisterReceiverCongestionControlObjects(
+      congestion_controller->packet_router());
   if (config.combined_audio_video_bwe) {
     if (UseSendSideBwe(config)) {
       remote_bitrate_estimator_ =
@@ -134,7 +134,7 @@
 AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
-  channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
+  channel_proxy_->ResetCongestionControlObjects();
   if (remote_bitrate_estimator_) {
     remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
   }

webrtc/audio/audio_receive_stream_unittest.cc

diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index b241bed..fd88fc8 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -94,13 +94,12 @@
           EXPECT_CALL(*channel_proxy_, EnableReceiveTransportSequenceNumber(
                                            kTransportSequenceNumberId))
               .Times(1);
-          EXPECT_CALL(*channel_proxy_, SetCongestionControlObjects(
-                                           nullptr, nullptr, &packet_router_))
+          EXPECT_CALL(*channel_proxy_,
+                      RegisterReceiverCongestionControlObjects(&packet_router_))
               .Times(1);
           EXPECT_CALL(congestion_controller_, packet_router())
               .WillOnce(Return(&packet_router_));
-          EXPECT_CALL(*channel_proxy_,
-                      SetCongestionControlObjects(nullptr, nullptr, nullptr))
+          EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
               .Times(1);
           return channel_proxy_;
         }));

webrtc/audio/audio_send_stream.cc

diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 35a6552..62121c6 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -68,7 +68,7 @@
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
-  channel_proxy_->SetCongestionControlObjects(
+  channel_proxy_->RegisterSenderCongestionControlObjects(
       congestion_controller->pacer(),
       congestion_controller->GetTransportFeedbackObserver(),
       congestion_controller->packet_router());
@@ -92,7 +92,7 @@
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
-  channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
+  channel_proxy_->ResetCongestionControlObjects();
 }
 
 void AudioSendStream::Start() {

webrtc/audio/audio_send_stream_unittest.cc

diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 466c157..a01ef02 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -83,13 +83,12 @@
                                            kTransportSequenceNumberId))
               .Times(1);
           EXPECT_CALL(*channel_proxy_,
-                      SetCongestionControlObjects(
+                      RegisterSenderCongestionControlObjects(
                           congestion_controller_.pacer(),
                           congestion_controller_.GetTransportFeedbackObserver(),
                           congestion_controller_.packet_router()))
               .Times(1);
-          EXPECT_CALL(*channel_proxy_,
-                      SetCongestionControlObjects(nullptr, nullptr, nullptr))
+          EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
               .Times(1);
           return channel_proxy_;
         }));