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

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 4dd952295667f7a231497b1202cd70ab10f5ed53..ab5c047b16d1758aa1293a6ecd7c7e32ef472e5e 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -17,6 +17,10 @@
 #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/pacing/packet_router.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"
@@ -55,7 +59,8 @@ std::string AudioSendStream::Config::ToString() const {
 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);
@@ -66,20 +71,33 @@ AudioSendStream::AudioSendStream(
   channel_proxy_->SetRTCPStatus(true);
   channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
   channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
+
+  TransportFeedbackObserver* transport_feedback_observer = nullptr;
+

[the sun, 2015/11/30 12:37:20]

RTC_DCHECK(congestion_controller);
since you dereference below.

[stefan-webrtc, 2015/11/30 15:22:02]

Done.

   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) {

[the sun, 2015/11/30 12:37:20]

You have 2x "if (extension.name == RtpExtension::kTransportSequenceNumber)"
A unit test should have caught this?

[stefan-webrtc, 2015/11/30 15:22:02]

Ah, merge problem. Adding the same test that you have added for the other
extensions.

+      channel_proxy_->SetSendTransportSequenceNumber(extension.id);
+    } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
+      channel_proxy_->SetSendTransportSequenceNumber(extension.id);
+      transport_feedback_observer =
+          congestion_controller->GetTransportFeedbackObserver();
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
+  channel_proxy_->SetCongestionControlObjects(

[the sun, 2015/11/30 12:37:20]

Would it be easier to just call this SetCongestionControl() and send in the cc*?

[stefan-webrtc, 2015/11/30 15:22:02]

It would be a bit more tricky since voice engine can't depend on the "webrtc"
target in which files in webrtc/call/ are included. I could break it out into a
separate lib, but I'm not sure it's really worth it.

[the sun, 2015/12/01 10:25:35]

No, likely not.

+      congestion_controller->pacer(), transport_feedback_observer,
+      congestion_controller->packet_router());
 }
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
+  channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
 }
 
 void AudioSendStream::Start() {