Add send-side BWE to WebRtcVoiceEngine under a finch experiment.

webrtc/audio/audio_receive_stream.cc

Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index ebe271367b6a9ec4d7a2e56c71e00cfefdea0d5f..9d84f9a4eb75bf9b99854c65e85b32281140799c 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -73,8 +73,6 @@ std::string AudioReceiveStream::Config::ToString() const {
   if (!sync_group.empty()) {
     ss << ", sync_group: " << sync_group;
   }
-  ss << ", combined_audio_video_bwe: "
-     << (combined_audio_video_bwe ? "true" : "false");
   ss << '}';
   return ss.str();
 }
@@ -119,15 +117,9 @@ AudioReceiveStream::AudioReceiveStream(
   // Configure bandwidth estimation.
   channel_proxy_->SetCongestionControlObjects(
       nullptr, nullptr, congestion_controller->packet_router());
-  if (config.combined_audio_video_bwe) {
-    if (UseSendSideBwe(config)) {
-      remote_bitrate_estimator_ =
-          congestion_controller->GetRemoteBitrateEstimator(true);
-    } else {
-      remote_bitrate_estimator_ =
-          congestion_controller->GetRemoteBitrateEstimator(false);
-    }
-    RTC_DCHECK(remote_bitrate_estimator_);
+  if (UseSendSideBwe(config)) {
+    remote_bitrate_estimator_ =
+        congestion_controller->GetRemoteBitrateEstimator(true);

[the sun, 2016/01/22 10:45:16]

The logic here has changed - is there no need to get the RBE in case
!UseSendSideBwe()? Sorry for being lazy; I'll look into more detail later, just
wanted to bring it up in case it is connected to the issue with the failing
peerconnection_unittests...

[stefan-webrtc, 2016/01/25 10:21:22]

No there is no need for that, it should only be used if UseSendSideBwe(config)
returns true.

   }
 }