Parse FlexFEC RTP headers in Call and add integration with BWE.

webrtc/modules/congestion_controller/include/congestion_controller.h

Index: webrtc/modules/congestion_controller/include/congestion_controller.h
diff --git a/webrtc/modules/congestion_controller/include/congestion_controller.h b/webrtc/modules/congestion_controller/include/congestion_controller.h
index f2bec76f2523c175b90577f24c72f3a1c56040c2..58f012770d1ba8109f4dab2e007f874d150dc5ed 100644
--- a/webrtc/modules/congestion_controller/include/congestion_controller.h
+++ b/webrtc/modules/congestion_controller/include/congestion_controller.h
@@ -36,6 +36,7 @@ class RateLimiter;
 class RemoteBitrateEstimator;
 class RemoteBitrateObserver;
 class RtcEventLog;
+class RtpHeaderExtensionMap;
 class TransportFeedbackObserver;
 
 class CongestionController : public CallStatsObserver, public Module {
@@ -100,6 +101,11 @@ class CongestionController : public CallStatsObserver, public Module {
   RateLimiter* GetRetransmissionRateLimiter();
   void EnablePeriodicAlrProbing(bool enable);
 
+  // Are the necessary RTCP and RTP header extensions enabled for send-side BWE?
+  static bool UseSendSideBwe(
+      bool transport_cc,

[danilchap, 2016/12/06 15:04:30]

what does transport_cc mean?

[brandtr, 2016/12/12 13:51:07]

Before this comment, I actually didn't know. Now I know that it stands for
"transport-wide congestion control". The variable name |transport_cc| is used in
the configs of AudioReceiveStream and VideoReceiveStream, so I just reused the
same here.

+      const RtpHeaderExtensionMap& rtp_header_extensions);
+
   // SetAllocatedSendBitrateLimits sets bitrates limits imposed by send codec
   // settings.
   // |min_send_bitrate_bps| is the total minimum send bitrate required by all