Initial implementation of RtpTransportControllerReceive and related interfaces.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 343b3ef41fbb54e2903c6420b2f43a6eef95ef1e..37ca23fcb1272bf65d6f95c47f52b3e69eb0aa9a 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -919,6 +919,9 @@ Call::Stats Call::GetStats() const {
       &send_bandwidth);
   std::vector<unsigned int> ssrcs;
   uint32_t recv_bandwidth = 0;
+
+  // TODO(nisse): Is this thread safe? Most access to |receive_side_cc_| is done
+  // via |*_rtp_transport_receive_|, and protected by |receive_crit_|.
   receive_side_cc_.GetRemoteBitrateEstimator(false)->LatestEstimate(
       &ssrcs, &recv_bandwidth);
   stats.send_bandwidth_bps = send_bandwidth;