Initial implementation of RtpTransportControllerReceive and related interfaces.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 901 matching lines @@
   // TODO(solenberg): Some test cases in EndToEndTest use this from a different
   // thread. Re-enable once that is fixed.
   // RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
   Stats stats;
   // Fetch available send/receive bitrates.
   uint32_t send_bandwidth = 0;
   transport_send_->send_side_cc()->GetBitrateController()->AvailableBandwidth(
       &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;
   stats.recv_bandwidth_bps = recv_bandwidth;
   stats.pacer_delay_ms =
       transport_send_->send_side_cc()->GetPacerQueuingDelayMs();
   stats.rtt_ms = call_stats_->rtcp_rtt_stats()->LastProcessedRtt();
   {
     rtc::CritScope cs(&bitrate_crit_);
     stats.max_padding_bitrate_bps = configured_max_padding_bitrate_bps_;
@@ skipping 503 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc