Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.

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 1193 matching lines @@
       if (status == DELIVERY_OK)
         event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
       return status;
     }
   }
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     auto it = video_receive_ssrcs_.find(ssrc);
     if (it != video_receive_ssrcs_.end()) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
-      // TODO(brandtr): Notify the BWE of received media packets here.
-      auto status = it->second->DeliverRtp(packet, length, packet_time)
+      auto status = it->second->OnRtpPacket(*parsed_packet)
                         ? DELIVERY_OK
                         : DELIVERY_PACKET_ERROR;
-      // Deliver media packets to FlexFEC subsystem. RTP header extensions need
-      // not be parsed, as FlexFEC is oblivious to the semantic meaning of the
-      // packet contents beyond the 12 byte RTP base header. The BWE is fed
-      // information about these media packets from the regular media pipeline.
+      // Deliver media packets to FlexFEC subsystem.
       if (parsed_packet) {
         auto it_bounds = flexfec_receive_ssrcs_media_.equal_range(ssrc);
         for (auto it = it_bounds.first; it != it_bounds.second; ++it)
           it->second->AddAndProcessReceivedPacket(*parsed_packet);
       }
       if (status == DELIVERY_OK)
         event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
       return status;
     }
   }
@@ skipping 65 matching lines @@
   if (media_type != MediaType::AUDIO ||
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     congestion_controller_->OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc