Replace AudioReceiveStream::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 1180 matching lines @@
 
   NotifyBweOfReceivedPacket(*parsed_packet, media_type);
 
   uint32_t ssrc = parsed_packet->Ssrc();
 
   if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
     auto it = audio_receive_ssrcs_.find(ssrc);
     if (it != audio_receive_ssrcs_.end()) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_audio_bytes_per_second_counter_.Add(static_cast<int>(length));
-      auto status = it->second->DeliverRtp(packet, length, packet_time)
-                        ? DELIVERY_OK
-                        : DELIVERY_PACKET_ERROR;
-      if (status == DELIVERY_OK)
-        event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
-      return status;
+      it->second->OnRtpPacket(*parsed_packet);
+      event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
+      return DELIVERY_OK;
     }
   }
   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));
       it->second->OnRtpPacket(*parsed_packet);
 
       // Deliver media packets to FlexFEC subsystem.
@@ skipping 71 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