Don't hardcode MediaType::ANY in FakeNetworkPipe.

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 1154 matching lines @@
 
   return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                 const uint8_t* packet,
                                                 size_t length,
                                                 const PacketTime& packet_time) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtp");
 
+  RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO);
+
   ReadLockScoped read_lock(*receive_crit_);
   // TODO(nisse): We should parse the RTP header only here, and pass
   // on parsed_packet to the receive streams.
   rtc::Optional<RtpPacketReceived> parsed_packet =
       ParseRtpPacket(packet, length, packet_time);
 
   if (!parsed_packet)
     return DELIVERY_PACKET_ERROR;
 
   NotifyBweOfReceivedPacket(*parsed_packet, media_type);
 
   uint32_t ssrc = parsed_packet->Ssrc();
 
-  if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
+  if (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));
       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) {
+  if (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.
       auto it_bounds = flexfec_receive_ssrcs_media_.equal_range(ssrc);
       for (auto it = it_bounds.first; it != it_bounds.second; ++it)
         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) {
+  if (media_type == MediaType::VIDEO) {
     received_bytes_per_second_counter_.Add(static_cast<int>(length));
     // TODO(brandtr): Update here when FlexFEC supports protecting audio.
     received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
     auto it = flexfec_receive_ssrcs_protection_.find(ssrc);
     if (it != flexfec_receive_ssrcs_protection_.end()) {
       it->second->OnRtpPacket(*parsed_packet);
       event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
       return DELIVERY_OK;
     }
   }
@@ skipping 52 matching lines @@
   if (media_type != MediaType::AUDIO ||
       (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