Don't hardcode MediaType::ANY in FakeNetworkPipe.

webrtc/media/engine/fakewebrtccall.cc

 /*
  *  Copyright (c) 2015 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 502 matching lines @@
 webrtc::PacketReceiver* FakeCall::Receiver() {
   return this;
 }
 
 FakeCall::DeliveryStatus FakeCall::DeliverPacket(
     webrtc::MediaType media_type,
     const uint8_t* packet,
     size_t length,
     const webrtc::PacketTime& packet_time) {
   EXPECT_GE(length, 12u);
+  RTC_DCHECK(media_type == webrtc::MediaType::AUDIO ||
+             media_type == webrtc::MediaType::VIDEO);
+
   uint32_t ssrc;
   if (!GetRtpSsrc(packet, length, &ssrc))
     return DELIVERY_PACKET_ERROR;
 
-  if (media_type == webrtc::MediaType::ANY ||
-      media_type == webrtc::MediaType::VIDEO) {
+  if (media_type == webrtc::MediaType::VIDEO) {
     for (auto receiver : video_receive_streams_) {
       if (receiver->GetConfig().rtp.remote_ssrc == ssrc)
         return DELIVERY_OK;
     }
   }
-  if (media_type == webrtc::MediaType::ANY ||
-      media_type == webrtc::MediaType::AUDIO) {
+  if (media_type == webrtc::MediaType::AUDIO) {
     for (auto receiver : audio_receive_streams_) {
       if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
         receiver->DeliverRtp(packet, length, packet_time);
         return DELIVERY_OK;
       }
     }
   }
   return DELIVERY_UNKNOWN_SSRC;
 }
 
@@ skipping 51 matching lines @@
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
   if (sent_packet.packet_id >= 0) {
     last_sent_nonnegative_packet_id_ = sent_packet.packet_id;
   }
 }
 
 }  // namespace cricket