Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 9be5d2ee70fc78d5373a6944cbf5b5fb6006964c..6e620b6a6eda0889441f43811f32e6d37db7e12b 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -1172,8 +1172,6 @@
                                                 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.
@@ -1187,7 +1185,7 @@
 
   uint32_t ssrc = parsed_packet->Ssrc();
 
-  if (media_type == MediaType::AUDIO) {
+  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));
@@ -1197,7 +1195,7 @@
       return DELIVERY_OK;
     }
   }
-  if (media_type == MediaType::VIDEO) {
+  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));
@@ -1213,7 +1211,7 @@
       return DELIVERY_OK;
     }
   }
-  if (media_type == MediaType::VIDEO) {
+  if (media_type == MediaType::ANY || 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));