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

webrtc/test/call_test.h

Index: webrtc/test/call_test.h
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
index 78fbbfbe4372d98a8a3ad0172472723dda8a0786..b4f665821ee2ffa420a6465835a4dc310e8b3961 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -59,6 +59,23 @@ class CallTest : public ::testing::Test {
   static const int kNackRtpHistoryMs;
 
  protected:
+  // Needed for tests sending both audio and video on the same
+  // FakeNetworkPipe. We then need to set correct MediaType based on
+  // packet payload type, before passing the packet on to Call.
+  class PayloadDemuxer : public PacketReceiver {

[stefan-webrtc, 2017/03/29 12:27:25]

It seems like PacketReceiver is the wrong type for this, since we don't actually
know the MediaType of the packet at this point. Should it insetad inherit from
something else, e.g., webrtc::Transport?

[nisse-webrtc, 2017/03/29 12:37:55]

We can't, since this object is intended to be passed to
FakeNetworkPipe::SetReceiver.

[stefan-webrtc, 2017/03/30 06:47:11]

Ah, ok. I guess we could if we changed the FakeNetworkPipe? If you think that
makes sense maybe you could leave a TODO here?

[nisse-webrtc, 2017/03/30 06:56:45]

For now, I think it makes sense for most tests to configure FakeNetworkPipe with
a media type. If we do any refactorings there, it might make sense to make media
type be a per-packet property in 
FakeNetworkPipe. But it's unclear what changes would be needed at input to the
FakeNetworkPipe to get that right.

+   public:
+    PayloadDemuxer() = default;
+
+    void SetReceiver(PacketReceiver* receiver);
+    DeliveryStatus DeliverPacket(MediaType media_type,
+                                 const uint8_t* packet,
+                                 size_t length,
+                                 const PacketTime& packet_time) override;
+
+   private:
+    PacketReceiver* receiver_ = nullptr;
+  };
+
   // RunBaseTest overwrites the audio_state and the voice_engine of the send and
   // receive Call configs to simplify test code and avoid having old VoiceEngine
   // APIs in the tests.
@@ -124,6 +141,9 @@ class CallTest : public ::testing::Test {
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
   test::FakeVideoRenderer fake_renderer_;
 
+  PayloadDemuxer receive_demuxer_;
+  PayloadDemuxer send_demuxer_;
+
  private:
   // TODO(holmer): Remove once VoiceEngine is fully refactored to the new API.
   // These methods are used to set up legacy voice engines and channels which is
@@ -172,6 +192,7 @@ class BaseTest : public RtpRtcpObserver {
   virtual Call::Config GetReceiverCallConfig();
   virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
 
+  // The default implementation creates MediaType::VIDEO transports.
   virtual test::PacketTransport* CreateSendTransport(Call* sender_call);
   virtual test::PacketTransport* CreateReceiveTransport();