Making FakeNetworkPipe demux audio and video packets.

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 b96e5e6e112a7e831c08f15ea8b5eb2b87d95259..6d7d9c548202bbef6555eccbd11a56645594c327 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -57,25 +57,9 @@ class CallTest : public ::testing::Test {
   static const uint32_t kReceiverLocalVideoSsrc;
   static const uint32_t kReceiverLocalAudioSsrc;
   static const int kNackRtpHistoryMs;
+  static const std::map<uint8_t, MediaType> payload_type_map_;

[stefan-webrtc, 2017/04/10 07:45:26]

kPayloadTypeMap

[minyue-webrtc, 2017/04/10 07:58:46]

Ok, I would like it but I learnt that only POD const can be kCamelCase.

[stefan-webrtc, 2017/04/10 12:04:15]

Acknowledged.

 
  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 {
-   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.
@@ -141,9 +125,6 @@ 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
@@ -192,10 +173,6 @@ class BaseTest : public RtpRtcpObserver {
   virtual Call::Config GetReceiverCallConfig();
   virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
 
-  // Returns VIDEO for video-only tests, AUDIO for audio-only tests,
-  // and ANY for tests sending audio and video over the same
-  // transport.
-  virtual MediaType SelectMediaType();
   virtual test::PacketTransport* CreateSendTransport(Call* sender_call);
   virtual test::PacketTransport* CreateReceiveTransport();