Making FakeNetworkPipe demux audio and video packets.

webrtc/test/call_test.h

 /*
  *  Copyright (c) 2014 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.
  */
 #ifndef WEBRTC_TEST_CALL_TEST_H_
@@ skipping 39 matching lines @@
   static const uint8_t kUlpfecPayloadType;
   static const uint8_t kFlexfecPayloadType;
   static const uint8_t kAudioSendPayloadType;
   static const uint32_t kSendRtxSsrcs[kNumSsrcs];
   static const uint32_t kVideoSendSsrcs[kNumSsrcs];
   static const uint32_t kAudioSendSsrc;
   static const uint32_t kFlexfecSendSsrc;
   static const uint32_t kReceiverLocalVideoSsrc;
   static const uint32_t kReceiverLocalAudioSsrc;
   static const int kNackRtpHistoryMs;
+  static const std::map<uint8_t, MediaType> payload_type_map_;
 
  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.
   void RunBaseTest(BaseTest* test);
 
   void CreateCalls(const Call::Config& sender_config,
                    const Call::Config& receiver_config);
   void CreateSenderCall(const Call::Config& config);
   void CreateReceiverCall(const Call::Config& config);
   void DestroyCalls();
@@ skipping 45 matching lines @@
 
   std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
   test::FakeEncoder fake_encoder_;
   std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders_;
   size_t num_video_streams_;
   size_t num_audio_streams_;
   size_t num_flexfec_streams_;
   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
   // necessary while voice engine is being refactored to the new stream API.
   struct VoiceEngineState {
     VoiceEngineState()
         : voice_engine(nullptr),
           base(nullptr),
           channel_id(-1) {}
 
@@ skipping 28 matching lines @@
 
   virtual std::unique_ptr<FakeAudioDevice::Capturer> CreateCapturer();
   virtual std::unique_ptr<FakeAudioDevice::Renderer> CreateRenderer();
   virtual void OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device,
                                          FakeAudioDevice* recv_audio_device);
 
   virtual Call::Config GetSenderCallConfig();
   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();
 
   virtual void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config);
   virtual void ModifyVideoCaptureStartResolution(int* width,
                                                  int* heigt,
                                                  int* frame_rate);
@@ skipping 33 matching lines @@
   EndToEndTest();
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_