Add FlexFEC to CallTest.

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 30 matching lines @@
   static const int kDefaultHeight = 180;
   static const int kDefaultFramerate = 30;
   static const int kDefaultTimeoutMs;
   static const int kLongTimeoutMs;
   static const uint8_t kVideoSendPayloadType;
   static const uint8_t kSendRtxPayloadType;
   static const uint8_t kFakeVideoSendPayloadType;
   static const uint8_t kRedPayloadType;
   static const uint8_t kRtxRedPayloadType;
   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 uint32_t kReceiverLocalFlexfecSsrc;
   static const int kNackRtpHistoryMs;
 
  protected:
   // 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();
 
   void CreateSendConfig(size_t num_video_streams,
                         size_t num_audio_streams,
+                        size_t num_flexfec_streams,
                         Transport* send_transport);
   void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
 
   void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
                                              float speed,
                                              int framerate,
                                              int width,
                                              int height);
   void CreateFrameGeneratorCapturer(int framerate, int width, int height);
   void CreateFakeAudioDevices();
 
   void CreateVideoStreams();
   void CreateAudioStreams();
+  void CreateFlexfecStreams();
   void Start();
   void Stop();
   void DestroyStreams();
   void SetFakeVideoCaptureRotation(VideoRotation rotation);
 
   Clock* const clock_;
 
   webrtc::RtcEventLogNullImpl event_log_;
   std::unique_ptr<Call> sender_call_;
   std::unique_ptr<PacketTransport> send_transport_;
   VideoSendStream::Config video_send_config_;
   VideoEncoderConfig video_encoder_config_;
   VideoSendStream* video_send_stream_;
   AudioSendStream::Config audio_send_config_;
   AudioSendStream* audio_send_stream_;
 
   std::unique_ptr<Call> receiver_call_;
   std::unique_ptr<PacketTransport> receive_transport_;
   std::vector<VideoReceiveStream::Config> video_receive_configs_;
   std::vector<VideoReceiveStream*> video_receive_streams_;
   std::vector<AudioReceiveStream::Config> audio_receive_configs_;
   std::vector<AudioReceiveStream*> audio_receive_streams_;
+  std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
+  std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;
 
   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_;
 
  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),
@@ skipping 19 matching lines @@
 class BaseTest : public RtpRtcpObserver {
  public:
   explicit BaseTest(unsigned int timeout_ms);
   virtual ~BaseTest();
 
   virtual void PerformTest() = 0;
   virtual bool ShouldCreateReceivers() const = 0;
 
   virtual size_t GetNumVideoStreams() const;
   virtual size_t GetNumAudioStreams() const;
+  virtual size_t GetNumFlexfecStreams() const;
 
   virtual Call::Config GetSenderCallConfig();
   virtual Call::Config GetReceiverCallConfig();
   virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
 
   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);
   virtual void OnVideoStreamsCreated(
       VideoSendStream* send_stream,
       const std::vector<VideoReceiveStream*>& receive_streams);
 
   virtual void ModifyAudioConfigs(
       AudioSendStream::Config* send_config,
       std::vector<AudioReceiveStream::Config>* receive_configs);
   virtual void OnAudioStreamsCreated(
       AudioSendStream* send_stream,
       const std::vector<AudioReceiveStream*>& receive_streams);
 
+  virtual void ModifyFlexfecConfigs(
+      std::vector<FlexfecReceiveStream::Config>* receive_configs);
+  virtual void OnFlexfecStreamsCreated(
+      const std::vector<FlexfecReceiveStream*>& receive_streams);
+
   virtual void OnFrameGeneratorCapturerCreated(
       FrameGeneratorCapturer* frame_generator_capturer);
 
   webrtc::RtcEventLogNullImpl event_log_;
 };
 
 class SendTest : public BaseTest {
  public:
   explicit SendTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 class EndToEndTest : public BaseTest {
  public:
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_