Call OnSentPacket for all packets sent in the test framework.

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_COMMON_CALL_TEST_H_
@@ skipping 28 matching lines @@
   static const uint8_t kRedPayloadType;
   static const uint8_t kRtxRedPayloadType;
   static const uint8_t kUlpfecPayloadType;
   static const uint32_t kSendRtxSsrcs[kNumSsrcs];
   static const uint32_t kSendSsrcs[kNumSsrcs];
   static const uint32_t kReceiverLocalSsrc;
   static const int kNackRtpHistoryMs;
   static const int kAbsSendTimeExtensionId;
 
  protected:
-  void RunBaseTest(BaseTest* test);
+  void RunBaseTest(BaseTest* test, const FakeNetworkPipe::Config& config);
 
   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_streams, Transport* send_transport);
   void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
 
   void CreateFrameGeneratorCapturer();
 
   void CreateStreams();
   void Start();
   void Stop();
   void DestroyStreams();
 
   Clock* const clock_;
 
   rtc::scoped_ptr<Call> sender_call_;
+  rtc::scoped_ptr<PacketTransport> send_transport_;
   VideoSendStream::Config send_config_;
   VideoEncoderConfig encoder_config_;
   VideoSendStream* send_stream_;
 
   rtc::scoped_ptr<Call> receiver_call_;
+  rtc::scoped_ptr<PacketTransport> receive_transport_;
   std::vector<VideoReceiveStream::Config> receive_configs_;
   std::vector<VideoReceiveStream*> receive_streams_;
 
   rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
   test::FakeEncoder fake_encoder_;
   ScopedVector<VideoDecoder> allocated_decoders_;
 };
 
 class BaseTest : public RtpRtcpObserver {
  public:
   explicit BaseTest(unsigned int timeout_ms);
-  BaseTest(unsigned int timeout_ms, const FakeNetworkPipe::Config& config);
   virtual ~BaseTest();
 
   virtual void PerformTest() = 0;
   virtual bool ShouldCreateReceivers() const = 0;
 
   virtual size_t GetNumStreams() const;
 
   virtual Call::Config GetSenderCallConfig();
   virtual Call::Config GetReceiverCallConfig();
   virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
 
   virtual void ModifyConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config);
   virtual void OnStreamsCreated(
       VideoSendStream* send_stream,
       const std::vector<VideoReceiveStream*>& receive_streams);
 
   virtual void OnFrameGeneratorCapturerCreated(
       FrameGeneratorCapturer* frame_generator_capturer);
 };
 
 class SendTest : public BaseTest {
  public:
   explicit SendTest(unsigned int timeout_ms);
-  SendTest(unsigned int timeout_ms, const FakeNetworkPipe::Config& config);
 
   bool ShouldCreateReceivers() const override;
 };
 
 class EndToEndTest : public BaseTest {
  public:
   explicit EndToEndTest(unsigned int timeout_ms);
-  EndToEndTest(unsigned int timeout_ms, const FakeNetworkPipe::Config& config);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_COMMON_CALL_TEST_H_