Reland of Use RtxReceiveStream.

webrtc/video/rtp_video_stream_receiver_unittest.cc

 /*
  *  Copyright 2017 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.
  */
 
@@ skipping 108 matching lines @@
 }  // namespace
 
 class RtpVideoStreamReceiverTest : public testing::Test {
  public:
   RtpVideoStreamReceiverTest()
       : config_(CreateConfig()),
         timing_(Clock::GetRealTimeClock()),
         process_thread_(ProcessThread::Create("TestThread")) {}
 
   void SetUp() {
-    rtp_video_stream_receiver_.reset(new RtpVideoStreamReceiver(
+    rtp_receive_statistics_ =
+        rtc::WrapUnique(ReceiveStatistics::Create(Clock::GetRealTimeClock()));
+    rtp_video_stream_receiver_ = rtc::MakeUnique<RtpVideoStreamReceiver>(
         &mock_transport_, nullptr, &packet_router_, &config_,
-        nullptr, process_thread_.get(), &mock_nack_sender_,
+        rtp_receive_statistics_.get(), nullptr, process_thread_.get(),
+        &mock_nack_sender_,
         &mock_key_frame_request_sender_, &mock_on_complete_frame_callback_,
-        &timing_));
+        &timing_);
   }
 
   WebRtcRTPHeader GetDefaultPacket() {
     WebRtcRTPHeader packet;
     memset(&packet, 0, sizeof(packet));
     packet.type.Video.codec = kRtpVideoH264;
     return packet;
   }
 
   // TODO(Johan): refactor h264_sps_pps_tracker_unittests.cc to avoid duplicate
@@ skipping 45 matching lines @@
   webrtc::test::ScopedFieldTrials override_field_trials_{
       kNewJitterBufferFieldTrialEnabled};
   VideoReceiveStream::Config config_;
   MockNackSender mock_nack_sender_;
   MockKeyFrameRequestSender mock_key_frame_request_sender_;
   MockTransport mock_transport_;
   MockOnCompleteFrameCallback mock_on_complete_frame_callback_;
   PacketRouter packet_router_;
   VCMTiming timing_;
   std::unique_ptr<ProcessThread> process_thread_;
+  std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
   std::unique_ptr<RtpVideoStreamReceiver> rtp_video_stream_receiver_;
 };
 
 TEST_F(RtpVideoStreamReceiverTest, GenericKeyFrame) {
   WebRtcRTPHeader rtp_header;
   const std::vector<uint8_t> data({1, 2, 3, 4});
   memset(&rtp_header, 0, sizeof(rtp_header));
   rtp_header.header.sequenceNumber = 1;
   rtp_header.header.markerBit = 1;
   rtp_header.type.Video.is_first_packet_in_frame = true;
@@ skipping 243 matching lines @@
   rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink);
   EXPECT_DEATH(rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink),
                "");
 
   // Test tear-down.
   rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink);
 }
 #endif
 
 }  // namespace webrtc