Revert Make the new jitter buffer the default jitter buffer.

webrtc/video/rtp_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 86 matching lines @@
 
 class RtpStreamReceiverTest : public testing::Test {
  public:
   RtpStreamReceiverTest()
       : config_(CreateConfig()),
         timing_(Clock::GetRealTimeClock()),
         process_thread_(ProcessThread::Create("TestThread")) {}
 
   void SetUp() {
     rtp_stream_receiver_.reset(new RtpStreamReceiver(
-        &mock_transport_, nullptr, &packet_router_, nullptr, &config_,
-        nullptr, process_thread_.get(), &mock_nack_sender_,
-        &mock_key_frame_request_sender_, &mock_on_complete_frame_callback_,
-        &timing_));
+        nullptr, &mock_transport_, nullptr, &packet_router_,
+        nullptr, &config_, nullptr, process_thread_.get(),
+        &mock_nack_sender_, &mock_key_frame_request_sender_,
+        &mock_on_complete_frame_callback_, &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 169 matching lines @@
   mock_on_complete_frame_callback_.AppendExpectedBitstream(
       kH264StartCode, sizeof(kH264StartCode));
   mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
                                                            data.size());
   EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
   rtp_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
                                               &idr_packet);
 }
 
 }  // namespace webrtc