Revert: Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt

webrtc/pc/rtpsenderreceiver_unittest.cc

 /*
  *  Copyright 2012 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 106 matching lines @@
                                                    0, 1);
     params.codecs.push_back(kTelephoneEventCodec);
     voice_media_channel_->SetSendParameters(params);
   }
 
   void AddVideoTrack() { AddVideoTrack(false); }
 
   void AddVideoTrack(bool is_screencast) {
     rtc::scoped_refptr<VideoTrackSourceInterface> source(
         FakeVideoTrackSource::Create(is_screencast));
-    video_track_ =
-        VideoTrack::Create(kVideoTrackId, source, rtc::Thread::Current());
+    video_track_ = VideoTrack::Create(kVideoTrackId, source);
     EXPECT_TRUE(local_stream_->AddTrack(video_track_));
   }
 
   void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); }
 
   void CreateAudioRtpSender(rtc::scoped_refptr<LocalAudioSource> source) {
     audio_track_ = AudioTrack::Create(kAudioTrackId, source);
     EXPECT_TRUE(local_stream_->AddTrack(audio_track_));
     audio_rtp_sender_ =
         new AudioRtpSender(local_stream_->GetAudioTracks()[0],
@@ skipping 654 matching lines @@
 // Make sure the signal from "GetOnDestroyedSignal()" fires when the sender is
 // destroyed, which is needed for the DTMF sender.
 TEST_F(RtpSenderReceiverTest, TestOnDestroyedSignal) {
   CreateAudioRtpSender();
   EXPECT_FALSE(audio_sender_destroyed_signal_fired_);
   audio_rtp_sender_ = nullptr;
   EXPECT_TRUE(audio_sender_destroyed_signal_fired_);
 }
 
 }  // namespace webrtc