Replace interface VideoCapturerInput with VideoSinkInterface.

webrtc/test/call_test.cc

 /*
  *  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.
  */
 #include "webrtc/base/checks.h"
@@ skipping 222 matching lines @@
     audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
     audio_config.decoder_factory = decoder_factory_;
     audio_receive_configs_.push_back(audio_config);
   }
 }
 
 void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
                                                      float speed) {
   VideoStream stream = video_encoder_config_.streams.back();
   frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
-      video_send_stream_->Input(), stream.width, stream.height,
-      stream.max_framerate * speed, clock));
+      stream.width, stream.height, stream.max_framerate * speed, clock));
+  video_send_stream_->SetSource(frame_generator_capturer_.get());
 }
 
 void CallTest::CreateFrameGeneratorCapturer() {
   VideoStream stream = video_encoder_config_.streams.back();
   frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
-      video_send_stream_->Input(), stream.width, stream.height,
-      stream.max_framerate, clock_));
+      stream.width, stream.height, stream.max_framerate, clock_));
+  video_send_stream_->SetSource(frame_generator_capturer_.get());
 }
 
 void CallTest::CreateFakeAudioDevices() {
   fake_send_audio_device_.reset(new FakeAudioDevice(
       clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"),
       DriftingClock::kNoDrift));
   fake_recv_audio_device_.reset(new FakeAudioDevice(
       clock_, test::ResourcePath("voice_engine/audio_long16", "pcm"),
       DriftingClock::kNoDrift));
 }
@@ skipping 168 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc