Revert of Add framerate to VideoSinkWants and ability to signal on overuse

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.
  */
 
@@ skipping 267 matching lines @@
 
 void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
                                                      float speed,
                                                      int framerate,
                                                      int width,
                                                      int height) {
   frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
       width, height, framerate * speed, clock));
   video_send_stream_->SetSource(
       frame_generator_capturer_.get(),
-      VideoSendStream::DegradationPreference::kMaintainFramerate);
+      VideoSendStream::DegradationPreference::kBalanced);
 }
 
 void CallTest::CreateFrameGeneratorCapturer(int framerate,
                                             int width,
                                             int height) {
   frame_generator_capturer_.reset(
       test::FrameGeneratorCapturer::Create(width, height, framerate, clock_));
   video_send_stream_->SetSource(
       frame_generator_capturer_.get(),
-      VideoSendStream::DegradationPreference::kMaintainFramerate);
+      VideoSendStream::DegradationPreference::kBalanced);
 }
 
 void CallTest::CreateFakeAudioDevices() {
   fake_send_audio_device_.reset(new FakeAudioDevice(
       FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000), nullptr, 1.f));
   fake_recv_audio_device_.reset(new FakeAudioDevice(
       nullptr, FakeAudioDevice::CreateDiscardRenderer(48000), 1.f));
 }
 
 void CallTest::CreateVideoStreams() {
@@ skipping 189 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc