Let ViEEncoder express resolution requests as Sinkwants

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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/video/video_quality_test.h"
@@ skipping 1194 matching lines @@
   for (auto& config : video_receive_configs_)
     config.pre_decode_callback = &analyzer;
   RTC_DCHECK(!video_send_config_.post_encode_callback);
   video_send_config_.post_encode_callback = analyzer.encode_timing_proxy();
 
   if (params_.screenshare.enabled)
     SetupScreenshare();
 
   CreateVideoStreams();
   analyzer.SetSendStream(video_send_stream_);
-  video_send_stream_->SetSource(analyzer.OutputInterface());
+  video_send_stream_->SetSource(analyzer.OutputInterface(), false);
 
   CreateCapturer();
   rtc::VideoSinkWants wants;
   capturer_->AddOrUpdateSink(analyzer.InputInterface(), wants);
 
   StartEncodedFrameLogs(video_send_stream_);
   StartEncodedFrameLogs(video_receive_streams_[0]);
   video_send_stream_->Start();
   for (VideoReceiveStream* receive_stream : video_receive_streams_)
     receive_stream->Start();
@@ skipping 79 matching lines @@
   }
 
   if (params_.screenshare.enabled)
     SetupScreenshare();
 
   video_send_stream_ = call->CreateVideoSendStream(
       video_send_config_.Copy(), video_encoder_config_.Copy());
   VideoReceiveStream* video_receive_stream =
       call->CreateVideoReceiveStream(video_receive_configs_[stream_id].Copy());
   CreateCapturer();
-  video_send_stream_->SetSource(capturer_.get());
+  video_send_stream_->SetSource(capturer_.get(), false);
 
   AudioReceiveStream* audio_receive_stream = nullptr;
   if (params_.audio) {
     audio_send_config_ = AudioSendStream::Config(&transport);
     audio_send_config_.voe_channel_id = voe.send_channel_id;
     audio_send_config_.rtp.ssrc = kAudioSendSsrc;
 
     // Add extension to enable audio send side BWE, and allow audio bit rate
     // adaptation.
     audio_send_config_.rtp.extensions.clear();
@@ skipping 93 matching lines @@
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.common.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 
 }  // namespace webrtc