Let ViEEncoder express resolution requests as Sinkwants

webrtc/call/bitrate_estimator_tests.cc

 /*
  *  Copyright (c) 2013 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 <functional>
@@ skipping 160 matching lines @@
           fake_decoder_() {
       test_->video_send_config_.rtp.ssrcs[0]++;
       test_->video_send_config_.encoder_settings.encoder = &fake_encoder_;
       send_stream_ = test_->sender_call_->CreateVideoSendStream(
           test_->video_send_config_.Copy(),
           test_->video_encoder_config_.Copy());
       RTC_DCHECK_EQ(1u, test_->video_encoder_config_.number_of_streams);
       frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
           kDefaultWidth, kDefaultHeight, kDefaultFramerate,
           Clock::GetRealTimeClock()));
-      send_stream_->SetSource(frame_generator_capturer_.get());
+      send_stream_->SetSource(frame_generator_capturer_.get(), false);

[kthelgason, 2016/10/18 08:31:20]

I personally really dislike boolean arguments like this. There is no way to know
what the "false" means but looking up the definition of the function. IMO this
would be a lot clearer with a boolean enum, something like:

enum CpuAdaptation : bool {
    Off = true,
    On = false,
  };

Then this call would become:
SetSource(capturer, CpuAdaptation::On);

So much clearer. Whether it's worth the hassle on the other hand...

[perkj_webrtc, 2016/10/26 16:40:16]

ok- I agree, I added an enum that match the current spec and hope we can later
implement all aspects of it.  For now it is translated in webrtcvideoengine2.

       send_stream_->Start();
       frame_generator_capturer_->Start();
 
       if (receive_audio) {
         AudioReceiveStream::Config receive_config;
         receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0];
         // Bogus non-default id to prevent hitting a RTC_DCHECK when creating
         // the AudioReceiveStream. Every receive stream has to correspond to
         // an underlying channel id.
         receive_config.voe_channel_id = 0;
@@ skipping 131 matching lines @@
       RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc