Add framerate to VideoSinkWants and ability to signal on overuse

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 154 matching lines @@
       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(1, 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(),
-          VideoSendStream::DegradationPreference::kBalanced);
+          VideoSendStream::DegradationPreference::kMaintainFramerate);

[stefan-webrtc, 2017/03/21 15:37:36]

Isn't balanced a better default, or why do we change it?

[sprang_webrtc, 2017/03/21 15:59:18]

Balanced was never balanced. It was maintain-framerate (ie scale down on overuse
detection). I've thus renamed it everywhere. When we introduce a balanced mode
we will probably want to evaluate it before making it the default.

       send_stream_->Start();
       frame_generator_capturer_->Start();
 
       VideoReceiveStream::Decoder decoder;
       decoder.decoder = &fake_decoder_;
       decoder.payload_type =
           test_->video_send_config_.encoder_settings.payload_type;
       decoder.payload_name =
           test_->video_send_config_.encoder_settings.payload_name;
       test_->receive_config_.decoders.clear();
@@ skipping 111 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));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc