Add framerate to VideoSinkWants and ability to signal on overuse

webrtc/media/engine/fakewebrtccall.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.
  */
 
@@ skipping 90 matching lines @@
   gain_ = gain;
 }
 
 FakeVideoSendStream::FakeVideoSendStream(
     webrtc::VideoSendStream::Config config,
     webrtc::VideoEncoderConfig encoder_config)
     : sending_(false),
       config_(std::move(config)),
       codec_settings_set_(false),
       resolution_scaling_enabled_(false),
+      framerate_scaling_enalbed_(false),

[nisse-webrtc, 2017/03/14 09:00:28]

Spelling.

[sprang_webrtc, 2017/03/14 14:15:02]

Done.

       source_(nullptr),
       num_swapped_frames_(0) {
   RTC_DCHECK(config.encoder_settings.encoder != NULL);
   ReconfigureVideoEncoder(std::move(encoder_config));
 }
 
 FakeVideoSendStream::~FakeVideoSendStream() {
   if (source_)
     source_->RemoveSink(this);
 }
@@ skipping 124 matching lines @@
 }
 
 void FakeVideoSendStream::SetSource(
     rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
     const webrtc::VideoSendStream::DegradationPreference&
         degradation_preference) {
   RTC_DCHECK(source != source_);
   if (source_)
     source_->RemoveSink(this);
   source_ = source;
-  resolution_scaling_enabled_ =
-      degradation_preference !=
-      webrtc::VideoSendStream::DegradationPreference::kMaintainResolution;
+  switch (degradation_preference) {

[nisse-webrtc, 2017/03/14 09:00:28]

This is only ever called once in the object's lifetime?

Otherwise, you'd need to assign both flags in all cases.

[sprang_webrtc, 2017/03/14 14:15:02]

Right, thanks. I misread this code.

+    case DegradationPreference::kMaintainFramerate:
+      resolution_scaling_enabled_ = true;
+      break;
+    case DegradationPreference::kMaintainResolution:
+      framerate_scaling_enalbed_ = true;
+      break;
+    case DegradationPreference::kBalanced:
+      resolution_scaling_enabled_ = true;
+      framerate_scaling_enalbed_ = true;
+      break;
+    case DegradationPreference::kDegradationDisabled:
+      // No scaling enabled.
+      break;
+  }
   if (source)
     source->AddOrUpdateSink(this, resolution_scaling_enabled_
                                       ? sink_wants_
                                       : rtc::VideoSinkWants());
 }
 
 void FakeVideoSendStream::InjectVideoSinkWants(
     const rtc::VideoSinkWants& wants) {
   sink_wants_ = wants;
   source_->AddOrUpdateSink(this, wants);
@@ skipping 58 matching lines @@
 // TODO(brandtr): Implement when the stats have been designed.
 webrtc::FlexfecReceiveStream::Stats FakeFlexfecReceiveStream::GetStats() const {
   return webrtc::FlexfecReceiveStream::Stats();
 }
 
 FakeCall::FakeCall(const webrtc::Call::Config& config)
     : config_(config),
       audio_network_state_(webrtc::kNetworkUp),
       video_network_state_(webrtc::kNetworkUp),
       num_created_send_streams_(0),
-      num_created_receive_streams_(0) {}
+      num_created_receive_streams_(0),
+      audio_transport_overhead_(0),
+      video_transport_overhead_(0) {}
 
 FakeCall::~FakeCall() {
   EXPECT_EQ(0u, video_send_streams_.size());
   EXPECT_EQ(0u, audio_send_streams_.size());
   EXPECT_EQ(0u, video_receive_streams_.size());
   EXPECT_EQ(0u, audio_receive_streams_.size());
 }
 
 webrtc::Call::Config FakeCall::GetConfig() const {
   return config_;
@@ skipping 250 matching lines @@
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
   if (sent_packet.packet_id >= 0) {
     last_sent_nonnegative_packet_id_ = sent_packet.packet_id;
   }
 }
 
 }  // namespace cricket