New jitter buffer experiment.

webrtc/modules/video_coding/video_receiver.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.
  */
 
@@ skipping 272 matching lines @@
     LOG(LS_INFO) << "Received first "
                  << (frame->Complete() ? "complete" : "incomplete")
                  << " decodable video frame";
   }
 
   const int32_t ret = Decode(*frame);
   _receiver.ReleaseFrame(frame);
   return ret;
 }
 
+// Used for the WebRTC-NewVideoJitterBuffer experiment.
+// TODO(philipel): Clean up among the Decode functions as we replace
+//                 VCMEncodedFrame with FrameObject.
+int32_t VideoReceiver::Decode(const webrtc::VCMEncodedFrame* frame) {
+  rtc::CritScope lock(&receive_crit_);
+  return Decode(*frame);
+}
+
 int32_t VideoReceiver::RequestSliceLossIndication(
     const uint64_t pictureID) const {
   TRACE_EVENT1("webrtc", "RequestSLI", "picture_id", pictureID);
   rtc::CritScope cs(&process_crit_);
   if (_frameTypeCallback != nullptr) {
     const int32_t ret =
         _frameTypeCallback->SliceLossIndicationRequest(pictureID);
     if (ret < 0) {
       return ret;
     }
@@ skipping 193 matching lines @@
   _receiver.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
                             max_incomplete_time_ms);
 }
 
 int VideoReceiver::SetMinReceiverDelay(int desired_delay_ms) {
   return _receiver.SetMinReceiverDelay(desired_delay_ms);
 }
 
 }  // namespace vcm
 }  // namespace webrtc