Reland of Make the new jitter buffer the default jitter buffer.

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 38 matching lines @@
       _codecDataBase(nullptr),
       pre_decode_image_callback_(pre_decode_image_callback),
       _receiveStatsTimer(1000, clock_),
       _retransmissionTimer(10, clock_),
       _keyRequestTimer(500, clock_) {}
 
 VideoReceiver::~VideoReceiver() {}
 
 void VideoReceiver::Process() {
   // Receive-side statistics
+
+  // TODO(philipel): Remove this if block when we know what to do with
+  //                 ReceiveStatisticsProxy::QualitySample.
   if (_receiveStatsTimer.TimeUntilProcess() == 0) {
     _receiveStatsTimer.Processed();
     rtc::CritScope cs(&process_crit_);
     if (_receiveStatsCallback != nullptr) {
-      uint32_t bitRate;
-      uint32_t frameRate;
-      _receiver.ReceiveStatistics(&bitRate, &frameRate);
-      _receiveStatsCallback->OnReceiveRatesUpdated(bitRate, frameRate);
-    }
-
-    if (_decoderTimingCallback != nullptr) {
-      int decode_ms;
-      int max_decode_ms;
-      int current_delay_ms;
-      int target_delay_ms;
-      int jitter_buffer_ms;
-      int min_playout_delay_ms;
-      int render_delay_ms;
-      if (_timing->GetTimings(&decode_ms, &max_decode_ms, &current_delay_ms,
-                              &target_delay_ms, &jitter_buffer_ms,
-                              &min_playout_delay_ms, &render_delay_ms)) {
-        _decoderTimingCallback->OnDecoderTiming(
-            decode_ms, max_decode_ms, current_delay_ms, target_delay_ms,
-            jitter_buffer_ms, min_playout_delay_ms, render_delay_ms);
-      }
+      _receiveStatsCallback->OnReceiveRatesUpdated(0, 0);
     }
   }
 
   // Key frame requests
   if (_keyRequestTimer.TimeUntilProcess() == 0) {
     _keyRequestTimer.Processed();
     bool request_key_frame = false;
     {
       rtc::CritScope cs(&process_crit_);
       request_key_frame = _scheduleKeyRequest && _frameTypeCallback != nullptr;
@@ skipping 191 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.
+// Used for the new jitter buffer.
 // 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_);
   if (pre_decode_image_callback_) {
     EncodedImage encoded_image(frame->EncodedImage());
     int qp = -1;
     if (qp_parser_.GetQp(*frame, &qp)) {
       encoded_image.qp_ = qp;
     }
@@ skipping 209 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