New jitter buffer experiment.

webrtc/modules/video_coding/rtp_frame_reference_finder.h

 /*
  *  Copyright (c) 2016 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 39 matching lines @@
   void ManageFrame(std::unique_ptr<RtpFrameObject> frame);
 
   // Notifies that padding has been received, which the reference finder
   // might need to calculate the references of a frame.
   void PaddingReceived(uint16_t seq_num);
 
   // Clear all stashed frames that include packets older than |seq_num|.
   void ClearTo(uint16_t seq_num);
 
  private:
-  static const uint16_t kPicIdLength = 1 << 7;
+  static const uint16_t kPicIdLength = 1 << 15;
   static const uint8_t kMaxTemporalLayers = 5;
-  static const int kMaxLayerInfo = 10;
-  static const int kMaxStashedFrames = 10;
-  static const int kMaxNotYetReceivedFrames = 20;
-  static const int kMaxGofSaved = 15;
+  static const int kMaxLayerInfo = 50;

[philipel, 2016/11/08 09:27:13]

In general we threw away info to early causing loss on a medium/high rtt link to
result in a keyframe request (since we couldn't continue the stream even if we
got the lost packet). Fixed by simply storing more history.

[stefan-webrtc, 2016/11/08 10:41:34]

Acknowledged.

+  static const int kMaxStashedFrames = 50;
+  static const int kMaxNotYetReceivedFrames = 100;
+  static const int kMaxGofSaved = 50;
   static const int kMaxPaddingAge = 100;
 
 
   struct GofInfo {
     GofInfo(GofInfoVP9* gof, uint16_t last_picture_id)
         : gof(gof), last_picture_id(last_picture_id) {}
     GofInfoVP9* gof;
     uint16_t last_picture_id;
   };
 
@@ skipping 119 matching lines @@
   // |cleared_to_seq_num_|.
   int cleared_to_seq_num_ GUARDED_BY(crit_);
 
   OnCompleteFrameCallback* frame_callback_;
 };
 
 }  // namespace video_coding
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_RTP_FRAME_REFERENCE_FINDER_H_