New jitter buffer experiment.

webrtc/modules/video_coding/timing.cc

 /*
  *  Copyright (c) 2011 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 136 matching lines @@
     if (frame_timestamp < 0x0000ffff && prev_frame_timestamp_ > 0xffff0000) {
       // wrap
       max_change_ms = kDelayMaxChangeMsPerS *
                       (frame_timestamp + (static_cast<int64_t>(1) << 32) -
                        prev_frame_timestamp_) /
                       90000;
     } else {
       max_change_ms = kDelayMaxChangeMsPerS *
                       (frame_timestamp - prev_frame_timestamp_) / 90000;
     }
+
     if (max_change_ms <= 0) {
       // Any changes less than 1 ms are truncated and
       // will be postponed. Negative change will be due
       // to reordering and should be ignored.
       return;
     }
     delay_diff_ms = std::max(delay_diff_ms, -max_change_ms);
     delay_diff_ms = std::min(delay_diff_ms, max_change_ms);
 
     current_delay_ms_ = current_delay_ms_ + delay_diff_ms;
@@ skipping 131 matching lines @@
   *decode_ms = last_decode_ms_;
   *max_decode_ms = RequiredDecodeTimeMs();
   *current_delay_ms = current_delay_ms_;
   *target_delay_ms = TargetDelayInternal();
   *jitter_buffer_ms = jitter_delay_ms_;
   *min_playout_delay_ms = min_playout_delay_ms_;
   *render_delay_ms = render_delay_ms_;
 }
 
 }  // namespace webrtc