Add histogram stats for AV sync stream offset:

webrtc/video/stream_synchronization.cc

 /*
  *  Copyright (c) 2012 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 42 matching lines @@
 
 StreamSynchronization::~StreamSynchronization() {
   delete channel_delay_;
 }
 
 bool StreamSynchronization::ComputeRelativeDelay(
     const Measurements& audio_measurement,
     const Measurements& video_measurement,
     int* relative_delay_ms) {
   assert(relative_delay_ms);
-  if (audio_measurement.rtcp.size() < 2 || video_measurement.rtcp.size() < 2) {
-    // We need two RTCP SR reports per stream to do synchronization.
-    return false;
-  }
   int64_t audio_last_capture_time_ms;
   if (!RtpToNtpMs(audio_measurement.latest_timestamp,
                   audio_measurement.rtcp,
                   &audio_last_capture_time_ms)) {
     return false;
   }
   int64_t video_last_capture_time_ms;
   if (!RtpToNtpMs(video_measurement.latest_timestamp,
                   video_measurement.rtcp,
                   &video_last_capture_time_ms)) {
@@ skipping 140 matching lines @@
       target_delay_ms - base_target_delay_ms_;
 
   channel_delay_->extra_video_delay_ms +=
       target_delay_ms - base_target_delay_ms_;
 
   // Video is already delayed by the desired amount.
   base_target_delay_ms_ = target_delay_ms;
 }
 
 }  // namespace webrtc