Add histogram stats for AV sync stream offset:

webrtc/system_wrappers/source/rtp_to_ntp.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 77 matching lines @@
   return true;
 }
 
 // Converts |rtp_timestamp| to the NTP time base using the NTP and RTP timestamp
 // pairs in |rtcp|. The converted timestamp is returned in
 // |rtp_timestamp_in_ms|. This function compensates for wrap arounds in RTP
 // timestamps and returns false if it can't do the conversion due to reordering.
 bool RtpToNtpMs(int64_t rtp_timestamp,
                 const RtcpList& rtcp,
                 int64_t* rtp_timestamp_in_ms) {
-  assert(rtcp.size() == 2);
+  if (rtcp.size() != 2)
+    return false;
+
   int64_t rtcp_ntp_ms_new = Clock::NtpToMs(rtcp.front().ntp_secs,
                                            rtcp.front().ntp_frac);
   int64_t rtcp_ntp_ms_old = Clock::NtpToMs(rtcp.back().ntp_secs,
                                            rtcp.back().ntp_frac);
   int64_t rtcp_timestamp_new = rtcp.front().rtp_timestamp;
   int64_t rtcp_timestamp_old = rtcp.back().rtp_timestamp;
   if (!CompensateForWrapAround(rtcp_timestamp_new,
                                rtcp_timestamp_old,
                                &rtcp_timestamp_new)) {
     return false;
@@ skipping 32 matching lines @@
     }
   } else if (static_cast<int32_t>(old_timestamp - new_timestamp) > 0) {
     // This difference should be less than -2^31 if we have had a backward wrap
     // around. Since it is cast to a int32_t, it should be positive.
     return -1;
   }
   return 0;
 }
 
 }  // namespace webrtc