NetEq jitter calculation now done in int64_t.

webrtc/modules/audio_coding/neteq/rtcp.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 29 matching lines @@
       cycles_++;
     }
     max_seq_no_ = rtp_header.sequenceNumber;
   }
 
   // Calculate jitter according to RFC 3550, and update previous timestamps.
   // Note that the value in |jitter_| is in Q4.
   if (received_packets_ > 1) {
     int32_t ts_diff = receive_timestamp - (rtp_header.timestamp - transit_);
     ts_diff = WEBRTC_SPL_ABS_W32(ts_diff);
-    int32_t jitter_diff = (ts_diff << 4) - jitter_;
+    uint64_t jitter_diff = (uint64_t{ts_diff} << 4) - jitter_;

[ossu, 2016/10/31 14:11:01]

Uh... I spoke too soon. Clearly this could be negative...

     // Calculate 15 * jitter_ / 16 + jitter_diff / 16 (with proper rounding).
     jitter_ = jitter_ + ((jitter_diff + 8) >> 4);
   }
   transit_ = rtp_header.timestamp - receive_timestamp;
 }
 
 void Rtcp::GetStatistics(bool no_reset, RtcpStatistics* stats) {
   // Extended highest sequence number received.
   stats->extended_max_sequence_number =
       (static_cast<int>(cycles_) << 16) + max_seq_no_;
@@ skipping 26 matching lines @@
   if (expected_since_last == 0 || lost <= 0 || received_packets_ == 0) {
     stats->fraction_lost = 0;
   } else {
     stats->fraction_lost = std::min(0xFFU, (lost << 8) / expected_since_last);
   }
 
   stats->jitter = jitter_ >> 4;  // Scaling from Q4.
 }
 
 }  // namespace webrtc