Add histogram stats for AV sync stream offset:

webrtc/video_receive_stream.h

 /*
  *  Copyright (c) 2013 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.
  */
 
 #ifndef WEBRTC_VIDEO_RECEIVE_STREAM_H_
 #define WEBRTC_VIDEO_RECEIVE_STREAM_H_
 
+#include <limits.h>
+
 #include <map>
 #include <string>
 #include <vector>
 
 #include "webrtc/common_types.h"
 #include "webrtc/config.h"
 #include "webrtc/frame_callback.h"
 #include "webrtc/stream.h"
 #include "webrtc/transport.h"
 #include "webrtc/video_renderer.h"
@@ skipping 35 matching lines @@
     int target_delay_ms = 0;
     int jitter_buffer_ms = 0;
     int min_playout_delay_ms = 0;
     int render_delay_ms = 10;
 
     int current_payload_type = -1;
 
     int total_bitrate_bps = 0;
     int discarded_packets = 0;
 
+    int sync_offset_ms = INT_MIN;

[stefan-webrtc, 2016/03/09 15:59:30]

std::numeric_limits<int>::max()

[åsapersson, 2016/03/10 09:08:35]

Done.

[stefan-webrtc, 2016/03/10 14:25:09]

Or min if you wanted that. I didn't see it was MIN before.

+
     uint32_t ssrc = 0;
     std::string c_name;
     StreamDataCounters rtp_stats;
     RtcpPacketTypeCounter rtcp_packet_type_counts;
     RtcpStatistics rtcp_stats;
   };
 
   struct Config {
     Config() = delete;
     explicit Config(Transport* rtcp_send_transport)
@@ skipping 90 matching lines @@
     int target_delay_ms = 0;
   };
 
   // TODO(pbos): Add info on currently-received codec to Stats.
   virtual Stats GetStats() const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RECEIVE_STREAM_H_