Add statistics gathering for packet loss.

webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h

 /*
  *  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 336 matching lines @@
 class NullRtpAudioFeedback : public RtpAudioFeedback {
  public:
   virtual ~NullRtpAudioFeedback() {}
 
   void OnPlayTelephoneEvent(const int32_t id,
                             const uint8_t event,
                             const uint16_t lengthMs,
                             const uint8_t volume) override {}
 };
 
+// Statistics about packet loss for a single directional connection. All values
+// are totals since the connection initiated.
+struct RtpPacketLossStats {
+  // The number of packets lost in events where no adjacent packets were also
+  // lost.
+  uint64_t single_packet_loss_count;
+  // The number of events in which more than one adjacent packet was lost.
+  uint64_t multiple_packet_loss_event_count;
+  // The number of packets lost in events where more than one adjacent packet
+  // was lost.
+  uint64_t multiple_packet_loss_packet_count;
+};
+
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_