Add statistics gathering for packet loss.

webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h

Index: webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
index 1587762763b636d5faf22e962af4f25fd728e97e..6fa1b0679689b4a682e049495fb9ce3b09ed0a71 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
@@ -354,5 +354,18 @@ class NullRtpAudioFeedback : public RtpAudioFeedback {
                             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_