Add statistics gathering for packet loss.

webrtc/modules/rtp_rtcp/interface/rtp_rtcp.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 421 matching lines @@
         uint32_t* packetsSent) const = 0;
 
     /*
     *   Get send statistics for the RTP and RTX stream.
     */
     virtual void GetSendStreamDataCounters(
         StreamDataCounters* rtp_counters,
         StreamDataCounters* rtx_counters) const = 0;
 
     /*
+     *  Get packet loss statistics for the RTP stream.
+     */
+    virtual void GetRtpPacketLossStats(
+        struct RtpPacketLossStats* incoming_loss,
+        struct RtpPacketLossStats* outgoing_loss) const = 0;
+
+    /*
     *   Get received RTCP sender info
     *
     *   return -1 on failure else 0
     */
     virtual int32_t RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0;
 
     /*
     *   Get received RTCP report block
     *
     *   return -1 on failure else 0
@@ skipping 186 matching lines @@
 
     /*
     *   send a request for a keyframe
     *
     *   return -1 on failure else 0
     */
     virtual int32_t RequestKeyFrame() = 0;
 };
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_