Add statistics gathering for packet loss.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 
 #include <list>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/test/testsupport/gtest_prod_util.h"
 
 namespace webrtc {
 
 class ModuleRtpRtcpImpl : public RtpRtcp {
  public:
   explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
@@ skipping 137 matching lines @@
   int32_t ResetSendDataCountersRTP() override;
 
   // Statistics of the amount of data sent and received.
   int32_t DataCountersRTP(size_t* bytes_sent,
                           uint32_t* packets_sent) const override;
 
   void GetSendStreamDataCounters(
       StreamDataCounters* rtp_counters,
       StreamDataCounters* rtx_counters) const override;
 
+  void GetRtpPacketLossStats(
+      struct RtpPacketLossStats* incoming_loss,
+      struct RtpPacketLossStats* outgoing_loss) const override;
+
   // Get received RTCP report, sender info.
   int32_t RemoteRTCPStat(RTCPSenderInfo* sender_info) override;
 
   // Get received RTCP report, report block.
   int32_t RemoteRTCPStat(
       std::vector<RTCPReportBlock>* receive_blocks) const override;
 
   // (REMB) Receiver Estimated Max Bitrate.
   bool REMB() const override;
 
@@ skipping 184 matching lines @@
   uint32_t nack_last_time_sent_full_prev_;
   uint16_t nack_last_seq_number_sent_;
 
   VideoCodec send_video_codec_;
   KeyFrameRequestMethod key_frame_req_method_;
 
   RemoteBitrateEstimator* remote_bitrate_;
 
   RtcpRttStats* rtt_stats_;
 
+  PacketLossStats send_loss_stats_;
+  PacketLossStats receive_loss_stats_;
+
   // The processed RTT from RtcpRttStats.
   rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
   int64_t rtt_ms_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_