RTCInboundRTPStreamStats added.

webrtc/stats/rtcstats_objects.cc

 /*
  *  Copyright 2016 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 329 matching lines @@
       fir_count(other.fir_count),
       pli_count(other.pli_count),
       nack_count(other.nack_count),
       sli_count(other.sli_count) {
 }
 
 RTCRTPStreamStats::~RTCRTPStreamStats() {
 }
 
 WEBRTC_RTCSTATS_IMPL(
+    RTCInboundRTPStreamStats, RTCRTPStreamStats, "inbound-rtp",
+    &packets_received,
+    &bytes_received,
+    &packets_lost,
+    &jitter,
+    &fraction_lost,
+    &packets_discarded,
+    &packets_repaired,
+    &burst_packets_lost,
+    &burst_packets_discarded,
+    &burst_loss_count,
+    &burst_discard_count,
+    &burst_loss_rate,
+    &burst_discard_rate,
+    &gap_loss_rate,
+    &gap_discard_rate);
+
+RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
+    const std::string& id, int64_t timestamp_us)
+    : RTCInboundRTPStreamStats(std::string(id), timestamp_us) {
+}
+
+RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
+    std::string&& id, int64_t timestamp_us)
+    : RTCRTPStreamStats(std::move(id), timestamp_us),
+      packets_received("packetsReceived"),
+      bytes_received("bytesReceived"),
+      packets_lost("packetsLost"),
+      jitter("jitter"),
+      fraction_lost("fractionLost"),
+      packets_discarded("packetsDiscarded"),
+      packets_repaired("packetsRepaired"),
+      burst_packets_lost("burstPacketsLost"),
+      burst_packets_discarded("burstPacketsDiscarded"),
+      burst_loss_count("burstLossCount"),
+      burst_discard_count("burstDiscardCount"),
+      burst_loss_rate("burstLossRate"),
+      burst_discard_rate("burstDiscardRate"),
+      gap_loss_rate("gapLossRate"),
+      gap_discard_rate("gapDiscardRate") {
+}
+
+RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
+    const RTCInboundRTPStreamStats& other)
+    : RTCRTPStreamStats(other),
+      packets_received(other.packets_received),
+      bytes_received(other.bytes_received),
+      packets_lost(other.packets_lost),
+      jitter(other.jitter),
+      fraction_lost(other.fraction_lost),
+      packets_discarded(other.packets_discarded),
+      packets_repaired(other.packets_repaired),
+      burst_packets_lost(other.burst_packets_lost),
+      burst_packets_discarded(other.burst_packets_discarded),
+      burst_loss_count(other.burst_loss_count),
+      burst_discard_count(other.burst_discard_count),
+      burst_loss_rate(other.burst_loss_rate),
+      burst_discard_rate(other.burst_discard_rate),
+      gap_loss_rate(other.gap_loss_rate),
+      gap_discard_rate(other.gap_discard_rate) {
+}
+
+RTCInboundRTPStreamStats::~RTCInboundRTPStreamStats() {
+}
+
+WEBRTC_RTCSTATS_IMPL(
     RTCOutboundRTPStreamStats, RTCRTPStreamStats, "outbound-rtp",
     &packets_sent,
     &bytes_sent,
     &target_bitrate,
     &round_trip_time);
 
 RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(
     const std::string& id, int64_t timestamp_us)
     : RTCOutboundRTPStreamStats(std::string(id), timestamp_us) {
 }
@@ skipping 54 matching lines @@
       active_connection(other.active_connection),
       selected_candidate_pair_id(other.selected_candidate_pair_id),
       local_certificate_id(other.local_certificate_id),
       remote_certificate_id(other.remote_certificate_id) {
 }
 
 RTCTransportStats::~RTCTransportStats() {
 }
 
 }  // namespace webrtc