Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

webrtc/api/stats/rtcstats_objects.h

 /*
  *  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 312 matching lines @@
   // SLI count is only defined for |media_type == "video"|.
   RTCStatsMember<uint32_t> sli_count;
   RTCStatsMember<uint64_t> qp_sum;
 
  protected:
   RTCRTPStreamStats(const std::string& id, int64_t timestamp_us);
   RTCRTPStreamStats(std::string&& id, int64_t timestamp_us);
 };
 
 // https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
-// Tracking bug crbug.com/657855
-// TODO(hbos): Support the remote case |is_remote = true|. crbug.com/657855
+// TODO(hbos): Support the remote case |is_remote = true|.
+// https://bugs.webrtc.org/7065
 class RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
  public:
   WEBRTC_RTCSTATS_DECL();
 
   RTCInboundRTPStreamStats(const std::string& id, int64_t timestamp_us);
   RTCInboundRTPStreamStats(std::string&& id, int64_t timestamp_us);
   RTCInboundRTPStreamStats(const RTCInboundRTPStreamStats& other);
   ~RTCInboundRTPStreamStats() override;
 
   RTCStatsMember<uint32_t> packets_received;
   RTCStatsMember<uint64_t> bytes_received;
   RTCStatsMember<uint32_t> packets_lost;
-  // TODO(hbos): Not collected in the "video" case by |RTCStatsCollector|.
-  // crbug.com/657855
+  // TODO(hbos): Collect and populate this value for both "audio" and "video",
+  // currently not collected for "video". https://bugs.webrtc.org/7065
   RTCStatsMember<double> jitter;
   RTCStatsMember<double> fraction_lost;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
+  RTCStatsMember<double> round_trip_time;
+  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
   RTCStatsMember<uint32_t> packets_discarded;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
   RTCStatsMember<uint32_t> packets_repaired;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
   RTCStatsMember<uint32_t> burst_packets_lost;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
   RTCStatsMember<uint32_t> burst_packets_discarded;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
   RTCStatsMember<uint32_t> burst_loss_count;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
   RTCStatsMember<uint32_t> burst_discard_count;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
   RTCStatsMember<double> burst_loss_rate;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
   RTCStatsMember<double> burst_discard_rate;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
   RTCStatsMember<double> gap_loss_rate;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
   RTCStatsMember<double> gap_discard_rate;
   RTCStatsMember<uint32_t> frames_decoded;
 };
 
 // https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
-// Tracking bug crbug.com/657856
-// TODO(hbos): Support the remote case |is_remote = true|. crbug.com/657856
+// TODO(hbos): Support the remote case |is_remote = true|.
+// https://bugs.webrtc.org/7066
 class RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
  public:
   WEBRTC_RTCSTATS_DECL();
 
   RTCOutboundRTPStreamStats(const std::string& id, int64_t timestamp_us);
   RTCOutboundRTPStreamStats(std::string&& id, int64_t timestamp_us);
   RTCOutboundRTPStreamStats(const RTCOutboundRTPStreamStats& other);
   ~RTCOutboundRTPStreamStats() override;
 
   RTCStatsMember<uint32_t> packets_sent;
   RTCStatsMember<uint64_t> bytes_sent;
-  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657856
+  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7066
   RTCStatsMember<double> target_bitrate;
-  RTCStatsMember<double> round_trip_time;
   RTCStatsMember<uint32_t> frames_encoded;
 };
 
 // https://w3c.github.io/webrtc-stats/#transportstats-dict*
 class RTCTransportStats final : public RTCStats {
  public:
   WEBRTC_RTCSTATS_DECL();
 
   RTCTransportStats(const std::string& id, int64_t timestamp_us);
   RTCTransportStats(std::string&& id, int64_t timestamp_us);
   RTCTransportStats(const RTCTransportStats& other);
   ~RTCTransportStats() override;
 
   RTCStatsMember<uint64_t> bytes_sent;
   RTCStatsMember<uint64_t> bytes_received;
   RTCStatsMember<std::string> rtcp_transport_stats_id;
   // TODO(hbos): Support enum types? "RTCStatsMember<RTCDtlsTransportState>"?
   RTCStatsMember<std::string> dtls_state;
   RTCStatsMember<std::string> selected_candidate_pair_id;
   RTCStatsMember<std::string> local_certificate_id;
   RTCStatsMember<std::string> remote_certificate_id;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_STATS_RTCSTATS_OBJECTS_H_