RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected.

webrtc/pc/rtcstatscollector.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 669 matching lines @@
       pc_->session()->GetStats(*channel_name_pairs_);
   if (session_stats) {
     std::map<std::string, CertificateStatsPair> transport_cert_stats =
         PrepareTransportCertificateStats_n(*session_stats);
 
     ProduceCertificateStats_n(
         timestamp_us, transport_cert_stats, report.get());
     ProduceCodecStats_n(
         timestamp_us, *track_media_info_map_, report.get());
     ProduceIceCandidateAndPairStats_n(
-        timestamp_us, *session_stats, report.get());
+        timestamp_us, *session_stats, track_media_info_map_->video_media_info(),
+        report.get());
     ProduceRTPStreamStats_n(
         timestamp_us, *session_stats, *track_media_info_map_, report.get());
     ProduceTransportStats_n(
         timestamp_us, *session_stats, transport_cert_stats, report.get());
   }
 
   AddPartialResults(report);
 }
 
 void RTCStatsCollector::AddPartialResults(
@@ skipping 109 matching lines @@
     data_channel_stats->messages_sent = data_channel->messages_sent();
     data_channel_stats->bytes_sent = data_channel->bytes_sent();
     data_channel_stats->messages_received = data_channel->messages_received();
     data_channel_stats->bytes_received = data_channel->bytes_received();
     report->AddStats(std::move(data_channel_stats));
   }
 }
 
 void RTCStatsCollector::ProduceIceCandidateAndPairStats_n(
       int64_t timestamp_us, const SessionStats& session_stats,
+      const cricket::VideoMediaInfo* video_media_info,
       RTCStatsReport* report) const {
   RTC_DCHECK(network_thread_->IsCurrent());
   for (const auto& transport_stats : session_stats.transport_stats) {
     for (const auto& channel_stats : transport_stats.second.channel_stats) {
       std::string transport_id = RTCTransportStatsIDFromTransportChannel(
           transport_stats.second.transport_name, channel_stats.component);
       for (const cricket::ConnectionInfo& info :
            channel_stats.connection_infos) {
         std::unique_ptr<RTCIceCandidatePairStats> candidate_pair_stats(
             new RTCIceCandidatePairStats(
@@ skipping 18 matching lines @@
         candidate_pair_stats->writable = info.writable;
         candidate_pair_stats->bytes_sent =
             static_cast<uint64_t>(info.sent_total_bytes);
         candidate_pair_stats->bytes_received =
             static_cast<uint64_t>(info.recv_total_bytes);
         // TODO(hbos): The |info.rtt| measurement is smoothed. It shouldn't be
         // smoothed according to the spec. crbug.com/633550. See
         // https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-currentrtt
         candidate_pair_stats->current_round_trip_time =
             static_cast<double>(info.rtt) / rtc::kNumMillisecsPerSec;
+        if (info.best_connection && video_media_info &&

[stefan-webrtc, 2017/02/06 16:08:17]

Is best_connection always the connection in used?

[hbos, 2017/02/06 16:12:37]

To my understanding, yes. +deadbeef, do you know?

[Taylor Brandstetter, 2017/02/06 19:32:13]

Yes, it is. It was even renamed from "best_connection" to "selected_connection"
in some places, but not here.

+            !video_media_info->bw_estimations.empty()) {
+          // The bandwidth estimations we have are for the selected candidate
+          // pair ("info.best_connection").
+          RTC_DCHECK_EQ(video_media_info->bw_estimations.size(), 1);
+          RTC_DCHECK_GE(
+              video_media_info->bw_estimations[0].available_send_bandwidth, 0);
+          RTC_DCHECK_GE(
+              video_media_info->bw_estimations[0].available_recv_bandwidth, 0);
+          if (video_media_info->bw_estimations[0].available_send_bandwidth) {
+            candidate_pair_stats->available_outgoing_bitrate =
+                static_cast<double>(video_media_info->bw_estimations[0]
+                                        .available_send_bandwidth);
+          }
+          if (video_media_info->bw_estimations[0].available_recv_bandwidth) {
+            candidate_pair_stats->available_incoming_bitrate =
+                static_cast<double>(video_media_info->bw_estimations[0]
+                                        .available_recv_bandwidth);
+          }
+        }
         candidate_pair_stats->requests_received =
             static_cast<uint64_t>(info.recv_ping_requests);
         candidate_pair_stats->requests_sent = static_cast<uint64_t>(
             info.sent_ping_requests_before_first_response);
         candidate_pair_stats->responses_received =
             static_cast<uint64_t>(info.recv_ping_responses);
         candidate_pair_stats->responses_sent =
             static_cast<uint64_t>(info.sent_ping_responses);
         RTC_DCHECK_GE(info.sent_ping_requests_total,
                       info.sent_ping_requests_before_first_response);
@@ skipping 354 matching lines @@
     const std::string& type) {
   return CandidateTypeToRTCIceCandidateType(type);
 }
 
 const char* DataStateToRTCDataChannelStateForTesting(
     DataChannelInterface::DataState state) {
   return DataStateToRTCDataChannelState(state);
 }
 
 }  // namespace webrtc