RTCMediaStream[Track]Stats 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 249 matching lines @@
 
 RTCRemoteIceCandidateStats::RTCRemoteIceCandidateStats(
     std::string&& id, int64_t timestamp_us)
     : RTCIceCandidateStats(std::move(id), timestamp_us) {
 }
 
 const char* RTCRemoteIceCandidateStats::type() const {
   return kType;
 }
 
+// TODO(hbos): According to spec this should be "track", but there isn't any
+// string for RTCMediaStreamTrackStats. Shouldn't that be "track"? Calling this
+// "stream" instead. crbug.com/660827, crbug.com/659137

[hta-webrtc, 2016/11/03 14:34:22]

Spec bug. Will fix.

[hbos, 2016/11/03 18:10:31]

Acknowledged.

+WEBRTC_RTCSTATS_IMPL(RTCMediaStreamStats, RTCStats, "stream",
+    &stream_identifier,
+    &track_ids);
+
+RTCMediaStreamStats::RTCMediaStreamStats(
+    const std::string& id, int64_t timestamp_us)
+    : RTCMediaStreamStats(std::string(id), timestamp_us) {
+}
+
+RTCMediaStreamStats::RTCMediaStreamStats(
+    std::string&& id, int64_t timestamp_us)
+    : RTCStats(std::move(id), timestamp_us),
+      stream_identifier("streamIdentifier"),
+      track_ids("trackIds") {
+}
+
+RTCMediaStreamStats::RTCMediaStreamStats(
+    const RTCMediaStreamStats& other)
+    : RTCStats(other.id(), other.timestamp_us()),
+      stream_identifier(other.stream_identifier),
+      track_ids(other.track_ids) {
+}
+
+RTCMediaStreamStats::~RTCMediaStreamStats() {
+}
+
+// TODO(hbos): The spec doesn't have a label for this stat, using "track" but
+// there is a risk that "track" should be used for RTCMediaStreamStats instead.
+// crbug.com/659137, crbug.com/660827

[hta-webrtc, 2016/11/03 14:34:22]

 https://github.com/w3c/webrtc-stats/issues/80

[hbos, 2016/11/03 18:10:30]

Acknowledged.

+WEBRTC_RTCSTATS_IMPL(RTCMediaStreamTrackStats, RTCStats, "track",
+    &track_identifier,
+    &remote_source,
+    &ended,
+    &detached,
+    &ssrc_ids,
+    &frame_width,
+    &frame_height,
+    &frames_per_second,
+    &frames_sent,
+    &frames_received,
+    &frames_decoded,
+    &frames_dropped,
+    &frames_corrupted,
+    &partial_frames_lost,
+    &full_frames_lost,
+    &audio_level,
+    &echo_return_loss,
+    &echo_return_loss_enhancement);
+
+RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
+    const std::string& id, int64_t timestamp_us)
+    : RTCMediaStreamTrackStats(std::string(id), timestamp_us) {
+}
+
+RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
+    std::string&& id, int64_t timestamp_us)
+    : RTCStats(std::move(id), timestamp_us),
+      track_identifier("trackIdentifier"),
+      remote_source("remoteSource"),
+      ended("ended"),
+      detached("detached"),
+      ssrc_ids("ssrcIds"),
+      frame_width("frameWidth"),
+      frame_height("frameHeight"),
+      frames_per_second("framesPerSecond"),
+      frames_sent("framesSent"),
+      frames_received("framesReceived"),
+      frames_decoded("framesDecoded"),
+      frames_dropped("framesDropped"),
+      frames_corrupted("framesCorrupted"),
+      partial_frames_lost("partialFramesLost"),
+      full_frames_lost("fullFramesLost"),
+      audio_level("audioLevel"),
+      echo_return_loss("echoReturnLoss"),
+      echo_return_loss_enhancement("echoReturnLossEnhancement") {
+}
+
+RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
+    const RTCMediaStreamTrackStats& other)
+    : RTCStats(other.id(), other.timestamp_us()),
+      track_identifier(other.track_identifier),
+      remote_source(other.remote_source),
+      ended(other.ended),
+      detached(other.detached),
+      ssrc_ids(other.ssrc_ids),
+      frame_width(other.frame_width),
+      frame_height(other.frame_height),
+      frames_per_second(other.frames_per_second),
+      frames_sent(other.frames_sent),
+      frames_received(other.frames_received),
+      frames_decoded(other.frames_decoded),
+      frames_dropped(other.frames_dropped),
+      frames_corrupted(other.frames_corrupted),
+      partial_frames_lost(other.partial_frames_lost),
+      full_frames_lost(other.full_frames_lost),
+      audio_level(other.audio_level),
+      echo_return_loss(other.echo_return_loss),
+      echo_return_loss_enhancement(other.echo_return_loss_enhancement) {
+}
+
+RTCMediaStreamTrackStats::~RTCMediaStreamTrackStats() {
+}
+
 WEBRTC_RTCSTATS_IMPL(RTCPeerConnectionStats, RTCStats, "peer-connection",
     &data_channels_opened,
     &data_channels_closed);
 
 RTCPeerConnectionStats::RTCPeerConnectionStats(
     const std::string& id, int64_t timestamp_us)
     : RTCPeerConnectionStats(std::string(id), timestamp_us) {
 }
 
 RTCPeerConnectionStats::RTCPeerConnectionStats(
@@ skipping 200 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