Adding stats that can be used to compute output audio levels.

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 330 matching lines @@
 RTCMediaStreamStats::RTCMediaStreamStats(
     const RTCMediaStreamStats& other)
     : RTCStats(other.id(), other.timestamp_us()),
       stream_identifier(other.stream_identifier),
       track_ids(other.track_ids) {
 }
 
 RTCMediaStreamStats::~RTCMediaStreamStats() {
 }
 
-WEBRTC_RTCSTATS_IMPL(RTCMediaStreamTrackStats, RTCStats, "track",
-    &track_identifier,
-    &remote_source,
-    &ended,
-    &detached,
-    &kind,
-    &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);
+WEBRTC_RTCSTATS_IMPL(RTCMediaStreamTrackStats,
+                     RTCStats,
+                     "track",

[hbos, 2017/07/07 12:52:24]

Place RTCMediaStreamTrackStats, RTCStats, "track" on the same line followed by
one member per line. Style guide of this file with the weird macros. :)

The WEBRTC_RTCSTATS_IMPL defines all members of the stats objects so that they
are iterable ("for each stat in stats object"). As a result, not having a member
listed here will mean it is missed for example when exporting stats to Android
or other platforms that get stats by iterate all stats.

It's not meant to be a whitelist though (update CL description) and not listing
a stat here would be an error, whitelisted or not.

Speaking of whitelists, there is a test in chromium which tries to make sure we
don't expose any stats that have not been whitelisted with a chromium repo CL
(guard against non-standard stats), see javascript test:
https://cs.chromium.org/chromium/src/chrome/test/data/webrtc/peerconnection_g...

If this CL lands without the chromium whitelist having been updated first the
roll into chromium will fail if the browser test detects the stat. Please make a
chromium CL whitelisting it, or if you don't work with chromium or don't know
what to do, I can whitelist it for you no problem.

[Zach Stein, 2017/07/07 21:27:37]

Done. "git cl format" changed it and I didn't notice.
Got it, thanks. Description updated.
Thanks for the heads up. I'll try to whitelist in Chromium before submitting.

+                     &track_identifier,
+                     &remote_source,
+                     &ended,
+                     &detached,
+                     &kind,
+                     &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,
+                     &total_audio_energy,
+                     &total_samples_duration,
+                     &echo_return_loss,
+                     &echo_return_loss_enhancement);
 
 RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
     const std::string& id, int64_t timestamp_us, const char* kind)
     : RTCMediaStreamTrackStats(std::string(id), timestamp_us, kind) {
 }
 
-RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
-    std::string&& id, int64_t timestamp_us, const char* kind)
+RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(std::string&& id,
+                                                   int64_t timestamp_us,
+                                                   const char* kind)
     : RTCStats(std::move(id), timestamp_us),
       track_identifier("trackIdentifier"),
       remote_source("remoteSource"),
       ended("ended"),
       detached("detached"),
       kind("kind", kind),
       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"),
+      total_audio_energy("totalAudioEnergy"),
+      total_samples_duration("totalSamplesDuration"),
       echo_return_loss("echoReturnLoss"),
       echo_return_loss_enhancement("echoReturnLossEnhancement") {
   RTC_DCHECK(kind == RTCMediaStreamTrackKind::kAudio ||
              kind == RTCMediaStreamTrackKind::kVideo);
 }
 
 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),
       kind(other.kind),
       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),
+      total_audio_energy(other.total_audio_energy),
+      total_samples_duration(other.total_samples_duration),
       echo_return_loss(other.echo_return_loss),
-      echo_return_loss_enhancement(other.echo_return_loss_enhancement) {
-}
+      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)
@@ skipping 213 matching lines @@
       dtls_state(other.dtls_state),
       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