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

webrtc/stats/rtcstats_objects.cc

Index: webrtc/stats/rtcstats_objects.cc
diff --git a/webrtc/stats/rtcstats_objects.cc b/webrtc/stats/rtcstats_objects.cc
index 5397ef782cca502ff4508e2118b38a251c0b4af3..22debda3ee622fa6b7c4b06141672701d8f8c398 100644
--- a/webrtc/stats/rtcstats_objects.cc
+++ b/webrtc/stats/rtcstats_objects.cc
@@ -348,33 +348,38 @@ RTCMediaStreamStats::RTCMediaStreamStats(
 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"),
@@ -392,6 +397,8 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
       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 ||
@@ -417,9 +424,10 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
       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() {
 }