Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
BUG=webrtc:7982
Review-Url: https://codereview.webrtc.org/2964593002
Cr-Commit-Position: refs/heads/master@{#19027}
diff --git a/webrtc/pc/rtcstats_integrationtest.cc b/webrtc/pc/rtcstats_integrationtest.cc
index 3d1e7d6..ea1f8df 100644
--- a/webrtc/pc/rtcstats_integrationtest.cc
+++ b/webrtc/pc/rtcstats_integrationtest.cc
@@ -475,6 +475,8 @@
verifier.TestMemberIsUndefined(media_stream_track.echo_return_loss);
verifier.TestMemberIsUndefined(
media_stream_track.echo_return_loss_enhancement);
+ verifier.TestMemberIsUndefined(media_stream_track.total_audio_energy);
+ verifier.TestMemberIsUndefined(media_stream_track.total_samples_duration);
} else {
RTC_DCHECK_EQ(*media_stream_track.kind,
RTCMediaStreamTrackKind::kAudio);
@@ -491,6 +493,10 @@
verifier.TestMemberIsUndefined(media_stream_track.full_frames_lost);
// Audio-only members
verifier.TestMemberIsNonNegative<double>(media_stream_track.audio_level);
+ verifier.TestMemberIsNonNegative<double>(
+ media_stream_track.total_audio_energy);
+ verifier.TestMemberIsNonNegative<double>(
+ media_stream_track.total_samples_duration);
// TODO(hbos): |echo_return_loss| and |echo_return_loss_enhancement| are
// flaky on msan bot (sometimes defined, sometimes undefined). Should the
// test run until available or is there a way to have it always be