| commit | e76bd3aa43821ff10f5cef5854836ea782b2464f | [log] [tgz] |
|---|---|---|
| author | zstein <zstein@webrtc.org> | Fri Jul 14 19:17:49 2017 |
| committer | Commit Bot <commit-bot@chromium.org> | Fri Jul 14 19:17:49 2017 |
| tree | a4419b5735a17886ec45541e0262de2d6fcac3e7 | |
| parent | 06f3aae345854ba9dcc5ae3b603de1f86505acf9 [diff] [blame] |
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/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc index 6dedca5..d9829c7 100644 --- a/webrtc/audio/audio_receive_stream.cc +++ b/webrtc/audio/audio_receive_stream.cc
@@ -187,6 +187,8 @@ } stats.delay_estimate_ms = channel_proxy_->GetDelayEstimate(); stats.audio_level = channel_proxy_->GetSpeechOutputLevelFullRange(); + stats.total_output_energy = channel_proxy_->GetTotalOutputEnergy(); + stats.total_output_duration = channel_proxy_->GetTotalOutputDuration(); // Get jitter buffer and total delay (alg + jitter + playout) stats. auto ns = channel_proxy_->GetNetworkStatistics();