| commit | b0a020783890e526263ef9ecbbeb4048653559d0 | [log] [tgz] |
|---|---|---|
| author | Gustaf Ullberg <gustaf@webrtc.org> | Mon Oct 02 12:00:34 2017 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Oct 02 10:47:00 2017 +0000 |
| tree | 546f03215501b9b6213bbfa0363a0a75432f0b3b | |
| parent | 652cc84069d3b2ff8d86405fe0b150cd1cb23264 [diff] [blame] |
Added RTCMediaStreamTrackStats.jitterBufferDelay for audio Description of this stat can be found here: https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-jitterbufferdelay Bug: webrtc:8281 Change-Id: Ib2e8174f3449e68ad419ae2d58d5565fc9854023 Reviewed-on: https://webrtc-review.googlesource.com/3381 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20069}
diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h index b349f20..e6cafa8 100644 --- a/modules/audio_coding/neteq/include/neteq.h +++ b/modules/audio_coding/neteq/include/neteq.h
@@ -66,6 +66,7 @@ uint64_t total_samples_received = 0; uint64_t concealed_samples = 0; uint64_t concealment_events = 0; + uint64_t jitter_buffer_delay_ms = 0; }; enum NetEqPlayoutMode {