commit b0a020783890e526263ef9ecbbeb4048653559d0
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

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}

api/stats/rtcstats_objects.h

diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index 903d266..97a7d3c 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -255,6 +255,10 @@
   RTCStatsMember<bool> detached;
   // See |RTCMediaStreamTrackKind| for valid values.
   RTCStatsMember<std::string> kind;
+  // TODO(gustaf): Implement jitter_buffer_delay for video (currently
+  // implemented for audio only).
+  // https://crbug.com/webrtc/8318
+  RTCStatsMember<double> jitter_buffer_delay;
   // Video-only members
   RTCStatsMember<uint32_t> frame_width;
   RTCStatsMember<uint32_t> frame_height;