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Added RTCMediaStreamTrackStats.jitterBufferDelay for audio

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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}
This commit is contained in:
Gustaf Ullberg
2017-10-02 12:00:34 +02:00
committed by Commit Bot
parent 652cc84069
commit b0a0207838
20 changed files with 113 additions and 12 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
pc/rtcstats_integrationtest.cc
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@ -562,8 +562,11 @@ class RTCStatsReportVerifier {
}
// totalSamplesReceived, concealedSamples and concealmentEvents are only
// present on inbound audio tracks.
// jitterBufferDelay is currently only implemented for audio.
if (*media_stream_track.kind == RTCMediaStreamTrackKind::kAudio &&
*media_stream_track.remote_source) {
verifier.TestMemberIsNonNegative<double>(
media_stream_track.jitter_buffer_delay);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.total_samples_received);
verifier.TestMemberIsNonNegative<uint64_t>(
@ -571,6 +574,7 @@ class RTCStatsReportVerifier {
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.concealment_events);
} else {
verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_delay);
verifier.TestMemberIsUndefined(media_stream_track.total_samples_received);
verifier.TestMemberIsUndefined(media_stream_track.concealed_samples);
verifier.TestMemberIsUndefined(media_stream_track.concealment_events);