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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
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5
modules/audio_coding/neteq/neteq_impl.cc
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@ -1950,7 +1950,8 @@ int NetEqImpl::ExtractPackets(size_t required_samples,
assert(false); // Should always be able to extract a packet here.
return -1;
}
stats_.StoreWaitingTime(packet->waiting_time->ElapsedMs());
const uint64_t waiting_time_ms = packet->waiting_time->ElapsedMs();
stats_.StoreWaitingTime(waiting_time_ms);
RTC_DCHECK(!packet->empty());
if (first_packet) {
@ -1990,6 +1991,8 @@ int NetEqImpl::ExtractPackets(size_t required_samples,
}
extracted_samples = packet->timestamp - first_timestamp + packet_duration;
stats_.JitterBufferDelay(extracted_samples, waiting_time_ms);
packet_list->push_back(std::move(*packet)); // Store packet in list.
packet = rtc::Optional<Packet>(); // Ensure it's never used after the move.