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}
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

View File

@ -337,6 +337,7 @@ void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
acm_stat->totalSamplesReceived = neteq_lifetime_stat.total_samples_received;
acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
}
int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,