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