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Fix unscaled timestamps passed to nack_tracker

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If timestamp_scaler_ is used, then rtp_header.timestamp, passed to UpdateLastDecodedPacket, will advance at a different rate than the scaled timestamp packet->timestamp, passed to UpdateLastDecodedPacket.

NackTracker::EstimateTimestamp uses timestamp_last_received_rtp_, and NackTracker::TimeToPlay uses timestamp_last_decoded_rtp_.

This difference causes TimeToPlay to continuously increase to huge values, so that every missing packet will be returned from GetNackList, even if RTT > real time to play.

Change-Id: Ie6ca347972edea98a202c9cdd26c6ab3f45a73c4
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222841
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34361}
This commit is contained in:
Jared Siskin
2021-06-18 10:45:16 -07:00
committed by WebRTC LUCI CQ
parent 9233af3e22
commit f2ed401679
2 changed files with 71 additions and 2 deletions
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3
modules/audio_coding/neteq/neteq_impl.cc
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@ -628,8 +628,7 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
if (update_sample_rate_and_channels) {
nack_->Reset();
}
nack_->UpdateLastReceivedPacket(rtp_header.sequenceNumber,
rtp_header.timestamp);
nack_->UpdateLastReceivedPacket(main_sequence_number, main_timestamp);
}
// Check for RED payload type, and separate payloads into several packets.