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Change playoutdelay to jitterbuffertarget #753
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jitterBufferDelay is the actual delay, the total time spent for each sample emitted by the jitter buffer. That does not need to change. The reason the actual delay may be different than the target delay, jitterBufferTargetDelay, is that when the target changes we accelerate or decelerate samples to avoid glitches. The RTCRtpReceiver.jitterBufferTarget API sets the desired target of the app. The reason why the API target and the current target may be different is due to, as the API describes, there could be network conditions or resource constraints that make the UA not able to achieve it right now. And the jitterBufferMinimumDelay is just exposing the internal view of what the minimum delay could be. The way I see it, there are no webrtc-stats changes needed and it all makes sense. @jan-ivar |
Sorry ignore me, I thought you wanted to rename the stats, but you just wanted to update a reference. Yes let's do that |
* fixs #753 change playout delay to jitterbuffertarget * add prefix of "dom" * Update webrtc-stats.html Co-authored-by: henbos <hbos@google.com> * Update webrtc-stats.html Co-authored-by: henbos <hbos@google.com> --------- Co-authored-by: 何超 <hechao.0520@bytedance.com> Co-authored-by: henbos <hbos@google.com>
As the playoutDelay changed to jitterbuffertarget in this PR , playoutDelay used in webrtc-stats should be changed accordingly.
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