1592988 - Support RTCRtpReceiver. jitterBufferTarget (formerly playoutDelay)

Status: RESOLVED FIXED

(Core :: WebRTC: Signaling, enhancement, P4)

Description

[Byron Campen [:bwc]]

Maybe needs some spec discussion first.

Comment 1

[Andrew]

Hi all,

Chromium implemented this starting in M79. It would be a great addition to FF.

For some more details, please see the following:
https://github.com/w3ctag/design-reviews/issues/441
https://discourse.wicg.io/t/hint-attribute-in-webrtc-to-influence-underlying-audio-video-buffering/4038

Also see the following example jsfiddle:
https://jsfiddle.net/75cnfojy/

Below, I've pasted the "Objective" and "Use Cases" sections from the discourse link above:
Objective:
We want to provide means for javascript applications to set their preferences on how fast they want to render audio or video data. As fast as possible might be beneficial for applications which concentrates on real time experience. For others additional data buffering may provide smother experience in case of network issues.

Use cases
Cloud gaming is a good example where application requires a very high level of interactivity. 60 FPS corresponds to 16.6 milliseconds delay and we would like to provide User Agent a hint to render media as fast as possible without hurting the user experience. On the other hand, for live streaming additional 500~1500 milliseconds probably won’t hurt live experience but it would allow smoother playback because when small network issues happen you would still have some data to play.

Comment 2

[Jan-Ivar Bruaroey [:jib] (needinfo? me)]

This is now called playoutDelay in the spec.

Comment 4

[Dan Baker[:dbaker]]

Attachment: Bug 1592988 - Enable wpt tests and renamed from playoutDelayHint to playoutDelay to match spec.r?jib

Comment 5

[Dan Baker[:dbaker]]

Attachment: Bug 1592988 - Implement RtcRtpReceiver jitterBufferTarget.r?bwc,jib

Depends on D174700

Comment 6

[Dan Baker[:dbaker]]

Attachment: Bug 1592988 - Updated wpt tests to match playoutDelay spec.r?jib

Depends on D174701

Comment 7

[Dan Baker[:dbaker]]

Attachment: Bug 1592988 - Renamed from tests from playoutDelay to jitterBufferTarget and spec changes.r?jib

Depends on D174702

Comment 8

[Jim Mathies [:jimm]]

Waiting on a libwebrtc change which we'll cherry pick or roll in from a regular update.

https://bugs.chromium.org/p/webrtc/issues/detail?id=15085

Comment 9

[Dan Baker[:dbaker]]

Attachment: Bug 1592988 - Add wpt to use stats to measure jitterBufferTarget.r?jib

Depends on D176450

Comment 10

[Pulsebot]

Pushed by [e-mail]:
https://hg.mozilla.org/integration/autoland/rev/17b11131ae66
Enable wpt tests and renamed from playoutDelayHint to playoutDelay to match spec.r=jib
https://hg.mozilla.org/integration/autoland/rev/32e7e07d957c
Implement RtcRtpReceiver jitterBufferTarget.r=bwc,jib,webidl,smaug,saschanaz
https://hg.mozilla.org/integration/autoland/rev/11d59d8a50fc
Updated wpt tests to match playoutDelay spec.r=jib
https://hg.mozilla.org/integration/autoland/rev/9fa26b09323b
Renamed from tests from playoutDelay to jitterBufferTarget and spec changes.r=jib
https://hg.mozilla.org/integration/autoland/rev/38320c197609
Add wpt to use stats to measure jitterBufferTarget.r=jib

Comment 11

[Web Platform Test Sync Bot (Matrix: #interop:mozilla.org)]

Created web-platform-tests PR https://github.com/web-platform-tests/wpt/pull/40128 for changes under testing/web-platform/tests

Comment 12

[Noemi Erli[:noemi_erli]]

https://hg.mozilla.org/mozilla-central/rev/17b11131ae66
https://hg.mozilla.org/mozilla-central/rev/32e7e07d957c
https://hg.mozilla.org/mozilla-central/rev/11d59d8a50fc
https://hg.mozilla.org/mozilla-central/rev/9fa26b09323b
https://hg.mozilla.org/mozilla-central/rev/38320c197609

Comment 13

[Web Platform Test Sync Bot (Matrix: #interop:mozilla.org)]

Upstream PR merged by moz-wptsync-bot

Comment 14

[Dan Baker[:dbaker]]

Release Note Request (optional, but appreciated)
[Why is this notable]: We will be one of the first browsers to ship jitterBufferTarget. The feature allows an application to modify the jitter buffer which can improve quality for users experiencing network issues such as packet loss.
[Affects Firefox for Android]: Yes
[Suggested wording]: WebRTC application developers can now specify a target in milliseconds of media for the jitter buffer to hold. Altering the target value allows applications to control the tradeoff between playout delay and the risk of running out of audio or video frames due to network jitter.
[Links (documentation, blog post, etc)]: https://w3c.github.io/webrtc-extensions/#rtcrtpreceiver-jitterbuffertarget-attributes

Comment 15

[Donal Meehan [:dmeehan]]

Thanks, added the note to the Nightly release notes. Keeping the relnote? flag open to keep it on the radar for inclusion in our final release notes.