LGTM2
On 9/21/23 5:31 AM, 'Philipp Hancke' via blink-dev wrote:
Am Do., 21. Sept. 2023 um 10:43 Uhr schrieb Florent Castelli
<orp...@chromium.org>:
On Wed, Sep 20, 2023 at 5:21 PM Yoav Weiss
<yoavwe...@chromium.org> wrote:
LGTM1
On Wed, Sep 20, 2023 at 11:48 AM Florent Castelli
<orp...@chromium.org> wrote:
Contact emails
orp...@chromium.org
Explainer
https://github.com/Orphis/webrtc-extensions/blob/explainer/explainers/codec-selection.md
Specification
https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-codec
Summary
This new API extends WebRTC encoding parameters to allow
developers to choose a specific negotiated codec to be
used for encoding an RTP stream.
Blink component
Blink>WebRTC>PeerConnection
<https://bugs.chromium.org/p/chromium/issues/list?q=component:Blink%3EWebRTC%3EPeerConnection>
TAG review
https://github.com/w3ctag/design-reviews/issues/836
TAG review status
Issues addressed
Risks
Interoperability and Compatibility
Interoperability risks are low. This is a new dictionary
member and API surface, if it isn't used by current
applications, it should not cause any impact on compatibility.
/Gecko/: Positive
(https://github.com/mozilla/standards-positions/issues/789)
/WebKit/: Positive
(https://github.com/WebKit/standards-positions/issues/179)
/Web developers/: No signals
Presumably this is something WebRTC developers are interested
in. Any ways you can think of to extract such a signal for
WebRTC things in the future?
WebRTC developers are indeed interested but have nothing to link
to. Google Meet is interested in using the API and other
conferencing solutions have been interested in advanced scenarios
this API allows as well.
Same for MSFT Teams which I work on. Switching between codecs on the
fly is becoming important (e.g. because AV1 has encoding tools which
very much benefit static content for slide sharing).
While this is already possible with SDP renegotiation (even locally),
the API is much more ergonomic and avoids some race conditions.
The API also enables use-cases where multiple simulcast streams (i.e.
same picture but different resolutions) can use different codecs, e.g.
a more modern one like AV1 for small resolutions where software
encoding is feasible and a hardware encoded one (H264, VP9 I think)
for large resolutions
What I heard from Jitsi Meet via Saul (cc'd in case he wants to say
more) sounded similar.
Ergonomics
No ergonomic risks. This API is part of WebRTC and will be
used with other encoding parameters. One of its goals is
to improve the WebRTC ergonomics by removing the need to
make several calls with possible side effects to change
the active codec, which should improve performance as well.
Debuggability
WebRTC is not supported by DevTools at the moment.
chrome://webrtc-internals will reflect in the RTP stream
statistics which codec is currently used.
Will this feature be supported on all six Blink
platforms (Windows, Mac, Linux, Chrome OS,
Android, and Android WebView)?
Yes
Is this feature fully tested by web-platform-tests
<https://chromium.googlesource.com/chromium/src/+/main/docs/testing/web_platform_tests.md>?
Yes
Flag name on chrome://flags
Finch feature name
RTCRtpEncodingParametersCodec
Requires code in //chrome?
False
Tracking bug
https://bugs.chromium.org/p/chromium/issues/detail?id=1442194
Estimated milestones
Shipping on desktop 119
Shipping on Android 119
Shipping on WebView 119
Shipping on WebView 119
Link to entry on the Chrome Platform Status
https://chromestatus.com/feature/5200982281027584
Links to previous Intent discussions
Intent to prototype:
https://groups.google.com/a/chromium.org/d/msgid/blink-dev/CADRnnSXEi2i3DzkXQo1ptX-KR%2BULh-4LOt5zpOi14BTstZN_9g%40mail.gmail.com
This intent message was generated by Chrome Platform
Status <https://chromestatus.com/>.
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