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→WebRTC / WebAudio: Updated headings and move current plan to previous.
(→Teams: added a definition of done.) |
(→WebRTC / WebAudio: Updated headings and move current plan to previous.) |
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! style="text-align: center;" width: "50%;" | Current Plans !! style="text-align: center;" width: "50%;" | Previous Plans | ! style="text-align: center;" width: "50%;" | Current Plans !! style="text-align: center;" width: "50%;" | Previous Plans | ||
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* Rewrite SDP/JSEP and add BUNDLE support | |||
* Multistream, renegotiation support | |||
* VP9 codec support in WebRTC (we may ship VP9 pref'd off by default) | |||
* Web Audio and WebRTC working well together | |||
* OpenH264 for Android supported (no sandbox) | |||
------ Items above done before or during Q1 ------ | |||
* Add support for audio sharing (when there is audio to share during screen/window/etc sharing) | |||
* e10s blocker problems fixed | |||
* Web Audio performance improvements - phase 1 | |||
* ICE TCP support | |||
------ Items above done before or during Q2 ------ | |||
* | |||
------ Items above planned to be done before or during Q3 ------ | |||
* | |||
------ Items above planned to be done before or during Q4 ------ | |||
| style="vertical-align:top;" | | | style="vertical-align:top;" | | ||
* Rewrite SDP/JSEP and add BUNDLE support | * Rewrite SDP/JSEP and add BUNDLE support | ||
| Line 618: | Line 634: | ||
* Compliant with "v1.0" (or equivalent) of the WebRTC spec | * Compliant with "v1.0" (or equivalent) of the WebRTC spec | ||
* getUserMedia and audio output sandboxing working (support sandboxing team, audio sandboxing may be a stretch for end of year) | * getUserMedia and audio output sandboxing working (support sandboxing team, audio sandboxing may be a stretch for end of year) | ||
------ Items above planned to be done before or during Gecko 45 (End 2015) ------ | ------ Items above planned to be done before or during Gecko 45 (End 2015) ------ | ||