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Platform/Roadmap

1,092 bytes added, 15:23, 27 October 2015
WebRTC / WebAudio: Updated after Q3
* VP9 codec support in WebRTC - 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 ------
* e10s blocker problems fixed
* Web Audio performance improvements - phase 1: significant perf wins on all benchmarks
* ICE TCP support landed in Nightly - pref'd off
* IPv6 support landed in Nightly
* Initial Identity support landed in Nightly
* WebRTC fully integrated with Web Audio: Web Audio can now feed a PeerConnection and receive audio back from that PC
 
------ Items above done before or during Q2 ------
* Capturing video from a canvas (pref'd on)
* Ability to change camera resolution with MediaStreamTrack.applyConstraint
* Simulcast - phase 1: largely focused on signaling and spec work
** We chose not to land it, but it was essential for setting us up for landing simulcast in Nightly in Q4
* Audio/Video refactor - paving the way for Full Duplex support and for resolving existing audio/video stack shortcomings
* Web Audio performance improvements shipping in Firefox 44
* IPV6 support shipping in Firefox 42
 
------ Items above done before or during Q3 ------
* Simulcast support (as described in the IETF spec proposal) landed in Nightly
* Full duplex audio support on Windows landed in Nightly
** stretch goal: land full duplex audio support for a second platform (likely Mac)
* New Delay Agnostic AEC landed in Nightly
* Media Stream add/remove track, constructions, and clone landed in Nightly (big part of spec compliance goal)
* ICE TCP pref'd on in Nightly
 
------ Items above planned to be done before or during Q4 ------
* VP9 support in WebRTC pref'd on by default (may or may not be the preferred codec when we pref on)
* Full duplex audio support on all platforms
* Audio output selection
* Considerably better NAT traversal (including ICE proxy and UPnP support)
* Audio Workers (dependent on achieving working group consensus)
* Data Channels in Workers (dependent on achieving working group consensus)
* Compliant with "v1.0" (or equivalent) of the WebRTC spec
** Finish implementation of MediaStreamTrack constraints
* getUserMedia and audio output sandboxing working (support sandboxing team, audio sandboxing may be a stretch for end of year)
 
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* Rewrite SDP/JSEP and add BUNDLE support
* Multistream, renegotiation support
* VP9 codec support in WebRTC - pref'd off by default
* Web Audio and WebRTC working well together
* OpenH264 for Android supported (no sandbox)
------ Items above done before or during Q2 ------
* Add support for audio sharing (when there is audio to share during screen/window/etc sharing)
** landing pref'd off in Fx42
** pref on before the end of the quarter
* Audio/Video refactor - phase 1: paving the way for Full Duplex support and for resolving existing audio/video stack shortcomings
* Web Audio perf - phase 2
** resolve remaining bugs on the web audio perf meta bug** identify if there are any games (as of the start of Q3) that must work well by the end of Q3 on a specific (targeted) device that do not currently work well enough and then deliver the performance needed
------ Items above planned to be done before or during Q3 ------
* Simulcast support (as described in the IETF spec proposal) landed in Nightly
* Audio/Video refactor - phase 2 (second of an estimated three phases)
* Considerably better NAT traversal (including ICE proxy and UPnP support)
* Audio Workers (dependent on achieving working group consensus)
* Data Channels in Workers (dependent on achieving working group consensus)
------ Items above planned to be done before or during Q4 ------
 
* Full duplex audio support
* Audio output selection
* 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)
 
| style="vertical-align:top;" |
* 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 were done before or during Gecko 39 (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 planned to be done before or during Gecko 42 (Q2) ------
* Resolve (modulo hardware limitations) reproducible audio problems in WebRTC 1:1 calls
* Excellent Web Audio performance, especially for games ("phase 2")
* Audio workers support
* Simulcast support
* IPv6 support
* Full duplex audio support
* Considerably better NAT traversal (including ICE proxy and UPnP support)
* Audio output selection
* Data channels in workers
* 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 ------ Items above planned to be done before or during Gecko 45 (End 2015) ------|}
==Web Compatibility==
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