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=Firefox 40 WebRTC/WebAudio Release Notes:= | |||
Full listing of all WebAudio bugs marked as Fixed in Firefox 40: | ==Full listing of all WebRTC/WebAudio bugs marked as Fixed in Firefox 40:== | ||
'''''WebRTC bugs:''''' | |||
[http://mzl.la/1FmwxEJ Bugzilla search for WebRTC bugs marked Fixed in Firefox 40] | |||
'''''WebAudio bugs:''''' | |||
[http://mzl.la/1FmwNUh Bugzilla search for WebAudio bugs marked Fixed in Firefox 40] | [http://mzl.la/1FmwNUh Bugzilla search for WebAudio bugs marked Fixed in Firefox 40] | ||
===Core:=== | ==Noteworthy bugs marked fixed in Firefox 40:== | ||
===Core (General) WebRTC:=== | |||
{{Bug|1093934}}, {{Bug|1097804}}, {{bug|1101651}}: standalone library for using | {{Bug|1093934}}, {{Bug|1097804}}, {{bug|1101651}}: standalone library for using | ||
WebRTC based on Firefox's implementation. Major new package for supporting 3rd party applications independent of libjingle/etc | WebRTC based on Firefox's implementation. Major new package for supporting 3rd party applications independent of libjingle/etc | ||
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{{Bug|1152538}}: enable WebRTC Identity by default | {{Bug|1152538}}: enable WebRTC Identity by default | ||
{{Bug|1100502}}: about:webrtc | {{Bug|1100502}}: about:webrtc works fully in e10s. All sessions are now visible regardless of process. | ||
{{Bug|1137614}}: crashes in libvpx in vp8_diamond_search_sadx4 | {{Bug|1137614}}: crashes in libvpx in vp8_diamond_search_sadx4 | ||
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{{Bug|1159659}}: Allow tab sharing gUM requests on Windows XP and OSX 10.6 | {{Bug|1159659}}: Allow tab sharing gUM requests on Windows XP and OSX 10.6 | ||
{{Bug|1161079}}: Multistream landing broke encoder statistics | {{Bug|1161079}}: Multistream landing broke encoder statistics - '''Uplifted to 39 and ESR 38.''' | ||
===Audio/Video:=== | ===Audio/Video:=== | ||
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{{Bug|1146637}}: Firefox OS (flame) ignores constraints, chooses very low video resolution | {{Bug|1146637}}: Firefox OS (flame) ignores constraints, chooses very low video resolution | ||
{{Bug|1155089}}: RTPSender.replaceTrack()ing a gUM audio track with a WebAudio track yields silence | {{Bug|1155089}}: RTPSender.replaceTrack()ing a gUM audio track with a WebAudio track yields silence - This bug fix and the bug fix for {{Bug|1081819}} means that it is now possible to replaceTrack() a getUserMedia audio track with a WebAudio track and send it over WebRTC! | ||
{{Bug|1106958}}: Use android.media.MediaCodec for decoding in WebRTC stack | {{Bug|1106958}}: Use android.media.MediaCodec for decoding in WebRTC stack- Major new support for hardware codecs on Android contributed by Intel '''(Thanks, Intel!)''' | ||
{{Bug|1151628}}, {{bug|1152016}} - MJPEG getUserMedia sources don't work (regression) | {{Bug|1151628}}, {{bug|1152016}} - MJPEG getUserMedia sources don't work (regression) - This was fixed in Firefox 29 and regressed in Firefox 30. Enables high resolution getUserMedia captures on many webcams. - '''Uplifted to 38''' | ||
{{Bug|1162412}} - FacingMode regression (android, b2g) - | {{Bug|1162412}} - FacingMode regression (android, b2g) - '''Uplifted to 38.0.5''' | ||
{{Bug|1149494}}: video.onloadedmetadata handler doesn't seem to work with MediaStream input since Firefox37 | {{Bug|1149494}}: video.onloadedmetadata handler doesn't seem to work with MediaStream input since Firefox37 - '''Uplifted to 38''' | ||
{{Bug|1159300}} - GMP OpenH264 fails to decode on reload on Windows | {{Bug|1159300}} - GMP OpenH264 fails to decode on reload on Windows - Works first time, fails on second. - '''Uplifted to 38''' | ||
===Networking:=== | ===Networking:=== | ||
{{Bug|1162251}}: H264 packetization was incompatible with sliced mode-1 | {{Bug|1162251}}: H264 packetization was incompatible with sliced mode-1 - H.264 and also with mode 0. - '''Fix landed in 40 and uplifted to 38.0''' | ||
{{Bug|996238}}: ALPN identifiers - Supports stream isolation from JS content | |||
{{Bug| | {{Bug|1157766}}: JSEP rewrite in 37 had regressed datachannels past max 16 (8 started by each side) - '''Fix landed in 40 and uplifted to 38.''' | ||
{{Bug|1131779}}: Webrtc stops using relay port after permission error response - After a TURN server responded to a permission request with denied (403) the relay will be tried for other candidates | |||
{{Bug|1131779}}: Webrtc stops using relay port after permission error response | |||
{{Bug|1161317}}: Incorrect encryption of RTCP Packets when using unidirectional PeerConnections | {{Bug|1161317}}: Incorrect encryption of RTCP Packets when using unidirectional PeerConnections - Typo fix - '''Uplifted to 38''' | ||
Some edge-condition crash fixes | Some edge-condition crash fixes | ||
===Signaling:=== | ===Signaling:=== | ||
{{Bug|1096795}}: Put a=rtcp in SDP when gathering ends. | |||
{{Bug|1146529}}: fix preferred_codec (only used for debugging) and HW H.264 priority on B2G | {{Bug|1146529}}: fix preferred_codec (only used for debugging) and HW H.264 priority on B2G | ||
{{Bug|1149838}}: We should not suppress negotiationneeded before the first offer/answer exchange | {{Bug|1149838}}: We should not suppress negotiationneeded before the first offer/answer exchange - negotiationneeded event fires also for initial negotiation, not only for re-negotiations. | ||
{{Bug|1149298}}: null candidate never fires on pc.onicecandidate (regression) | {{Bug|1149298}}: null candidate never fires on pc.onicecandidate (regression) - '''Uplifted to 39''' | ||
{{Bug|1161619}}: fix leaks/races in StatsQueries in edge cases | {{Bug|1161619}}: fix leaks/races in StatsQueries in edge cases | ||
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{{Bug|1161136}}: JsepSessionImpl does not remove mid from bundle group when rejecting an m-section in some cases | {{Bug|1161136}}: JsepSessionImpl does not remove mid from bundle group when rejecting an m-section in some cases | ||
{{Bug| | {{Bug|1164061}}: TMMBR is now enabled for WebRTC. We will be putting it behind pref when we uplift the patch on this bug by or before May 22nd. | ||
{{Bug|1153056}}: about:webrtc blanks whenever allocated PeerConnections goes to zero - '''Uplifted to 38''' | |||
{{Bug| | {{Bug|1152093}}: Made SDP codec name comparisons case-insensitive (regression) - '''Uplifted to 38''' | ||
===WebAudio=== | ===WebAudio=== | ||
{{Bug|926838}}: Import new FFT kernel for ARM; major performance improvement | {{Bug|926838}}: Import new FFT kernel for ARM; major performance improvement | ||
{{Bug|1094764}}: Implement suspend and resume | {{Bug|1094764}}: Implement suspend and resume - Provides major power/CPU use improvements, especially on mobile | ||
Provides major power/CPU use improvements, especially on mobile | |||
{{Bug|1140448}}, {{Bug|1027864}}, etc: Various WebAudio performance improvements | {{Bug|1140448}}, {{Bug|1027864}}, etc: Various WebAudio performance improvements | ||
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{{Bug|1152359}}: Fix bug in doppler calculations | {{Bug|1152359}}: Fix bug in doppler calculations | ||
{{Bug| | {{Bug|1157137}}: Reduce delay/improve performance in ScriptProcessorNodes | ||