Confirm
654
edits
Changes
initial commit
=Firefox 45 WebRTC/WebAudio Release Notes:=
==Full listing of all WebRTC/WebAudio bugs marked as Fixed in Firefox 45:==
'''''WebRTC bugs:'''''
[http://mzl.la/1OEoKVm Bugzilla search for WebRTC bugs marked Fixed in Firefox 45]
'''''WebAudio bugs:'''''
[http://mzl.la/1OEoHZN Bugzilla search for WebAudio bugs marked Fixed in Firefox 45]
== Noteworthy Changes: ==
===<big>'''IMPORTANT: New Delay-Agnostic AEC'''</big>:===
'''We've replaced the AEC we were using with the new Delay-Agnostic AEC. It is currently enabled by a pref. If we hear of regressions, we may pref off before Fx45 goes to Release. (See {{Bug|1226347}}.)'''
'''''Other noteworthy changes:'''''
* Updated webrtc.org code to branch 43 (See {{Bug|1198458}}.)
==Bug tickets fixed in Firefox 45 (full list):==
===Core (General) WebRTC:===
{{Bug|1175609}}: onnegotiationneeded fires synchronously
{{Bug|1190574}}: Test harness functions like test.chain.replace aren't asserting inputs, hiding bad tests.
{{Bug|1198458}}: Update WebRTC code to webrtc.org stable branch 43
{{Bug|1215769}}: TrackIDs of tracks in MediaStreams captured from a media element change when looping
{{Bug|1219480}}: Replace PRLogModuleInfo usage with LazyLogModule in dom/media
{{Bug|1219566}}: Add aarch64 macro to webrtc/trunk/build/build_config.h
{{Bug|1220679}}: JSAPI handling in nsContentPermissionProxy::Allow is busted
{{Bug|1227481}}: [Static Analysis][Result of operation is garbage or undefined] Function OpenCoreDebugFiles from media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core.c
{{Bug|1227708}}: Differentiate ICE_SUCCESS_RATE telemetry for loop vs webrtc
{{Bug|1227781}}: Crash creating RTCPeerConnection
{{Bug|1231106}}: Check PlatformThreadId is non-zero before returning on BSDs as well (for --enable-debug)
===Audio/Video:===
{{Bug|800564}}: In libyuv, MJPGToI420() throws away the return code in many cases
{{Bug|911450}}: sndio webrtc audio backend
{{Bug|1193075}}: Allow setting & changing the viewport on gUM tab sharing.
{{Bug|1202087}: WebRTC screen capture partially broken on windows 10
{{Bug|1207546}}: Integrate WebRTC with audio channels
{{Bug|1211656}}: GUM Constraints for screen sharing don't affect stream resolution
{{Bug|1218799}}: MediaEngine::Shutdown() should not be called on the main thread
{{Bug|1219711}}: Disabling/muting Media tracks no longer works
{{Bug|1226146}}: Fix sndio backend after webrtc 43 landing in 1198458
{{Bug|1226347}}: Cherry-pick Delay-Agnostic AEC from later upstream versions of webrtc
{{Bug|1227407}}: Correct a logic error in camera code that could be causing a crash
{{Bug|1228788}}: mozGetUserMedia — FATAL ERROR: AsyncShutdown timeout (Media shutdown: blocking on media thread)
{{Bug|1229413}}: Remove MediaManager's xpcom-will-shutdown observer
{{Bug|1229926}}: Attempting to start a "browser" (aka tab) share with fake streams gives "OverconstrainedError: Constraints could be not satisfied., line 0"
===Networking:===
{{Bug|1006809}}: Bring triggered check behavior up-to-date with final ICE RFC
{{Bug|1125947}}: Fix -Wunreachable-code warning in netwerk/sctp/datachannel/DataChannel.cpp
{{Bug|1192403}}: ice_unittest reports error 13 for TCP sockets
{{Bug|1208278}}: ICE TCP connections time out after 1.6s
{{Bug|1219557}}: Only pair addresses from 1918 ranges with others from the same range
{{Bug|1220441}}: ice unit tests TestStun[Tcp]ServerTrickle are at least unreliable
===Signaling:===
{{Bug|1192813}}: WebRTC SDP sets 0.0.0.0 in c line with multiple NICs
{{Bug|1220043}}: Building WebrtcGlobalInformation.cpp causes C1001 error on VS2015 x64
{{Bug|1221837}}: sdp without rtpmap for static codec IDs => "Failed to negotiate codec details for all codecs" error.
{{Bug|1223160}}: Create a SDP file reader for fuzzing
{{Bug|1231196}}: signaling_unittests hits an assertion in PeerConnectionImpl.cpp
-----
===WebAudio:===
{{Bug|930257}}: AudioBufferSourceNode and OscillatorNode don't dispatch ended event when OfflineAudioContext completes soon after source end
{{Bug|1094925}}: Web Audio ChannelMergerNode ignores inputs from GainNodes with zero gain
{{Bug|1171436}}: Support at least 8192 elements for PeriodicWave
{{Bug|1214239}}: [Web Audio] StereoPannerNode ignores input gain when pan is 0
{{Bug|1217625}}: suspend inactive AudioNodeStreams
{{Bug|1220037}}: corrupted convolution from out-of-position nyquist DFT coefficient
{{Bug|1220041}}: some optimizations for copying libav FFT data
{{Bug|1220042}}: make AlignedTArray base class inheritance private
{{Bug|1221830}}: double maximum realtime convolver stage size
{{Bug|1221831}}: use initial input buffer offset to control when convolver stages perform their FFT
{{Bug|1221833}}: remove direct convolution stage; instead use largest initial FFT stage possible
{{Bug|1221836}}: return output pointer from FFTConvolver::process() to save a buffer copy
{{Bug|1221855}}: Crash [@ mozilla::dom::ScriptProcessorNodeEngine::SizeOfExcludingThis const ]
{{Bug|1222202}}: "ASSERTION: QueryInterface needed" with web audio memory report
{{Bug|1222405}}: Build band limited tables lazily in PeriodicWave
{{Bug|1223520}}: "Assertion failure: mOutputBuffer[halfSize].i == 0" with NaN buffer
{{Bug|1224022}}: Intermittent test_WebAudioMemoryReporting.html | Non-zero usage for explicit/webaudio/audio-node/ScriptProcessorNode/dom-nodes
{{Bug|1224102}}: reduce the size of the final FFT convolution stage if possible
{{Bug|1225003}}: Crash [@ mozilla::WebAudioDecodeJob::SizeOfExcludingThis]
{{Bug|1227411}}:Add some logging of Web Audio API use
{{Bug|1230692}}: "Assertion failure: !Failed(), at ErrorResult.h:103" - mozilla::WebAudioDecodeJob::OnFailure
-----
===Audio/Video:Cubeb :===
{{Bug|1225703}}: Update in-tree libcubeb
-----
===Audio/Video:GMP (Gecko Media Plugin):===
{{Bug|1220326}}: [EME] Wipe stack after generating machine ID on MacOSX
{{Bug|1224442}}: Armor GMP Parent against late Shmem messages from the child
{{Bug|1228215}}: [EME] Segregate GMP storage by GMP
-----
===Audio/Video:MediaStreamGraph (MSG):===
{{Bug|1214018}}: [EME] Generate machine ID on MacOSX
{{Bug|1216417}}: Resuming a paused media element playing a stream does not play audio
{{Bug|1218593}}: Dialer touch tones sounds intermittently stop playing audio in Dialer.
{{Bug|1223655}}: Intermittent "###!!! ASSERTION: Track not found" in MediaEngineDefaultAudioSource::NotifyPull
{{Bug|1223696}}: The MediaElement will not be played after removeTrack() and addTrack() back.
{{Bug|1228559}}: Don't allow media stream cross different type MSG
-----
===Audio/Video:Media Recording: ===
{{Bug|1154213}}: Same timestamps in audio recorded with media recorder api (in audio+video mode)(with vorbis)
{{Bug|1225327}}: Crash [@ mozilla::dom::MediaRecorder::Session::SizeOfExcludingThis]
==Full listing of all WebRTC/WebAudio bugs marked as Fixed in Firefox 45:==
'''''WebRTC bugs:'''''
[http://mzl.la/1OEoKVm Bugzilla search for WebRTC bugs marked Fixed in Firefox 45]
'''''WebAudio bugs:'''''
[http://mzl.la/1OEoHZN Bugzilla search for WebAudio bugs marked Fixed in Firefox 45]
== Noteworthy Changes: ==
===<big>'''IMPORTANT: New Delay-Agnostic AEC'''</big>:===
'''We've replaced the AEC we were using with the new Delay-Agnostic AEC. It is currently enabled by a pref. If we hear of regressions, we may pref off before Fx45 goes to Release. (See {{Bug|1226347}}.)'''
'''''Other noteworthy changes:'''''
* Updated webrtc.org code to branch 43 (See {{Bug|1198458}}.)
==Bug tickets fixed in Firefox 45 (full list):==
===Core (General) WebRTC:===
{{Bug|1175609}}: onnegotiationneeded fires synchronously
{{Bug|1190574}}: Test harness functions like test.chain.replace aren't asserting inputs, hiding bad tests.
{{Bug|1198458}}: Update WebRTC code to webrtc.org stable branch 43
{{Bug|1215769}}: TrackIDs of tracks in MediaStreams captured from a media element change when looping
{{Bug|1219480}}: Replace PRLogModuleInfo usage with LazyLogModule in dom/media
{{Bug|1219566}}: Add aarch64 macro to webrtc/trunk/build/build_config.h
{{Bug|1220679}}: JSAPI handling in nsContentPermissionProxy::Allow is busted
{{Bug|1227481}}: [Static Analysis][Result of operation is garbage or undefined] Function OpenCoreDebugFiles from media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core.c
{{Bug|1227708}}: Differentiate ICE_SUCCESS_RATE telemetry for loop vs webrtc
{{Bug|1227781}}: Crash creating RTCPeerConnection
{{Bug|1231106}}: Check PlatformThreadId is non-zero before returning on BSDs as well (for --enable-debug)
===Audio/Video:===
{{Bug|800564}}: In libyuv, MJPGToI420() throws away the return code in many cases
{{Bug|911450}}: sndio webrtc audio backend
{{Bug|1193075}}: Allow setting & changing the viewport on gUM tab sharing.
{{Bug|1202087}: WebRTC screen capture partially broken on windows 10
{{Bug|1207546}}: Integrate WebRTC with audio channels
{{Bug|1211656}}: GUM Constraints for screen sharing don't affect stream resolution
{{Bug|1218799}}: MediaEngine::Shutdown() should not be called on the main thread
{{Bug|1219711}}: Disabling/muting Media tracks no longer works
{{Bug|1226146}}: Fix sndio backend after webrtc 43 landing in 1198458
{{Bug|1226347}}: Cherry-pick Delay-Agnostic AEC from later upstream versions of webrtc
{{Bug|1227407}}: Correct a logic error in camera code that could be causing a crash
{{Bug|1228788}}: mozGetUserMedia — FATAL ERROR: AsyncShutdown timeout (Media shutdown: blocking on media thread)
{{Bug|1229413}}: Remove MediaManager's xpcom-will-shutdown observer
{{Bug|1229926}}: Attempting to start a "browser" (aka tab) share with fake streams gives "OverconstrainedError: Constraints could be not satisfied., line 0"
===Networking:===
{{Bug|1006809}}: Bring triggered check behavior up-to-date with final ICE RFC
{{Bug|1125947}}: Fix -Wunreachable-code warning in netwerk/sctp/datachannel/DataChannel.cpp
{{Bug|1192403}}: ice_unittest reports error 13 for TCP sockets
{{Bug|1208278}}: ICE TCP connections time out after 1.6s
{{Bug|1219557}}: Only pair addresses from 1918 ranges with others from the same range
{{Bug|1220441}}: ice unit tests TestStun[Tcp]ServerTrickle are at least unreliable
===Signaling:===
{{Bug|1192813}}: WebRTC SDP sets 0.0.0.0 in c line with multiple NICs
{{Bug|1220043}}: Building WebrtcGlobalInformation.cpp causes C1001 error on VS2015 x64
{{Bug|1221837}}: sdp without rtpmap for static codec IDs => "Failed to negotiate codec details for all codecs" error.
{{Bug|1223160}}: Create a SDP file reader for fuzzing
{{Bug|1231196}}: signaling_unittests hits an assertion in PeerConnectionImpl.cpp
-----
===WebAudio:===
{{Bug|930257}}: AudioBufferSourceNode and OscillatorNode don't dispatch ended event when OfflineAudioContext completes soon after source end
{{Bug|1094925}}: Web Audio ChannelMergerNode ignores inputs from GainNodes with zero gain
{{Bug|1171436}}: Support at least 8192 elements for PeriodicWave
{{Bug|1214239}}: [Web Audio] StereoPannerNode ignores input gain when pan is 0
{{Bug|1217625}}: suspend inactive AudioNodeStreams
{{Bug|1220037}}: corrupted convolution from out-of-position nyquist DFT coefficient
{{Bug|1220041}}: some optimizations for copying libav FFT data
{{Bug|1220042}}: make AlignedTArray base class inheritance private
{{Bug|1221830}}: double maximum realtime convolver stage size
{{Bug|1221831}}: use initial input buffer offset to control when convolver stages perform their FFT
{{Bug|1221833}}: remove direct convolution stage; instead use largest initial FFT stage possible
{{Bug|1221836}}: return output pointer from FFTConvolver::process() to save a buffer copy
{{Bug|1221855}}: Crash [@ mozilla::dom::ScriptProcessorNodeEngine::SizeOfExcludingThis const ]
{{Bug|1222202}}: "ASSERTION: QueryInterface needed" with web audio memory report
{{Bug|1222405}}: Build band limited tables lazily in PeriodicWave
{{Bug|1223520}}: "Assertion failure: mOutputBuffer[halfSize].i == 0" with NaN buffer
{{Bug|1224022}}: Intermittent test_WebAudioMemoryReporting.html | Non-zero usage for explicit/webaudio/audio-node/ScriptProcessorNode/dom-nodes
{{Bug|1224102}}: reduce the size of the final FFT convolution stage if possible
{{Bug|1225003}}: Crash [@ mozilla::WebAudioDecodeJob::SizeOfExcludingThis]
{{Bug|1227411}}:Add some logging of Web Audio API use
{{Bug|1230692}}: "Assertion failure: !Failed(), at ErrorResult.h:103" - mozilla::WebAudioDecodeJob::OnFailure
-----
===Audio/Video:Cubeb :===
{{Bug|1225703}}: Update in-tree libcubeb
-----
===Audio/Video:GMP (Gecko Media Plugin):===
{{Bug|1220326}}: [EME] Wipe stack after generating machine ID on MacOSX
{{Bug|1224442}}: Armor GMP Parent against late Shmem messages from the child
{{Bug|1228215}}: [EME] Segregate GMP storage by GMP
-----
===Audio/Video:MediaStreamGraph (MSG):===
{{Bug|1214018}}: [EME] Generate machine ID on MacOSX
{{Bug|1216417}}: Resuming a paused media element playing a stream does not play audio
{{Bug|1218593}}: Dialer touch tones sounds intermittently stop playing audio in Dialer.
{{Bug|1223655}}: Intermittent "###!!! ASSERTION: Track not found" in MediaEngineDefaultAudioSource::NotifyPull
{{Bug|1223696}}: The MediaElement will not be played after removeTrack() and addTrack() back.
{{Bug|1228559}}: Don't allow media stream cross different type MSG
-----
===Audio/Video:Media Recording: ===
{{Bug|1154213}}: Same timestamps in audio recorded with media recorder api (in audio+video mode)(with vorbis)
{{Bug|1225327}}: Crash [@ mozilla::dom::MediaRecorder::Session::SizeOfExcludingThis]
