Media/WebRTC/ReleaseNotes/54
Contents
Firefox 54 WebRTC/WebAudio Release Notes:
Full listing of all WebRTC & WebAudio bugs marked as Fixed in Firefox 54:
WebRTC bugs: Bugzilla search for WebRTC related bugs marked Fixed in Firefox 54
WebAudio bugs: Bugzilla search for WebAudio bugs marked Fixed in Firefox 54
Noteworthy Changes:
Full-duplex audio is now turned on for all primary supported platforms by default:
- bug 1333438 Pref on full-duplex for Android
- bug 1325577 Use aggregate devices API in audiounit backend
- This means that issues with drift when using full-duplex on Mac should be solved, and full-duplex will be the default on Mac. (It became the default in linux a number of releases ago, and the default on Windows in 53)
bug 1335939 Pref on ICE TCP support
bug 1340718 If a audio device (Citrix) disappears and fails on re-open attempts, output can be hung until restart You never want to leave the audio system hung, and this fixes a problem for a major user of 52ESR. (Fix uplifted to 53 and 52ESR)
bug 1320170 Screensharing previews are broken in current nightlies
bug 1300665 Missing support for RTP toffset, abs-time header extensions Note: we did not yet add support for 3gpp:video-orientation
bug 1219468 Replace PRLogModuleInfo usage with LazyLogModule in webrtc This means that all logging can be done through MOZ_LOG (NSPR_LOG_MODULES is no longer needed); see [Media/WebRTC/Logging]
Bug tickets fixed in Firefox 54 that affect WebRTC or Web Audio (full list):
Audio/Video:Cubeb :
bug 1333438 Pref on full-duplex for Android
bug 1337328 Uplift cubeb to revision 927877
bug 1332937 Turn libcubeb's XASSERT into a synonym for MOZ_ASSERT or MOZ_RELEASE_ASSERT
bug 1337805 Update cubeb from upstream to 21e96ac
bug 1339816 Update cubeb from upstream to 8977c13b (uplifted to 53)
bug 1342212 Include date of commit in cubeb README_MOZILLA
bug 1343780 Update cubeb from upstream to 0753297
bug 1342363 Update cubeb from upstream to 25b593f (uplifted to 53)
bug 1325577 Use aggregate devices API in audiounit backend
Audio/Video:GMP (Gecko Media Plugin):
bug 1321871 Replace use of opens and bridges in GMP protocols with endpoints
bug 1334079 [Static Analysis][Resource leak] In function ClearKeySessionManager::CreateSession
bug 1332530 Remove GMP device binding
bug 1316650 7,900 instances of "CDM returned NoKeyErr" emitted from dom/media/gmp/GMPCDMProxy.cpp during linux64 debug testing
bug 1337518 Remove PCrashReporter use from GMP
bug 1341135 Rename Log() in WidevineUtils.h to CDM_LOG
bug 1341138 Move LogToConsole from GMPCDMProxy to GMPUtils
bug 1341497 Move WidevineBuffer and WidevineDecryptedBlock out into WidevineUtils.h
bug 1341441 GMPVideoDecoderParent.cpp should include nsPrintfCString.h
bug 1337778 Widevine plugin crashes when opening Amazon videos in background tabs
bug 1342822 Widevine throughput limiting causes MacOS playback to fail (uplifted to 53)
Audio/Video:MediaStreamGraph (MSG):
bug 1336945 Modernize MSG/GraphDriver logging
bug 1340718 If a audio device (Citrix) disappears and fails on re-open attempts, output can be hung until restart (uplifted to 53)
Audio/Video:Media Recording:
bug 1332619 VP8TrackEncoder shortens the encoded track's duration when skipping frames (uplifted to 53)
bug 1333341 VideoTrackEncoder is subject to an accumulating rounding error (uplifted to 53)
bug 1335066 Fix logging in TrackEncoder.cpp
bug 1331839 Intermittent dom/media/test/test_mediarecorder_bitrate.html | seek-short.webm encoded@1000000=6380 > encoded@100000=6041 (uplifted to 53)
Core (General) WebRTC:
bug 1219468 Replace PRLogModuleInfo usage with LazyLogModule in webrtc
bug 1335515 MediaEngine.h fails to compile with --disable-webrtc
bug 1333686 Fix compiler warnings and enable warnings-as-errors in webrtc/signaling
bug 1334840 We can Reconfigure the send codec before we create the SendStream (uplifted to 53)
bug 1335250 nightly 20170130 compile fails with webrtc disabled MediaEngine.h:229:28: error: reference to ‘ipc’ is ambiguous
bug 1335075 RTCCertificate.cpp: output truncated before the last format character
bug 1338000 error: no 'float webrtc::SincResampler::Convolve_NEON(const float*, const float*, const float*, double)' member function declared in class 'webrtc::SincResampler' (uplifted to 53)
bug 1337468 RID RTP header extensions never gets send
bug 1333110 Remove dom/media/standalone
bug 1340054 RID extmap checks in sdputils.js assume RID is the only extension in use
bug 1301286 Intermittent dom/media/tests/mochitest/test_peerConnection_simulcastOffer.html | Error in test execution: Error: Timed out waiting for frames timeout/<@... (uplifted to 53)
bug 1332516 Intermittent browser/base/content/test/webrtc/browser_devices_get_user_media_unprompted_access_tear_off_tab.js | expected notification getUserMedia:response:allow - Got undefined, expected 1 (uplifted to 53)
bug 1332622 Remove MOZILLA_INTERNAL_API code from webrtc
bug 1342388 'Clear History' and 'Clear Log' at about:webrtc not localizable (hardcoded English strings)
bug 1339244 Code to set the minimum bitrate estimate to Try runs in webrtc was lost in the 49 update (uplifted to 53)
bug 1337525 Add mochitests for WebRtc inbound-rtp stats
bug 1332402 Shutdown abort after using BrowserStack, with fatal "Assertion failure: !mThread (SingletonThreads should be Released and shut down before exit!), at media/mtransport/nr_socket_prsock.cpp:181"
bug 1336329 media/webrtc/ fails to build with --enable-warnings-as-errors on Tier3 platforms (uplifted to 53)
bug 1337777 Audio/video on Ciscospark is not working (uplifted to 53)
WebRTC:Audio/Video:
bug 1331648 Only one event is registered when plugging / unplugging camera on Ubuntu for mediaDevices.ondevicechange
bug 1332664 media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h:20: fatal error: 'vpx/svc_context.h' file not found (uplifted to 53)
bug 1320170 Screensharing previews are broken in current nightlies
bug 1341409 Advanced constraints stopped working in FF50
bug 1330350 The "mediaDevices.ondevicechange" devices counter is faulty (uplifted to 53)
WebRTC:Networking:
bug 1333185 [e10s] Default route detection fails to connect for IPv6
bug 1334682 Start trickle grace timeout only when remote candidates are available
bug 1160558 Update SDP for datachannels to match draft 21
bug 1334265 'find' called with a string literal consisting of a single character; consider using the more effective overload accepting a character
bug 1325536 Add ice telemetry to begin replacing stats provided by Hello calls
bug 1317946 [e10s] Zero size UDP packets close sockets
bug 1335939 Pref on ICE TCP
bug 1336182 Need telemetry for DTLS handshake time
bug 1338273 Field "stats.stun_retransmits" is uninitialized
bug 1339270 Upstream webrtc code doesn't add padding packets to retransmission history (Issue 7143)
bug 1300665 Missing support for RTP toffset, abs-time and 3gpp:video-orientation
bug 1340030 Remove necko_standalone lib (netwerk/standalone/)
bug 1338696 Crash in nr_stun_server_process_request (probably caused by failure in nr_stun_message_create2) (uplifted to 53)
bug 1341995 WebRTC: payload type of RED in stream is fixed as 122, which is not match the one in SDP.
bug 1342727 TIAS issues with the webrtc49 update (FF53+)
bug 1338384 TURN responses with bandwidth attr result in failure (uplifted to 53)
bug 1341374 Intermittent TEST-UNEXPECTED-TIMEOUT | dom/media/tests/mochitest/test_peerConnection_basicAudioNATSrflx.html | application timed out after 330 seconds with no output
bug 1340734 NAT simulator doesn't detect the use of TLS properly anymore (uplifted to 53)
WebRTC:Signaling:
bug 1333002 Fix a misleading indentation in PeerConnectionCtx.cpp (uplifted to 53)
bug 1332826 Return InvalidStateError when calling addIceCandidate() too early
bug 1322358 All media lines get disabled in answer SDP when all m-lines in offer are inactive
bug 1341162 Fix -Wunreachable-code-return warning in webrtc/signaling
bug 1336507 Local relay transport is not properly displayed on about:webrtc (uplifted to 53)
