Media/WebRTC/ReleaseNotes/53

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Firefox 53 WebRTC/WebAudio Release Notes:

Full listing of all WebRTC & WebAudio bugs marked as Fixed in Firefox 53:

WebRTC bugs: Bugzilla search for WebRTC related bugs marked Fixed in Firefox 53

WebAudio bugs: Bugzilla search for WebAudio bugs marked Fixed in Firefox 53

Noteworthy Changes:

  • bug 1250356 Updated core webrtc.org code to stable branch 49
    • Required major rewrite of interfaces for for video
  • bug 1331498 Updated libvpx to 1.6.1 (major update)
    • Our thanks to Johann Koenig of Google for contributing the patches to update it and improve the update and build process!
  • bug 1056934 TURN/TLS is now supported
  • bug 1231848 and bug 1330676 etc - MediaRecorder improvements - supports >30fps, significantly improved encoding quality
  • bug 1221574 Full-duplex audio for android is landed and available; currently pref'd on by default in 54
  • bug 1308481/bug 1320101/bug 1330318 Resolved TIAS bitrate limit issues after renegotiation
  • bug 1322503 Fixed RTCStatsType to be spec-compatible (missing hyphens in most enum names)

Bug tickets fixed in Firefox 53 that affect WebRTC or Web Audio (full list):

Audio/Video:CubebĀ :

bug 1310224 OOM crash in output-only scenario on Windows/WASAPI

bug 1314514 gtestify the cubeb unit tests

bug 1317234 audiounit_stream_init() sometimes gets stuck forever on OSX 10.10

bug 1318619 Update cubeb from upstream to 7f74039f92

bug 1319623 Valgrind reports uninitialized memory use in pulse_stream_set_panning running cubeb.run_panning_volume_test_short gtest

bug 1221574 Write a full-duplex Android OpenSL ES cubeb backend

bug 1331869 Update cubeb from upstream to d96e35f02d

bug 1326176 Crash in jemalloc_crash | arena_dalloc_small | je_free | `anonymous namespace::wasapi_stream_render_loop

bug 1332354 Allow enabling cubeb log by flipping a pref

bug 1332905 Crash in abort | `anonymous namespace::wasapi_stream_init


Audio/Video:GMP (Gecko Media Plugin):

bug 1273372 [EME] Crash in mozilla::gmp::GMPChild::ProcessingError

bug 1313258 Intermittent FATAL ERROR: AsyncShutdown timeout in xpcom-will-shutdown Conditions: [{"name":"MediaShutdownManager: shutdown","state":"(none)","filename":"/builds/slave/autoland-m64-d-000000000000000/build/src/dom/media/MediaShutdownManager.cpp","lineNumbe

bug 1316215 Convert GMPService to MozPromise

bug 1317473 GMPServiceParent::AddOnGMPThread(path) can't handle a mixture of dir separators in path

bug 1317822 Move GMPCrashHelper into its own file

bug 1318965 Convert gmp-clearkey to use Chromium ContentDecryptionModule8 interface

bug 1319197 Remove audio decoding from gmp-clearkey

bug 1325185 Fix operator precedence in GMPUtils' ToHexString()

bug 1331829 Remove GMP async shutdown

bug 1332149 Don't expose plugin-container or sandbox vouchers to GMPs.


Audio/Video:MediaStreamGraph (MSG):

bug 1305949 Do some cleaning around direct listeners and video sinks

bug 1314886 Intermittent dom/media/test/test_streams_element_capture_reset.html | checking vout has not ended - got true, expected false

bug 1319445 Disable direct audio listeners for RTCPeerConnection with full duplex

bug 1321235 Can not remove a stopped media track using removeTrack on Firefox version 52 onwards

bug 1329075 Null-deref in [@ HTMLMediaElement::StreamCaptureTrackSource::GetCORSMode]

bug 1330212 Intermittent dom/media/tests/mochitest/test_getUserMedia_mediaStreamTrackClone.html | application crashed [@ mozilla::CycleCollectedJSContext::ProcessMetastableStateQueue]

bug 1330696 Add profiler labels to Canvas frame capturing

bug 1330919 Set proper timestamps on video frames from canvas.captureStream()


Audio/Video:Media Recording:

bug 1231848 CanvasStream + MediaRecorder does not create variable framerate video

bug 1322745 VP8TrackEncoder::GetSourceSurface can be improved

bug 1326311 The Media Recorder API crash when we do a lot of stop/start and we destroy the Session. It also leaks a Listener

bug 1330676 MediaRecorder's CBR setting causes really bad perceived video quality

bug 1330918 Make MediaRecorder use timestamps for video

bug 1332584 MediaRecorder doesn't record the last frame of a video track

bug 1332585 Add some VideoTrackEncoder unit tests

bug 1332598 Improve logging of VP8TrackEncoder


Core (General) WebRTC:

bug 1197021 Remove last remnants of already retired backwards compatible RTCOfferOptions

bug 1250356 Update WebRTC code to webrtc.org stable branch 49

bug 1263312 Have addIceCandidate, setLocalDescription et al take dictionary (spec update)

bug 1308481 TIAS bitrate limitation does not work when resolution changes

bug 1310355 Improve resiliency for using webrtc permission hooks

bug 1313966 RTCSessionDescription interface doesn't match spec

bug 1318163 Remove unimplemented and non-spec getStreamById from RTCPeerConnection.

bug 1319268 Extend WebRTC ICE Telemetry probes

bug 1319542 Update pc.createDataChannel's RTCDataChannelInit dict to spec.

bug 1320891 Make some webrtc tests build with gcc 7.0 and --enable-warnings-as-errors

bug 1322274 Overhaul PeerConnection.js with modern JavaScript

bug 1322338 Point out lack of STUN/TURN server in ICE failure message

bug 1322503 Firefox's RTCStatsType is not spec-compatible (missing hyphens in most enum names)

bug 1322659 Too many STUN/TURN servers don't help with conectivity

bug 1323079 Intermittent dom/media/tests/mochitest/test_peerConnection_trackDisabling_clones.html | Test timed out.

bug 1323095 Add deprecation warnings to callback-based pc.getStats()

bug 1326011 webrtc/trunk/webrtc/base/platform_thread.cc:44:47: error: cast from 'pthread_t {aka pthread*}' to 'pid_t {aka int}' loses precision [-fpermissive]

bug 1328440 Legacy PeerConnection.getStats should return a legacy stats compatible object

bug 1329193 More overhaul PeerConnection.js with modern JavaScript

bug 1329762 Strengthen deprecation warning of legacy PeerConnection.getStats

bug 1330091 Renegotiation doesn't actually change the codec configuration after 49 update landing

bug 1331158 Renegotiation doesn't actually change the receive codec configuration after 49 update landing


WebRTC:Audio/Video:

bug 1223692 Update libvpx to 1.6.0

bug 1270572 While page already has a live track, getUserMedia should allow un-prompted re-access to same device.

bug 1277037 MOZ_CRASH: Could not start cubeb stream for MSG.

bug 1306359 Stop using Scoped.h NSS types in RTCCertificate.(cpp|h)

bug 1307754 Webrtc. FF Beta 50.0b4. No signal from microphone.

bug 1313758 WebRTC getUserMedia mediaSource 'browser' broken: Cause: webrtcUI.jsm (listScreenShareDevices) ==> getString() NS_ERROR_FAILURE

bug 1317660 Fix CID 1394336: Resource leaks in TestAudioPacketizer.cpp

bug 1317714 port mediaconduit_unittests to xul gtest

bug 1318132 Coverity issue in CamerasChild

bug 1319566 Crash in nsTArray_Impl<T>::DestructRange | nsTArray_Impl<T>::RemoveElementsAt | mozilla::MediaEngineSource::Deallocate

bug 1321609 PeerConnection tests sometimes expect media flow on received tracks that ended due to renegotiation

bug 1326288 VP9 decoding broken by webrtc.org 49 update - YCbCr pointers are off

bug 1326386 webrtc.org 49 update mismerged away a mochitest

bug 1326442 VideoConduit code should simply reconfigure the VideoSendStream when possible on a configuration change

bug 1326463 Build failure in webrtc with sndio after bug 1250356

bug 1328330 vp8 error concealment should be removed

bug 1329562 Remove WebRTC checks for Windows Vista

bug 1329976 getUserMedia(audio, video) when already capturing audio fails

bug 1329922 [DTMF] Tones are not heard when duration is set to lowest (70)

bug 1320101 Setting b=TIAS caps us at 2kbps

bug 1330318 Setting b=TIAS caps us at 2kbps

bug 1331498 Update libvpx to 1.6.1

bug 1332139 Drop ifdefs in webrtc vp9 interface code for handling old versions of libvpx


WebRTC:Networking:

bug 1056934 Support TURN TLS in WebRTC

bug 1266667 [e10s] active ICE TCP fails because multiple connections with identical TCP SRC port fail

bug 1316261 System CA's cause big and fragmented DTLS messages

bug 1318180 Cannot createOffer after network change

bug 1318803 Provide IPC reason for STUN filter blocking

bug 1320150 ICE consent signals connected too earlier for non bundled transports

bug 1321628 add ice restart and rollback counts to about:webrtc

bug 1322438 Change ICE failed message depending on presence of relay candidates

bug 1322546 Cannot compile nrappkit with WINVER=0x0600 or later

bug 1323998 Stop using Scoped.h NSS types in dtlsidentity.(cpp|h) and nricectx.cpp

bug 1324608 RtpStreamId RTP header extension indicates incorrect header length

bug 1324995 Crash in jemalloc_crash | je_free | r_free | stun_get_win32_addrs

bug 1329932 Remove unneeded nsXPCOMGlue includes

WebRTC:Signaling:

bug 1193731 onicegatheringstatechange doesn't work

bug 1271681 Move SDP-related test cases from signaling_unittests to sdp_unittest

bug 1271682 Move JSEP-related tests from signaling_unittests to jsep_session_unittest

bug 1316886 Port sdp_file_parser unit test to standalone binary

bug 1316888 Port sdp_unittest to xul gtest

bug 1317009 Port jsep_session_unittest and jsep_track_unittest to xul gtest

bug 1317044 Intermittent mediapipeline_unittest | test failed with return code -139 due to MediaPipelineTest.TestAudioSendNoMux failure

bug 1317726 sdp_file_parser still depends upon xpcom glue

bug 1317764 --disable-tests fails to build: media/webrtc/signaling/fuzztest/sdp_file_parser.cpp:12:25: fatal error: gtest/gtest.h: No such file or directory

bug 1322707 Stop building signaling_unittest and mediapipeline_unittests

bug 1328142 Webrtc.org 49 update broke simulcast

bug 1328429 When no redundant encodings are specified for RED in offer, do not output "empty" fmtp line for RED payload type in answer

bug 1307461 Intermittent mediapipeline_unittest | test failed with return code 1 due to MediaPipelineTest.TestAudioSendMux failure