Media/WebRTC/ReleaseNotes/47

From MozillaWiki
Jump to navigation Jump to search
The printable version is no longer supported and may have rendering errors. Please update your browser bookmarks and please use the default browser print function instead.

Firefox 47 WebRTC/WebAudio Release Notes:

Full listing of all WebRTC/WebAudio bugs marked as Fixed in Firefox 47:

WebRTC bugs: Bugzilla search for WebRTC related bugs marked Fixed in Firefox 47

WebAudio bugs: Bugzilla search for WebAudio bugs marked Fixed in Firefox 47

Noteworthy Changes:

  • Recording:
    • Periodic blinking is fixed (See bug 1234230)
    • Fire error if mime type is unsupported (See bug 963238)
    • Recorded videos of longer than a few seconds would not play (See bug 1213775)
  • Clear screen and other privacy improvements to about:webrtc (See bug 1209252)
  • Make RTCRtpSender.setParameters return a promise. (See bug 1247622)
  • Add support for RTCRtpEncodingParameters.scaleResolutionDownBy (See bug 1244913)
  • Increase buffer sizes on older Windows to avoid packet loss at very high (HD) bitrates (bug 1217677); allow Jumbo ethernet frames when in e10s mode (bug 1252755)
  • MediaStream in video element doesn't guarantee video size on "loadedmetadata" (See bug 1240478)
  • Firefox always sends full band Opus audio even though the negotiation specifies wideband audio (bug 1249098); support negotiation of mono Opus (bug 818618). Also fixed no audio on browser side when Opus payload types in offer and answer do not match (bug 1221473).
  • High CPU usage on some systems when IPV6 interfaces are configured (See bug 1224845)

Bug tickets fixed in Firefox 47 (full list):

Core (General) WebRTC:

bug 1209252: about:webrtc should have a clear screen button

bug 1217677: Video Packet drop when sending video at 1920 Kbps

bug 1221786: about:webrtc includes calls from private browsing mode even after last pb session is closed

bug 1241064: getStats API returns 0 values for some attributes for audio

bug 1241321: No RTCP stats for audio streams

bug 1242199: Intermittent test_peerConnection_simulcastOffer.html | Test timed out

bug 1244913: Add support for RTCRtpEncodingParameters.scaleResolutionDownBy

bug 1247622: Make RTCRtpSender.setParameters return a promise.

bug 1249860: Add simulcast test for resolution scaling

bug 1250990: Make RTCRtpEncodingParameters.maxBitrate and scaleResolutionDownBy work with H.264 in unicast at least.

bug 1251714: use UniquePtr instead of ScopedDeletePtr in media/

bug 1251737: Remove remaining references to MOZILLA_XPCOMRT_API from media

bug 1252163: TestNrSocketTest::CheckTcpConnectivity does not wait for readable/writeable when necessary

bug 1252699: Build error on ubuntu 15.04 with an arm-hf toolchain

bug 1252901: don't use ScopedFreePtrTraits in MediaEngineWrapper.h


WebRTC:Audio/Video:

bug 1177242: Video device access needs to re-verify UX permissions

bug 1237816: Crash in MediaStreamGraph GraphDriver::SetGraphTime when accepting separate video then audio gum calls

bug 1240478: MediaStream in video element doesn't guarantee video size on "loadedmetadata"

bug 1242991: aurora branch fails to build against system libvpx after this week's merge

bug 1243607: Re-enable webrtc bandwidth prefs (to override adaptive bandwidth settings)

bug 1245216: full_duplex getUserMedia audio wasn't getting properly plumbed through to MediaStreams

bug 1248335: Configure's version check of libvpx is not enough

bug 1249098: Firefox always sends full band Opus audio even though the negotiation specifies wideband audio

bug 1249244: Fix b2g bustage introduced by bug 1177242

bug 1250540: Screen share stops working in Firefox Nightly

bug 1251357: MediaStreamTrack.stop does not stop the track

bug 1252737: WebrtcMediaCodecVP8VideoEncoder::InitEncode is invalid function define


WebRTC:Networking:

bug 1214269: WebRTC file transfer, data channel stale

bug 1224845: High CPU usage while using WebRTC

bug 1231971: Teach mtransport NAT simulator to support e10s (and also TCP)

bug 1231973: Hook the mtransport NAT simulator up to the pref system, to allow it to be used in browser testing

bug 1239870: Convert mtransport CppUnitTests to mozilla gtests

bug 1244338: crash in nr_ice_ctx_destroy_cb

bug 1246011: SDP answer chooses dynamic payload type without rtmap over lower priority codec

bug 1246363: relay_only option can cause major confusion

bug 1246925: [Static Analysis][Explicit null dereferenced] In function RefPtr<NrIceCtx> NrIceCtx::Create

bug 1247536: Fix -Wunreachable-code warning in media/mtransport

bug 1247547: WEBRTC_ICE_ADD_CANDIDATE* reports errors twice

bug 1247619: Warn app devs when they try to use OAuth for STUN

bug 1251214: R_WOULDBLOCK can cause STUN client failure in nr_stun_client_send_request

bug 1252171: TestNrSocket does not update last-used timestamps for TCP port mappings

bug 1252585: TestStunServer::Reset does not clear received_ct_

bug 1252755: e10s max packet size doesn't allow jumbo frames

bug 1252777: Disable passive ICE TCP sockets until bug 1179345 lands

bug 1253198: Add common prefix to all ice unit tests


WebRTC:Signaling:

bug 818618: Implement Opus stereo/mono codec configuration and SDP handling

bug 1221473: No audio on browser side when Opus payload types in offer and answer do not match

bug 1241153: Find some way to make setParameters and GMP init less fragile

bug 1246501: jsep_track_unittest missing LockImpl signals

bug 1247656: Validate that a reoffer does not reorder m-sections, or change their media type

bug 1248770: sdp_unittests fails when libc++ is used

bug 1249369: use UniquePtr instead of nsAutoArrayPtr in MediaPipeline.cpp

bug 1252073: Uninitialised value uses in mozilla::EncodingConstraints::operator==

WebAudio:

bug 1158741: Implement a version of omxSP_FFTInv_CCSToR_F32_Sfs in openmax DL's FFT that is not scaled

bug 1241499: Initialize the HRTF database lazily

Audio/Video:Cubeb :

bug 1243275: Update in-tree cubeb to pick up device-change-notification changes and a pulse fix


Audio/Video:GMP (Gecko Media Plugin):

bug 1236680: Result "not reached" crash in mozilla::gmp::GMPChild::ProcessingError()

bug 1246763: Refactor GMP info file parsing into helper class

bug 1246835: Reduce #ifdefs pain in GMPChild.cpp

bug 1247556: Use GMPInfoFileParser in GMPParent::ReadGMPMetaData()

bug 1247763: Remove info file parsing from GMP child process

bug 1249424: remove nsAutoArrayPtr usage from dom/media/gmp/

Audio/Video:MediaStreamGraph (MSG):

bug 1239893: Add gonk SidebandStream handling to DOMHwMediaStream

bug 1247395: use UniquePtr for control messages in MediaStreamGraphImpl

Audio/Video:Media Recording:

bug 963238: Media Recording - If the file's mime type cannot be recorded on the target platform, then fire an onerror callback indicating that the mime type is not supported

bug 1213775: VP8 automatic resizing breaks ffmpeg-based players

bug 1234230: MediaRecorder video blob has blinking when recording both image and audio

bug 1243611: VP8TrackEncoder is not calling vpx_codec_encode correctly on EOS

Retrieved from "https://wiki.mozilla.org/index.php?title=Media/WebRTC/ReleaseNotes/47&oldid=1123377"