Changes

===Audio/Video:Cubeb :===

{{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