[Pkg-voip-commits] [janus] annotated tag upstream/0.2.5 created (now b9659e2)

Jonas Smedegaard dr at jones.dk
Sat Oct 28 01:22:29 UTC 2017


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js pushed a change to annotated tag upstream/0.2.5
in repository janus.

        at  b9659e2   (tag)
   tagging  2aca1b30bb7f4c54269e6ee5f16f9451f391ebc6 (commit)
  replaces  upstream/0.2.4
 tagged by  Jonas Smedegaard
        on  Sat Oct 28 01:48:17 2017 +0200

- Log -----------------------------------------------------------------
Upstream version 0.2.5
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Aleksey Ivanov (1):
      User forward declaration of json_t in plugin.h

Alessandro Toppi (5):
      Check for iceloop status before quitting the loop.
      Merge branch 'master' into fix-descriptors-leak
      Append latest glib.supp to janus-valgrind.supp
      Fix a regression in websocket ACL.
      Merge pull request #1020 from meetecho/fix-ws-acl

Alex (1):
      Fix compilation issue with libavcodec < 55.28

Gábor Tóth (2):
      documentation: add default value of videoroom publishers
      documentation: fix for a settings name

Johan Ouwerkerk (5):
      Decouple `janus.js` from its dependencies.
      Added documentation updates for decoupled dependencies:
      Add support for building JavaScript modules from html/janus.js.
      Integrate support for building JavaScript modules within the build system.
      Document available JavaScript module formats/syntaxes and how to build them.

Jonas Smedegaard (1):
      New upstream version 0.2.5

Lorenzo Miniero (127):
      First integration of VP9 SVC support in the VideoRoom plugin
      Removed excessive verbosity
      Merge branch 'master' into vp9-svc
      Merge branch 'master' into vp9-svc
      Merge branch 'master' into vp9-svc
      Merge branch 'master' into vp9-svc
      First experiments with VP8 simulcasting (Chrome and Firefox, WIP)
      Several simulcast related fixes (third substream, REMB, etc.)
      Use tracks instead of streams in janus.js
      Merge branch 'master' into js-track
      Fixed a couple of typos, and improved a couple of methods
      Initial experiments with VP8 temporal scalability
      Merge branch 'master' into vp8-simulcast
      Experimenting with VP8 simulcasting in VideoRoom plugin as well
      Fixed typo, and enriched VideoRoom specific Admin API info when simulcasting
      Updated reference to webrtc-adapter (4.1.1)
      Fixed SDP munging when simulcasting
      Experiments with automatic simulcast substream changes (EchoTest)
      Reorganized simulcast-related JS code to make integration cleaner
      Reorganized how we munge SDP when simulcasting as a method, and added to createAnswer (but not for Chrome, it breaks video there)
      Added simulcasting support to VideoCall plugin
      Some fixes and enhancements in VideoRoom simulcasting
      Merge branch 'master' into vp8-simulcast
      Added placeholder code for simulcasting in RecordPlay and SIP plugins
      Don't enable simulcasting if the negotiated codec is not VP8
      Version bump for plugins
      Added simulcast support to RTP live mountpoints in the Streaming plugin
      Extended RTP forwarding in VideoRoom plugin to support simulcasting
      Merge branch 'master' into vp9-svc
      Updated plugin to send events at layer changes, and demo to use those (as VP8 simulcasting stuff does)
      Merge branch 'master' into js-track
      Merge branch 'master' into vp8-simulcast
      Merge branch 'master' into vp9-svc
      Merge branch 'master' into vp8-simulcast
      Merge branch 'master' into vp9-svc
      Moved VP9 SVC parsing to utils, so that they can be used in other plugins if needed
      Merge branch 'master' into vp9-svc
      Merge branch 'master' into vp8-simulcast
      Merge branch 'master' into vp8-simulcast
      Merge branch 'master' into vp9-svc
      Bumped to version 0.2.5
      Merge pull request #944 from meetecho/vp8-simulcast
      Aligned the VP9 SVC branch to the simulcast stuff merged in master
      Merge pull request #925 from meetecho/vp9-svc
      Added link to blog post in the VP9 SVC demo
      Merge branch 'master' into js-track
      Reduced verbosity of a few warnings
      Make sure handles are detached immediately when a session is destroyed
      Added ability for VideoRoom listeners to not negotiate support for media the publishers are sending
      Added option for Streaming viewers to subscribe to a subset of a mountpoint media
      Added a new paper to the Janus-related bibliography
      Fixed some typos, leaks, and broken checks
      Added some checks on the availability of an SDP in the SIP plugin
      Merge branch 'master' into js-track
      Implemented way to edit some room properties dynamically in AudioBridge, VideoRoom and TextRoom (see #939)
      Make sure handles are removed right away in case of a session timeout (see #963)
      Merge branch 'master' into js-track
      Fixed typo in AudioBridge docs
      Merge pull request #962 from tmatth/static_mutex_init
      Merge pull request #961 from longsleep/honor-trickle-with-external-stream
      Only do NACKs for a specific medium if they were negotiated
      Merge pull request #967 from tgabi333/patch-5
      Merge pull request #968 from tgabi333/patch-6
      Fixed RTP forwarding not working for video in VideoRoom plugin (fixes #975)
      Merge branch 'master' of github.com:meetecho/janus-gateway
      Don't offer medium to VideoRoom subscribers if recvonly/inactive (fixes #985)
      Updated VP9 SVC demo to use tracks, and bumped adapter to 5.0.1
      Merge pull request #942 from meetecho/js-track
      Added configurable ping/pong to WebSockets transport (see #986)
      Better management of session timeouts and destruction (see #986)
      Only use new WebSockets ping/pong stuff if libwebsockets >= 2.1
      Merge pull request #989 from bebo/rtt-fix
      Removed typo/leftover from stats evaluation in janus.js (fixes #976)
      Merge pull request #971 from uasan/patch-1
      Allow rtpmap/fmtp to be overridden when creating RTSP mountpoints (fixes #972)
      Wrap build date in quotes, when creating version.c (see #990)
      Fixed copy/paste error (see #972)
      Fixed static mutex initialization in HTTP transport, as per recent changes
      Made Record&Play plugin codec-agnostic (was opus/vp8 only before)
      Added text2pcap support for handles traffic
      Added option to truncate packets when dumping via text2pcap
      Added printf format to text2pcap dump method, and session/handle info when dumping
      Added G.722 support to Record&Play and janus-pp-rec
      Added some more video material to the FAQ on Janus in the docs
      Added some more video material to the FAQ on Janus in the docs
      Added index and anchors to the Janus FAQ in the docs
      Fixed the contact link in the Support HTML page
      Merge pull request #994 from mvnrsa/PHPClass
      Notify VideoRoom publishers and subscribers about codecs
      Changed the commented sample values for the WebSockets pingpong (see #988)
      Make sure there's only one a=end-of-candidates per m-line (fixes #995)
      Fixed playout of G.711 and G.722 recordings in Record&Play
      Fixed race condition when starting WebSockets threads and initialized still 0
      Fix for the SSL+WebSockets crashes (see #913)
      Changed check for libwebsockets in configure.ac, to make sure vhosts are supported
      Merge pull request #996 from meetecho/wss-fix
      Fixed typo where we sent the wrong info about video codecs in the Videoroom (fixes #1000)
      Merge pull request #993 from meetecho/text2pcap
      Ignore EINTR for Streaming mountpoints (fixes #1004)
      Fixed leak in Record&Play, and dead code in VideoRoom
      Fixed occasional deadlock when shutting down Janus
      Merge pull request #1005 from meetecho/fix-descriptors-leak
      Ignore ontrack event if there's no stream info
      Changed flags to dlopen to load modules
      Moved websockets ping/pong setup from the vhosts to the main context
      Fixed flags for NACK management after SDP parsing
      Fixed broken stats/NACKs for incoming streams (see #999)
      Merge pull request #991 from cmacq2/janus-js-dependencies
      Move SRTP specific definitions from rtp.h to a new header
      Don't ignore ICMP errors on RTP packets in the SIP plugin
      Fixed typo in comment
      Merge pull request #1024 from meetecho/rtp-srtp-header
      Merge pull request #1027 from mikaelnousiainen/feature/rtp-forwarder-static-config
      Made clearer that the new options added in #1027 are optional
      Fixed typo in rtp.c when resetting sequence numbers (should fix #992 and #1032)
      Changed check when enforcing RTP switching header changes
      Added configurable port range for RTP/RTCP ports in the SIP plugin
      Merge pull request #1031 from tugtugtug/notify_joining
      Merge pull request #1029 from cmacq2/build-js-modules
      Fixed a couple of nits in the new js-modules documentation (see #1029)
      Add js modules to dist target too, if built
      Added configure flag (--enable-all-js-modules) to enable all JS modules
      Ugly hack to fix nits (double dashes) in documentation
      Merge pull request #1036 from op/fix-adapter-dep
      Merge pull request #1016 from ivanovaleksey/plugin-h
      Added minijanus.js project to the resources in the docs
      Additional changes before iterating on hashtables

Marnus van Niekerk (1):
      Add PHPClass to Resources

Mikael Nousiainen (3):
      Add support for static configuration of RTP forwarder in audiobridge plugin
      Mention static RTP forwarder in the documentation
      Allow configuration of static RTP forwarder ID

Pete Hicks (1):
      fix rtcp delay calculation

S.A.N (1):
      Explicit mode non-interleaved

Simon Eisenmann (1):
      Honor trickle configuration when using ext. stream

Steven Tang (3):
      Fix issue #1030 by introducing a new flag to videoroom for joining events
      Add documentation to notify_joining
      Update change based on review and feedback by @lminiero

Tristan Matthews (4):
      mutex: add alias for PTHREAD_MUTEX_INITIALIZER
      plugins: use JANUS_MUTEX_INITIALIZER
      transports: use JANUS_MUTEX_INITIALIZER
      turnrest: use JANUS_MUTEX_INITIALIZER

Örjan Fors (1):
      Use supplied WebRTC adapter throughout file

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