792 lines
32 KiB
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792 lines
32 KiB
Plaintext
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-------------------------------------------------------------------
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Thu Feb 1 10:59:09 UTC 2024 - Antonio Larrosa <alarrosa@suse.com>
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- Update to version 1.22.9:
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+ No changes, stable bump only.
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- Rebase reduce-required-meson.patch.
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-------------------------------------------------------------------
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Thu Jan 4 08:00:21 UTC 2024 - Antonio Larrosa <alarrosa@suse.com>
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- Update to version 1.22.8:
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+ No changes, stable bump only.
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- Rebase reduce-required-meson.patch.
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-------------------------------------------------------------------
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Wed Nov 15 09:36:27 UTC 2023 - Antonio Larrosa <alarrosa@suse.com>
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- Update to version 1.22.7:
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+ rtspclientsink: Don't leak previous server_ip
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- Rebase reduce-required-meson.patch.
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-------------------------------------------------------------------
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Fri Sep 22 16:46:22 UTC 2023 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.22.6:
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+ No changes, stable bump only.
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- Rebase reduce-required-meson.patch.
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-------------------------------------------------------------------
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Tue Jul 25 11:21:29 UTC 2023 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.22.5:
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+ No changes
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- Rebase reduce-required-meson.patch.
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-------------------------------------------------------------------
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Mon Jun 26 14:57:30 UTC 2023 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.22.4:
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+ No changes.
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- Rebase reduce-required-meson.patch.
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-------------------------------------------------------------------
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Wed May 24 15:29:52 UTC 2023 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.22.3:
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+ No changes.
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- Rebase patch.
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-------------------------------------------------------------------
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Wed Apr 12 13:51:29 UTC 2023 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.22.2:
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+ rtsp-server: fix deadlock on shutdown with non-live pipeline if
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media isn't playing/prerolled yet and eos-shutdown is enabled
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for the media
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- Rebase patch.
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-------------------------------------------------------------------
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Thu Mar 9 18:32:26 UTC 2023 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.22.1:
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+ No changes.
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- Rebase patch with quilt.
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-------------------------------------------------------------------
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Wed Mar 1 13:01:19 UTC 2023 - Antonio Larrosa <alarrosa@suse.com>
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- Add patch to reduce the required meson version to 0.61.0 since
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that's what we have in SLE 15:
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* reduce-required-meson.patch
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-------------------------------------------------------------------
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Wed Jan 25 22:33:01 UTC 2023 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.22.0:
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+ Please see changes in gstreamer main package, major version
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bump.
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-------------------------------------------------------------------
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Fri Dec 23 19:33:02 UTC 2022 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.20.5:
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+ rtsp-server: Free client if no connection could be created
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-------------------------------------------------------------------
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Sat Oct 22 09:10:27 UTC 2022 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.20.4:
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+ gst-rtsp-server: Fix pushing backlog to client.
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+ rtsp-server: stream: Don't loop forever if binding to the
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multicast address fails.
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-------------------------------------------------------------------
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Wed Jun 22 10:44:53 UTC 2022 - Aaron Stern <ukbeast89@protonmail.com>
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- Update to version 1.20.3:
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+ No changes.
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-------------------------------------------------------------------
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Mon May 9 11:06:24 UTC 2022 - Antonio Larrosa <alarrosa@suse.com>
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- Update to version 1.20.2:
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+ rtspclientsink: fix possible shutdown deadlock in
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collect_streams()
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+ Minor spelling fixes
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-------------------------------------------------------------------
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Wed Apr 6 07:06:41 UTC 2022 - Antonio Larrosa <alarrosa@suse.com>
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- Remove BuildRequires: hotdoc and disable the doc generation.
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It's really not used at all.
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-------------------------------------------------------------------
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Fri Mar 18 07:46:41 UTC 2022 - Antonio Larrosa <alarrosa@suse.com>
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- Update to version 1.20.1:
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+ Fix race in rtsp-client when tunneling over HTTP
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-------------------------------------------------------------------
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Wed Feb 9 22:38:30 UTC 2022 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.20.0:
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+ GstRTSPMediaFactory gained API to disable RTCP
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(gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp"
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property). Previously RTCP was always allowed for all RTSP
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medias. With this change it is possible to disable RTCP
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completely, irrespective of whether the client wants to do RTCP
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or not.
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+ Make a mount point of / work correctly. While not allowed by
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the RTSP 2 spec, the RTSP 1 spec is silent on this and it is
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used in the wild. It is now possible to use / as a mount path
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in gst-rtsp-server, e.g. rtsp://example.com/ would work with
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this now. Note that query/fragment parts of the URI are not
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necessarily correctly handled, and behaviour will differ
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between various client/server implementations; so use it if you
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must but don't bug us if it doesn't work with third party
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clients as you'd hoped.
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+ multithreading fixes (races, refcounting issues, deadlocks).
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+ ONVIF audio backchannel fixes.
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+ ONVIF trick mode optimisations.
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+ rtspclientsink: new "update-sdp" signal that allows updating
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the SDP before sending it to the server via ANNOUNCE. This can
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be used to add additional metadata to the SDP, for example. The
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order and number of medias must not be changed, however.
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-------------------------------------------------------------------
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Fri Feb 4 19:45:17 UTC 2022 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.18.6:
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+ rtsp-stream: fix get_rates raciness
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+ rtsp-media: Only unprepare a media if it was not already
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unpreparing anyway
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+ rtsp-media: Unprepare suspended medias too
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+ rtsp-client: make sure sessmedia will not get freed while used
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+ rtsp-media: Also mark receive-only (RECORD) medias as prepared
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when unsuspending
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+ rtsp-session: Don't unref medias twice if it is removed inside
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+ examples: Fix leak in appsrc2 example
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- Drop service, use source url, upstream changes in git.
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-------------------------------------------------------------------
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Thu Jan 20 16:59:00 UTC 2022 - Dominique Leuenberger <dimstar@opensuse.org>
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- Fix parameters passed to meson: with meson 60, the parameters are
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strictly checked, which helps in identifying those wrong
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parameters.
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-------------------------------------------------------------------
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Wed Sep 15 10:02:59 UTC 2021 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.18.5:
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+ rtsp-media:
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- Ensure the bus watch is removed during unprepare
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- Add one more case to seek avoidance
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- Improve skipping trickmode seek
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+ Fix a few memory leaks
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-------------------------------------------------------------------
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Wed Mar 31 16:21:16 UTC 2021 - Antonio Larrosa <alarrosa@suse.com>
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- Update to version 1.18.4:
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+ rtspclientsink: fix deadlock on shutdown if no data has been
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received yet
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+ rtspclientsink: fix leaks in unit tests
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+ rtsp-stream: avoid deadlock in send_func
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+ rtsp-client: cleanup transports during TEARDOWN
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-------------------------------------------------------------------
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Sat Jan 16 20:00:07 UTC 2021 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.18.3:
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+ rtsp-media: Only count senders when counting blocked streams
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+ rtsp-client: Only unref client watch context on finalize, to
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avoid deadlock
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-------------------------------------------------------------------
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Thu Dec 10 08:41:39 UTC 2020 - Bjørn Lie <bjorn.lie@gmail.com>
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- Update to version 1.18.2:
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+ stream: collect a clock_rate when blocking
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+ media:
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- Ignore GstRTSPStreamBlocking from incomplete streams, to
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prevent cases with prerolling when the inactive stream
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prerolls first and the server proceeds without waiting for
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the active stream. When there are no complete streams (during
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DESCRIBE), we will listen to all streams.
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- Use guint64 for setting the size-time property on rtpstorage,
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fixes potential crashes or memory corruption.
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- Get rates only on sender streams, fixing issue with ONVIF
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audio backchannel streams
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- Plug memory leak
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- Fix the _service file and spec to really use the tarball
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generated by service.
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-------------------------------------------------------------------
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Wed Oct 28 10:34:58 UTC 2020 - Antonio Larrosa <alarrosa@suse.com>
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- Update to 1.18.1:
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+ Highlighted bugfixes in 1.18.1
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- important security fixes
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- bug fixes and memory leak fixes
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- various stability and reliability improvements
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+ gst-rtsp-server changes:
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- rtsp-stream: collect rtp info when blocking
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- rtsp-media: set a 0 storage size for TCP receivers
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- rtsp-stream: preroll on gap events
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- rtsp-media: do not unblock on unsuspend
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-------------------------------------------------------------------
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Thu Sep 17 15:09:30 UTC 2020 - Antonio Larrosa <alarrosa@suse.com>
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- Update to 1.18.0:
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+ Highlights:
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- GstTranscoder: new high level API for applications to
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transcode media files from one format to another
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- High Dynamic Range (HDR) video information representation
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and signalling enhancements
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- Instant playback rate change support
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- Active Format Description (AFD) and Bar Data support
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- RTSP server and client implementations gained ONVIF trick
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modes support
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- Hardware-accelerated video decoding on Windows via
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DXVA2/Direct3D11
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- Microsoft Media Foundation plugin for video capture and
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hardware-accelerated video encoding on Windows
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- qmlgloverlay: New overlay element that renders a QtQuick
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|
scene over the top of an input video stream
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|
- imagesequencesrc: New element to easily create a video
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|
stream from a sequence of jpeg or png images
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- dashsink: New sink to produce DASH content
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- dvbsubenc: New DVB Subtitle encoder element
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|
- MPEG-TS muxing now also supports TV broadcast compliant
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|
muxing with constant bitrate muxing and SCTE-35 support
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|
- rtmp2: New RTMP client source and sink element from-scratch
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|
implementation
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|
- svthevcenc: New SVT-HEVC-based H.265 video encoder
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- vaapioverlay: New compositor element using VA-API
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|
- rtpmanager gained support for Google's Transport-Wide
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Congestion Control (twcc) RTP extension
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|
- splitmuxsink and splitmuxsrc gained support for auxiliary
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|
video streams
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- webrtcbin now contains some initial support for
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|
renegotiation involving stream addition and removal
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- RTP support was enhanced with new RTP source and sink
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elements to easily set up RTP streaming via rtp:// URIs
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- avtp: New Audio Video Transport Protocol (AVTP) plugin for
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Time-Sensitive Applications
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- Support for the Video Services Forum's Reliable Internet
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Stream Transport (RIST) TR-06-1 Simple Profile
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- Universal Windows Platform (UWP) support
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- rpicamsrc: New element for capturing from the Raspberry Pi
|
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camera
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- RTSP Server TCP interleaved backpressure handling
|
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|
improvements as well as support for Scale/Speed headers
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- GStreamer Editing Services gained support for nested
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timelines, per-clip speed rate control and the OpenTimelineIO
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format.
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- Autotools build system has been removed in favour of Meson
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||
|
- Drop patches already included upstream:
|
||
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* gst-rtsp-Fix-NULL-pointer.patch
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||
|
* gst-rtsp-fix-token-leak.patch
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* gst-rtsp-replace-G_TYPE_INSTANCE_GET_PRIVATE.patch
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|
|
||
|
-------------------------------------------------------------------
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|
Sun Apr 12 18:40:20 UTC 2020 - Bjørn Lie <bjorn.lie@gmail.com>
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|
|
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- Fix boo#1168026, CVE-2020-6095 and TALOS-2020-1018:
|
||
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+ Add gst-rtsp-Fix-NULL-pointer.patch: rtsp-auth: Fix NULL
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|
pointer dereference when handling an invalid basic
|
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|
Authorization header.
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|
- Add upstream bug fix patches:
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||
|
+ Add gst-rtsp-fix-token-leak.patch: rtsp-auth: Fix default token
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|
leak.
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||
|
+ Add gst-rtsp-replace-G_TYPE_INSTANCE_GET_PRIVATE.patch:
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||
|
rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's
|
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|
been deprecated.
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|
|
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|
|
||
|
-------------------------------------------------------------------
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|
Wed Dec 4 13:21:03 UTC 2019 - Bjørn Lie <bjorn.lie@gmail.com>
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|
|
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|
- Update to version 1.16.2:
|
||
|
+ rtsp-media: Use lock in gst_rtsp_media_is_receive_only
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|
+ rtsp-client:
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|
- RTP Info when completed_sender
|
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|
- Fix location uri-format by getting uri directly from context
|
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|
instead
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|
|
||
|
-------------------------------------------------------------------
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||
|
Tue Sep 24 15:01:29 UTC 2019 - Bjørn Lie <bjorn.lie@gmail.com>
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|
|
||
|
- Update to version 1.16.1:
|
||
|
+ See main gstreamer package for changelog.
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||
|
|
||
|
-------------------------------------------------------------------
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||
|
Tue Jun 25 11:47:07 UTC 2019 - Bjørn Lie <bjorn.lie@gmail.com>
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|
|
||
|
- Update to version 1.16.0:
|
||
|
+ Highlights:
|
||
|
- GStreamer WebRTC stack gained support for data channels for
|
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|
peer-to-peer communication based on SCTP, BUNDLE support,
|
||
|
as well as support for multiple TURN servers.
|
||
|
- AV1 video codec support for Matroska and QuickTime/MP4
|
||
|
containers and more configuration options and supported
|
||
|
input formats for the AOMedia AV1 encoder
|
||
|
- Support for Closed Captions and other Ancillary Data in video
|
||
|
- Support for planar (non-interleaved) raw audio
|
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|
- GstVideoAggregator, compositor and OpenGL mixer elements are
|
||
|
now in -base
|
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|
- New alternate fields interlace mode where each buffer carries
|
||
|
a single field
|
||
|
- WebM and Matroska ContentEncryption support in the Matroska
|
||
|
demuxer
|
||
|
- new WebKit WPE-based web browser source element
|
||
|
- Video4Linux: HEVC encoding and decoding, JPEG encoding, and
|
||
|
improved dmabuf import/export
|
||
|
- Hardware-accelerated Nvidia video decoder gained support for
|
||
|
VP8/VP9 decoding, whilst the encoder gained support for
|
||
|
H.265/HEVC encoding.
|
||
|
- Many improvements to the Intel Media SDK based
|
||
|
hardware-accelerated video decoder and encoder plugin
|
||
|
(msdk): dmabuf import/export for zero-copy integration with
|
||
|
other components; VP9 decoding; 10-bit HEVC encoding; video
|
||
|
post-processing (vpp) support including deinterlacing; and
|
||
|
the video decoder now handles dynamic resolution changes.
|
||
|
- The ASS/SSA subtitle overlay renderer can now handle multiple
|
||
|
subtitles that overlap in time and will show them on screen
|
||
|
simultaneously
|
||
|
- The Meson build is now feature-complete (*) and it is now the
|
||
|
recommended build system on all platforms. The Autotools
|
||
|
build is scheduled to be removed in the next cycle.
|
||
|
- The GStreamer Rust bindings and Rust plugins module are now
|
||
|
officially part of upstream GStreamer.
|
||
|
- The GStreamer Editing Services gained a gesdemux element
|
||
|
that allows directly playing back serialized edit list with
|
||
|
playbin or (uri)decodebin
|
||
|
- Many performance improvements.
|
||
|
- Updated options passed to meson following upstream changes.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Fri May 31 22:28:53 UTC 2019 - Bjørn Lie <bjorn.lie@gmail.com>
|
||
|
|
||
|
- Update to version 1.14.5:
|
||
|
+ rtsp-client: Fix crash in close handler and remove timeout
|
||
|
GSource on cleanup.
|
||
|
+ rtsp-media:
|
||
|
- Handle set state when preparing.
|
||
|
- Fix race condition in finish_unprepare.
|
||
|
+ rtsp-stream:
|
||
|
- Use cached address when allocating sockets.
|
||
|
- Use seqnum-offset for rtpinfo.
|
||
|
- Add source elements to the pipeline before activation for
|
||
|
stream-status create message.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Wed Oct 3 16:01:19 UTC 2018 - bjorn.lie@gmail.com
|
||
|
|
||
|
- Update to version 1.14.4:
|
||
|
+ Bugfix release, please see .changes in gstreamer main package.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Wed Sep 26 20:33:14 UTC 2018 - bjorn.lie@gmail.com
|
||
|
|
||
|
- Update to version 1.14.3:
|
||
|
+ Bugfix release, please see .changes in gstreamer main package.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Tue Jul 24 08:25:37 UTC 2018 - bjorn.lie@gmail.com
|
||
|
|
||
|
- Update to version 1.14.2:
|
||
|
+ rtsp-media:
|
||
|
- unref clock (if set) when finalizing.
|
||
|
- add gst_rtsp_media_*_set_clock to docs.
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||
|
+ media-factory:
|
||
|
- unref old clock when setting new clock.
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||
|
- unref clock in finalize.
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||
|
+ rtsp-onvif-media:
|
||
|
- fix g-ir-scanner warnings.
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||
|
- export gst_rtsp_onvif_media_factory_requires_backchannel.
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||
|
+ client: Strip transport parts as whitespaces could be around
|
||
|
commas.
|
||
|
+ rtsp-stream: avoid pushing data on unlinked udpsrc pad during
|
||
|
setup.
|
||
|
+ rtspclientsink: fix waiting for multiple streams.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Sat Jun 23 09:55:55 UTC 2018 - bjorn.lie@gmail.com
|
||
|
|
||
|
- Switch to meson build system:
|
||
|
+ Add meson, pkgconfig(glib-2.0),pkgconfig(gstreamer-app-1.0),
|
||
|
pkgconfig(gstreamer-net-1.0), pkgconfig(gstreamer-rtp-1.0),
|
||
|
pkgconfig(gstreamer-rtsp-1.0) and pkgconfig(gstreamer-sdp-1.0)
|
||
|
BuildRequires.
|
||
|
+ Add meson macros, replacing autotools ones.
|
||
|
+ Pass disable_introspection=false,
|
||
|
with-package-name='openSUSE GStreamer-rtsp-server package',
|
||
|
with-package-origin='http://download.opensuse.org' and
|
||
|
tests=false and examples=false to meson, ensure we build the
|
||
|
features we want. Tests have always been disabled, be explicit
|
||
|
about it, as they need a working network connection.
|
||
|
+ Drop pkgconfig(gstreamer-plugins-base-1.0) BuildRequires.
|
||
|
+ No longer rm la files, not needed when building with meson.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Fri Jun 22 11:17:45 UTC 2018 - bjorn.lie@gmail.com
|
||
|
|
||
|
- Drop gstreamer-plugins-good and
|
||
|
pkgconfig(gstreamer-plugins-bad-1.0) BuildRequires: Only needed
|
||
|
for unit tests and we do not build or run those tests.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Sun May 20 09:58:11 UTC 2018 - bjorn.lie@gmail.com
|
||
|
|
||
|
- Update to version 1.14.1:
|
||
|
+ GstPad: Fix race condition causing the same probe to be called
|
||
|
multiple times
|
||
|
+ Fix occasional deadlocks on windows when outputting debug
|
||
|
logging
|
||
|
+ Fix debug levels being applied in the wrong order
|
||
|
+ GIR annotation fixes for bindings
|
||
|
+ audiomixer, audioaggregator: fix some negotiation issues
|
||
|
+ gst-play-1.0: fix leaving stdin in non-blocking mode after exit
|
||
|
+ flvmux: wait for caps on all input pads before writing header
|
||
|
even if source is live
|
||
|
+ flvmux: don't wake up the muxer unless there is data, fixes
|
||
|
busy looping if there's no input data
|
||
|
+ flvmux: fix major leak of input buffers
|
||
|
+ rtspsrc, rtsp-server: revert to RTSP RFC handling of
|
||
|
sendonly/recvonly attributes
|
||
|
+ rtpvrawpay: fix payloading with very large mtu sizes where
|
||
|
everything fits into a single RTP packet
|
||
|
+ v4l2: Fix hard-coded enabled v4l2 probe on Linux/ARM
|
||
|
+ v4l2: Disable DMABuf for emulated formats when using libv4l2
|
||
|
+ v4l2: Always set colorimetry in S_FMT
|
||
|
+ asfdemux: Set stream-format field for H264 streams and handle
|
||
|
H.264 in bytestream format
|
||
|
+ x265enc: Fix tagging of keyframes on output buffers
|
||
|
+ ladspa: Fix critical during plugin load on Windows
|
||
|
+ decklink: Fix COM initialisation on Windows
|
||
|
+ h264parse: fix re-use across pipeline stop/restart
|
||
|
+ mpegtsmux: fix force-keyframe event handling and PCR/PMT
|
||
|
changes that would confuse some players with generated HLS
|
||
|
streams
|
||
|
+ adaptivedemux: Support period change in live playlist
|
||
|
+ rfbsrc: Fix support for applevncserver and support NULL pool in
|
||
|
decide_allocation
|
||
|
+ jpegparse: Fix APP1 marker segment parsing
|
||
|
+ h265parse: Make caps writable before modifying them, fixes
|
||
|
criticals
|
||
|
+ fakevideosink: request an extra buffer if enable-last-sample is
|
||
|
enabled
|
||
|
+ wasapisrc: Don't provide a clock based on WASAPI's clock
|
||
|
+ wasapi: Only use audioclient3 when low-latency, as it might
|
||
|
otherwise glitch with slow CPUs or VMs
|
||
|
+ wasapi: Don't derive device period from latency time, should
|
||
|
make it more robust against glitches
|
||
|
+ audiolatency: Fix wave detection in buffers and avoid bogus pts
|
||
|
values while starting
|
||
|
+ msdk: fix plugin load on implementations with only HW support
|
||
|
+ msdk: dec: set framerate to the driver only if provided, not in
|
||
|
0/1 case
|
||
|
+ msdk: Don't set extended coding options for JPEG encode
|
||
|
+ rtponviftimestamp: fix state change function init/reset causing
|
||
|
races/crashes on shutdown
|
||
|
+ decklink: fix initialization failure in windows binary
|
||
|
+ ladspa: Fix critical warnings during plugin load on Windows and
|
||
|
fix dependencies in meson build
|
||
|
+ gl: fix cross-compilation error with viv-fb
|
||
|
+ qmlglsink: make work with eglfs_kms
|
||
|
+ rtspclientsink: Don't deadlock in preroll on early close
|
||
|
+ rtspclientsink: Fix client ports for the RTCP backchannel
|
||
|
+ rtsp-server: Fix session timeout when streaming data to client
|
||
|
over TCP
|
||
|
+ vaapiencode: h264: find best profile in those available, fixing
|
||
|
negotiation errors
|
||
|
+ vaapi: remove custom GstGL context handling, use GstGL instead.
|
||
|
Fixes GL Context sharing with WebkitGtk on wayland
|
||
|
+ gst-editing-services: various fixes
|
||
|
+ gst-python: bump pygobject req to 3.8;
|
||
|
fix GstPad.set_query_function(); dist autogen.sh and
|
||
|
configure.ac in tarball
|
||
|
+ g-i: pick up GstVideo-1.0.gir from local build directory in
|
||
|
GstGL build
|
||
|
+ g-i: update constant values for bindings
|
||
|
+ avoid duplicate symbols in plugins across modules in static
|
||
|
builds
|
||
|
+ ... and many, many more!
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Tue Apr 17 10:56:41 UTC 2018 - bjorn.lie@gmail.com
|
||
|
|
||
|
- Update to version 1.14.0:
|
||
|
+ Highlights:
|
||
|
- WebRTC support: real-time audio/video streaming to and from
|
||
|
web browsers;
|
||
|
- Experimental support for the next-gen royalty-free AV1 video
|
||
|
codec
|
||
|
- Video4Linux: encoding support, stable element names and
|
||
|
faster device probing;
|
||
|
- Support for the Secure Reliable Transport (SRT) video
|
||
|
streaming protocol;
|
||
|
- RTP Forward Error Correction (FEC) support (ULPFEC);
|
||
|
- RTSP 2.0 support in rtspsrc and gst-rtsp-server;
|
||
|
- ONVIF audio backchannel support in gst-rtsp-server and
|
||
|
rtspsrc;
|
||
|
- playbin3 gapless playback and pre-buffering support;
|
||
|
- Tee, our stream splitter/duplication element, now does
|
||
|
allocation query aggregation which is important for efficient
|
||
|
data handling and zero-copy;
|
||
|
- QuickTime muxer has a new prefill recording mode that allows
|
||
|
file import in Adobe Premiere and FinalCut Pro while the file
|
||
|
is still being written;
|
||
|
- rtpjitterbuffer fast-start mode and timestamp offset
|
||
|
adjustment smoothing;
|
||
|
- souphttpsrc connection sharing, which allows for connection
|
||
|
reuse, cookie sharing, etc;
|
||
|
- nvdec: new plugin for hardware-accelerated video decoding
|
||
|
using the NVIDIA NVDEC API;
|
||
|
- Adaptive DASH trick play support;
|
||
|
- ipcpipeline: new plugin that allows splitting a pipeline
|
||
|
across multiple processes;
|
||
|
- Major gobject-introspection annotation improvements for large
|
||
|
parts of the library API;
|
||
|
- GStreamer C# bindings have been revived and seen many updates
|
||
|
and fixes;
|
||
|
- The externally maintained GStreamer Rust bindings had many
|
||
|
usability improvements and cover most of the API now.
|
||
|
Coinciding with the 1.14 release, a new release with the 1.14
|
||
|
API additions is happening.
|
||
|
+ Updated translations.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Fri Mar 30 12:13:18 UTC 2018 - bjorn.lie@gmail.com
|
||
|
|
||
|
- Update to version 1.12.5:
|
||
|
+ Bugs fixed: bgo#789646, bgo#791743.
|
||
|
- Drop upstream fixed patches:
|
||
|
+ gst-rtsp-server-add-annotations-and-API-guards.patch.
|
||
|
+ gst-rtsp-server-gst_rtsp_context_get_current.patch.
|
||
|
+ gst-rtsp-server-rtsp-client-add-type-annotations.patch.
|
||
|
+ gst-rtsp-server-Set-udpsink_out-ttl-mc-property.patch.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Mon Mar 26 13:32:35 UTC 2018 - dimstar@opensuse.org
|
||
|
|
||
|
- Drop pkgconfig(libcgroup) BuildRequires: libcgroup's
|
||
|
functionality is largely deprecated by systemd and the two
|
||
|
actually clash in some ways which cause bug reports.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Wed Feb 28 16:31:55 UTC 2018 - dimstar@opensuse.org
|
||
|
|
||
|
- Modernize spec-file by calling spec-cleaner
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Mon Feb 12 22:32:46 UTC 2018 - bjorn.lie@gmail.com
|
||
|
|
||
|
- Add upstream bug fix patches:
|
||
|
+ gst-rtsp-server-rtsp-client-add-type-annotations.patch.
|
||
|
+ gst-rtsp-server-gst_rtsp_context_get_current.patch.
|
||
|
+ gst-rtsp-server-add-annotations-and-API-guards.patch.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Tue Jan 9 11:39:09 UTC 2018 - zaitor@opensuse.org
|
||
|
|
||
|
- Add gst-rtsp-server-Set-udpsink_out-ttl-mc-property.patch: rtsp:
|
||
|
Set udpsink_out ttl-mc property on creation (bgo#791743).
|
||
|
- Clean up spec, silence some rpmlint warnings.
|
||
|
- Drop explicit libgstrtspserver-1_0-0 and
|
||
|
typelib-1_0-GstRtspServer-1_0 Obsoletes and Provides: Not needed
|
||
|
and only leads to a rpmlint warning.
|
||
|
- Add gstreamer-rtsp-server-rpmlintrc: Filter out bogus warning
|
||
|
about missing dependencies in devel package.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Mon Dec 11 20:24:14 UTC 2017 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.12.4:
|
||
|
+ Bugs fixed: bgo#789646, bgo#769521.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Mon Sep 18 21:56:48 UTC 2017 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.12.3:
|
||
|
+ Bugs fixed: bgo#784094, bgo#786457.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Fri Jul 14 12:01:51 UTC 2017 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.12.2:
|
||
|
+ No changes, stable version bump only.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Wed Jun 21 08:57:36 UTC 2017 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.12.1:
|
||
|
+ No changes, stable version bump only.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Wed May 10 12:38:06 UTC 2017 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.12.0:
|
||
|
+ No changes, stable version bump only.
|
||
|
- Changes from version 1.11.91:
|
||
|
+ gi: Fix some annotations and docstrings.
|
||
|
+ Automatic update of common submodule.
|
||
|
- Changes from version 1.11.90:
|
||
|
+ examples: make test-launch pipeline shared by default as well.
|
||
|
+ gstreamer-rtsp-server: Add both srcdir and builddir to the
|
||
|
include path.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Sat Feb 25 00:29:34 UTC 2017 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.11.2:
|
||
|
+ Meson build fixes.
|
||
|
+ Minor changes and fixes.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Thu Feb 23 20:25:06 UTC 2017 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.11.1:
|
||
|
+ Bugs fixed: bgo#758062, bgo#771830, bgo#774173, bgo#774640,
|
||
|
bgo#776867, bgo#777037, bgo#774416.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Thu Feb 23 20:25:05 UTC 2017 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.10.4:
|
||
|
+ Minor tweaks and fixes.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Mon Jan 30 16:48:37 UTC 2017 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.10.3:
|
||
|
+ Bugs fixed: bgo#755329, bgo#776343, bgo#776345.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Sun Jan 1 13:31:30 UTC 2017 - jengelh@inai.de
|
||
|
|
||
|
- Summary updates.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Sat Dec 3 19:07:31 UTC 2016 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.10.2:
|
||
|
+ Bugs fixed: bgo#765673, bgo#770239.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Sun Nov 27 12:42:14 UTC 2016 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.10.1:
|
||
|
+ Meson update.
|
||
|
- Changes from version 1.10.0:
|
||
|
+ Bugs fixed: bgo#771983, bgo#772478, bgo#773640.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Fri Aug 19 19:45:29 UTC 2016 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.8.3 (boo#996937):
|
||
|
+ g-i: pass compiler env to g-ir-scanner.
|
||
|
- Changes from version 1.8.2:
|
||
|
+ rtsp-session: RFC2326 does not allow a space between ; and
|
||
|
timeout in the Session header.
|
||
|
+ rtsp-stream:
|
||
|
- Fix crash on cleanup with shared media and multiple udpsrc.
|
||
|
- Always bind to ANY when address is a multicast address and
|
||
|
not only on Windows.
|
||
|
- Rename package to gstreamer-rtsp-server. Align with the other
|
||
|
gstreamer packages. Also obsolete and provide the previous ones
|
||
|
to ease updates.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Wed Jun 15 14:27:03 UTC 2016 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.8.1:
|
||
|
+ bgo#764744: Crashes when multiple udpsrc are created for each
|
||
|
client on a shared media, misses tracking and cleanup.
|
||
|
+ bgo#766619: Space between ; and timeout= in session header is
|
||
|
not RFC2326 compliant.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Thu Apr 21 08:48:19 UTC 2016 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.8.1:
|
||
|
+ No changes, version bump only.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Sat Mar 26 20:52:00 UTC 2016 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.8.0:
|
||
|
+ Hardware-accelerated zero-copy video decoding on Android
|
||
|
+ New video capture source for Android using the
|
||
|
android.hardware.Camera API.
|
||
|
+ Windows Media reverse playback support (ASF/WMV/WMA).
|
||
|
+ New tracing system provides support for more sophisticated
|
||
|
debugging tools.
|
||
|
+ New high-level GstPlayer playback convenience API.
|
||
|
+ Initial support for the new Vulkan API, see Matthew Waters'
|
||
|
blog post for more details.
|
||
|
+ Improved Opus audio codec support: Support for more than two
|
||
|
channels; MPEG-TS demuxer/muxer can now handle Opus;
|
||
|
sample-accurate encoding/decoding/transmuxing with Ogg,
|
||
|
Matroska, ISOBMFF (Quicktime/MP4), and MPEG-TS as container;
|
||
|
new codec utility functions for Opus header and caps handling
|
||
|
in pbutils library. The Opus encoder/decoder elements were
|
||
|
also moved to gst-plugins-base (from -bad), and the opus RTP
|
||
|
depayloader/payloader to -good.
|
||
|
+ GStreamer VAAPI module now released and maintained as part of
|
||
|
the GStreamer project.
|
||
|
+ Asset proxy support in the GStreamer Editing Services.
|
||
|
+ Bugs fixed: bgo#740509.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Tue Dec 15 11:40:41 UTC 2015 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.6.2:
|
||
|
+ rtsp-server: Change the logic so we don't pop a NULL context.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Sun Nov 1 08:23:57 UTC 2015 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.6.1:
|
||
|
+ gst-rtsp-server: Retain reference to rtsp-media when preparing.
|
||
|
+ rtsp-stream: GstBin leak in udp-mcast case.
|
||
|
- Changes from version 1.6.0:
|
||
|
+ For changelog, see mainpackage changes, everything is condensed
|
||
|
there.
|
||
|
- Drop grs-rtsp-fix-double-unlock-in_get_buffer_size.patch: Fixed
|
||
|
upstream.
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Wed Aug 5 13:59:12 UTC 2015 - zaitor@opensuse.org
|
||
|
|
||
|
- Add grs-rtsp-fix-double-unlock-in_get_buffer_size.patch: Fixes an
|
||
|
abort when calling gst_rtsp_media_get_buffer_size() because of
|
||
|
double g_mutex_unlock () usage (bgo#745434).
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Fri Dec 26 11:15:05 UTC 2014 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.4.5:
|
||
|
+ rtsp-stream: leaks srtp decoder when leaving rtpbin
|
||
|
(bgo#739481).
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Fri Nov 14 11:33:10 UTC 2014 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.4.4:
|
||
|
+ rtsp-client: mikey memory leaks (bgo#739383).
|
||
|
- Changes from version 1.4.3:
|
||
|
+ No changes.
|
||
|
- Changes from version 1.4.2:
|
||
|
+ rtsp-media: Make sure that sequence numbers are monotonic after
|
||
|
pause (bgo#736017).
|
||
|
+ rtsp-client: Protect saved clients watch with a mutex
|
||
|
(bgo#735570).
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Thu Aug 28 20:15:57 UTC 2014 - zaitor@opensuse.org
|
||
|
|
||
|
- Update to version 1.4.1:
|
||
|
+ RTSP PLAY with specified range replies with wrong range
|
||
|
(bgo#732644).
|
||
|
|
||
|
-------------------------------------------------------------------
|
||
|
Fri Mar 14 10:15:18 UTC 2014 - zaitor@opensuse.org
|
||
|
|
||
|
- Initial packaging, version 1.4.0.
|
||
|
|