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Release 1.19.1
This commit is contained in:
Tim-Philipp Müller
parent
a5b30f179b
commit
a3c3afbf56
6 changed files with 492 additions and 1995 deletions
402
ChangeLog
402
ChangeLog
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@ -1,11 +1,409 @@
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=== release 1.18.0 ===
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=== release 1.19.1 ===
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2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
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2021-06-01 00:15:08 +0100 Tim-Philipp Müller <tim@centricular.com>
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* ChangeLog:
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* ChangeLog:
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* NEWS:
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* NEWS:
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* RELEASE:
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* RELEASE:
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* gst-rtsp-server.doap:
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* gst-rtsp-server.doap:
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* meson.build:
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Release 1.19.1
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2021-05-24 18:58:00 +0100 Tim-Philipp Müller <tim@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: use new gst_buffer_new_memdup()
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/208>
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2021-05-04 20:47:18 -0400 Doug Nazar <nazard@nazar.ca>
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* gst/rtsp-server/rtsp-media-factory-uri.c:
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rtsp-media: fix leak when adding converter
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Free the previous caps before reusing the variable for the converter caps.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
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2021-05-04 20:45:19 -0400 Doug Nazar <nazard@nazar.ca>
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* gst/rtsp-server/rtsp-client.c:
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rtsp-client: fix leak adding headers
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gst_rtsp_message_add_header() makes a copy of the header, instead
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of taking ownership.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
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2021-04-21 10:43:41 +0200 François Laignel <fengalin@free.fr>
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* gst/rtsp-server/rtsp-stream.c:
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Use gst_element_request_pad_simple...
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Instead of the deprecated gst_element_get_request_pad.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/195>
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2021-04-29 03:07:42 -0400 Doug Nazar <nazard@nazar.ca>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Ensure the bus watch is removed during unprepare
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It's possible for the destruction of the source to be delayed.
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Instead of relying on the dispose() to remove the bus watch, do
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it ourselves.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/202>
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2021-04-27 09:22:21 +0200 Marc Leeman <m.leeman@televic.com>
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* docs/README:
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docs: minor spelling correction in README
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
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2021-04-27 09:05:39 +0200 Marc Leeman <m.leeman@televic.com>
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* examples/test-replay-server.c:
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test-replay-server: minor spelling corrections
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Bumped on these while investigating the example code.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
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2021-04-22 23:26:02 -0400 Doug Nazar <nazard@nazar.ca>
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* tests/check/gst/stream.c:
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tests: Don't fail tests if IPv6 not available.
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On computers with IPv6 disabled it shouldn't result in a test failure.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
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2021-04-23 07:18:48 +0200 Edward Hervey <edward@centricular.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Add one more case to seek avoidance
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This is an extension to the previous commit. There can also be cases where the
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start position is not specified, in those cases we should also avoid doing
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seeking unless it's forced.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/197>
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2021-04-16 14:35:02 -0400 Doug Nazar <nazard@nazar.ca>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Improve skipping trickmode seek.
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We can also skip the seek if the end range is already
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correct.
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Avoids initial seek on play start if playing full stream.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/194>
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2021-03-19 10:36:01 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-sink/gstrtspclientsink.c:
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rtspclientsink: Don't run signal class handlers during the CLEANUP stage
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It's sufficient to run them during the FIRST stage instead of in both.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/193>
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2021-02-15 12:07:15 +0000 Tim-Philipp Müller <tim@centricular.com>
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* tests/check/gst/rtspclientsink.c:
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tests: rtspclientsink: fix some leaks
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
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2021-02-15 12:26:30 +0000 Tim-Philipp Müller <tim@centricular.com>
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* gst/rtsp-sink/gstrtspclientsink.c:
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rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
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2021-02-15 12:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
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* tests/check/gst/rtspclientsink.c:
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rtspclientsink: add unit test for potential shutdown deadlock
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
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2021-02-15 12:01:34 +0000 Tim-Philipp Müller <tim@centricular.com>
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* gst/rtsp-sink/gstrtspclientsink.c:
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rtspclientsink: fix deadlock on shutdown before preroll
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Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
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2021-02-01 12:16:46 +0100 Branko Subasic <branko@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: avoid deadlock in send_func
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Currently the send_func() runs in a thread of its own which is started
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the first time we enter handle_new_sample(). It runs in an outer loop
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until priv->continue_sending is FALSE, which happens when a TEARDOWN
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request is received. We use a local variable, cont, which is initialized
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to TRUE, meaning that we will always enter the outer loop, and at the
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end of the outer loop we assign it the value of priv->continue_sending.
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Within the outer loop there is an inner loop, where we wait to be
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signaled when there is more data to send. The inner loop is exited when
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priv->send_cookie has changed value, which it does when more data is
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available or when a TEARDOWN has been received.
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But if we get a TEARDOWN before send_func() is entered we will get stuck
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in the inner loop because no one will increase priv->session_cookie
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anymore.
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By not entering the outer loop in send_func() if priv->continue_sending
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is FALSE we make sure that we do not get stuck in send_func()'s inner
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loop should we receive a TEARDOWN before the send thread has started.
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Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/187>
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2021-01-22 08:58:23 +0100 Branko Subasic <branko@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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rtsp-client: cleanup transports during TEARDOWN
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When tunneling RTP over RTSP the stream transports are stored in a hash
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table in the GstRTSPClientPrivate struct. They are used for, among other
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things, mapping channel id to stream transports when receiving data from
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the client. The stream tranports are created and added to the hash table
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in handle_setup_request(), but unfortuately they are not removed in
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handle_teardown_request(). This means that if the client sends data on
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the RTSP connection after it has sent the TEARDOWN, which is often the
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case when audio backchannel is enabled, handle_data() will still be able
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to map the channel to a session transport and pass the data along to it.
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Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
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because the stream is no longer joined to a bin.
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We avoid this by removing the stream transports from the hash table when
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we handle the TEARDOWN request.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184>
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2020-12-15 11:07:01 +0200 Sebastian Dröge <sebastian@centricular.com>
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* docs/gst_plugins_cache.json:
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* gst/rtsp-sink/gstrtspclientsink.c:
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rtspclientsink: Add "update-sdp" signal that allows updating the SDP before sending it to the server
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/178>
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2020-12-23 13:54:54 -0500 John Lindgren <john.lindgren@avasure.com>
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* tests/check/gst/client.c:
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Add test cases for mountpoint of '/'
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
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2020-11-05 16:02:49 -0500 John Lindgren <john.lindgren@avasure.com>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-mount-points.c:
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* gst/rtsp-server/rtsp-session-media.c:
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Make a mount point of "/" work correctly.
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As far as I can tell, this is neither explicitly allowed nor
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forbidden by RFC 7826.
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Meanwhile, URLs such as rtsp://<IP>:554 or rtsp://<IP>:554/ are in
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use in the wild (presumably with non-GStreamer servers).
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GStreamer's prior behavior was confusing, in that
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gst_rtsp_mount_points_add_factory() would appear to accept a mount
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path of "" or "/", but later connection attempts would fail with a
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"media not found" error.
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This commit makes a mount path of "/" work for either form of URL,
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while an empty mount path ("") is rejected and logs a warning.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
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2020-12-15 10:18:16 +0200 Sebastian Dröge <sebastian@centricular.com>
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* docs/gst_plugins_cache.json:
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* gst/rtsp-sink/gstrtspclientsink.c:
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rtspclientsink: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/177>
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2020-12-17 15:27:27 +0100 Tobias Ronge <tobiasr@axis.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Only count senders when counting blocked streams
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Only sender streams sends the GstRTSPStreamBlocking message, so only
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these should be counted before setting media status to prepared.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/180>
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2020-10-21 15:38:43 +0200 Jimmi Holst Christensen <jimmi.christensen@aivero.com>
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* gst/rtsp-sink/gstrtspclientsink.c:
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rtspclientsink add proper support for uri queries
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/166>
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2020-12-14 14:12:38 +1300 Lawrence Troup <lawrence.troup@teknique.com>
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* gst/rtsp-server/rtsp-client.c:
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rtsp-client: Only unref client watch context on finalize, to avoid deadlock
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Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/176>
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2020-11-18 20:36:50 +0100 Mathieu Duponchelle <mathieu@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: collect a clock_rate when blocking
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This lets us provide a clock_rate in a fashion similar to the
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other code paths in get_rtpinfo()
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/174>
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2020-11-16 10:34:41 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Use guint64 for setting the size-time property on rtpstorage
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Otherwise this will cause memory corruption as the property expects a 64
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bit integer.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/169>
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2020-11-03 16:56:28 +0100 David Phung <davidph@axis.com>
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-media: Ignore GstRTSPStreamBlocking from incomplete streams
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To prevent cases with prerolling when the inactive stream prerolls first
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and the server proceeds without waiting for the active stream, we will
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ignore GstRTSPStreamBlocking messages from incomplete streams. When
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there are no complete streams (during DESCRIBE), we will listen to all
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streams.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
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2020-10-28 21:48:06 +0100 Kristofer Björkström <kristofb@axis.com>
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* tests/check/gst/media.c:
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* tests/check/meson.build:
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* tests/files/test.avi:
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media test: Add test for seeking one active stream with a demuxer
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Add another seek_one_active_stream test but with a demuxer. The demuxer
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will flush both streams in opposed to the existing test which only
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flushes the active stream. This will help exposing problems with the
|
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prerolling process after a flushing seek.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
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|
|
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2018-10-29 09:19:33 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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|
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* gst/rtsp-server/meson.build:
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* meson.build:
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* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
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* pkgconfig/gstreamer-rtsp-server.pc.in:
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* pkgconfig/meson.build:
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Meson: Use pkg-config generator
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/1>
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|
|
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2020-10-19 11:25:25 +0300 Sebastian Dröge <sebastian@centricular.com>
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* meson.build:
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meson: update glib minimum version to 2.56
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/164>
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|
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2020-09-04 21:14:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
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|
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* examples/test-launch.c:
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* gst/rtsp-server/rtsp-media-factory.c:
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* gst/rtsp-server/rtsp-media-factory.h:
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-server-internal.h:
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* gst/rtsp-server/rtsp-stream.c:
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* tests/check/gst/client.c:
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rtsp-media-factory: expose API to disable RTCP
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This is supported by the RFC, and can be useful on systems where
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allocating two consecutive ports is problematic, and RTCP is not
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necessary.
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
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2020-10-08 23:45:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
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|
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* hooks/pre-commit.hook:
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* meson.build:
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git: use our standard pre commit hook
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/162>
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|
|
||||||
|
2020-10-08 22:17:16 +0200 Mathieu Duponchelle <mathieu@centricular.com>
|
||||||
|
|
||||||
|
* gst/rtsp-server/rtsp-stream.c:
|
||||||
|
rtsp-stream: make use of blocked_running_time in query_position
|
||||||
|
When blocking, the sink element will not have received a buffer
|
||||||
|
yet and the position query will fail. Instead, we make use of
|
||||||
|
the running time of the buffer we blocked on.
|
||||||
|
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
|
||||||
|
|
||||||
|
2020-10-06 00:04:17 +0200 Mathieu Duponchelle <mathieu@centricular.com>
|
||||||
|
|
||||||
|
* gst/rtsp-server/rtsp-stream.c:
|
||||||
|
rtsp-stream: collect rtp info when blocking
|
||||||
|
We don't unblock the stream anymore before replying to the
|
||||||
|
play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443),
|
||||||
|
so the sinks don't have a last-sample after potentially flush
|
||||||
|
seeking. seek_trickmode waits for preroll however, which means
|
||||||
|
the stream will block and wait for a first buffer. Subsequent
|
||||||
|
calls to get_rtpinfo() can thus make use of the information.
|
||||||
|
See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115
|
||||||
|
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
|
||||||
|
|
||||||
|
2020-09-27 20:09:22 +0900 Seungha Yang <seungha@centricular.com>
|
||||||
|
|
||||||
|
* examples/meson.build:
|
||||||
|
* examples/test-replay-server.c:
|
||||||
|
* examples/test-replay-server.h:
|
||||||
|
examples: Add an example for loop playback
|
||||||
|
This demo example shows a way of file loop playback of a given source.
|
||||||
|
Note that client seek request is not properly implemented yet.
|
||||||
|
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/154>
|
||||||
|
|
||||||
|
2020-09-28 22:03:47 +0200 David Phung <davidph@axis.com>
|
||||||
|
|
||||||
|
* gst/rtsp-server/rtsp-media.c:
|
||||||
|
rtsp-media: Plug memory leak
|
||||||
|
The get-storage signal of rtpbin increases the ref count of the storage.
|
||||||
|
So we have to unref it after usage.
|
||||||
|
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/155>
|
||||||
|
|
||||||
|
2020-09-11 15:46:41 +0200 Guiqin Zou <guiqinzu@axis.com>
|
||||||
|
|
||||||
|
* gst/rtsp-server/rtsp-media.c:
|
||||||
|
rtsp-media: Get rates only on sender streams
|
||||||
|
When play a media with both sender and receiver stream, like ONVIF
|
||||||
|
back channel audio in, gst_rtsp_media_get_rates call
|
||||||
|
gst_rtsp_stream_get_rates for each stream to set the rates. But
|
||||||
|
gst_rtsp_stream_get_rates return false for the receiver steam, which
|
||||||
|
lead a g_assert crash.
|
||||||
|
Instead to get rates on all streams, now just get rates on sender
|
||||||
|
streams.
|
||||||
|
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/150>
|
||||||
|
|
||||||
|
2020-09-05 00:30:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
|
||||||
|
|
||||||
|
* gst/rtsp-server/rtsp-media.c:
|
||||||
|
* gst/rtsp-server/rtsp-server-internal.h:
|
||||||
|
* gst/rtsp-server/rtsp-stream.c:
|
||||||
|
rtsp-media: set a 0 storage size for TCP receivers
|
||||||
|
ulpfec correction is obviously useless when receiving a stream
|
||||||
|
over TCP, and in TCP modes the rtp storage receives non
|
||||||
|
timestamped buffers, causing it to queue buffers indefinitely,
|
||||||
|
until the queue grows so large that sanity checks kick in and
|
||||||
|
warnings start to get emitted.
|
||||||
|
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/149>
|
||||||
|
|
||||||
|
2020-08-21 03:02:40 +0200 Mathieu Duponchelle <mathieu@centricular.com>
|
||||||
|
|
||||||
|
* gst/rtsp-server/rtsp-stream.c:
|
||||||
|
rtsp-stream: preroll on gap events
|
||||||
|
This allows negotiating a SDP with all streams present, but only
|
||||||
|
start sending packets at some later point in time
|
||||||
|
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/146>
|
||||||
|
|
||||||
|
2020-08-25 16:10:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
|
||||||
|
|
||||||
|
* gst/rtsp-server/rtsp-media.c:
|
||||||
|
rtsp-media: do not unblock on unsuspend
|
||||||
|
rtsp_media_unsuspend() is called from handle_play_request()
|
||||||
|
before sending the play response. Unblocking the streams here
|
||||||
|
was causing data to be sent out before the client was ready
|
||||||
|
to handle it, with obvious side effects such as initial packets
|
||||||
|
getting discarded, causing decoding errors.
|
||||||
|
Instead we can simply let the media streams be unblocked when
|
||||||
|
the state of the media is set to PLAYING, which occurs after
|
||||||
|
sending the play response.
|
||||||
|
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
|
||||||
|
|
||||||
|
2020-09-08 17:30:49 +0100 Tim-Philipp Müller <tim@centricular.com>
|
||||||
|
|
||||||
|
* .gitlab-ci.yml:
|
||||||
|
ci: include template from gst-ci master branch again
|
||||||
|
|
||||||
|
2020-09-08 16:58:58 +0100 Tim-Philipp Müller <tim@centricular.com>
|
||||||
|
|
||||||
|
* docs/gst_plugins_cache.json:
|
||||||
|
* meson.build:
|
||||||
|
Back to development
|
||||||
|
|
||||||
|
=== release 1.18.0 ===
|
||||||
|
|
||||||
|
2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
|
||||||
|
|
||||||
|
* .gitlab-ci.yml:
|
||||||
|
* ChangeLog:
|
||||||
|
* NEWS:
|
||||||
|
* RELEASE:
|
||||||
|
* docs/gst_plugins_cache.json:
|
||||||
|
* gst-rtsp-server.doap:
|
||||||
* meson.build:
|
* meson.build:
|
||||||
Release 1.18.0
|
Release 1.18.0
|
||||||
|
|
||||||
|
|
|
||||||
15
RELEASE
15
RELEASE
|
|
@ -1,18 +1,15 @@
|
||||||
This is GStreamer gst-rtsp-server 1.18.0.
|
This is GStreamer gst-rtsp-server 1.19.1.
|
||||||
|
|
||||||
The GStreamer team is thrilled to announce a new major feature release
|
GStreamer 1.19 is the development branch leading up to the next major
|
||||||
of your favourite cross-platform multimedia framework!
|
stable version which will be 1.20.
|
||||||
|
|
||||||
As always, this release is again packed with new features, bug fixes and
|
The 1.19 development series adds new features on top of the 1.18 series and is
|
||||||
other improvements.
|
|
||||||
|
|
||||||
The 1.18 release series adds new features on top of the 1.16 series and is
|
|
||||||
part of the API and ABI-stable 1.x release series of the GStreamer multimedia
|
part of the API and ABI-stable 1.x release series of the GStreamer multimedia
|
||||||
framework.
|
framework.
|
||||||
|
|
||||||
Full release notes can be found at:
|
Full release notes will one day be found at:
|
||||||
|
|
||||||
https://gstreamer.freedesktop.org/releases/1.18/
|
https://gstreamer.freedesktop.org/releases/1.20/
|
||||||
|
|
||||||
Binaries for Android, iOS, Mac OS X and Windows will usually be provided
|
Binaries for Android, iOS, Mac OS X and Windows will usually be provided
|
||||||
shortly after the release.
|
shortly after the release.
|
||||||
|
|
|
||||||
|
|
@ -321,7 +321,7 @@
|
||||||
"construct": false,
|
"construct": false,
|
||||||
"construct-only": false,
|
"construct-only": false,
|
||||||
"controllable": false,
|
"controllable": false,
|
||||||
"default": "GStreamer/1.19.0.1",
|
"default": "GStreamer/1.19.1",
|
||||||
"mutable": "null",
|
"mutable": "null",
|
||||||
"readable": true,
|
"readable": true,
|
||||||
"type": "gchararray",
|
"type": "gchararray",
|
||||||
|
|
|
||||||
|
|
@ -30,6 +30,16 @@ RTSP server library based on GStreamer
|
||||||
</GitRepository>
|
</GitRepository>
|
||||||
</repository>
|
</repository>
|
||||||
|
|
||||||
|
<release>
|
||||||
|
<Version>
|
||||||
|
<revision>1.19.1</revision>
|
||||||
|
<branch>master</branch>
|
||||||
|
<name></name>
|
||||||
|
<created>2021-06-01</created>
|
||||||
|
<file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.19.1.tar.xz" />
|
||||||
|
</Version>
|
||||||
|
</release>
|
||||||
|
|
||||||
<release>
|
<release>
|
||||||
<Version>
|
<Version>
|
||||||
<revision>1.18.0</revision>
|
<revision>1.18.0</revision>
|
||||||
|
|
|
||||||
|
|
@ -1,5 +1,5 @@
|
||||||
project('gst-rtsp-server', 'c',
|
project('gst-rtsp-server', 'c',
|
||||||
version : '1.19.0.1',
|
version : '1.19.1',
|
||||||
meson_version : '>= 0.54',
|
meson_version : '>= 0.54',
|
||||||
default_options : ['warning_level=1', 'buildtype=debugoptimized'])
|
default_options : ['warning_level=1', 'buildtype=debugoptimized'])
|
||||||
|
|
||||||
|
|
|
||||||
Loading…
Reference in a new issue