mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-31 20:48:56 +00:00
Code Issues 1 Projects Releases Wiki Activity

Release 1.19.1

Browse source
This commit is contained in:
Tim-Philipp Müller 2021-06-01 00:15:08 +01:00
parent a5b30f179b
commit a3c3afbf56
6 changed files with 492 additions and 1995 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

402
ChangeLog
View file

@ -1,11 +1,409 @@
=== release 1.18.0 ===
=== release 1.19.1 ===
2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
2021-06-01 00:15:08 +0100 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* gst-rtsp-server.doap:
* meson.build:
Release 1.19.1
2021-05-24 18:58:00 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: use new gst_buffer_new_memdup()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/208>
2021-05-04 20:47:18 -0400 Doug Nazar <nazard@nazar.ca>
* gst/rtsp-server/rtsp-media-factory-uri.c:
rtsp-media: fix leak when adding converter
Free the previous caps before reusing the variable for the converter caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
2021-05-04 20:45:19 -0400 Doug Nazar <nazard@nazar.ca>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: fix leak adding headers
gst_rtsp_message_add_header() makes a copy of the header, instead
of taking ownership.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
2021-04-21 10:43:41 +0200 François Laignel <fengalin@free.fr>
* gst/rtsp-server/rtsp-stream.c:
Use gst_element_request_pad_simple...
Instead of the deprecated gst_element_get_request_pad.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/195>
2021-04-29 03:07:42 -0400 Doug Nazar <nazard@nazar.ca>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Ensure the bus watch is removed during unprepare
It's possible for the destruction of the source to be delayed.
Instead of relying on the dispose() to remove the bus watch, do
it ourselves.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/202>
2021-04-27 09:22:21 +0200 Marc Leeman <m.leeman@televic.com>
* docs/README:
docs: minor spelling correction in README
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
2021-04-27 09:05:39 +0200 Marc Leeman <m.leeman@televic.com>
* examples/test-replay-server.c:
test-replay-server: minor spelling corrections
Bumped on these while investigating the example code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
2021-04-22 23:26:02 -0400 Doug Nazar <nazard@nazar.ca>
* tests/check/gst/stream.c:
tests: Don't fail tests if IPv6 not available.
On computers with IPv6 disabled it shouldn't result in a test failure.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
2021-04-23 07:18:48 +0200 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Add one more case to seek avoidance
This is an extension to the previous commit. There can also be cases where the
start position is not specified, in those cases we should also avoid doing
seeking unless it's forced.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/197>
2021-04-16 14:35:02 -0400 Doug Nazar <nazard@nazar.ca>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Improve skipping trickmode seek.
We can also skip the seek if the end range is already
correct.
Avoids initial seek on play start if playing full stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/194>
2021-03-19 10:36:01 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Don't run signal class handlers during the CLEANUP stage
It's sufficient to run them during the FIRST stage instead of in both.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/193>
2021-02-15 12:07:15 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspclientsink.c:
tests: rtspclientsink: fix some leaks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
2021-02-15 12:26:30 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
2021-02-15 12:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspclientsink.c:
rtspclientsink: add unit test for potential shutdown deadlock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
2021-02-15 12:01:34 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: fix deadlock on shutdown before preroll
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
2021-02-01 12:16:46 +0100 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: avoid deadlock in send_func
Currently the send_func() runs in a thread of its own which is started
the first time we enter handle_new_sample(). It runs in an outer loop
until priv->continue_sending is FALSE, which happens when a TEARDOWN
request is received. We use a local variable, cont, which is initialized
to TRUE, meaning that we will always enter the outer loop, and at the
end of the outer loop we assign it the value of priv->continue_sending.
Within the outer loop there is an inner loop, where we wait to be
signaled when there is more data to send. The inner loop is exited when
priv->send_cookie has changed value, which it does when more data is
available or when a TEARDOWN has been received.
But if we get a TEARDOWN before send_func() is entered we will get stuck
in the inner loop because no one will increase priv->session_cookie
anymore.
By not entering the outer loop in send_func() if priv->continue_sending
is FALSE we make sure that we do not get stuck in send_func()'s inner
loop should we receive a TEARDOWN before the send thread has started.
Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/187>
2021-01-22 08:58:23 +0100 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: cleanup transports during TEARDOWN
When tunneling RTP over RTSP the stream transports are stored in a hash
table in the GstRTSPClientPrivate struct. They are used for, among other
things, mapping channel id to stream transports when receiving data from
the client. The stream tranports are created and added to the hash table
in handle_setup_request(), but unfortuately they are not removed in
handle_teardown_request(). This means that if the client sends data on
the RTSP connection after it has sent the TEARDOWN, which is often the
case when audio backchannel is enabled, handle_data() will still be able
to map the channel to a session transport and pass the data along to it.
Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
because the stream is no longer joined to a bin.
We avoid this by removing the stream transports from the hash table when
we handle the TEARDOWN request.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184>
2020-12-15 11:07:01 +0200 Sebastian Dröge <sebastian@centricular.com>
* docs/gst_plugins_cache.json:
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Add "update-sdp" signal that allows updating the SDP before sending it to the server
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/178>
2020-12-23 13:54:54 -0500 John Lindgren <john.lindgren@avasure.com>
* tests/check/gst/client.c:
Add test cases for mountpoint of '/'
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
2020-11-05 16:02:49 -0500 John Lindgren <john.lindgren@avasure.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-session-media.c:
Make a mount point of "/" work correctly.
As far as I can tell, this is neither explicitly allowed nor
forbidden by RFC 7826.
Meanwhile, URLs such as rtsp://<IP>:554 or rtsp://<IP>:554/ are in
use in the wild (presumably with non-GStreamer servers).
GStreamer's prior behavior was confusing, in that
gst_rtsp_mount_points_add_factory() would appear to accept a mount
path of "" or "/", but later connection attempts would fail with a
"media not found" error.
This commit makes a mount path of "/" work for either form of URL,
while an empty mount path ("") is rejected and logs a warning.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
2020-12-15 10:18:16 +0200 Sebastian Dröge <sebastian@centricular.com>
* docs/gst_plugins_cache.json:
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/177>
2020-12-17 15:27:27 +0100 Tobias Ronge <tobiasr@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Only count senders when counting blocked streams
Only sender streams sends the GstRTSPStreamBlocking message, so only
these should be counted before setting media status to prepared.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/180>
2020-10-21 15:38:43 +0200 Jimmi Holst Christensen <jimmi.christensen@aivero.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink add proper support for uri queries
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/166>
2020-12-14 14:12:38 +1300 Lawrence Troup <lawrence.troup@teknique.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Only unref client watch context on finalize, to avoid deadlock
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/176>
2020-11-18 20:36:50 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: collect a clock_rate when blocking
This lets us provide a clock_rate in a fashion similar to the
other code paths in get_rtpinfo()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/174>
2020-11-16 10:34:41 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Use guint64 for setting the size-time property on rtpstorage
Otherwise this will cause memory corruption as the property expects a 64
bit integer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/169>
2020-11-03 16:56:28 +0100 David Phung <davidph@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-media: Ignore GstRTSPStreamBlocking from incomplete streams
To prevent cases with prerolling when the inactive stream prerolls first
and the server proceeds without waiting for the active stream, we will
ignore GstRTSPStreamBlocking messages from incomplete streams. When
there are no complete streams (during DESCRIBE), we will listen to all
streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
2020-10-28 21:48:06 +0100 Kristofer Björkström <kristofb@axis.com>
* tests/check/gst/media.c:
* tests/check/meson.build:
* tests/files/test.avi:
media test: Add test for seeking one active stream with a demuxer
Add another seek_one_active_stream test but with a demuxer. The demuxer
will flush both streams in opposed to the existing test which only
flushes the active stream. This will help exposing problems with the
prerolling process after a flushing seek.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
2018-10-29 09:19:33 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/meson.build:
* meson.build:
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
* pkgconfig/gstreamer-rtsp-server.pc.in:
* pkgconfig/meson.build:
Meson: Use pkg-config generator
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/1>
2020-10-19 11:25:25 +0300 Sebastian Dröge <sebastian@centricular.com>
* meson.build:
meson: update glib minimum version to 2.56
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/164>
2020-09-04 21:14:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/test-launch.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-server-internal.h:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/client.c:
rtsp-media-factory: expose API to disable RTCP
This is supported by the RFC, and can be useful on systems where
allocating two consecutive ports is problematic, and RTCP is not
necessary.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
2020-10-08 23:45:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* hooks/pre-commit.hook:
* meson.build:
git: use our standard pre commit hook
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/162>
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:
Release 1.18.0

2056
NEWS
View file

File diff suppressed because it is too large Load diff

15
RELEASE
View file

@ -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
of your favourite cross-platform multimedia framework!
GStreamer 1.19 is the development branch leading up to the next major
stable version which will be 1.20.
As always, this release is again packed with new features, bug fixes and
other improvements.
The 1.18 release series adds new features on top of the 1.16 series and is
The 1.19 development series adds new features on top of the 1.18 series and is
part of the API and ABI-stable 1.x release series of the GStreamer multimedia
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
shortly after the release.

2
docs/gst_plugins_cache.json
View file

@ -321,7 +321,7 @@
"construct": false,
"construct-only": false,
"controllable": false,
"default": "GStreamer/1.19.0.1",
"default": "GStreamer/1.19.1",
"mutable": "null",
"readable": true,
"type": "gchararray",

10
gst-rtsp-server.doap
View file

@ -30,6 +30,16 @@ RTSP server library based on GStreamer
</GitRepository>
</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>
<Version>
<revision>1.18.0</revision>

2
meson.build
View file

@ -1,5 +1,5 @@
project('gst-rtsp-server', 'c',
version : '1.19.0.1',
version : '1.19.1',
meson_version : '>= 0.54',
default_options : ['warning_level=1', 'buildtype=debugoptimized'])