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Release 1.13.90

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Tim-Philipp Müller 2018-03-03 22:49:34 +00:00
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=== release 1.12.0 === === release 1.13.90 ===
2017-05-04 Sebastian Dröge <slomo@coaxion.net> 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.13.90
2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-permissions.c:
permissions: add Since tags and example for new API
2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-permissions.c:
* gst/rtsp-server/rtsp-permissions.h:
* tests/check/gst/permissions.c:
permissions: more bindings-friendly API
https://bugzilla.gnome.org/show_bug.cgi?id=793975
2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* meson.build:
meson: enable more warnings
2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Place netaddress meta on packets received via TCP
This allows us to later map signals from rtpbin/rtpsource back to the
corresponding stream transport, and allows to do keep-alive based on
RTCP packets in case of TCP media transport.
https://bugzilla.gnome.org/show_bug.cgi?id=789646
2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: if OPEN failed, unqueue next command
As READY_TO_PAUSED can no longer return async, the RECORD
command will be queued before the OPEN command fails
(for example in case the server could not be connected),
and record then waits for ever.
https://bugzilla.gnome.org/show_bug.cgi?id=793896
2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: fix retrieval of custom payloader caps
If a bin is passed as the custom payloader, the caps of
its factory will be empty, the correct way to obtain the caps
is to query its sinkpad.
2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: fix extra unref of custom payloader
2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rspclientsink: fix recent code indentation
2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: add missing get_type prototype
2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: allow setting payloader as pad property
This was a FIXME item, and can be quite useful, also
allowing to specify payloader properties from the command
line, which is always nice.
https://bugzilla.gnome.org/show_bug.cgi?id=793776
2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Replace g_print() log line
https://bugzilla.gnome.org/show_bug.cgi?id=793838
2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/rtspclientsink.c:
rtsp-media: fix RECORD getting stuck
The test_record case was working because async=false had
been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
but that was incorrect, as it should not be needed.
Removing async=false made the test fail as expected, this is
fixed by not trying to preroll when preparing the media for
RECORD, as start_prepare is called upon receiving ANNOUNCE,
and our peer will not start sending media until it has received
a response to that request, and sent and received a response
to RECORD as well, thus obviously preventing preroll.
https://bugzilla.gnome.org/show_bug.cgi?id=793738
2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-auth.c:
rtsp-auth: fix set_tls_authentication_mode annotation
2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
* gst/rtsp-server/rtsp-onvif-media.c:
rtp-server: remove redefined variable
res is a boolean variable which is defined in the function scope and
redefined, with no reason, in the loop scope. This patch removes the
redefinition.
https://bugzilla.gnome.org/show_bug.cgi?id=793592
2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: Add functions for checking if stream is receiver or sender
...and replace all checks for RECORD in GstRTSPMedia which are really
for "sender-only". This way the code becomes more generic and introducing
support for onvif-backchannel later on will require no changes in
GstRTSPMedia.
2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-onvif-media-factory.c:
* gst/rtsp-server/rtsp-onvif-media-factory.h:
onvif: Make requires_backchannel() public
...in order to let subclasses building the onvif part of the pipeline
check whether backchannel shall be included or not.
2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-onvif-media.c:
rtsp-server: Switch around sendonly/recvonly attributes
They are wrong in the ONVIF streaming spec. The backchannel should be
recvonly and the normal media should be sendonly: direction is always
from the point of view of the SDP offerer (the server) according to
RFC 3264.
2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
* docs/libs/gst-rtsp-server-docs.sgml:
* docs/libs/gst-rtsp-server-sections.txt:
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-onvif-backchannel.c:
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-onvif-client.c:
* gst/rtsp-server/rtsp-onvif-client.h:
* gst/rtsp-server/rtsp-onvif-media-factory.c:
* gst/rtsp-server/rtsp-onvif-media-factory.h:
* gst/rtsp-server/rtsp-onvif-media.c:
* gst/rtsp-server/rtsp-onvif-media.h:
* gst/rtsp-server/rtsp-onvif-server.c:
* gst/rtsp-server/rtsp-onvif-server.h:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-sdp.h:
rtsp: Add support for ONVIF backchannel
This adds a new RTSP server, client, media-factory and media subclass
for handling the specifics of the backchannel. Ideally this later can be
extended with other ONVIF specific features.
2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Add support for sending+receiving medias
We need to add an appsrc/appsink in that case because otherwise the
media bin will be a sink and a source for rtpbin, causing a pipeline
loop.
https://bugzilla.gnome.org/show_bug.cgi?id=788950
2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
* configure.ac: * configure.ac:
releasing 1.12.0 * meson.build:
Back to development
=== release 1.13.1 ===
2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
* NEWS:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.13.1
2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-session-pool.c:
session-pool: remove nullable return annotation
create_watch can only return NULL from the API guards, no
need for nullable.
2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
set_clock functions: Add nullable annotations
2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-thread-pool.c:
All around: add annotations and API guards
2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* tests/test-cleanup.c:
test-cleanup: bind any port
The meson test suite runs tests in parallel, trying to bind
a single port made the test fail.
2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: make version numbers ints and fix int/string comparison
WARNING: Trying to compare values of different types (str, int).
The result of this is undefined and will become a hard error
in a future Meson release.
2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-context.c:
gst_rtsp_context_get_current: add (skip) annotation
The return value type is defined with G_DEFINE_POINTER_TYPE,
and gi emits the following warning:
Invalid non-constant return of bare structure or union; register as
boxed type or (skip)
2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: add type annotations
gi doesn't seem to be able to figure out the type of the
signal parameters when defined with G_DEFINE_POINTER_TYPE
2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
autotools: use -fno-strict-aliasing where supported
https://bugzilla.gnome.org/show_bug.cgi?id=769183
2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: use -fno-strict-aliasing where supported
https://bugzilla.gnome.org/show_bug.cgi?id=769183
2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-mount-points.c:
mount-points: bail out of loop again when matching mount points
Previous patch led to us iterating the entire sequence. Bail out
of the loop again if we have a match but are moving away from it.
https://bugzilla.gnome.org/show_bug.cgi?id=771555
2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/mountpoints.c:
tests: mountpoints: add more checks for mount point path matching
https://bugzilla.gnome.org/show_bug.cgi?id=771555
2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
* gst/rtsp-server/rtsp-mount-points.c:
mount-points: fix matching of paths where there's also an entry with a common prefix
e.g. with the following mount points
/raw
/raw/snapshot
/raw/video
_match() would not match /raw/video and /raw/snapshot correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=771555
2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-permissions.c:
* gst/rtsp-server/rtsp-permissions.h:
* tests/check/gst/permissions.c:
permissions: add some new API to make this usable from bindings
https://bugzilla.gnome.org/show_bug.cgi?id=787073
2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-token.c:
rtsp-token: annotate constructors for bindings
This maps _new_empty() to _new(), which also makes RTSPToken()
work properly now. Since this API wasn't usable from bindings
before, this should hopefully be fine.
https://bugzilla.gnome.org/show_bug.cgi?id=787073
2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-token.c:
* gst/rtsp-server/rtsp-token.h:
* tests/check/gst/token.c:
rtsp-token: add some API to set fields from bindings
The existing functions are all vararg-based and as such
not usable from bindings.
https://bugzilla.gnome.org/show_bug.cgi?id=787073
2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspclientsink.c:
* tests/check/gst/rtspserver.c:
* tests/check/gst/sessionpool.c:
* tests/check/gst/stream.c:
tests: fix indentation
Fix and "fix".
2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspserver.c:
tests: rtspserver: fix another ref leak
Even if this didn't show up in valgrind.
2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspclientsink.c:
tests: rtspclientsink: fix leak
2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
* tests/check/gst/rtspserver.c:
test: rtspserver: plug memory leak in test_no_session_timeout
In test_no_session_timeout, unref the rtsp session object when the
test is done.
https://bugzilla.gnome.org/show_bug.cgi?id=792127
2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtpsclientsink: Initialize and clear newly added mutex and cond
While it *did* work, glib would automatically create new mutex and cond
... which never got freed
2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Set multicast TTL on the multicast sockets
And not if we do unicast UDP.
https://bugzilla.gnome.org/show_bug.cgi?id=791743
2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
In the multicast case (as in test-multicast, not test-multicast2), the
address could be allocated/reserved (and thus set) already without
allocating the actual socket. We need to allocate the socket here still
instead of just claiming that it was already allocated.
See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
rtspclientsink: Use the new rtsp-stream API
https://bugzilla.gnome.org/show_bug.cgi?id=790412
2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
rtspclientsink: Wait until OPEN has been scheduled
Make sure that the sink thread has started opening connection
to the server before continuing.
https://bugzilla.gnome.org/show_bug.cgi?id=790412
2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
* common:
Automatic update of common submodule
From e8c7a71 to 3fa2c9e
2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-server: Minor doc fixes
Mostly for g-i
2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
* Makefile.am:
* tests/Makefile.am:
tests: disable all tests when --disable-tests is used
Move conditional subdir include into top level.
Based on patch by: Joel Holdsworth
https://bugzilla.gnome.org/show_bug.cgi?id=757703
2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
* meson_options.txt:
* tests/meson.build:
meson: build more tests and add options to disable tests and examples
2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
* gst/rtsp-server/rtsp-session.c:
Fix build when -Werror=deprecated-declarations is on
As gst_rtsp_session_next_timeout is deprecated.
```
../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:760:3: error: ‘gst_rtsp_session_next_timeout’ is deprecated: Use 'gst_rtsp_session_next_timeout_usec' instead [-Werror=deprecated-declarations]
res = (gst_rtsp_session_next_timeout (session, now) == 0);
^~~
../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
```
2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
* common:
Automatic update of common submodule
From 3f4aa96 to e8c7a71
2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
* tests/check/gst/media.c:
check/media: Add seekability test case: not all streams are active
Media contains two streams but only one is complete and prepared
for playing.
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Do not reset 'blocking' if stream is already blocked
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: remove vs_module_defs_dir variable which is no longer needed
2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-session.h:
rtsp: fix distcheck
2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
* Makefile.am:
* gst/rtsp-server/meson.build:
* win32/MANIFEST:
* win32/common/libgstrtspserver.def:
win32: remove .def file with exports
They're no longer needed, symbol exporting is now explicit
via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
autotools: stop controlling symbol visibility with -export-symbols-regex
Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
This should result in consistent behaviour for the autotools and
Meson builds.
2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
rtsp-server: add missing GST_EXPORT and export deprecated funcs
2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
* tests/check/gst/media.c:
check: Add seekability testing on medias
Make sure that once GstRTSPMedia are prepared they returned
the expected seekability results
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* win32/common/libgstrtspserver.def:
rtsp-media: Enable seeking query before pipeline is complete
SDP are now provided *before* the pipeline is fully complete. In order
to know whether a media is seekable or not therefore requires asking
the invididual streams.
API: gst_rtsp_stream_seekable
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix handling in default_unsuspend()
Handle the case when streams are not blocked and media
is suspended from PAUSED.
Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
* tests/check/gst/media.c:
check/media: Fix thread pool leak.
Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Removed fakesink elements
There is not need of adding fakesink elements to the media
pipeline in the dynamic-payloader case.
The media pipeline itself is dynamically updated with
the receiver and sender parts that are based on the client
transport information known after SETUP has been received.
Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Corrected ASYNC_DONE handling
Media is complete when all the transport based parts are
added to the media pipeline. At this point ASYNC_DONE is
posted by the media pipeline and media is ready to enter
the PREPARED state.
Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
* tests/check/gst/media.c:
check/media: Check that prepared media can provide a SDP
Whenever a RTSPMedia is prepared, it should be able to provide a SDP
2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Don't leak addr
CID #1422260
2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream.c:
Run gst-indent
2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't unblock with remaining dynamic payloaders
If we still have some dynamic paylaoders which haven't posted
no-more-pads yet, don't go to PREPARED if one of the streams
blocked.
The risk was that we would end up not exposing/using all specified
streams.
The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
then it will take a bit more time to start. But only if those 3
conditions are present.
https://bugzilla.gnome.org/show_bug.cgi?id=769521
2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix doc
2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't set float on a gint64 variable
Just use 0. Fixes 'undefined' behaviour from clang
2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix previous commit
We only want to count dynamic payloaders
2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
rtsp-media: Handle multiple dynamic elements
If we have more than one dynamic payloader in the pipeline, we need
to wait until the *last* one emits 'no-more-pads' before switching
to PREPARED.
Failure to do so would result in a race where some of the streams
wouldn't properly be prepared
https://bugzilla.gnome.org/show_bug.cgi?id=769521
2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
* win32/common/libgstrtspserver.def:
win32: Fix exported symbols list
2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Only update the RTP udpsink if it actually exists
For send-only streams it does not exist, but the RTCP udpsink might.
2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
* win32/common/libgstrtspserver.def:
win32: Update exports
2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-media: seek on media pipelines that are complete
Make sure that a seek is performed on pipelines that
contain at least one sink element.
Change-Id: Icf398e10add3191d104b1289de612412da326819
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/media.c:
* tests/check/gst/rtspserver.c:
* tests/check/gst/stream.c:
Dynamically reconfigure pipeline in PLAY based on transports
The initial pipeline does not contain specific transport
elements. The receiver and the sender parts are added
after PLAY.
If the media is shared, the streams are dynamically
reconfigured after each PLAY.
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: obtain stream position from pad
If no sinks have been added yet, obtain the current and
the stop position of the stream from the send_src pad.
Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
rtsp-session-media: add function to get a list of transports
Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: add functions to get rtp and rtcp multicast sockets
Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
stream: set async=sync=false only for RTCP appsink
Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: return minimum value in query position case
The minimum position should be returned as we are interested
in the whole interval.
Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
* gst/rtsp-server/rtsp-session.c:
* tests/check/gst/rtspserver.c:
rtsp-session: Handle the case when timeout=0
According to the documentation, a timeout of value 0 means
that the session never timeouts. This adds handling of that.
If timeout=0 we just return with a -1 from
gst_rtsp_session_next_timeout_usec ().
https://bugzilla.gnome.org/show_bug.cgi?id=785058
2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
https://bugzilla.gnome.org/show_bug.cgi?id=785024
2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media-factory.c:
docs: add media factory transport mode accessors
and fix the documentation for the return value of the getter
2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: unref 'pipelined_requests' in finalize
The hash table priv->pipelined_requests is not unref:ed in the
finalize funktion. Make sure it is.
https://bugzilla.gnome.org/show_bug.cgi?id=788704
2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Initialize scalar variable
CID 1418985
2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
* win32/common/libgstrtspserver.def:
win32: Update export file
2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
Start support for RTSP 2.0
This adds basic support for new 2.0 features, though the protocol is
subposdely backward incompatible, most semantics are the sames.
This commit adds:
- features:
* version negotiation
* pipelined requests support
* Media-Properties support
* Accept-Ranges support
- APIs:
* gst_rtsp_media_seekable
The RTSP methods that have been removed when using 2.0 now return
BAD_REQUEST.
https://bugzilla.gnome.org/show_bug.cgi?id=781446
2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
* gst/rtsp-server/rtsp-stream.c:
stream: Use stream duration as stream-stop if segment was not configured with a stop
Allowing client to know stream duration when no seeking happened.
https://bugzilla.gnome.org/show_bug.cgi?id=783435
2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
rtsp-media-factory: Don't cache any media if NULL was returned as key
The docs already mentioned this, but we actually stored it in the hash
table with key==NULL and leaked its reference forever.
2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
rtspclientsink: Use a mutex for protecting against concurrent send/receives
This is a simple port of:
* a722f6e8329032c6eda4865d6a07f4ba5981d7ea
* c438545dc9e2f14f657bc0ef261fff726449867b
* cd17c71dcea5c9310d21f1347c7520983e5869ac
in gst-plugins-good.
2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
* gst/rtsp-server/rtsp-sdp.c:
sdp: fix Memory leak in error case
https://bugzilla.gnome.org/show_bug.cgi?id=787059
2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
* pkgconfig/meson.build:
meson: don't install -uninstalled.pc file
https://bugzilla.gnome.org/show_bug.cgi?id=786457
2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From 48a5d85 to 3f4aa96
2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Fix typo in debug message
2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: hide symbols by default unless explicitly exported
2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
Fixes meson warning about undefined @srcdir@.
2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/meson.build:
meson: skip tests on windows for now
As we do in the other modules. As libgstcheck is currently not
built on windows. Fixes "Fallback variable 'gst_check_dep' in
the subproject 'gstreamer' does not exist"" Meson error.
2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fix connection delay due to wrong assumption on last-sample
Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
multiudpsink's last-sample always comes from the payloader. Which
is wrong if auxiliary streams are multiplexed in the same stream.
So check the buffer's ssrc against the caps'ssrc before to use its
seqnum. If not the same ssrc just use the payloader as done prior
the commit above or when there is no last-sample yet.
https://bugzilla.gnome.org/show_bug.cgi?id=784094
2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
* meson.build:
meson: Allow using glib as a subproject
2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: fix with-package-name option
https://bugzilla.gnome.org/show_bug.cgi?id=784082
2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* Makefile.am:
Distribute meson_options.txt
2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* Makefile.am:
And config.h.meson is no longer dist either
2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
* config.h.meson:
* meson.build:
meson: config.h.meson is no longer needed
2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
* tests/check/meson.build:
* tests/meson.build:
meson: Fix building tests and activate them again
2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
* tests/check/meson.build:
meson: Do not use path separator in test names
Avoiding warnings like:
WARNING: Target "elements/audioamplify" has a path separator in its name.
2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
* meson_options.txt:
meson: add options to set package name and origin
https://bugzilla.gnome.org/show_bug.cgi?id=782172
2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-address-pool.h:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-context.h:
* gst/rtsp-server/rtsp-media-factory-uri.h:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-mount-points.h:
* gst/rtsp-server/rtsp-params.h:
* gst/rtsp-server/rtsp-permissions.h:
* gst/rtsp-server/rtsp-sdp.h:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.h:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.h:
* gst/rtsp-server/rtsp-thread-pool.h:
* gst/rtsp-server/rtsp-token.h:
Mark symbols explicitly for export with GST_EXPORT
2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* configure.ac:
* gst/rtsp-sink/Makefile.am:
Remove plugin specific static build option
Static and dynamic plugins now have the same interface. The standard
--enable-static/--enable-shared toggle are sufficient.
2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
* meson.build:
Back to development
=== release 1.12.0 ===
2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.12.0
=== release 1.11.91 === === release 1.11.91 ===

281
NEWS
View file

@ -1,174 +1,235 @@
# GStreamer 1.14 Release Notes
GSTREAMER 1.14 RELEASE NOTES
GStreamer 1.14.0 has not been released yet. It is scheduled for release GStreamer 1.14.0 has not been released yet. It is scheduled for release
in late February / early March 2018. in early March 2018.
There are unstable pre-releases available for testing and development purposes. There are unstable pre-releases available for testing and development
The latest pre-release is version 1.13.1 and was released on 15 February 2018. purposes. The latest pre-release is version 1.13.90 (rc1) and was
released on 03 March 2018.
See [https://gstreamer.freedesktop.org/releases/1.14/][latest] for the latest See https://gstreamer.freedesktop.org/releases/1.14/ for the latest
version of this document. version of this document.
*Last updated: Thursday 15 February 2018, 16:30 UTC [(log)][gitlog]* _Last updated: Saturday 03 March 2018, 16:30 UTC (log)_
[latest]: https://gstreamer.freedesktop.org/releases/1.14/
[gitlog]: https://cgit.freedesktop.org/gstreamer/www/log/src/htdocs/releases/1.14/release-notes-1.14.md
## Introduction
The GStreamer team is proud to announce a new major feature release in the
stable 1.x API series of your favourite cross-platform multimedia framework!
As always, this release is again packed with new features, bug fixes and other
improvements.
## Highlights
- this section will be completed shortly
## Major new features and changes
### Noteworthy new API
- this section will be filled in shortly
### New Elements
- this section will be filled in shortly
### New element features and additions
- this section will be filled in shortly
### Plugin and library moves
- this section will be filled in shortly
### Plugin removals
- this section will be filled in shortly
## Miscellaneous API additions Introduction
- this section will be filled in shortly The GStreamer team is proud to announce a new major feature release in
the stable 1.x API series of your favourite cross-platform multimedia
framework!
### GstPlayer As always, this release is again packed with new features, bug fixes and
other improvements.
- this section will be filled in shortly
## Miscellaneous changes Highlights
- this section will be filled in shortly - this section will be completed shortly
### OpenGL integration
- this section will be filled in shortly Major new features and changes
## Tracing framework and debugging improvements Noteworthy new API
- this section will be filled in shortly - this section will be filled in shortly
## Tools New Elements
- this section will be filled in shortly - this section will be filled in shortly
## GStreamer RTSP server New element features and additions
- this section will be filled in shortly - this section will be filled in shortly
## GStreamer VAAPI Plugin and library moves
- this section will be filled in shortly - this section will be filled in shortly
## GStreamer Editing Services and NLE Plugin removals
- this section will be filled in shortly - this section will be filled in shortly
## GStreamer validate
- this section will be filled in shortly Miscellaneous API additions
## GStreamer Python Bindings - this section will be filled in shortly
- this section will be filled in shortly GstPlayer
## Build and Dependencies - this section will be filled in shortly
- this section will be filled in shortly
## Platform-specific improvements Miscellaneous changes
### Android - this section will be filled in shortly
- this section will be filled in shortly OpenGL integration
### macOS and iOS - this section will be filled in shortly
- this section will be filled in shortly
### Windows Tracing framework and debugging improvements
- this section will be filled in shortly - this section will be filled in shortly
## Contributors
- this section will be filled in shortly Tools
- this section will be filled in shortly
GStreamer RTSP server
- this section will be filled in shortly
GStreamer VAAPI
- this section will be filled in shortly
GStreamer Editing Services and NLE
- this section will be filled in shortly
GStreamer validate
- this section will be filled in shortly
GStreamer Python Bindings
- this section will be filled in shortly
Build and Dependencies
- this section will be filled in shortly
Platform-specific improvements
Android
- this section will be filled in shortly
macOS and iOS
- this section will be filled in shortly
Windows
- this section will be filled in shortly
Contributors
Aaron Boxer, Adrián Pardini, Adrien SCH, Akinobu Mita, Alban Bedel,
Alessandro Decina, Alex Ashley, Alicia Boya García, Alistair Buxton,
Alvaro Margulis, Anders Jonsson, Andreas Frisch, Andrejs Vasiljevs,
Andrew Bott, Antoine Jacoutot, Antonio Ospite, Antoni Silvestre, Anton
Obzhirov, Anuj Jaiswal, Arjen Veenhuizen, Arnaud Bonatti, Arun Raghavan,
Ashish Kumar, Aurélien Zanelli, Ayaka, Branislav Katreniak, Branko
Subasic, Brion Vibber, Carlos Rafael Giani, Cassandra Rommel, Chris
Bass, Chris Paulson-Ellis, Christoph Reiter, Claudio Saavedra, Clemens
Lang, Cyril Lashkevich, Daniel van Vugt, Dave Craig, Dave Johnstone,
David Evans, David Schleef, Deepak Srivastava, Dimitrios Katsaros,
Dmitry Zhadinets, Dongil Park, Dustin Spicuzza, Eduard Sinelnikov,
Edward Hervey, Enrico Jorns, Eunhae Choi, Ezequiel Garcia, fengalin,
Filippo Argiolas, Florent Thiéry, Florian Zwoch, Francisco Velazquez,
François Laignel, fvanzile, George Kiagiadakis, Georg Lippitsch, Graham
Leggett, Guillaume Desmottes, Gurkirpal Singh, Gwang Yoon Hwang, Gwenole
Beauchesne, Haakon Sporsheim, Haihua Hu, Håvard Graff, Heekyoung Seo,
Heinrich Fink, Holger Kaelberer, Hoonhee Lee, Hosang Lee, Hyunjun Ko,
Ian Jamison, James Stevenson, Jan Alexander Steffens (heftig), Jan
Schmidt, Jason Lin, Jens Georg, Jeremy Hiatt, Jérôme Laheurte, Jimmy
Ohn, Jochen Henneberg, John Ludwig, John Nikolaides, Jonathan Karlsson,
Josep Torra, Juan Navarro, Juan Pablo Ugarte, Julien Isorce, Jun Xie,
Jussi Kukkonen, Justin Kim, Lasse Laursen, Lubosz Sarnecki, Luc
Deschenaux, Luis de Bethencourt, Marcin Lewandowski, Mario Alfredo
Carrillo Arevalo, Mark Nauwelaerts, Martin Kelly, Matej Knopp, Mathieu
Duponchelle, Matteo Valdina, Matt Fischer, Matthew Waters, Matthieu
Bouron, Matthieu Crapet, Matt Staples, Michael Catanzaro, Michael
Olbrich, Michael Shigorin, Michael Tretter, Michał Dębski, Michał Górny,
Michele Dionisio, Miguel París, Mikhail Fludkov, Munez, Nael Ouedraogo,
Neos3452, Nicholas Panayis, Nick Kallen, Nicola Murino, Nicolas
Dechesne, Nicolas Dufresne, Nirbheek Chauhan, Ognyan Tonchev, Ole André
Vadla Ravnås, Oleksij Rempel, Olivier Crête, Omar Akkila, Orestis
Floros, Patricia Muscalu, Patrick Radizi, Paul Kim, Per-Erik Brodin,
Peter Seiderer, Philip Craig, Philippe Normand, Philippe Renon, Philipp
Zabel, Pierre Pouzol, Piotr Drąg, Ponnam Srinivas, Pratheesh Gangadhar,
Raimo Järvi, Ramprakash Jelari, Ravi Kiran K N, Reynaldo H. Verdejo
Pinochet, Rico Tzschichholz, Robert Rosengren, Roland Peffer, Руслан
Ижбулатов, Sam Hurst, Sam Thursfield, Sangkyu Park, Sanjay NM, Satya
Prakash Gupta, Scott D Phillips, Sean DuBois, Sebastian Cote, Sebastian
Dröge, Sebastian Rasmussen, Sejun Park, Sergey Borovkov, Seungha Yang,
Shakin Chou, Shinya Saito, Simon Himmelbauer, Sky Juan, Song Bing,
Sreerenj Balachandran, Stefan Kost, Stefan Popa, Stefan Sauer, Stian
Selnes, Thiago Santos, Thibault Saunier, Thijs Vermeir, Tim Allen,
Tim-Philipp Müller, Ting-Wei Lan, Tomas Rataj, Tom Bailey, Tonu Jaansoo,
U. Artie Eoff, Umang Jain, Ursula Maplehurst, VaL Doroshchuk, Vasilis
Liaskovitis, Víctor Manuel Jáquez Leal, vijay, Vincent Penquerc'h,
Vineeth T M, Vivia Nikolaidou, Wang Xin-yu (王昕宇), Wei Feng, Wim
Taymans, Wonchul Lee, Xabier Rodriguez Calvar, Xavier Claessens,
XuGuangxin, Yasushi SHOJI, Yi A Wang, Youness Alaoui,
... and many others who have contributed bug reports, translations, sent ... and many others who have contributed bug reports, translations, sent
suggestions or helped testing. suggestions or helped testing.
## Bugs fixed in 1.14
- this section will be filled in shortly Bugs fixed in 1.14
More than [704 bugs][bugs-fixed-in-1.14] have been fixed during - this section will be filled in shortly
the development of 1.14.
More than 704 bugs have been fixed during the development of 1.14.
This list does not include issues that have been cherry-picked into the This list does not include issues that have been cherry-picked into the
stable 1.12 branch and fixed there as well, all fixes that ended up in the stable 1.12 branch and fixed there as well, all fixes that ended up in
1.12 branch are also included in 1.14. the 1.12 branch are also included in 1.14.
This list also does not include issues that have been fixed without a bug This list also does not include issues that have been fixed without a
report in bugzilla, so the actual number of fixes is much higher. bug report in bugzilla, so the actual number of fixes is much higher.
[bugs-fixed-in-1.14]: https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&classification=Platform&limit=0&list_id=213265&order=bug_id&product=GStreamer&query_format=advanced&resolution=FIXED&target_milestone=1.12.1&target_milestone=1.12.2&target_milestone=1.12.3&target_milestone=1.12.4&target_milestone=1.13.1&target_milestone=1.13.2&target_milestone=1.13.3&target_milestone=1.13.4&target_milestone=1.13.90&target_milestone=1.13.91&target_milestone=1.14.0
## Stable 1.14 branch Stable 1.14 branch
After the 1.14.0 release there will be several 1.14.x bug-fix releases which After the 1.14.0 release there will be several 1.14.x bug-fix releases
will contain bug fixes which have been deemed suitable for a stable branch, which will contain bug fixes which have been deemed suitable for a
but no new features or intrusive changes will be added to a bug-fix release stable branch, but no new features or intrusive changes will be added to
usually. The 1.14.x bug-fix releases will be made from the git 1.14 branch, a bug-fix release usually. The 1.14.x bug-fix releases will be made from
which is a stable branch. the git 1.14 branch, which is a stable branch.
### 1.14.0 1.14.0
1.14.0 is scheduled to be released in late February / early March 2018. 1.14.0 is scheduled to be released in early March 2018.
## Known Issues
- The `webrtcdsp` element is currently not shipped as part of the Windows Known Issues
binary packages due to a [build system issue][bug-770264].
[bug-770264]: https://bugzilla.gnome.org/show_bug.cgi?id=770264 - The webrtcdsp element is currently not shipped as part of the
Windows binary packages due to a build system issue.
## Schedule for 1.16
Our next major feature release will be 1.16, and 1.15 will be the unstable Schedule for 1.16
development version leading up to the stable 1.16 release. The development
of 1.15/1.16 will happen in the git master branch.
The plan for the 1.16 development cycle is yet to be confirmed, but it is Our next major feature release will be 1.16, and 1.15 will be the
expected that feature freeze will be around August 2017 unstable development version leading up to the stable 1.16 release. The
followed by several 1.15 pre-releases and the new 1.16 stable release development of 1.15/1.16 will happen in the git master branch.
in September.
1.16 will be backwards-compatible to the stable 1.14, 1.12, 1.10, 1.8, 1.6, 1.4, The plan for the 1.16 development cycle is yet to be confirmed, but it
1.2 and 1.0 release series. is expected that feature freeze will be around August 2017 followed by
several 1.15 pre-releases and the new 1.16 stable release in September.
- - - 1.16 will be backwards-compatible to the stable 1.14, 1.12, 1.10, 1.8,
1.6, 1.4, 1.2 and 1.0 release series.
*These release notes have been prepared by Tim-Philipp Müller.* ------------------------------------------------------------------------
*License: [CC BY-SA 4.0](http://creativecommons.org/licenses/by-sa/4.0/)* _These release notes have been prepared by Tim-Philipp Müller._
_License: CC BY-SA 4.0_

67
RELEASE
View file

@ -1,26 +1,63 @@
This is GStreamer gst-rtsp-server 1.13.90.
Release notes for GStreamer RTSP Server Library 1.12.0 The GStreamer team is pleased to announce the first release candidate for the
upcoming stable 1.14 release series.
The GStreamer team is pleased to announce the first release in the stable 1.12 The 1.14 release series adds new features on top of the 1.0, 1.2, 1.4, 1.6,
release series. The 1.12 release series is adding new features on top of the 1.8, 1.10 and 1.12 series and is part of the API and ABI-stable 1.x release
1.0, 1.2, 1.4, 1.6, 1.8 and 1.10 series and is part of the API and ABI-stable series of the GStreamer multimedia framework.
1.x release series of the GStreamer multimedia framework.
Full release notes can be found at:
Full release notes can be found here https://gstreamer.freedesktop.org/releases/1.14/
Binaries for Android, iOS, Mac OS X and Windows will be provided shortly
after the release.
Binaries for Android, iOS, Mac OS X and Windows will be provided in the next days. This module will not be very useful by itself and should be used in conjunction
with other GStreamer modules for a complete multimedia experience.
- gstreamer: provides the core GStreamer libraries and some generic plugins
- gst-plugins-base: a basic set of well-supported plugins and additional
media-specific GStreamer helper libraries for audio,
video, rtsp, rtp, tags, OpenGL, etc.
- gst-plugins-good: a set of well-supported plugins under our preferred
license
- gst-plugins-ugly: a set of well-supported plugins which might pose
problems for distributors
- gst-plugins-bad: a set of plugins of varying quality that have not made
their way into one of core/base/good/ugly yet, for one
reason or another. Many of these are are production quality
elements, but may still be missing documentation or unit
tests; others haven't passed the rigorous quality testing
we expect yet.
- gst-libav: a set of codecs plugins based on the ffmpeg library. This is
where you can find audio and video decoders and encoders
for a wide variety of formats including H.264, AAC, etc.
- gstreamer-vaapi: hardware-accelerated video decoding and encoding using
VA-API on Linux. Primarily for Intel graphics hardware.
- gst-omx: hardware-accelerated video decoding and encoding, primarily for
embedded Linux systems that provide an OpenMax
implementation layer such as the Raspberry Pi.
- gst-rtsp-server: library to serve files or streaming pipelines via RTSP
- gst-editing-services: library an plugins for non-linear editing
==== Download ==== ==== Download ====
You can find source releases of gst-rtsp-server in the download You can find source releases of gstreamer in the download
directory: https://gstreamer.freedesktop.org/src/gst-rtsp-server/ directory: https://gstreamer.freedesktop.org/src/gstreamer/
The git repository and details how to clone it can be found at The git repository and details how to clone it can be found at
http://cgit.freedesktop.org/gstreamer/gst-rtsp-server/ http://cgit.freedesktop.org/gstreamer/gstreamer/
==== Homepage ==== ==== Homepage ====
@ -45,11 +82,3 @@ from there (see link above).
Interested developers of the core library, plugins, and applications should Interested developers of the core library, plugins, and applications should
subscribe to the gstreamer-devel list. subscribe to the gstreamer-devel list.
Contributors to this release
* Sebastian Dröge
* Thibault Saunier
* Tim-Philipp Müller
 

12
configure.ac
View file

@ -2,7 +2,7 @@ AC_PREREQ(2.69)
dnl initialize autoconf dnl initialize autoconf
dnl when going to/from release please set the nano (fourth number) right ! dnl when going to/from release please set the nano (fourth number) right !
dnl releases only do Wall, cvs and prerelease does Werror too dnl releases only do Wall, cvs and prerelease does Werror too
AC_INIT([GStreamer RTSP Server Library], [1.13.1.1], AC_INIT([GStreamer RTSP Server Library], [1.13.90],
[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer], [http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
[gst-rtsp-server]) [gst-rtsp-server])
AG_GST_INIT AG_GST_INIT
@ -53,13 +53,13 @@ dnl 1.2.5 => 205
dnl 1.10.9 (who knows) => 1009 dnl 1.10.9 (who knows) => 1009
dnl dnl
dnl sets GST_LT_LDFLAGS dnl sets GST_LT_LDFLAGS
AS_LIBTOOL(GST, 1301, 0, 1301) AS_LIBTOOL(GST, 1390, 0, 1390)
dnl *** required versions of GStreamer stuff *** dnl *** required versions of GStreamer stuff ***
GST_REQ=1.13.1.1 GST_REQ=1.13.90
GSTPB_REQ=1.13.1.1 GSTPB_REQ=1.13.90
GSTPG_REQ=1.13.1.1 GSTPG_REQ=1.13.90
GSTPD_REQ=1.13.1.1 GSTPD_REQ=1.13.90
dnl *** autotools stuff **** dnl *** autotools stuff ****

12
gst-rtsp-server.doap
View file

@ -28,7 +28,17 @@ RTSP server library based on GStreamer
<location rdf:resource="git://anongit.freedesktop.org/gstreamer/gst-rtsp-server"/> <location rdf:resource="git://anongit.freedesktop.org/gstreamer/gst-rtsp-server"/>
<browse rdf:resource="http://cgit.freedesktop.org/gstreamer/gst-rtsp-server"/> <browse rdf:resource="http://cgit.freedesktop.org/gstreamer/gst-rtsp-server"/>
</GitRepository> </GitRepository>
</repository> </repository>
<release>
<Version>
<revision>1.13.90</revision>
<branch>master</branch>
<name></name>
<created>2018-03-03</created>
<file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.13.90.tar.xz" />
</Version>
</release>
<release> <release>
<Version> <Version>

2
meson.build
View file

@ -1,5 +1,5 @@
project('gst-rtsp-server', 'c', project('gst-rtsp-server', 'c',
version : '1.13.1.1', version : '1.13.90',
meson_version : '>= 0.33.0', meson_version : '>= 0.33.0',
default_options : ['warning_level=1', 'buildtype=debugoptimized']) default_options : ['warning_level=1', 'buildtype=debugoptimized'])