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Release 1.5.1

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Sebastian Dröge 2015-06-07 11:20:01 +02:00
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@ -1,9 +1,921 @@
=== release 1.4.0 ===
=== release 1.5.1 ===
2014-07-19 Sebastian Dröge <slomo@coaxion.net>
2015-06-07 Sebastian Dröge <slomo@coaxion.net>
* configure.ac:
releasing 1.4.0
releasing 1.5.1
2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: No flush during Teardown.
When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
backlog is empty it can happen that just a part of a message will be
sent and rest is in backlog queue. If then flush during teardown
just a part of message will be sent.This can lead to client miss
teardown response since it expect to get the last part of message.
The flushing during teardown was introduced to fix a deadlock that now
is fixed more generally in handle_request by temporary setting backlog
size to unlimited.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/Makefile.am:
tests: Use AM_TESTS_ENVIRONMENT
Needed by the new automake test runner and the
current version of the common submodule.
2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.h:
rtsp-server: Use single-include rtsp header to make sure we get all definitions
2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Mark some more functions static
2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Only unblock the media in suspend() when actually changing the state
Otherwise we're going to lose a few packets for live streams during DESCRIBE.
2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-video-rtx.c:
examples: Use AVPF profile for the RTX example
2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: Only add RTX to the SDP when using a feedback profile
2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: get valid clock-rate from last-sample
clock-rate in last-sample's caps is integer, not unsigned.
To get this value properly, variable needs to be type-casted to int.
https://bugzilla.gnome.org/show_bug.cgi?id=747614
2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
* autogen.sh:
* common:
autogen.sh: only run autopoint if gettext requested in configure.ac
Not just because there happens to be a po directory.
https://bugzilla.gnome.org/show_bug.cgi?id=748058
2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
Revert "configure.ac: uncomment gettext version setup"
This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
We don't need a gettext setup here and there's no po
directory either, so no reason why autopoint would be
run in the first place.
See https://bugzilla.gnome.org/show_bug.cgi?id=748058
2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
* examples/test-multicast.c:
* examples/test-multicast2.c:
* examples/test-sdp.c:
* examples/test-video-rtx.c:
* examples/test-video.c:
* tests/test-cleanup.c:
* tests/test-reuse.c:
Fix timeout function signatures across tests and examples
2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/Makefile.am:
tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
Make sure the test environment is set up.
https://bugzilla.gnome.org//show_bug.cgi?id=747624
2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
configure: bump automake requirement to 1.14 and autoconf to 2.69
This is only required for builds from git, people can still
build tarballs if they only have older autotools.
https://bugzilla.gnome.org//show_bug.cgi?id=747624
2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* configure.ac:
configure.ac: uncomment gettext version setup
Fixes autogen.sh. It would run autopoint, which would complain
that it could not find the gettext version in configure.ac.
https://bugzilla.gnome.org/show_bug.cgi?id=748058
2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* examples/test-video-rtx.c:
test-video-rtx: set exact payload type to PCMA payloader
Setting wrong payload type causes failure to do retransmission through audio stream
https://bugzilla.gnome.org/show_bug.cgi?id=747839
2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: fix to get valid each stream data for request-aux-sender signal
Because of duplicated g_signal_connect for request-aux-sender signal,
wrong stream pointer is passed to the signal handler.
Instead of passing each stream, pass stream array and get the relevant stream.
https://bugzilla.gnome.org/show_bug.cgi?id=747839
2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
* acinclude.m4:
* autogen.sh:
Update autogen.sh to latest version from common
Fixes build after aclocal_check etc. helpers have been removed.
2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From bc76a8b to c8fb372
2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Limit the queues to 1 buffer
We only need them to be able to pre-roll, queueing up more data here
is only going to harm latency and memory usage.
2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Update comment and ASCII art to the latest code
We have a queue in front of the udpsink too to prevent the pipeline from
locking up.
2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-media: Properly return first rtptime
Instead we where returning first GstBuffer timestamp. This would result
in clock skew and unwanted behaviour in RTSP playback.
https://bugzilla.gnome.org/show_bug.cgi?id=746479
2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Don't leave buffer mapped
If the seq is NULL, the RTP buffer was left mapped. We should always
unmap the buffer.
2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
* README:
Fix typo in README
2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* tests/check/gst/client.c:
Fix double semicolons
2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
This gives more accurate values than asking the payloader. There might be
queueing happening between the payloader and the sink.
https://bugzilla.gnome.org/show_bug.cgi?id=745704
2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't seek for PLAY if the position will not change
https://bugzilla.gnome.org/show_bug.cgi?id=745704
2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't include payload type in the caps for framesize
When the sdp media attribute framesize are converted to caps
the <payload> should not be included.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: add payload type to the sdp framesize attribute
The sdp framesize attribute is desribed in RFC6064. It is specified
for payloading of H263 and has the following form
a=framesize:<payload type> <width>-<height>. The <width>-<height> part
should be added to the caps in a payloader and the <payload type> should
be added by the rtsp-server.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
* examples/test-uri.c:
examples: test-uri: fix tainted variable
Insignificant but this keeps Coverity happy.
CID #1268404
2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-netclock-client.c:
* examples/test-netclock.c:
examples: Add a simple example of network synch for live streams.
An example server and client that works for synchronising live streams
only - as it can't support pause/play.
2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
rtsp-media-factory: Add functions to set/get the media gtype
Allow specifying the GType of a GstRtspMedia subclass to create
as a simpler way to get the factory to create a custom
GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: fix double unlock in _get_buffer_size()
Fixes an abort when calling gst_rtsp_media_get_buffer_size()
because of double g_mutex_unlock () usage.
https://bugzilla.gnome.org/show_bug.cgi?id=745434
2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
rtsp-session: Use monotonic time for RTSP session timeout
Changed RTSP session timeout handling to monotonic time
and deprecating the API for current system time.
This fixes timeouts when the system time changes.
https://bugzilla.gnome.org/show_bug.cgi?id=743346
2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
rtsp-client: Only error out in PLAY if seeking actually failed
If the media was just not seekable, we continue from whatever position we are
and let the client decide if that is what is wanted or not.
Only if the actual seek failed, we can't really recover and should error out.
2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Add necessary queues between tee and multiudpsink
https://bugzilla.gnome.org/show_bug.cgi?id=744379
2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
rtsp-media: If seeking fails, don't wait forever for the media to preroll again
Instead error out properly the same way as if the SEEKING query already
failed.
2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: minor code formatting fix
2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: fix logic for collect_streams
Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
all streams it knows if it got any, and can check if the transport mode is OK.
CID #1268400
2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't set the transport mode based on what elements we find
Just print a warning if the one that was set before disagrees with what
elements we found. It must already be set to something before as this
function is called after we received the SDP from ANNOUNCE in RECORD mode,
and we would reject ANNOUNCE if the RECORD flag was not set.
2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspserver.c:
tests: rtspserver: rename shadowed variable
We have two different 'sink' variables here,
rename one of them for clarity.
2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: fix awkward if clause
2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/test-uri.c:
examples: test-uri: improve uri argument handling and accept file names
Print an error if the argument passed is not a URI and can't
be converted into one, or no arguments have been provided.
2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/test-uri.c:
examples: test-uri: don't remove mount point after 10 seconds
It's very irritating when trying to test stuff repeatedly
and serves no real purpose other than showing that it can
be done.
2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/.gitignore:
examples: add new test-record to .gitignore
2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/test-record.c:
* gst/rtsp-server/rtsp-client.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-media.h:
* tests/check/gst/rtspserver.c:
rtsp-media: Use flags to distinguish between PLAY and RECORD media
2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/test-record.c:
test-record: Set latency for playback-style example to 2s instead of 200ms
2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspserver.c:
tests: add some unit tests for ANNOUNCE and RECORD
https://bugzilla.gnome.org/show_bug.cgi?id=743175
2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: fix a couple of leaks in handle_announce
2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
rtsp-media: Expose latency setting for setting the rtpbin latency
2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/test-record.c:
test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/Makefile.am:
* examples/test-record.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
Add initial support for RECORD
We currently only support media that is RECORD or PLAY only, not both at once.
https://bugzilla.gnome.org/show_bug.cgi?id=743175
2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: RTCP and RTP transport cache cookies seperated
RTCP packets were not sent because the same tr_cache_cookie was used for
both RTP and RTCP. So only one of the tr_cache lists were populated
depending on which one was sent first. If the tr_cache list is not
populated then no packets can be sent. Most often this happened to be
RTCP. Now seperate RTCP and RTP transport cache cookies are added which
resulted in both the tr_cache_lists to be populated regardless of which
one was sent first.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fix false compiler warning
rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: log interleaved data received
2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Use a random session ID in the SDP
RFC4566 Section 5.2 says that it should make the username, session id,
nettype, addrtype and unicast address tuple globally unique. Always using
1188340656180883 is not going to guarantee that: https://xkcd.com/221/
Instead let's create a 64 bit random number, which at least brings us
closer to the goal of global uniqueness.
https://tools.ietf.org/html/rfc4566#section-5.2
2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/test-launch.c:
* examples/test-mp4.c:
* examples/test-ogg.c:
* examples/test-uri.c:
examples: Don't call gst_init() and gst_get_option_group()
The latter calls the former at the appropriate time.
2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Drop trailing \0 of RTSP DATA messages
We add a trailing \0 in GstRTSPConnection to make parsing of
string message bodies easier (e.g. the SDP from DESCRIBE) but
for actual data this means we have to drop it or otherwise
create invalid data.
2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
Fixes crash when two threads access handle_new_sample() at the same
time, one for RTP, one for RTCP.
Otherwise, when iterating over the transports cache, it might be modified by
another thread at the same time if the transports cookie has changed.
https://bugzilla.gnome.org/show_bug.cgi?id=742954
2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Set format=TIME on our app sources for TCP
2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
* gst/rtsp-server/rtsp-session-pool.c:
Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
RFC 2326 states that session IDs may consist of alphanumeric as well as
the safe characters $-_.+ -- N.B. the percent character is not allowed.
Previously the session ID was URI-escaped, this meant that any character
which was not alphanumeric or any of the characters +-._~ would be
percent encoded. While the RFC (surprisingly) mentions that linear white
space in session IDs should be URI-escaped, it does not say anything
about other characters. Moreover no white space is allowed in the
session ID. Finally the percent character which is the result of
URI-escaping is not allowed in a session ID.
So there is no reason to do any URI-escaping, and now it is removed.
https://bugzilla.gnome.org/show_bug.cgi?id=742869
2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From f2c6b95 to bc76a8b
2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
* Makefile.am:
Fix 'make check' from top-level directory
2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* examples/test-launch.c:
* examples/test-mp4.c:
* examples/test-ogg.c:
* examples/test-uri.c:
examples: Add command-line parsing and take a 'port' argument
This allows users to run multiple servers on different ports for testing.
Only done for examples that actually take arguments and hence are capable of
outputting different streams for each instance on each port.
https://bugzilla.gnome.org/show_bug.cgi?id=742115
2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
rtsp-client: Add a send_message default signal handler
This allows subclasses to easily hook into the response sending
mechanism without doing everything from a signal, which seems
awkward from subclasses.
2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From ef1ffdc to f2c6b95
2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* Makefile.am:
* configure.ac:
configure: add --disable-examples switch
https://bugzilla.gnome.org/show_bug.cgi?id=741678
2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-video-rtx.c:
examples: add a retransmisison example implementing RFC4588
Currently only SSRC-multiplexed rtx streams are supported
2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Fix some minor memory leaks
2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Some minor cleanup
2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Fix compiler warnings
rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
^
rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
^
2014-11-27 01:12:36 +1100 Matthew Waters <matthew@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-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
media: implement ssrc-multiplexed retransmission support
based off RFC 4588 and the server-rtpaux example in -good
2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp: Ref transports in hash table.
Also ref streams for transports.
This solves a crash when reciving a rtcp after teardown but before
client finalize.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From 7bb2bce to ef1ffdc
2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: refactor cleanup of cached media
2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
* tests/check/gst/client.c:
tests: Remove FIXME
The session leak is now fixed, lets remove those FIXME comments.
2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
* tests/check/gst/rtspserver.c:
tests: Test to setup two sessions on one connection
https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
* tests/check/gst/rtspserver.c:
tests: Test setup with tcp transport
https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: Configure transport after creating session media
The default implementation of configure_client_transport() in
rtsp-client uses the session media when it chooses channels for
interleaved traffic.
https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session-media.c:
client: Stop caching media in client when doing setup
If the media has been managed by a session media, it should not be
cached in the client any longer. The GstRTSPSessionMedia object is now
responsible for unpreparing the GstRTSPMedia object using
gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
session media.
https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: unref srtp decoder when leaving bin
https://bugzilla.gnome.org/show_bug.cgi?id=739481
2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: mikey memory leaks
https://bugzilla.gnome.org/show_bug.cgi?id=739383
2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From 84d06cd to 7bb2bce
2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
* Makefile.am:
Parallelise 'make check-valgrind'
2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From a8c8939 to 84d06cd
2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 36388a1 to a8c8939
2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: deactivate media when shutting down from paused
This was only done when going directly from playing.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-context.h:
rtsp-client: add stream transport to context
We add the stream transport to the context so we can get the configured
client stream transport in the setup request signal.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-stream.c:
stream: release lock even not all transports have been removed
We don't want to keep the lock even we return FALSE because not all the
transports have been removed. This could lead into a deadlock.
https://bugzilla.gnome.org/show_bug.cgi?id=737797
2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
These were renamed in GstRTPBasePayload in 1.0
2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
client: set session media to NULL without the lock
We need to set session medias to NULL without the client lock otherwise
we can end up in a deadlock if another thread is waiting for the lock
and media unprepare is also waiting for that thread to end.
https://bugzilla.gnome.org/show_bug.cgi?id=737690
2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Set state to UNPREPARING in all cases
2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
* gst/rtsp-server/rtsp-media.c:
media: set state to unpreparing when unprepare is initiated
https://bugzilla.gnome.org/show_bug.cgi?id=737675
2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Remove backlog limit while processings requests
If the backlog limit is kept two cases of deadlocks may be
encountered when streaming over TCP. Without the backlog
limit this deadlocks can not happen, at the expence of
memory usage.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: do not free main context before rtsp watch
https://bugzilla.gnome.org/show_bug.cgi?id=737110
2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
* tests/check/gst/rtspserver.c:
tests: Extend unit test timeout to accomodate for valgrind
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream-transport.c:
rtsp-*: Treat sending packets to clients as keepalive
As long as gst-rtsp-server can successfully send RTP/RTCP data to
clients then the client must be reading. This change makes the server
timeout the connection if the client stops reading.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Allow backlog to grow while expiring session
Allow the send backlog in the RTSP watch to grow to unlimited size while
attempting to bring the media pipeline to NULL due to a session
expiring. Without this change the appsink element cannot change state
because it is blocked while rendering data in the new_sample callback.
This callback will block until it has successfully put the data into the
send backlog. There is a chance that the send backlog is full at this
point which means that the callback may block for a long time, possibly
forever. Therefore the media pipeline may also be prevented from
changing state for a long time.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Make old compilers happy
rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
Just in case that guint8 doesn't fit in a pointer. Just in case ...
2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: raise the backlog limits before pausing
We need to raise the backlog limits before pausing the pipeline or else
the appsink might be blocking in the render method in wait_backlog() and
we would deadlock waiting for paused.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: make define for the WATCH_BACKLOG
See https://bugzilla.gnome.org/show_bug.cgi?id=736322
2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: simplify session transport handling
link/unlink of the transport in a session was done to keep track of all
TCP transports and to send RTP/RTCP data to the streams. We can simplify
that by putting all the TCP transports in a hashtable indexed with the
channel number.
We also don't need to link/unlink the transports when we pause/resume
the streams. The same effect is already achieved when we pause/play the
media. Indeed, when we pause the media, the transport is removed from
the media and the callbacks will not be called anymore.
See https://bugzilla.gnome.org/show_bug.cgi?id=736041
2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
stream-transport: make method to handle received data
Make a method to handle the data received on a channel. It sends the
data to the stream of the transport on the RTP or RTCP pads based on
the channel number.
2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
* examples/test-mp4.c:
test: add example of dumping RTCP reports
2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-media: Make sure that sequence numbers are monotonic after pause
The sequence number is not monotonic for RTP packets after pause. The
reason is basepayloader generates a randon sequence number when the
pipeline goes from ready to pause. With this fix generation of sequence
number will be monotonic when going from pause to play request.
https://bugzilla.gnome.org/show_bug.cgi?id=736017
2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Protect saved clients watch with a mutex
Fixes a crash when close() is called while merging clients
in handle_tunnel(). In that case close() would destroy the
watch while it is still being used in handle_tunnel().
https://bugzilla.gnome.org/show_bug.cgi?id=735570
2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Remove the multicast group udp sources when removing from the bin
2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-media: Query position and stop time only on the RTP parts of the pipeline
The RTCP parts, in specific the RTCP udpsinks, are not flushed when
seeking and will always continue counting the time. This leads to
the NPT after a backwards seek to be something completely different
to the actual seek position.
https://bugzilla.gnome.org/show_bug.cgi?id=732644
2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
* examples/test-appsrc.c:
examples: fix another reference leak
gst_rtsp_media_get_element() returns a new ref.
2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* examples/test-appsrc.c:
examples: unref element after usage
gst_bin_get_by_name_recurse_up() returns an element
reference that must be unreffed after usage.
https://bugzilla.gnome.org/show_bug.cgi?id=734546
2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
* gst/rtsp-server/rtsp-media.c:
signals: Fix copy-pasto in target-state signal offset
2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
* Makefile.am:
* common:
Makefile: Add usage of build-checks step
Allows building checks without running them
2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Listen on the multicast group for RTP/RTCP packets
When a UDP multicast transport is used it is expected that the server listens
for RTP and RTCP packets on the multicast group with the corresponding port.
Without this we will never get RTCP packets from clients in multicast mode.
https://bugzilla.gnome.org/show_bug.cgi?id=732238
2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.4.0 ===
2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.4.0
2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>

145
NEWS
View file

@ -1,145 +1,2 @@
This is GStreamer RTSP Server 1.4.0
This is GStreamer RTSP Server 1.5.1
Changes since 1.2:
New API:
• GstMessageType has GST_MESSAGE_EXTENDED added. All types before
that can be used together as a flags type as before, but from
that message onwards the types are just counted incrementally.
This was necessary to be able to add more message types.
In 2.0 GstMessageType will just become an enum and not a flags
type anymore.
• GstDeviceMonitor for device probing, e.g. to list all available
audio or video capture devices. This is the replacement for
GstPropertyProbe from 0.10.
• Events accumulate the running-time offset now when travelling
through pads, as set by the gst_pad_set_offset() function. This
allows to compensate for this in the QOS event for example.
• GstBuffer has a new flag "tag-memory" that is set automatically
when memory is added or removed to a buffer. This allows buffer
pools to detect if they can recycle a buffer or need to reset
it first.
• GstToc has new API to mark GstTocEntries as loops.
• A not-authorized resource error has been defined to notify
applications that accessing the resource has failed because
of missing authorization and to distinguish this case from others.
This change is actually already in 1.2.4.
• GstPad has a new flag "accept-intersect", that will let the default
ACCEPT_CAPS query handler do an intersection instead of subset check.
This is interesting for parser elements that can handle incomplete
caps.
• GstCollectPads has support for flushing and a default handler for
SEEK events now.
• New GstFlowAggregator helper object that simplifies handling of
flow returns in elements with multiple source pads. Additionally
GstPad now always stores the last flow return and provides an
API to retrieve it.
• GstSegment has new API to offset the running time by a specific
value and this is used in GstPad to allow positive and negative
offsets in gst_pad_set_offset() in all situations.
• Support for h265/HEVC and VP8 has been added to the codec utils and codec
parsers library, and was integrated into various elements.
• API for adjusting the TLS validation of RTSP connection has been added.
• The RTSP and SDP library has MIKEY (RFC 3830) support now, and
there is API to distinguish between the different RTSP profiles.
• API to access RTP time information and statistics.
• Support for auxiliary streams was added to rtpbin.
• Support for tiled, raw video formats has been added.
• GstVideoDecoder and GstAudioDecoder have API to help aggregating tag
events and merge custom tags into them consistently.
• GstBufferPool has support for flushing now.
• playbin/playsink has support for application provided audio and video
filters.
• GstDiscoverer has new and simplified API to get details about missing
plugins and information to pass to the plugin installer.
• The GL library was merged from gst-plugins-gl to gst-plugins-bad,
providing a generic infrastructure for handling GL inside GStreamer
pipelines and a plugin with some elements using these, especially
a video sink. Supported platforms currently are Android, Cocoa (OS X),
DispManX (Raspberry Pi), EAGL (iOS), WGL (Windows) and generic X11,
Wayland and EGL platforms.
This replaces eglglessink and also is supposed to replace osxvideosink.
• New GstAggregator base class in gst-plugins-bad. This is supposed to
replace GstCollectPads in the future and fix long-known shortcomings
in its API. Together with the base class some elements are provided
already, like a videomixer (compositor).
Major changes:
• New plugins and elements:
∘ v4l2videodec element for accessing hardware codecs on
platforms that make them accessible via V4L2, e.g.
Samsung Exynos. This comes together with major refactoring
of the existing V4L2 elements and the corresponding
infrastructure.
The v4l2videodec element replaces the mfcdec element.
∘ New downloadbuffer element that replaces the download
buffering feature of queue2. Compared to queue2's code
it is much simpler and only for this single use case.
A noteworthy new feature is that it's downloading gaps
in the already downloaded stream parts when nothing else
is to be downloaded.
This is now used by playbin when download buffering is
enabled.
∘ rtpstreampay and rtpstreamdepay elements for transmitting
RTP packets over a stream API (e.g. TCP) according to
RFC 4571.
∘ rtprtx elements for standard compliant implementation of
retransmissions, integrated into the rtpmanager plugin.
∘ audiomixer element that mixes multiple audio streams together
into a single one while keeping synchronization. This is
planned to become the replacement of the adder element.
∘ OpenNI2 plugin for 3D cameras like the Kinect camera.
∘ OpenEXR plugin for decoding high-dynamic-range EXR images.
∘ curlsshsink and curlsftpsink to write files via SSH/SFTP.
∘ videosignal, ivfparse and sndfile plugins ported from 0.10.
∘ avfvideosrc, vtdec and other elements were ported from 0.10 and
are available on OS X and iOS now.
• Other changes:
∘ gst-libav now uses libav 10.2, and gained support for H265/HEVC.
∘ Support for hardware codecs and special memory types has been
improved with bugfixes and feature additions in various plugins
and base classes.
∘ Various bugfixes and improvements to buffering in queue2 and
multiqueue elements.
∘ dvbsrc supports more delivery mechanisms and other features
now, including DVB S2 and T2 support.
∘ The MPEGTS library has support for many more descriptors.
∘ Major improvements to tsdemux and tsparse, especially time and
seeking related.
∘ souphttpsrc now has support for keep-alive connections,
compression, configurable number of retries and configuration
for SSL certificate validation.
∘ hlsdemux has undergone major refactoring and works more
reliable now and supports more HLS features like trick modes.
Also fragments are pushed downstream while they're downloaded
now instead of waiting for each fragment to finish.
∘ dashdemux and mssdemux are now also pushing fragments downstream
while they're downloaded instead of waiting for each fragment to
finish.
∘ videoflip can automatically flip based on the orientation tag.
∘ openjpeg supports the OpenJPEG2 API.
∘ waylandsink was refactored and should be more useful now. It also
includes a small library which most likely is going to be removed
in the future and will result in extensions to the GstVideoOverlay
interface.
∘ gst-rtsp-server supports SRTP and MIKEY now.
∘ gst-libav encoders are now negotiating any profile/level settings
with downstream via caps.
∘ Lots of fixes for coverity warnings all over the place.
∘ Negotiation related performance improvements.
∘ 800+ fixed bug reports, and many other bug fixes and other
improvements everywhere that had no bug report.
Things to look out for:
• The eglglessink element was removed and replaced by the glimagesink
element.
• The mfcdec element was removed and replaced by v4l2videodec.
• osxvideosink is only available in OS X 10.6 or newer.
• On Android the namespace of the automatically generated Java class
for initialization of GStreamer has changed from com.gstreamer to
org.freedesktop.gstreamer to prevent namespace pollution.
• On iOS you have to update your gst_ios_init.h and gst_ios_init.m in
your projects from the one included in the binaries if you used the
GnuTLS GIO module before. The loading mechanism has slightly changed.

82
RELEASE
View file

@ -1,30 +1,53 @@
Release notes for GStreamer RTSP Server Library 1.4.0
Release notes for GStreamer RTSP Server Library 1.5.1
The GStreamer team is pleased to announce the first release of
the stable 1.4 release series. The 1.4 release series is adding new
features on top of the 1.0 and 1.2 series and is part of the API and
ABI-stable 1.x release series of the GStreamer multimedia framework.
The GStreamer team is pleased to announce the first release of the unstable
1.5 release series. The 1.5 release series is adding new features on top of
the 1.0, 1.2 and 1.4 series and is part of the API and ABI-stable 1.x release
series of the GStreamer multimedia framework. The unstable 1.5 release series
will lead to the stable 1.6 release series in the next weeks, and newly added
API can still change until that point.
Binaries for Android, iOS, Mac OS X and Windows are provided together
with this release.
The stable 1.4 release series is API and ABI compatible with 1.0.x,
1.2.x and any other 1.x release series in the future. Compared to 1.2.x
it contains some new features and more intrusive changes that were
considered too risky as a bugfix.
Binaries for Android, iOS, Mac OS X and Windows will be provided separately
during the unstable 1.5 release series.
Features of this release
Bugs fixed in this release
* 733244 : Correct misspelled words
* 732238 : Listen on the multicast group for RTP/RTCP packets
* 734546 : tests: Unref element after usage
* 736041 : Protect rtsp transport data.
* 736647 : Tunneled RTSP sessions do not always timeout as expected
* 737110 : rtsp-client: race condition when closing client connection
* 737631 : gst-rtsp-server deadlock while sending response over TCP
* 737675 : media: media_unprepare() is kind of broken
* 737690 : rtsp-client: deadlock when setting session medias to NULL
* 737797 : rtsp-stream: lock not released when leaving bin and transports not removed
* 737829 : rtsp-server: deactivate media when shutting down from paused
* 738905 : rtsp-client: add stream transport to the context
* 739112 : rtsp-client: Can not allocate ports for interleaved traffic in setup
* 740752 : add retransmission support
* 740845 : crash when reciving a rtcp after teardown but before client finalize.
* 741678 : configure: add --disable-examples switch
* 742115 : Examples: Accept a 'port' argument for running multiple instances
* 742869 : Remove URI-escaping of RTSP session-id
* 742954 : Crash when two treads are in handle_new_sample at the same time.
* 743175 : Add support for RECORD
* 743346 : When system time is increased the ongoing RTSP sessions will time out.
* 743734 : RTCP packets not sent
* 744379 : gst-rtsp-server does not preroll when piping data into the media-pipeline
* 745704 : Losing the first packet
* 747614 : gst-rtsp-server: uninitialized clock rate causes critical warning
* 747839 : gst-rtsp-server: doesn't perform retransmission to both streams in test-video-rtx
* 748058 : autogen.sh fails due to autopoint erroring out due to missing gettext version in configure.ac
* 749845 : Client have problem to find the teardown response.
==== Download ====
@ -59,7 +82,34 @@ Interested developers of the core library, plugins, and applications should
subscribe to the gstreamer-devel list.
Applications
Contributors to this release
* Aleix Conchillo Flaqué
* Alistair Buxton
* Andreas Frisch
* Anila Balavan
* Arun Raghavan
* Branko Subasic
* Edward Hervey
* Gregor Boirie
* Göran Jönsson
* Hyunjun Ko
* Jan Schmidt
* Kent-Inge Ingesson
* Linus Svensson
* Luis de Bethencourt
* Matthew Waters
* Nicolas Dufresne
* Nirbheek Chauhan
* Ognyan Tonchev
* Olivier Crête
* Sebastian Dröge
* Sebastian Rasmussen
* Srimanta Panda
* Stefan Sauer
* Tim-Philipp Müller
* Vincent Penquerc'h
* Wim Taymans
 

10
configure.ac
View file

@ -2,7 +2,7 @@ AC_PREREQ(2.69)
dnl initialize autoconf
dnl when going to/from release please set the nano (fourth number) right !
dnl releases only do Wall, cvs and prerelease does Werror too
AC_INIT([GStreamer RTSP Server Library], [1.5.0.1],
AC_INIT([GStreamer RTSP Server Library], [1.5.1],
[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
[gst-rtsp-server])
AG_GST_INIT
@ -56,10 +56,10 @@ dnl sets GST_LT_LDFLAGS
AS_LIBTOOL(GST, 501, 0, 501)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.5.0.1
GSTPB_REQ=1.5.0.1
GSTPG_REQ=1.5.0.1
GSTPD_REQ=1.5.0.1
GST_REQ=1.5.1
GSTPB_REQ=1.5.1
GSTPG_REQ=1.5.1
GSTPD_REQ=1.5.1
dnl *** autotools stuff ****

10
gst-rtsp-server.doap
View file

@ -30,6 +30,16 @@ RTSP server library based on GStreamer
</GitRepository>
</repository>
<release>
<Version>
<revision>1.5.1</revision>
<branch>1.5</branch>
<name></name>
<created>2015-06-07</created>
<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.5.1.tar.xz" />
</Version>
</release>
<release>
<Version>
<revision>1.4.0</revision>