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Release 1.5.1
This commit is contained in:
Sebastian Dröge
parent
08e0c79cee
commit
e86bbbb66c
5 changed files with 997 additions and 168 deletions
918
ChangeLog
918
ChangeLog
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@ -1,9 +1,921 @@
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=== release 1.4.0 ===
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=== release 1.5.1 ===
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2014-07-19 Sebastian Dröge <slomo@coaxion.net>
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2015-06-07 Sebastian Dröge <slomo@coaxion.net>
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* configure.ac:
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* configure.ac:
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releasing 1.4.0
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releasing 1.5.1
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2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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rtsp-client: No flush during Teardown.
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When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
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backlog is empty it can happen that just a part of a message will be
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sent and rest is in backlog queue. If then flush during teardown
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just a part of message will be sent.This can lead to client miss
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teardown response since it expect to get the last part of message.
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The flushing during teardown was introduced to fix a deadlock that now
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is fixed more generally in handle_request by temporary setting backlog
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size to unlimited.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
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2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
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* tests/check/Makefile.am:
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tests: Use AM_TESTS_ENVIRONMENT
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Needed by the new automake test runner and the
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current version of the common submodule.
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2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-media.h:
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* gst/rtsp-server/rtsp-stream.h:
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rtsp-server: Use single-include rtsp header to make sure we get all definitions
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2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Mark some more functions static
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2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Only unblock the media in suspend() when actually changing the state
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Otherwise we're going to lose a few packets for live streams during DESCRIBE.
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2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
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* examples/test-video-rtx.c:
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examples: Use AVPF profile for the RTX example
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2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-sdp.c:
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rtsp-sdp: Only add RTX to the SDP when using a feedback profile
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2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: get valid clock-rate from last-sample
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clock-rate in last-sample's caps is integer, not unsigned.
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To get this value properly, variable needs to be type-casted to int.
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https://bugzilla.gnome.org/show_bug.cgi?id=747614
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2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
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* autogen.sh:
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* common:
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autogen.sh: only run autopoint if gettext requested in configure.ac
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Not just because there happens to be a po directory.
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https://bugzilla.gnome.org/show_bug.cgi?id=748058
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2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
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* configure.ac:
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Revert "configure.ac: uncomment gettext version setup"
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This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
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We don't need a gettext setup here and there's no po
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directory either, so no reason why autopoint would be
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run in the first place.
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See https://bugzilla.gnome.org/show_bug.cgi?id=748058
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2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
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* examples/test-multicast.c:
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* examples/test-multicast2.c:
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* examples/test-sdp.c:
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* examples/test-video-rtx.c:
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* examples/test-video.c:
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* tests/test-cleanup.c:
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* tests/test-reuse.c:
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Fix timeout function signatures across tests and examples
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2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
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* tests/check/Makefile.am:
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tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
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Make sure the test environment is set up.
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https://bugzilla.gnome.org//show_bug.cgi?id=747624
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2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
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* configure.ac:
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configure: bump automake requirement to 1.14 and autoconf to 2.69
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This is only required for builds from git, people can still
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build tarballs if they only have older autotools.
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https://bugzilla.gnome.org//show_bug.cgi?id=747624
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2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
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* configure.ac:
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configure.ac: uncomment gettext version setup
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Fixes autogen.sh. It would run autopoint, which would complain
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that it could not find the gettext version in configure.ac.
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https://bugzilla.gnome.org/show_bug.cgi?id=748058
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2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
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* examples/test-video-rtx.c:
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test-video-rtx: set exact payload type to PCMA payloader
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Setting wrong payload type causes failure to do retransmission through audio stream
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https://bugzilla.gnome.org/show_bug.cgi?id=747839
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2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-stream.c:
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* gst/rtsp-server/rtsp-stream.h:
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rtsp-stream: fix to get valid each stream data for request-aux-sender signal
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Because of duplicated g_signal_connect for request-aux-sender signal,
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wrong stream pointer is passed to the signal handler.
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Instead of passing each stream, pass stream array and get the relevant stream.
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https://bugzilla.gnome.org/show_bug.cgi?id=747839
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2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
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* acinclude.m4:
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* autogen.sh:
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Update autogen.sh to latest version from common
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Fixes build after aclocal_check etc. helpers have been removed.
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2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
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* common:
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Automatic update of common submodule
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From bc76a8b to c8fb372
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2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Limit the queues to 1 buffer
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We only need them to be able to pre-roll, queueing up more data here
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is only going to harm latency and memory usage.
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2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Update comment and ASCII art to the latest code
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We have a queue in front of the udpsink too to prevent the pipeline from
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locking up.
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2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-media: Properly return first rtptime
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Instead we where returning first GstBuffer timestamp. This would result
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in clock skew and unwanted behaviour in RTSP playback.
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https://bugzilla.gnome.org/show_bug.cgi?id=746479
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2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Don't leave buffer mapped
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If the seq is NULL, the RTP buffer was left mapped. We should always
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unmap the buffer.
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2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
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* README:
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Fix typo in README
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2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
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* gst/rtsp-server/rtsp-media-factory.c:
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* tests/check/gst/client.c:
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Fix double semicolons
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2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
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This gives more accurate values than asking the payloader. There might be
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queueing happening between the payloader and the sink.
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https://bugzilla.gnome.org/show_bug.cgi?id=745704
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2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Don't seek for PLAY if the position will not change
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https://bugzilla.gnome.org/show_bug.cgi?id=745704
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2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Don't include payload type in the caps for framesize
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When the sdp media attribute framesize are converted to caps
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the <payload> should not be included.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
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Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
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2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
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* gst/rtsp-server/rtsp-sdp.c:
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rtsp-sdp: add payload type to the sdp framesize attribute
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The sdp framesize attribute is desribed in RFC6064. It is specified
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for payloading of H263 and has the following form
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a=framesize:<payload type> <width>-<height>. The <width>-<height> part
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should be added to the caps in a payloader and the <payload type> should
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be added by the rtsp-server.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
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2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
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* examples/test-uri.c:
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examples: test-uri: fix tainted variable
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Insignificant but this keeps Coverity happy.
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CID #1268404
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2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
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* examples/.gitignore:
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* examples/Makefile.am:
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* examples/test-netclock-client.c:
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* examples/test-netclock.c:
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examples: Add a simple example of network synch for live streams.
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An example server and client that works for synchronising live streams
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only - as it can't support pause/play.
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2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
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* gst/rtsp-server/rtsp-media-factory.c:
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* gst/rtsp-server/rtsp-media-factory.h:
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rtsp-media-factory: Add functions to set/get the media gtype
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Allow specifying the GType of a GstRtspMedia subclass to create
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as a simpler way to get the factory to create a custom
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GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
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2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: fix double unlock in _get_buffer_size()
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Fixes an abort when calling gst_rtsp_media_get_buffer_size()
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because of double g_mutex_unlock () usage.
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https://bugzilla.gnome.org/show_bug.cgi?id=745434
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2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
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* gst/rtsp-server/rtsp-session-pool.c:
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* gst/rtsp-server/rtsp-session.c:
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* gst/rtsp-server/rtsp-session.h:
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rtsp-session: Use monotonic time for RTSP session timeout
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Changed RTSP session timeout handling to monotonic time
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and deprecating the API for current system time.
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This fixes timeouts when the system time changes.
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https://bugzilla.gnome.org/show_bug.cgi?id=743346
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2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-media.c:
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rtsp-client: Only error out in PLAY if seeking actually failed
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If the media was just not seekable, we continue from whatever position we are
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and let the client decide if that is what is wanted or not.
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Only if the actual seek failed, we can't really recover and should error out.
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2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Add necessary queues between tee and multiudpsink
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https://bugzilla.gnome.org/show_bug.cgi?id=744379
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2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: If seeking fails, don't wait forever for the media to preroll again
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Instead error out properly the same way as if the SEEKING query already
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failed.
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2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
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* gst/rtsp-server/rtsp-stream.h:
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rtsp-stream: minor code formatting fix
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2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: fix logic for collect_streams
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Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
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all streams it knows if it got any, and can check if the transport mode is OK.
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CID #1268400
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2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Don't set the transport mode based on what elements we find
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Just print a warning if the one that was set before disagrees with what
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elements we found. It must already be set to something before as this
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function is called after we received the SDP from ANNOUNCE in RECORD mode,
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and we would reject ANNOUNCE if the RECORD flag was not set.
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2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
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* tests/check/gst/rtspserver.c:
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tests: rtspserver: rename shadowed variable
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We have two different 'sink' variables here,
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rename one of them for clarity.
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2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
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* gst/rtsp-server/rtsp-client.c:
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rtsp-client: fix awkward if clause
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2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
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* examples/test-uri.c:
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examples: test-uri: improve uri argument handling and accept file names
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Print an error if the argument passed is not a URI and can't
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be converted into one, or no arguments have been provided.
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||||||
|
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>
|
2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
|
||||||
|
|
||||||
|
|
|
||||||
145
NEWS
145
NEWS
|
|
@ -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
82
RELEASE
|
|
@ -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 GStreamer team is pleased to announce the first release of the unstable
|
||||||
the stable 1.4 release series. The 1.4 release series is adding new
|
1.5 release series. The 1.5 release series is adding new features on top of
|
||||||
features on top of the 1.0 and 1.2 series and is part of the API and
|
the 1.0, 1.2 and 1.4 series and is part of the API and ABI-stable 1.x release
|
||||||
ABI-stable 1.x release series of the GStreamer multimedia framework.
|
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 will be provided separately
|
||||||
Binaries for Android, iOS, Mac OS X and Windows are provided together
|
during the unstable 1.5 release series.
|
||||||
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.
|
|
||||||
|
|
||||||
|
|
||||||
|
|
||||||
|
|
||||||
|
Features of this release
|
||||||
|
|
||||||
|
|
||||||
Bugs fixed in 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 ====
|
==== Download ====
|
||||||
|
|
||||||
|
|
@ -59,7 +82,34 @@ Interested developers of the core library, plugins, and applications should
|
||||||
subscribe to the gstreamer-devel list.
|
subscribe to the gstreamer-devel list.
|
||||||
|
|
||||||
|
|
||||||
|
Applications
|
||||||
|
|
||||||
Contributors to this release
|
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
|
* 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
10
configure.ac
|
|
@ -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], [1.5.0.1],
|
AC_INIT([GStreamer RTSP Server Library], [1.5.1],
|
||||||
[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
|
||||||
|
|
@ -56,10 +56,10 @@ dnl sets GST_LT_LDFLAGS
|
||||||
AS_LIBTOOL(GST, 501, 0, 501)
|
AS_LIBTOOL(GST, 501, 0, 501)
|
||||||
|
|
||||||
dnl *** required versions of GStreamer stuff ***
|
dnl *** required versions of GStreamer stuff ***
|
||||||
GST_REQ=1.5.0.1
|
GST_REQ=1.5.1
|
||||||
GSTPB_REQ=1.5.0.1
|
GSTPB_REQ=1.5.1
|
||||||
GSTPG_REQ=1.5.0.1
|
GSTPG_REQ=1.5.1
|
||||||
GSTPD_REQ=1.5.0.1
|
GSTPD_REQ=1.5.1
|
||||||
|
|
||||||
dnl *** autotools stuff ****
|
dnl *** autotools stuff ****
|
||||||
|
|
||||||
|
|
|
||||||
|
|
@ -30,6 +30,16 @@ RTSP server library based on GStreamer
|
||||||
</GitRepository>
|
</GitRepository>
|
||||||
</repository>
|
</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>
|
<release>
|
||||||
<Version>
|
<Version>
|
||||||
<revision>1.4.0</revision>
|
<revision>1.4.0</revision>
|
||||||
|
|
|
||||||
Loading…
Reference in a new issue