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ChangeLog
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=== release 1.4.0 ===
2014-07-19 Sebastian Dröge <slomo@coaxion.net>
* configure.ac:
releasing 1.4.0
2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
* gst/rtsp-server/rtsp-media.h:
media: correct misspelled words in description
https://bugzilla.gnome.org/show_bug.cgi?id=733244
=== release 1.3.91 ===
2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.3.91
2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
* docs/libs/gst-rtsp-server-sections.txt:
docs: update docs
2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-server.c:
server: implement client REMOVE filter
2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: expose _close() method
Expose a previously internal close method to close the client
connection.
2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-session-pool.c:
session-pool: signal session-removed outside of the lock
Release the lock before emiting the session-removed signal.
2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream.c:
filter: Release lock in filter functions
Release the object lock before calling the filter functions. We need to
keep a cookie to detect when the list changed during the filter
callback. We also keep a hashtable to make sure we only call the filter
function once for each object in case of concurrent modification.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: check if watch is set in handle_teardown()
The unit tests run without a watch
2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
* tests/check/gst/client.c:
client tests: send teardown to cleanup session
2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
* tests/check/gst/rtspserver.c:
server tests: send teardown to cleanup session
2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: keep ref to client for the session removed handler
This extra ref will be dropped when all client sessions have been
removed. A session is removed when a client sends teardown, closes its
endpoint of the TCP connection or the sessions expires.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session.c:
* tests/check/gst/client.c:
client: manage media in session as a last step
Once we manage a media in a session, we can't unmanage it anymore
without destroying it. Therefore, first check everything before we
manage the media, otherwise if something is wrong we have no way to
unmanage the media.
If we created a new session and something went wrong, remove the session
again. Fixes a leak in the unit test.
2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
* examples/test-mp4.c:
* examples/test-ogg.c:
examples: print 'stream ready at url' for mp4 and ogg example
2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-sdp.c:
rtsp: fix for MIKEY api change
2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: free watch context only once
The watch context is freed when the source is destroyed. Avoids
a CRITICAL when we try to unref the context twice.
2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: fix build
2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: protect sessions with lock
Protect the list of sessions with the lock.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
Client: keep a ref to the session
Don't just keep a weak ref to the session objects but use a hard ref. We
will be notified when a session is removed from the pool (expired) with
the new session-removed signal.
Don't automatically close the RTSP connection when all the sessions of
a client are removed, a client can continue to operate and it can create
a new session if it wants. If you want to remove the client from the
server, you have to use gst_rtsp_server_client_filter() now.
Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
See https://bugzilla.gnome.org/show_bug.cgi?id=732226
2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session-pool.h:
session-pool: add session-removed signal
Add a signal to be notified when a session is removed from the pool.
2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-server.h:
Make rtsp-server.h a single-include header, use it for G-I
https://bugzilla.gnome.org/show_bug.cgi?id=732411
=== release 1.3.90 ===
2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.3.90
2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: crypto can be NULL
2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-mount-points.c:
introspection: add missing allow-none annotations
https://bugzilla.gnome.org/show_bug.cgi?id=730952
2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-token.c:
introspection: add (nullable) annotations to return values
https://bugzilla.gnome.org/show_bug.cgi?id=730952
2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
gi: improve annotations
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-server.c:
signals: use generic marshal function
Use the generic C marshal function.
Use more explicit type instead of G_TYPE_POINTER
2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-context.h:
context: add type macro
2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-sdp.h:
sdp: hide key length defines
They don't have a namespace.
2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.3.3 ===
2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.3.3
2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-sdp.h:
mikey: add different key length parameters
Add encryption and authentication key length parameters to MIKEY. For
the encoders, the key lengths are obtained from the cipher and auth
algorithms set in the caps. For the decoders, they are obtained while
parsing the key management from the client.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
* tests/check/gst/stream.c:
stream tests: Make sure we get right multicast address from stream
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: ref the context until rtsp watch is alive
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: Destroy the rtsp watch after connection close
2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: fix confusing comment
2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-session.c:
rtsp-session: Timeout in header.
Adding the possbilty to always have timout in header.
This is configurabe with setting "timeout-always-visible".
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.3.2 ===
2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* common:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.3.2
2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From 211fa5f to 1f5d3c3
2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: store TCP ports in transport
Store the TCP ports in the transport when we are doing RTSP over TCP.
This way, we can easily get to the ports from the transport.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-stream.c:
stream: add signals for new RTP/RTCP encoders
New signals to allow the user to configure the dynamically created
encoders.
https://bugzilla.gnome.org/show_bug.cgi?id=730228
2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: Make suspend()/unsuspend() virtual
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
client: fix send-message signal marshaller
Use generic marshalling for the send-message signal. It has
two POINTER arguments, not just one.
https://bugzilla.gnome.org/show_bug.cgi?id=729900
2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
* tests/check/gst/media.c:
tests: add and remove pads only once
In this test we simulate a dynamic pad by watching the caps event.
Because of renegotiation in the base payloader now, this caps is sent
multiple times but we can only deal with 1 invocation, use a variable to
only 'add and remove' the pad once.
2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspserver.c:
tests: add unit test for correct handling of Require headers
https://bugzilla.gnome.org/show_bug.cgi?id=729426
2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
Servers must handle Require headers and must report a failure
if they don't handle any of the Required options, see RFC 2326,
section 12.32: https://tools.ietf.org/html/rfc2326#page-54
https://bugzilla.gnome.org/show_bug.cgi?id=729426
2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.3.1 ===
2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.3.1
2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From bcb1518 to 211fa5f
2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
* .gitignore:
Update .gitignore
2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/sessionmedia.c:
tests: fix memory leak in sessionmedia unit test
2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: emit a signal before sending a message
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: pass context to send_message
Pass the current context to send_message, we will need it later.
2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: fix typo in comment
2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
media: Do not stop thread twice if default_prepare() fails
2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: set the watch to flushing before going to NULL
First set the watch to flushing so that we unblock any current and
future attempt to send data on the watch, Then set the pipeline to
NULL.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
* gst/rtsp-server/rtsp-session-pool.c:
* tests/check/gst/sessionpool.c:
rtsp-session-pool: Fixes annotation
Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
in the sessionpool test.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: make media_prepare virtual
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
media: stop the thread in more error cases
2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
media: allow NULL as the thread
Use the default context whan passing a NULL thread.
2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: indent cleanup
Coverity was moaning about unreachable code, and I think it was just
confused by { being before the label. We'll see if it pops up again.
Coverity 1197705
2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
client: Add drop-backlog property
When we have too many messages queued for a client (currently hardcoded
to 100) we overflow and drop the messages. Add a drop-backlog property
to control this behaviour. Setting this property to FALSE will retry
to send the messages to the client by waiting for more room in the
backlog.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: support for POST before GET when setting up a tunnel
2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: remove watch of the second client after http tunnel setup
The second client will be freed after the HTTP tunnel has been set up.
Make sure it's RTSP watch is never dispatched again.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
media: Make media_prepare() fail if port allocation fails
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
* tests/check/gst/media.c:
media test: cleanup the thread pool in tests
2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
rtsp-media: Unblock blocked streams in unprepare
The streams will be blocked when a live media is prepared.
The streams should be unblocked in gst_rtsp_media_unprepare.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: release the state lock when going to NULL
Set our state to UNPREPARING and release the state-lock before
setting the pipeline to the NULL state. This way, any pad-added
callback will be able to take the state-lock and check that we are now
unpreparing instead of deadlocking.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: protect status with lock
Make sure we only update the status with the lock.
2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-sdp.c:
rtsp: update for MIKEY API changes
2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: parse the mikey response from the client
Parse the mikey response from the client and update the policy for
each SSRC.
2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add method to set crypto info
Make a method to configure the crypto information of a stream.
Set udpsrc in READY instead of PAUSED so that we can configure caps
later.
2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: cleanup error paths
2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: fix docs
2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
* examples/test-video.c:
test: enable SRTP only on RTSPS
We only want to enable SRTP when doing rtsp over TLS so that we can
exchange the keys in a secure way.
2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
* examples/test-video.c:
test: print an error on failure
2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
* configure.ac:
* examples/test-video.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/Makefile.am:
stream: add SRTP support
Install srtp encoder and decoder elements in rtpbin
Add MIKEY in SDP
2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* tests/check/Makefile.am:
* tests/check/gst/sessionpool.c:
tests: Add unit tests for sessionpool
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* tests/check/gst/threadpool.c:
tests: Improve code coverage of rtsp-threadpool tests
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* tests/check/gst/sessionmedia.c:
tests: Improve code coverage for rtsp-session-media
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
gobject-introspection: Add annotations to support language bindings
In addition a few cosmetic changes:
* Adjust the order of arguments
* Fix typo: occured -> occurred
* Fix indentation after Return:-clauses
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Don't mix IPv4 and IPv6 addresses
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: take caps after the session manager
Take the caps for the SDP after they leave the rtpbin so that we can
also get the properties added by rtpbin elements.
2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: release lock while pushing out packets
Keep a cache of the transports and use this to iterate the transport
while pushing packets. This allows us to release the lock early.
See https://bugzilla.gnome.org/show_bug.cgi?id=725898
2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
rtsp-client: vmethod for modifying tunnel GET response
Add a vmethod tunnel_http_response where the response to the HTTP GET
for tunneled connections can be modified.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-sdp.c:
sdp: make 1 media line per profile
If we have multiple profiles (AVP or AVPF) for a stream, make one m=
line in the SDP for each profile. The client is then supposed to pick
one of the profiles in the SETUP request. Because the m= lines have the
same pt, the client also knows that only 1 option is possible.
2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
factory: add profile property and pass to media and streams
2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
* examples/test-multicast.c:
* gst/rtsp-server/rtsp-sdp.c:
sdp: pass multicast connection for multicast-only stream
Pass the multicast address of the stream in the connection info in the
SDP so that clients try a multicast connection first.
Only allow multicast connections in the test-multicast example. Also
increase the TTL a little.
2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* .gitignore:
.gitignore: Ignore gcov intermediate files
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: release some locks in error cases
2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
docs: Enable and fix gtk-doc warnings
* Makefile: Enable gtk-doc warnings, like the rest of GStreamer
* addresspool/mediafactory: Add missing annotation colon
* stream: Annotate return value
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From fe1672e to bcb1518
2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 1a07da9 to fe1672e
2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/Makefile.am:
examples: use LDADD for libs instead of LDFLAGS
2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
configure: make sure releases are in .doap file
2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/test-cgroups.c:
examples: test-cgroups: don't put code with side effects into g_assert()
The g_assert() might get compiled out with the right
compiler/preprocessor flags.
2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/.gitignore:
examples: add cgroup test binary to .gitignore
2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/test-cgroups.c:
examples: fix cgroup test build
Fixes build failure caused by compiler warning:
test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
* .gitignore:
.gitignore: ignore temp files created in the course of 'make check'
2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: don't loose frames handling new PLAY request
If client supplied a range check if the range specifies the start point.
If not, then do an accurate seek to the current position. If a start
point was specified do do a key unit seek to make sure the streaming
starts with decodeable frames.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
Revert "media: only flush when setting a new start position"
This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
We need to do the flush in all cases, demuxer block currently for
non-flushing seeks.
2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: only flush when setting a new start position
Only flush the pipeline when we change the start position with
a seek.
See https://bugzilla.gnome.org/show_bug.cgi?id=724611
2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
stream: set ttl-mc before adding the socket
Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
never be set on socket.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-media.c:
media: stop thread if media is already prepared
in gst_rtsp_media_prepare() the thread is not used if media is already
prepared (e.g. media shared) so we want to stop the thread. otherwise, a
leak occurs.
https://bugzilla.gnome.org/show_bug.cgi?id=724182
2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
* Makefile.am:
build: Ship gst-rtsp-server.doap file
2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
* tests/check/gst/rtspserver.c:
tests: Fix another compiler warning with gcc
2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/client.c:
rtsp-server: Fix lots of compiler warnings with clang
2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
* gst-rtsp-server.doap:
* tests/Makefile.am:
configure: Synchronise with the configure scripts of the other modules
2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
Revert "rtsp-server: support build against last stable release"
This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
Let us require 1.2.3 now, which is going to be released in a few
minutes.
2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream-transport.c:
session: improve RTP-Info
Ignore streams that can't generate RTP-Info instead of failing.
Don't return the empty string when all streams are unconfigured but
return NULL so that we don't generate and empty RTP-Info header.
Improve docs a little.
2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
* gst/rtsp-server/rtsp-session-media.c:
Don't free rtpinfo GString when it is NULL
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: only set keyframe flag when modifying start
Only set the keyframe flag when we modify the start position. The
keyframe flag should probably be ignored when no change is requested but
until we can claim this is all documented properly and all demuxer
implement this, avoid setting the flag.
See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-thread-pool.c:
thread-pool: Unref source after mainloop has quit to avoid races in GLib
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: handle NULL seqnum and rtptime arguments
2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-thread-pool.c:
* tests/check/gst/threadpool.c:
thread-pool: Unref reused threads in gst_rtsp_thread_stop()
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: add fallback for missing stats property
Use a fallback when the payloader does not have a stats property
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From f7bc1c3 to 1a07da9
2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: don't leak stats structure
Don't leak the stats structure and deal with NULL stats.
2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
stream: Get rtpinfo properties atomically from payloader
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: refactor state change functions and signals
Make functions to set the target state and the pipeline state and emit
the signals from those functions.
2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: add signal to notify of pending state changes
2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-server: support build against last stable release
Until 1.2.3 is out with the new get_type function and we
can require that.
2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: fix compilation
2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add property to configure profiles
2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: let stream check supported transport
Delegate the check if a transport is allowed to the stream.
See https://bugzilla.gnome.org/show_bug.cgi?id=720696
2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add method to check supported transport
Add a method to check if a transport is supported
2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
configure.ac: Only check for gstreamer-check, not check
We include check in gstreamer-check since quite some time now.
2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: return clock-rate from get_rtpinfo
And use it to correct the rtptime to the requested start-time.
See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
session-media: calculate start-time
2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: also return the running-time
Return the running-time in the rtpinfo as well.
2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
session-media: let the session-media make the RTPInfo
Add method to create the RTPInfo for a stream-transport.
Add method to create the RTPInfo for all stream-transports in a
session-media.
Use the session-media RTPInfo code in client. This allows us to refactor
another method to link the TCP callbacks.
2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
mount-points: sort sequence before g_sequence_lookup
* gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
sort sequence if dirty, otherwise lookup will fail.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
configure: rename package from gst-rtsp to gst-rtsp-server
To match git module name and avoid confusion with the
rtsp lib in gst-plugins-base and rtsp plugin in -good.
2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
configure: bump core/base/good requirement to 1.2.0
Bump to released stable version and make implicit
requirements explicit.
2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
* autogen.sh:
* common:
* configure.ac:
Fix broken gettext setup which is not used anyway
2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>