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ChangeLog
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=== release 1.17.90 ===
2020-08-20 16:15:06 +0100 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* gst-rtsp-server.doap:
* meson.build:
Release 1.17.90
2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
* gst/rtsp-server/rtsp-thread-pool.c:
rtsp-thread-pool.c: fix clang 10 warning
clang 10 is complaining about incompatible types due to the
glib typesystem.
```
../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-thread-pool.c:534:10: error: incompatible pointer types passing 'typeof ((((void *)0))) *' (aka 'void **') to parameter of type 'GThreadPool **' (aka 'struct _GThreadPool **') [-Werror,-Wincompatible-pointer-types]
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
* gst/rtsp-server/rtsp-thread-pool.c:
rtsp-thread-pool.c: fix clang 10 warning
clang 10 is complaining about incompatible types due to the
glib typesystem.
```
../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-thread-pool.c:534:10: error: incompatible pointer types passing 'typeof ((((void *)0))) *' (aka 'void **') to parameter of type 'GThreadPool **' (aka 'struct _GThreadPool **') [-Werror,-Wincompatible-pointer-types]
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
2020-07-15 11:19:40 +0200 Srimanta Panda <srimanta@axis.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: Fix resource leak in mikey messsage
Fixed a resource leak for mikey message while adding crypto session
failed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/144>
2020-07-08 17:28:57 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
* scripts/extract-release-date-from-doap-file.py:
meson: set release date from .doap file for releases
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/143>
2020-07-02 23:52:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: explicitly set caps on udpsrc elements
This causes them to send caps events before data flow, which is
usually a pretty correct thing to do!
Not doing so manifested in a bug where ssrcdemux wouldn't forward
the caps it had received with an extra ssrc field, as it hadn't
received any caps event.
Fixes #85
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/141>
2020-07-03 02:04:04 +0100 Tim-Philipp Müller <tim@centricular.com>
* docs/gst_plugins_cache.json:
* meson.build:
Back to development
=== release 1.17.2 ===
2020-07-03 00:33:54 +0100 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* docs/gst_plugins_cache.json:
* gst-rtsp-server.doap:
* meson.build:
Release 1.17.2
2020-06-19 22:55:54 -0400 Thibault Saunier <tsaunier@igalia.com>
* docs/gst_plugins_cache.json:
doc: Stop documenting properties from parents
2020-06-22 20:04:45 +0300 Sebastian Dröge <sebastian@centricular.com>
* docs/gst_plugins_cache.json:
docs: Fix version in the plugins cache
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
2020-06-22 12:33:32 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Don't call gst_ghost_pad_construct() anymore
It's deprecated, unneeded and doesn't do anything anymore.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
2020-06-20 00:28:28 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
Back to development
=== release 1.17.1 ===
2020-06-19 19:24:38 +0100 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* docs/gst_plugins_cache.json:
* gst-rtsp-server.doap:
* meson.build:
Release 1.17.1
2020-06-15 19:45:38 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Add/configure transports when completing the pipeline
Otherwise the transports are not set up yet during the PLAY request
handling when unsuspending (and thus unblocking) the media.
In case of live pipelines this then causes the first few packets to go
to the sinks before they know what to do with them, and they simply
discard them which is rather suboptimal in case of keyframes.
For non-live pipelines this is not a problem because the sink will still
be PAUSED and as such not send out the data yet but wait until it goes
to PLAYING, which is late enough.
Adding the transports multiple times is not a problem: if the transport
is already added it won't be added another time and TRUE will be
returned.
This fixes a regression introduced by a7732a68e8bc6b4ba15629c652056c240c624ff0
before 1.14.0.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/107
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
2020-06-15 19:45:21 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix misleading comment
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
2020-06-15 18:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Make sure to also unblock pads when going to PLAYING while buffering
The pad probes are not needed anymore at this point and later when
reaching buffering 100% only the state is changed, no unblocking
happens.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
2020-06-15 18:17:40 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Remove duplicated media_unblock() function
It does literally the same as media_streams_set_blocked(FALSE).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
2020-06-12 15:38:45 +0200 Lenny Jorissen <lennyjorissen@gmail.com>
* examples/test-onvif-server.c:
test-onvif-server: cast ntp-offset property value to 64 bit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/134>
2020-06-09 15:21:24 -0400 Thibault Saunier <tsaunier@igalia.com>
* docs/gst_plugins_cache.json:
docs: Update plugins cache
2020-06-10 13:45:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/test-onvif-server.c:
* examples/test-onvif-server.h:
* gst/rtsp-server/rtsp-onvif-media-factory.h:
onvif-media-factory: define autoptr cleanup function
And have the factory in the onvif-server example inherit from
GstRTSPOnvifMediaFactory.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/133>
2020-06-08 10:59:34 -0400 Thibault Saunier <tsaunier@igalia.com>
* docs/gst_plugins_cache.json:
docs: Update plugins cache
2020-06-08 09:45:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
* tests/check/gst/rtspserver.c:
tests: enforce I420 format
Test was not enforcing a video format on videotestsrc. I420 was picked as it
was the first format in GST_VIDEO_FORMATS_ALL which will no longer be
true (gst-plugins-base!689).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/129>
2020-06-06 00:41:51 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
plugins: uddate gst_type_mark_as_plugin_api() calls
2020-06-03 18:36:25 -0400 Thibault Saunier <tsaunier@igalia.com>
* docs/meson.build:
doc: Require hotdoc >= 0.11.0
2020-05-27 17:00:05 +0300 Sebastian Dröge <sebastian@centricular.com>
* docs/gst_plugins_cache.json:
docs: Update gst_plugins_cache.json
2020-05-30 23:23:51 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types
2020-05-27 23:38:06 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/meson.build:
meson: gir: remove bogus sources_top_dir kwarg
Doesn't actually exist. Was fixed differently in Meson
so that the user doesn't have to specify it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/127>
2020-05-27 17:43:43 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/meson.build:
tests: put registry into tests/check not the gst/ subdir
Underscorify the test name before setting GST_REGISTRY,
so the registry actually ends up in the current build dir
and not some subdir.
For consistency with the other modules, but should also
avoid problems on windows.
Also fix indentation of environment block.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
2020-05-27 17:33:24 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/meson.build:
tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
If core is built as a subproject (e.g. as in gst-build), make sure to use
the gst-plugin-scanner from the built subproject. Without this, gstreamer
might accidentally use the gst-plugin-scanner from the install prefix if
that exists, which in turn might drag in gst library versions we didn't
mean to drag in. Those gst library versions might then be older than
what our current build needs, and might cause our newly-built plugins
to get blacklisted in the test registry because they rely on a symbol
that the wrongly-pulled in gst lib doesn't have.
This should fix running of unit tests in gst-build when invoking
meson test or ninja test from outside the devenv for the case where
there is an older or different-version gst-plugin-scanner installed
in the install prefix.
In case no gst-plugin-scanner is installed in the install prefix, this
will fix "GStreamer-WARNING: External plugin loader failed. This most
likely means that the plugin loader helper binary was not found or
could not be run. You might need to set the GST_PLUGIN_SCANNER
environment variable if your setup is unusual." warnings when running
the unit tests.
In the case where we find GStreamer core via pkg-config we use
a newly-added pkg-config var "pluginscannerdir" to get the right
directory. This has the benefit of working transparently for both
installed and uninstalled pkg-config files/setups.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
2020-05-27 17:32:02 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/meson.build:
tests: gst-plugins-base and -bad plugins are required for the unit tests
Make hard requirement until we have more fine-grained control
in the unit tests. Of course the presence of the .pc file doesn't
imply that the plugins we need are actually there, but it's at
least a step in the right direction.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
2020-05-27 17:29:18 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/meson.build:
tests: pick up rtsp-server plugins from build directory only
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
2020-05-26 15:31:22 +0200 Ludvig Rappe <ludvigr@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: wait for all GstRTSPStreamBlocking messages
Make sure rtsp-media have received a GstRTSPStreamBlocking message from
each active stream when checking if all streams are blocked.
Without this change there will be a race condition when using two or
more streams and rtsp-media receives a GstRTSPStreamBlocking message
from one of the streams. This is because rtsp-media then checks if all
streams are blocked by calling gst_rtsp_stream_is_blocking() for each
stream. This function call returns TRUE if the stream has sent a
GstRTSPStreamBlocking message, however, rtsp-media may have yet to
receive this message. This would then result in that rtsp-media
erroneously thinks it is blocking all streams which could result in
rtsp-media changing state, from PREPARING to PREPARED. In the case of a
preroll, this could result in that rtsp-media thinks that the pipeline
is prerolled even though that might not be the case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/124>
2020-05-04 13:43:00 +0200 Ludvig Rappe <ludvigr@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: update expected_async_done during suspend
Set expected_async_done to FALSE in default_suspend() if a state change
occurs and the return value from set_target_state() is something other
than GST_STATE_CHANGE_ASYNC.
Without this change there is a risk that expected_async_done will be
TRUE even though no asynchronous state change is taking place. This
could happen if the pipeline is set to PAUSED using
media_set_pipeline_state_locked(), an asynchronous state change starts
and then the media is suspended (which could result in a state change,
aborting the asynchronous state change). If the media is suspended
before the asynchronous state change ends then expected_async_done will
be TRUE but no asynchronous state change is taking place.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/123>
2020-05-25 13:49:45 +0200 Kristofer Björkström <kristofb@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Fix race condition in rtsp ctrl timeout by WeakRef client
There was a race condition where client was being finalized and
concurrently in some other thread the rtsp ctrl timout was relying on
client data that was being freed.
When rtsp ctrl timeout is setup, a WeakRef on Client is set.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/121>
2015-03-03 14:42:07 +0100 Gregor Boirie <gregor.boirie@parrot.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:
media-factory: complete DSCP QoS setting support
add dscp_qos setting support at factory and media level to setup IP DSCP
field of bounded UDP sinks.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
2020-05-14 10:08:32 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Fix some race conditions around timeout source removal
We always need to take the lock while accessing it as otherwise another
thread might've removed it in the meantime. Also when destroying and
creating a new one, ensure that the mutex is not shortly unlocked in
between as during that time another one might potentially be created
already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/119>
2020-05-03 16:29:31 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-media: Mark out parameters accordingly in gst_rtsp_media_get_rates()
And the same for gst_rtsp_stream_get_rates().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/118>
2020-05-03 10:17:41 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/test-onvif-server.c:
examples: test-onvif-server: fix compiler warnings on raspbian
Fix printf format for 64-bit variables.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/117>
2020-05-01 10:42:17 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks
The old API is preserved now and new API was added that provides the
additional parameter to the callback.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
2020-04-28 23:33:49 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Store the timeout source by pointer instead of id
That way we don't have to retrieve it again from the main context when
destroying it but can directly do so.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
2020-04-28 23:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Clean up watch/watch context and related state consistently
And assert that it was cleaned up properly before the client is
finalized. If something is still around when the client is shut down
then something went very wrong before.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
2020-04-27 23:25:22 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/rtspserver.c:
rtsp-client: Combine the pre-session and post-session timeout
They previously used the same state but different mechanisms and
functions, which was difficult to follow, error prone and simply
confusing.
Also adjust the test for the post-session timeout a bit to be less racy
now that the timing has slightly changed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
2020-04-27 19:47:15 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Don't ever close the client connection directly when a session is torn down
There might be other sessions that are running over the same RTSP
connection and we should not simply close the client directly if one of
them is torn down.
By default the connection will be closed once the client closes it or
the OS does. This behaviour can be adjusted with the
post-session-timeout property, which allows to close it automatically
from the server side after all sessions are gone and the given timeout
is reached.
This reverts the previous commit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
2020-04-27 13:49:55 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: If the TEARDOWN response can be sent directly, directly close the client
Instead of closing it never at all. Previously there was only code that
closed the client asynchronously if sending the response happened
asynchrously at a later time.
Thanks to Christian M for debugging this issue.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/102
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/114>
2020-03-23 14:51:28 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: use mcast_udpsink[0] last-sample if available for rtpinfo
Otherwise no sink is found for multicast sreams and the less accurate
fallback is used to determine the current sequence number and timestamp.
2020-03-23 16:06:43 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-auth.c:
rtsp-auth: Fix NULL pointer dereference when handling an invalid basic Authorization header
When using the basic authentication scheme, we wouldn't validate that
the authorization field of the credentials is not NULL and pass it on
to g_hash_table_lookup(). g_str_hash() however is not NULL-safe and will
dereference the NULL pointer and crash.
A specially crafted (read: invalid) RTSP header can cause this to
happen.
As a solution, check for the authorization to be not NULL before
continuing processing it and if it is simply fail authentication.
This fixes CVE-2020-6095 and TALOS-2020-1018.
Discovered by Peter Wang of Cisco ASIG.
2020-03-09 14:17:34 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Use watch_context before unref
Move the usage of priv->watch_context to beginning of function
gst_rtsp_client_finalize. Instead of use it after
g_main_context_unref (priv->watch_context).
2020-02-14 14:59:43 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fix deadlock on transport removal
We cannot take the RTSPStream lock while holding a transport backlog
lock, as remove_transport may be called externally, which will
take first the RTSPStream lock then the transport backlog lock.
2020-02-14 14:59:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-server-internal.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: clear backlog when removing transport
This ensures we don't end up calling any of transports' callbacks
with a potentially unreffed user_data (in practice, a client that
may have been removed)
2020-02-06 22:46:18 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: marshal calls to send_tcp_message to a single thread
In order to address the race condition pointed out at
https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/merge_requests/108#note_403579
we get rid of the send thread pool, and instead spawn and manage
a single thread to pull samples from app sinks and add them to
the transport's backlogs.
Additionally, we now also always go through the backlogs in order
to simplify the logic.
2020-02-05 20:28:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-server-internal.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: properly protect TCP backlog access
Fixes #97
We cannot hold stream->lock while pushing data, but need
to consistently check the state of the backlog both from
the send_tcp_message function and the on_message_sent function,
which may or may not be called from the same thread.
This commit introduces internal API to allow for potentially
recursive locking of transport streams, addressing a race
condition where the RTSP stream could push items out of order
when popping them from the backlog.
2020-02-22 00:41:32 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Sink pipeline in gst_rtsp_media_take_pipeline()
It's taken ownership of by the media, and returned with `transfer none`
from the GstRTSPMedia::create_pipeline() vfunc. If we don't sink it
first then any bindings will wrongly take ownership of the pipeline once
it arrives in bindings code.
2020-02-05 16:51:14 +0100 Bastian Bouchardon <bastian.bouchardon@gmail.com>
* examples/test-onvif-client.c:
Add initialization for context and params (gchar *) Insert define (DEFAULT_*) into help to have to modify only the constants
2020-02-03 12:30:14 +0000 Marc Leeman <marc.leeman@gmail.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: fix default latency
2020-01-15 17:06:41 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: make closing more thread safe
+ Take the watch lock prior to using priv->watch
+ Flush both the watch and connection before closing / unreffing
gst_rtsp_connection_close() is not threadsafe on its own, this is
a workaround at the client level, where we control both the watch
and the connection
2020-01-23 16:41:26 +0200 Jordan Petridis <jordan@centricular.com>
* gst/rtsp-server/rtsp-latency-bin.c:
rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's been deprecated
from glib
```
Deprecated: 2.58: Use %G_ADD_PRIVATE and the generated
`your_type_get_instance_private()` function instead
```
2019-12-17 16:08:19 +0100 Zoltán Imets <zoltani@axis.com>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/rtspserver.c:
rtsp-client: add property post-session-timeout
This is a TCP connection timeout for client connections, in seconds.
If a positive value is set for this property, the client connection
will be kept alive for this amount of seconds after the last session
timeout. For negative values of this property the connection timeout
handling is delegated to the system (just as it was before).
Fixes #83
2020-01-11 22:58:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: check for NULL transports prior to ref'ing
2020-01-09 14:10:44 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-server-internal.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fix checking of TCP backpressure
The internal index of our appsinks, while it can be used to
determine whether a message is RTP or RTCP, is not necessarily
the same as the interleaved channel. Let the stream-transport
determine the channel to check backpressure for, the same way
it determines the channel according to whether it is sending
RTP or RTCP.
2019-12-10 19:16:51 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-session.c:
rtsp-session: Butcher the file to please gst-indent in the CI
This should be reverted once the CI has an updated gst-indent.
2019-12-10 18:39:32 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
rtsp-session & client: Remove deprecated GTimeVal
GTimeVal won't work past 2038
2019-12-12 17:56:18 +0100 Nicola Murino <nicola.murino@gmail.com>
* gst/rtsp-server/rtsp-auth.c:
rtsp-auth: fix default token leak
2019-12-09 14:17:05 +0100 Adam x Nilsson <adamni@axis.com>
* gst/rtsp-sink/gstrtspclientsink.c:
gstrtspclientsink: unref transports when closing bin
Fixes #91
2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Force seek when flush flag is set
The commit "rtsp-client: define all seek accuracy flags from
setup_play_mode" changed the behaviour of when doing a seek.
Before that commit, having the flush flag set would result in a seek
(forced seek).
Even if no seek was needed. One reason to force seek is to flush old buffers
created in Describe requests.
Thus adding force seek also for flush flag will result in play request
with fresh buffers.
2019-11-21 17:12:45 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Revitalize dead code
Leftover from 65d9aa327cd1844934836249cd4463edf09c725d
CID: 1455379
2019-11-27 15:22:35 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: Don't try to use non-initialized values
Only attempt to use the various timing values iif gst_rtsp_stream_get_info()
returns TRUE. Also avoid the whole clock signalling block if we're not
dealing with senders.
CID: 1439524
CID: 1439536
CID: 1439520
2019-11-01 12:01:41 +0100 Adam x Nilsson <adamni@axis.com>
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/stream.c:
rtsp-stream: Removing invalid transports returns false
When removing transports an assertion was that the transports passed in
for removal are present in the list, however that can't be assumed.
As an example if a transport was removed from a thread running
send_tcp_message, the main thread can try to remove the same transport
again if it gets a handle_pause_request. This will not effect the
transport list but it will effect n_tcp_transports as it will be
decrement and then have the wrong value.
2019-11-06 14:17:48 +0100 Zoltán Imets <zoltani@axis.com>
* tests/check/gst/client.c:
client test: add scale and speed negative tests
Negative tests for scale and speed should be done as well, verify that
the response code is "400 Bad request" when a bad request is done.
2019-08-29 07:34:26 +0200 Niels De Graef <nielsdegraef@gmail.com>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-sink/gstrtspclientsink.c:
Don't pass default GLib marshallers for signals
By passing NULL to `g_signal_new` instead of a marshaller, GLib will
actually internally optimize the signal (if the marshaller is available
in GLib itself) by also setting the valist marshaller. This makes the
signal emission a bit more performant than the regular marshalling,
which still needs to box into `GValue` and call libffi in case of a
generic marshaller.
Note that for custom marshallers, one would use
`g_signal_set_va_marshaller()` with the valist marshaller instead.
2019-09-05 19:51:06 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-mount-points.c:
GstRTSPMountPoints: Remove any existing factory before adding a new one
The documentation of gst_rtsp_mount_points_add_factory() says "Any
previous mount point will be freed" which was true when it was
implemented using a GHashTable. But in 2012 it got rewrote using a
GSequence and since then it could have 2 factories for the same path.
Which one gets used is random, depending on the sorting order of 2
identical items.
2019-10-15 19:08:32 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-server-internal.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
stream: refactor TCP backpressure handling
The previous implementation stopped sending TCP messages to
all clients when a single one stopped consuming them, which
obviously created problems for shared media.
Instead, we now manage a backlog in stream-transport, and slow
clients are removed once this backlog exceeds a maximum duration,
currently hardcoded.
Fixes #80
2019-10-18 00:42:12 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: build gir even when cross-compiling if introspection was enabled explicitly
This can be made to work in certain circumstances when
cross-compiling, so default to not building g-i stuff
when cross-compiling, but allow it if introspection was
enabled explicitly via -Dintrospection=enabled.
See gstreamer/gstreamer#454 and gstreamer/gstreamer#381.
2019-10-18 09:19:59 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-session.c:
rtsp-session: clean up comment extra-timeout
2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses
Instead of hardcoding the URI, take the actual URI (and especially the correct port)
from the RTSP context.
Fixes #84
2019-10-16 13:20:54 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
rtsp-client: Lock shared media
For shared media we got race conditions. Concurrently rtsp clients might
suspend or unsuspend the shared media and thus change the state without
the clients expecting that.
By introducing a lock that can be taken by callers such as rtsp_client
one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media,
to handle the media sequentially thus allowing one client to finish its
rtsp call before another client calls on the same media.
https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86
Fixes #86
2019-10-15 07:33:29 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-session.c:
rtsp-session: add property extra-timeout
Extra time to add to the timeout, in seconds. This only
affects the time until a session is considered timed out
and is not signalled in the RTSP request responses.
Only the value of the timeout property is signalled in the
request responses.
2019-10-07 12:13:47 +0200 Adam x Nilsson <adamni@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream : fix race condition in send_tcp_message
If one thread is inside the send_tcp_message function and are done
sending rtp or rtcp messages so the n_outstanding variable is zero
however have not exit the loop sending the messages. While sending its
messages, transports have been added or removed to the transport list,
so the cache should be updated. If now an additional thread comes to
the function send_tcp_message and trying to send rtp messages it will
first destroy the rtp cache that is still being iterated trough by the
first thread.
Fixes #81
2019-05-24 14:32:50 +0200 Tim-Philipp Müller <tim@centricular.com>
* .gitignore:
* .gitmodules:
* Makefile.am:
* autogen.sh:
* common:
* configure.ac:
* docs/.gitignore:
* examples/.gitignore:
* examples/Makefile.am:
* gst/Makefile.am:
* gst/rtsp-server/.gitignore:
* gst/rtsp-server/Makefile.am:
* gst/rtsp-sink/Makefile.am:
* pkgconfig/.gitignore:
* pkgconfig/Makefile.am:
* tests/.gitignore:
* tests/Makefile.am:
* tests/check/Makefile.am:
Remove autotools build
Replaced by Meson.
Maybe we can now use the meson pkgconfig module
for .pc files? (Does it support uninstalled now?)
2019-10-07 10:27:36 +0200 Göran Jönsson <goranjn@axis.com>
* tests/check/gst/client.c:
client: fix test mem leak in attach_rate_tweaking_probe
2019-10-07 10:14:52 +0200 Göran Jönsson <goranjn@axis.com>
* tests/check/gst/media.c:
media: remove memleak in test test_media_seek
2019-10-07 10:07:54 +0200 Göran Jönsson <goranjn@axis.com>
* tests/check/gst/rtspserver.c:
rtspserver: Remove memleak in test test_double_play
2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Use lock in gst_rtsp_media_is_receive_only
2018-10-29 17:02:41 +0100 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/rtspserver.c:
rtsp-media: Unblock all streams
When unsuspending and going to PLAYING, unblock all streams instead of
only those that are linked (the linked streams are the ones for which
SETUP has been called). GST_FLOW_NOT_LINKED will be returned when
pushing buffers on unlinked streams.
This change is because playback using single-threaded demuxers like
matroska-demux could be blocked if SETUP was not called for all media.
Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux,
gstflvdemux, qtdemux, and matroska-demux) will handle
GST_FLOW_NOT_LINKED automatically.
Fixes #39
2019-09-11 07:08:37 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/rtspserver.c:
rtsp-media: Wait on async when needed.
Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode.
In the unit test the pause from adjust_play_mode will cause a preroll
and after that async-done will be produced.
Without this patch there are no one consuming this async-done and when
later when seek fluch is done in gst_rtsp_media_seek_trickmode then it
wait for async-done. But then it wrongly find the async-done prodused by
adjus_play_mode and continue executing without waiting for the preroll
to finish.
2019-09-30 15:13:15 +0200 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: RTP Info when completed_sender
Change condition that should be fulfilled regarding RTPInfo.
Replace !gst_rtsp_media_is_receive_only with
gst_rtsp_media_has_completed_sender. It is more correct to actually look
for a sender pipeline that is complete. Only then a RTPInfo should
exist.
gst_rtsp_media_is_receive_only gives different answears depending on
state of server.
If Describe is called wth URL+options for backchannel SDP will give only
audio and only backchannel a=sendonly
If Describe is called on URL+options that gives both audio and video
direction from server to client, pipelines are created. Thus
receive_only will return false, even though Setup only would setup
backchannel.
RTP-Info is only for outgoing streams. Thus one should look if outgoing
streams are complete.
2019-09-25 09:14:08 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/client.c:
rtsp-client: RTP Info exists conditionally in PLAY
If RTP Info is missing and it is not a receiver only, eg. audio
backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR.
In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional.
Since 1.14 there is audio backchannel support. Thus RTP-info is
conditional now. When audio backchannel only mode, there is no RTP-info.
Fixes #82
2019-09-05 16:23:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/test-onvif-client.c:
test-onvif-client: remove unused query
2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: RTP Info must exist in PLAY response
If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
Fixes #76
2019-08-29 21:37:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/test-onvif-client.c:
test-onvif-client: perform accurate seeks
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/336
Also, modify how we compute the position: position queries in
PAUSED mode fail to account for the newly-prerolled frame, leading
to frame skips when performing seeks in that state. Instead,
compute the current position from the last sample.
2019-08-21 14:57:25 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* tests/check/gst/rtspserver.c:
Use complete streams for scale and speed.
Without this patch it's always stream0 that is used to get segment event
that is used to set scale and speed. This even if client not doing SETUP
for stream0. At least in suspend mode reset this not working since then
it's just random if send_rtp_sink have got any segment event. There are
no check if send_rtp_sink for stream0 got any data before media is
prerolled after PLAY request.
2019-08-26 22:24:12 +1000 Matthew Waters <matthew@centricular.com>
* examples/test-onvif-server.c:
* examples/test-onvif-server.h:
examples/onvif-server: fix werror build with clang
../subprojects/gst-rtsp-server/examples/test-onvif-server.c:346:65: warning: implicit conversion from enumeration type 'const GstSegmentFlags' to different enumeration type 'GstSeekFlags' [-Wenum-conversion]
self->incoming_segment->format, self->incoming_segment->flags,
~~~~~~~~~~~~~~~~~~~~~~~~^~~~~
../subprojects/gst-rtsp-server/examples/test-onvif-server.c:53:1: warning: unused function 'REPLAY_IS_BIN' [-Wunused-function]
G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin);
^
/usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
^
<scratch space>:77:1: note: expanded from here
REPLAY_IS_BIN
^
../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_FACTORY' [-Wunused-function]
G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY,
^
/usr/include/glib-2.0/gobject/gtype.h:1405:33: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
static inline ModuleObjName * MODULE##_##OBJ_NAME (gpointer ptr) { \
^
<scratch space>:9:1: note: expanded from here
ONVIF_FACTORY
^
../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_IS_FACTORY' [-Wunused-function]
/usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
^
<scratch space>:12:1: note: expanded from here
ONVIF_IS_FACTORY
^
2019-08-23 16:21:36 +1000 Matthew Waters <matthew@centricular.com>
* docs/meson.build:
meson: Don't generate doc cache when no plugins are enabled
Fixes gst-build with -Dauto-features=disabled -Drtsp_server=enabled
2019-08-16 13:38:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* examples/test-onvif-client.c:
test-onvif-client: stdin is not defined in MSVC
2019-08-12 18:03:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: add missing Since tag
2019-08-08 15:52:53 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/test-onvif-client.c:
test-onvif-client: STDIN_FILENO is not portable
If not defined, define it to _fileno(stdin) on Windows, 0
everywhere else
2019-08-07 21:04:33 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/test-onvif-server.c:
test-onvif-server: downgrade logging
2019-07-27 05:14:49 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/meson.build:
* examples/test-onvif-client.c:
* examples/test-onvif-server.c:
examples: add ONVIF client / server example
2019-07-27 05:14:28 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
rtsp-client: define all seek accuracy flags from setup_play_mode
We then pass those to adjust_play_mode, which needs to operate
on the "final" seek flags, as previously the code in rtsp-media
was assuming that accuracy seek flags (accurate / key_unit) should
not be set if the flags passed to the seek method were already set.
2019-07-22 19:32:43 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Try to get dynamic payloaders by name from their bin first
First try "pay", then "pay_%s" (where %s == pad name). And only then
fall back to the code that simply takes the first payloader that is
found.
The current code usually works (but is racy) because it will always take
the payloader that was last added (due to g_list_prepend() when adding
elements) in pad-added and that's usually the correct one. But if a new
payloader is added between pad-added and us trying to get it, we would
get the wrong payloader.
2019-07-17 15:51:08 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* tests/check/gst/client.c:
client test: expect any port in transport
setup_multicast_client sets a 5000-5010 range for the client
ports, it is incorrect to expect the transport to always use
5000-5001