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ChangeLog
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=== release 1.15.1 ===
2019-01-17 02:26:48 +0000 Tim-Philipp Müller <[email protected]>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.15.1
2018-12-05 15:07:25 +0100 Patricia Muscalu <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
Add source elements to the pipeline before activation
In plug_src we changed the element state before adding it to
the owner container. This prevented the pipeline from intercepting
a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
to assign a custom task pool.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
2018-12-05 17:24:59 -0300 Thibault Saunier <[email protected]>
* common:
Automatic update of common submodule
From ed78bee to 59cb678
2018-11-20 19:12:09 +0100 Ingo Randolf <[email protected]>
* examples/test-appsrc.c:
examples: test-appsrc: fix coding style error
2018-11-20 11:07:48 +0100 Ingo Randolf <[email protected]>
* examples/test-appsrc.c:
examples: test-appsrc: fix buffer leak
2018-11-17 19:19:54 +0100 Patricia Muscalu <[email protected]>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Update priv->blocked when linked streams are unblocked.
Media is considered to be blocked when all streams that belong to
that media are blocked.
This patch solves the problem of inconsistent updates of
priv->blocked that are not synchronized with the media state.
2018-11-17 18:18:27 +0100 Patricia Muscalu <[email protected]>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't block streams before seeking
Before the seek operation is performed on media, it's required that
its pipeline is prepared <=> the pipeline is in the PAUSED state.
At this stage, all transport parts (transport sinks) have been successfully
added to the pipeline and there is no need for blocking the streams.
2018-11-17 16:11:53 +0100 Patricia Muscalu <[email protected]>
* tests/check/gst/rtspserver.c:
tests: rtspserver: Add shared media test case for TCP
2018-11-06 18:21:54 +0100 Linus Svensson <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Use seqnum-offset for rtpinfo
The sequence number in the rtpinfo is supposed to be the first RTP
sequence number. The "seqnum" property on a payloader is supposed to be
the number from the last processed RTP packet. The sequence number for
payloaders that inherit gstrtpbasepayload will not be correct in case of
buffer lists. In order to fix the seqnum property on the payloaders
gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
"seqnum-offset" from the "stats" property contains the value of the
very first RTP packet in a stream. The server will, however, try to look
at the last simple in the sink element and only use properties on the
payloader in case there no sink elements yet, and by looking at the last
sample of the sink gives the server full control of which RTP packet it
looks at. If the payloader does not have the "stats" property, "seqnum"
is still used since "seqnum-offset" is only present in as part of
"stats" and this is still an issue not solved with this patch.
Needed for gst-plugins-base!17
2018-11-06 18:10:56 +0100 Linus Svensson <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Plug memory leak
Attaching a GSource to a context will increase the refcount. The idle
source will never be free'd since the initial reference is never
dropped.
2018-11-12 16:06:39 +0200 Jordan Petridis <[email protected]>
* .gitlab-ci.yml:
Add Gitlab CI configuration
This commit adds a .gitlab-ci.yml file, which uses a feature
to fetch the config from a centralized repository. The intent is
to have all the gstreamer modules use the same configuration.
The configuration is currently hosted at the gst-ci repository
under the gitlab/ci_template.yml path.
Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
2018-11-05 05:56:35 +0000 Matthew Waters <[email protected]>
* .gitmodules:
* gst-rtsp-server.doap:
Update git locations to gitlab
2018-11-01 14:20:16 +0100 Mathieu Duponchelle <[email protected]>
* gst/rtsp-server/meson.build:
meson: add new onvif types
2018-11-01 12:49:51 +0200 Sebastian Dröge <[email protected]>
* gst/rtsp-server/meson.build:
Add ONVIF subclass headers to the installed headers in meson.build too
2018-11-01 11:29:01 +0200 Sebastian Dröge <[email protected]>
* gst/rtsp-server/rtsp-server-object.h:
* gst/rtsp-server/rtsp-server.h:
rtsp-server: Declare GstRTSPServer struct before anything else
It's needed by all kinds of other headers, including the ones that are
required for defining the GstRTSPServer struct itself and its API.
2018-11-01 10:23:02 +0200 Sebastian Dröge <[email protected]>
* gst/rtsp-server/rtsp-onvif-client.h:
* gst/rtsp-server/rtsp-onvif-media-factory.h:
* gst/rtsp-server/rtsp-onvif-media.h:
* gst/rtsp-server/rtsp-onvif-server.h:
Mark all ONVIF-specific subclasses as Since 1.14
2018-11-01 10:18:22 +0200 Sebastian Dröge <[email protected]>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/meson.build:
* gst/rtsp-server/rtsp-context.h:
* gst/rtsp-server/rtsp-onvif-server.c:
* gst/rtsp-server/rtsp-onvif-server.h:
* gst/rtsp-server/rtsp-server-object.h:
* gst/rtsp-server/rtsp-server-prelude.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session.h:
Include ONVIF types from single-include rtsp-server.h
... by actually making it a single-include header and moving everything
related to the GstRTSPServer type to rtsp-server-object.h instead.
Otherwise there are too many circular includes.
https://bugzilla.gnome.org/show_bug.cgi?id=797361
2018-10-18 07:25:05 +0200 Göran Jönsson <[email protected]>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-latency-bin.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: use idle source in on_message_sent
When the underlying layers are running on_message_sent, this sometimes
causes the underlying layer to send more data, which will cause the
underlying layer to run callback on_message_sent again. This can go on
and on.
To break this chain, we introduce an idle source that takes care of
sending data if there are more to send when running callback
https://bugzilla.gnome.org/show_bug.cgi?id=797289
2018-10-20 16:14:53 +0200 Edward Hervey <[email protected]>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Remove timeout GSource on cleanup
Avoids ending up with races where a timeout would still be around
*after* a client was gone. This could happen rather easily in
RTSP-over-HTTP mode on a local connection, where each RTSP message
would be sent as a different HTTP connection with the same tunnelid.
If not properly removed, that timeout would then try to free again
a client (and its contents).
2018-10-04 14:31:59 +0100 Tim-Philipp Müller <[email protected]>
* gst/rtsp-server/Makefile.am:
autotools: fix distcheck
2018-09-12 11:55:15 +0200 Ognyan Tonchev <[email protected]>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/meson.build:
* gst/rtsp-server/rtsp-latency-bin.c:
* gst/rtsp-server/rtsp-latency-bin.h:
* gst/rtsp-server/rtsp-onvif-media.c:
onvif: encapsulate onvif part into a bin
...and thus do not let onvif affect pipelines latency
https://bugzilla.gnome.org/show_bug.cgi?id=797174
2018-09-27 19:57:13 +0200 Patricia Muscalu <[email protected]>
* tests/check/gst/client.c:
tests: client: Avoid bind() failures in tests
https://bugzilla.gnome.org/show_bug.cgi?id=797059
2018-09-06 16:17:33 +0200 Patricia Muscalu <[email protected]>
* 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-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/mediafactory.c:
New property for socket binding to mcast addresses
By default the multicast sockets are bound to INADDR_ANY,
as it's not allowed to bind sockets to multicast addresses
in Windows. This default behaviour can be changed by setting
bind-mcast-address property on the media-factory object.
https://bugzilla.gnome.org/show_bug.cgi?id=797059
2018-09-24 09:36:21 +0100 Tim-Philipp Müller <[email protected]>
* configure.ac:
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/meson.build:
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-context.c:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-params.c:
* gst/rtsp-server/rtsp-permissions.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-server-prelude.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-thread-pool.c:
* gst/rtsp-server/rtsp-token.c:
* meson.build:
libs: fix API export/import and 'inconsistent linkage' on MSVC
Export rtsp-server library API in headers when we're building the
library itself, otherwise import the API from the headers.
This fixes linker warnings on Windows when building with MSVC.
Fix up some missing config.h includes when building the lib which
is needed to get the export api define from config.h
https://bugzilla.gnome.org/show_bug.cgi?id=797185
2018-09-19 14:31:56 +0200 Edward Hervey <[email protected]>
* gst/rtsp-server/rtsp-media-factory.c:
rtsp-media-factory: Add missing break statements
This resulted in warnings/assertions whenever one accessed the
max-mcast-ttl property.
CID #1439515
CID #1439523
2018-09-19 12:21:30 +0100 Tim-Philipp Müller <[email protected]>
* meson.build:
* meson_options.txt:
meson: add gobject-cast-checks, glib-asserts, glib-checks options
2018-09-19 12:17:57 +0100 Tim-Philipp Müller <[email protected]>
* gst/meson.build:
* meson_options.txt:
* tests/check/meson.build:
meson: add option to disable build of rtspclientsink plugin
2018-09-19 12:10:14 +0100 Tim-Philipp Müller <[email protected]>
* meson_options.txt:
meson: re-arrange options
2018-09-01 11:21:15 +0530 Nirbheek Chauhan <[email protected]>
* meson.build:
* meson_options.txt:
* tests/check/meson.build:
* tests/meson.build:
meson: Use feature option for tests option
This was somehow missed the last time around.
2018-08-31 14:42:15 +0530 Nirbheek Chauhan <[email protected]>
* gst/rtsp-server/meson.build:
* meson.build:
meson: Maintain macOS ABI through dylib versioning
Requires Meson 0.48, but the feature will be ignored on older versions
so it's safe to add it without bumping the requirement.
Documentation:
https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2018-08-31 17:20:47 +1000 Matthew Waters <[email protected]>
* gst/rtsp-sink/meson.build:
* meson.build:
meson: add pkg-config file for the rtspclientsink plugin
2018-08-17 09:54:27 +0200 David Svensson Fors <[email protected]>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/client.c:
rtsp-client: Avoid reuse of channel numbers for interleaved
If a (strange) client would reuse interleaved channel numbers in
multiple SETUP requests, we should not accept them. The channel
numbers are used for looking up stream transports in the
priv->transports hash table, and transports disappear from the table
if channel numbers are reused.
RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
server to change the channel numbers suggested by the client.
https://bugzilla.gnome.org/show_bug.cgi?id=796988
2018-08-17 09:54:27 +0200 David Svensson Fors <[email protected]>
* tests/check/gst/client.c:
rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
Allow regex for matching transport header against expected pattern.
https://bugzilla.gnome.org/show_bug.cgi?id=796988
2018-08-15 18:57:27 +0530 Nirbheek Chauhan <[email protected]>
* tests/check/meson.build:
meson: There is no gstreamer-plugins-good-1.0.pc
There is no installed version of that, only an uninstalled version.
2018-08-14 14:31:55 +0300 Sebastian Dröge <[email protected]>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/stream.c:
Fix indentation again
2018-07-26 12:01:16 +0200 Patricia Muscalu <[email protected]>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/stream.c:
stream: Added a list of multicast client addresses
When media is shared, the same media stream can be sent
to multiple multicast groups. Currently, there is no API
to retrieve multicast addresses from the stream.
When calling gst_rtsp_stream_get_multicast_address() function,
only the first multicast address is returned.
With this patch, each multicast destination requested in SETUP
will be stored in an internal list (call to
gst_rtsp_stream_add_multicast_client_address()).
The list of multicast groups requested by the clients can be
retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
There still exist some problems with the current implementation
in the multicast case:
1) The receiving part is currently only configured with
regard to the first multicast client (see
https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2) Secondly, of security reasons, some constraints should be
put on the requested multicast destinations (see
https://bugzilla.gnome.org/show_bug.cgi?id=796916).
Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-25 15:33:18 +0200 Patricia Muscalu <[email protected]>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
stream: Choose the maximum ttl value provided by multicast clients
The maximum ttl value provided so far by the multicast clients
will be chosen and reported in the response to the current
client request.
Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-02-23 14:34:32 +0100 Patricia Muscalu <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/client.c:
rtsp-stream: Don't require address pool in the transport specific case
If "transport.client-settings" parameter is set to true, the client is
allowed to specify destination, ports and ttl.
There is no need for pre-configured address pool.
Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-24 14:02:40 +0200 Patricia Muscalu <[email protected]>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/client.c:
client: Don't reserve multicast address in the client setting case
When two multicast clients request specific transport
configurations, and "transport.client-settings" parameter is
set to true, it's wrong to actually require that these two
clients request the same multicast group.
Removed test_client_multicast_invalid_transport_specific test
cases as they wrongly require that the requested destination
address is supposed to be present in the address pool, also in
the case when "transport.client-settings" parameter is set to true.
Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-24 09:35:46 +0200 Patricia Muscalu <[email protected]>
* 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-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/mediafactory.c:
Add new API for setting/getting maximum multicast ttl value
Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-31 21:17:41 +0200 Mathieu Duponchelle <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: avoid duplicating the first multicast client
In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
clients were dynamically added and removed to the multicast
udp sinks, as such we should no longer add a first client in
set_multicast_socket_for_udpsink
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:25:53 +0300 Sebastian Dröge <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
Revert "rtsp-stream: avoid duplicating the first multicast client"
This reverts commit 33570944401747f44d8ebfec535350651413fb92.
Commits where accidentially squashed together
2018-08-14 14:25:42 +0300 Sebastian Dröge <[email protected]>
* 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:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/mediafactory.c:
Revert "Add new API for setting/getting maximum multicast ttl value"
This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
Commits where accidentially squashed together
2018-08-14 14:25:37 +0300 Sebastian Dröge <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/client.c:
Revert "rtsp-stream: Don't require address pool in the transport specific case"
This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
Commits where accidentially squashed together
2018-08-14 14:25:14 +0300 Sebastian Dröge <[email protected]>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/stream.c:
Revert "stream: Choose the maximum ttl value provided by multicast clients"
This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
Commits where accidentially squashed together
2018-08-14 14:10:56 +0300 Sebastian Dröge <[email protected]>
* examples/test-auth-digest.c:
examples: Fix indentation
2018-07-25 15:33:18 +0200 Patricia Muscalu <[email protected]>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/stream.c:
stream: Choose the maximum ttl value provided by multicast clients
The maximum ttl value provided so far by the multicast clients
will be chosen and reported in the response to the current
client request.
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-02-23 14:34:32 +0100 Patricia Muscalu <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/client.c:
rtsp-stream: Don't require address pool in the transport specific case
If "transport.client-settings" parameter is set to true, the client is
allowed to specify destination, ports and ttl.
There is no need for pre-configured address pool.
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-24 09:35:46 +0200 Patricia Muscalu <[email protected]>
* 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:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/mediafactory.c:
Add new API for setting/getting maximum multicast ttl value
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-31 21:17:41 +0200 Mathieu Duponchelle <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: avoid duplicating the first multicast client
In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
clients were dynamically added and removed to the multicast
udp sinks, as such we should no longer add a first client in
set_multicast_socket_for_udpsink
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-06 15:33:04 -0400 Thibault Saunier <[email protected]>
* gst/rtsp-server/Makefile.am:
rtsp-server: Add gstreamer-base gir dir in autotools
2018-07-25 19:54:55 +0200 Mathieu Duponchelle <[email protected]>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-client: always allocate both IPV4 and IPV6 sockets
multiudpsink does not support setting the socket* properties
after it has started, which meant that rtsp-server could no
longer serve on both IPV4 and IPV6 sockets since the patches
from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
merged.
When first connecting an IPV6 client then an IPV4 client,
multiudpsink fell back to using the IPV6 socket.
When first connecting an IPV4 client, then an IPV6 client,
multiudpsink errored out, released the IPV4 socket, then
crashed when trying to send a message on NULL nevertheless,
that is however a separate issue.
This could probably be fixed by handling the setting of
sockets in multiudpsink after it has started, that will
however be a much more significant effort.
For now, this commit simply partially reverts the behaviour
of rtsp-stream: it will continue to only create the udpsinks
when needed, as was the case since the patches were merged,
it will however when creating them, always allocate both
sockets and set them on the sink before it starts, as was
the case prior to the patches.
Transport configuration will only error out if the allocation
of UDP sockets fails for the actual client's family, this
also downgrades the GST_ERRORs in alloc_ports_one_family
to GST_WARNINGs, as failing to allocate is no longer
necessarily fatal.
https://bugzilla.gnome.org/show_bug.cgi?id=796875
2018-07-25 17:22:20 +0530 Nirbheek Chauhan <[email protected]>
* meson.build:
* meson_options.txt:
meson: Convert common options to feature options
These are necessary for gst-build to set options correctly. The
remaining automagic option is cgroup support in examples.
https://bugzilla.gnome.org/show_bug.cgi?id=795107
2018-07-23 18:03:51 +0300 Sebastian Dröge <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Slightly simplify locking
2018-06-28 11:22:21 +0200 David Svensson Fors <[email protected]>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
Limit queued TCP data messages to one per stream
Before, the watch backlog size in GstRTSPClient was changed
dynamically between unlimited and a fixed size, trying to avoid both
unlimited memory usage and deadlocks while waiting for place in the
queue. (Some of the deadlocks were described in a long comment in
handle_request().)
In the previous commit, we changed to a fixed backlog size of 100.
This is possible, because we now handle RTP/RTCP data messages differently
from RTSP request/response messages.
The data messages are messages tunneled over TCP. We allow at most one
queued data message per stream in GstRTSPClient at a time, and
successfully sent data messages are acked by sending a "message-sent"
callback from the GstStreamTransport. Until that ack comes, the
GstRTSPStream does not call pull_sample() on its appsink, and
therefore the streaming thread in the pipeline will not be blocked
inside GstRTSPClient, waiting for a place in the queue.
pull_sample() is called when we have both an ack and a "new-sample"
signal from the appsink. Then, we know there is a buffer to write.
RTSP request/response messages are not acked in the same way as data
messages. The rest of the 100 places in the queue are used for
them. If the queue becomes full of request/response messages, we
return an error and close the connection to the client.
Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
2018-06-28 11:22:13 +0200 David Svensson Fors <[email protected]>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Use fixed backlog size
Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
Preparation for the next commit, which changes to a different way of
avoiding both deadlocks and unlimited memory usage with the watch
backlog.
2018-07-16 21:57:08 +0200 Carlos Rafael Giani <[email protected]>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: unref clock (if set) when finalizing
https://bugzilla.gnome.org/show_bug.cgi?id=796814
2018-07-16 21:56:44 +0200 Carlos Rafael Giani <[email protected]>
* docs/libs/gst-rtsp-server-sections.txt:
rtsp-media: add gst_rtsp_media_*_set_clock to docs
https://bugzilla.gnome.org/show_bug.cgi?id=796814
2018-07-12 19:01:54 +0100 Tim-Philipp Müller <[email protected]>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: unref old clock when setting new clock
https://bugzilla.gnome.org/show_bug.cgi?id=796724
2018-06-29 15:20:57 -0700 Brendan Shanks <[email protected]>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: unref clock in finalize
https://bugzilla.gnome.org/show_bug.cgi?id=796724
2018-07-12 18:57:21 +0100 Tim-Philipp Müller <[email protected]>
* gst/rtsp-server/rtsp-onvif-media.c:
rtsp-onvif-media: fix g-ir-scanner warnings
2018-07-10 23:56:23 +0100 Tim-Philipp Müller <[email protected]>
* .gitignore:
.gitignore: add another example binary
2018-07-10 23:55:20 +0100 Tim-Philipp Müller <[email protected]>
* examples/meson.build:
meson: add new test-appsrc2 example to meson build
2018-07-10 23:53:41 +0100 Tim-Philipp Müller <[email protected]>
* examples/Makefile.am:
examples: fix build of new test-appsrc2 example
Need to link against libgstapp-1.0.
2018-07-11 01:25:51 +1000 Jan Schmidt <[email protected]>
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-appsrc2.c:
examples: Add test-appsrc2
Add an example of feeding both audio and video into an RTSP
pipeline via appsrc.
2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <[email protected]>
* gst/rtsp-server/rtsp-client.c:
client: Strip transport parts as whitespaces could be around commas
https://bugzilla.gnome.org/show_bug.cgi?id=758428
2018-06-27 08:30:42 +0200 Göran Jönsson <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
Fix race when setting up source elements.
Since we set the source element(s) to PLAYING state before hooking
them up to the downstream funnel, it's possible for the source element
to receive packets before we actually get to linking it to the funnel,
in which case buffers would be pushed out on an unlinked pad, causing
it to error out and stop receiving more data.
We fix this by blocking the source's srcpad until we have linked it.
https://bugzilla.gnome.org/show_bug.cgi?id=796160
2018-03-21 10:56:51 +0100 Ognyan Tonchev <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Fix mismatch between allowed and configured protocols
https://bugzilla.gnome.org/show_bug.cgi?id=796679
2017-02-01 09:44:50 +0100 Ulf Olsson <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Emit a signal when the SRTP decoder is created
https://bugzilla.gnome.org/show_bug.cgi?id=778080
2018-03-13 11:10:35 +0100 Patricia Muscalu <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Don't require presence of sinks in _get_*_socket()
Transport specific sink elements are added to the pipeline
in PLAY request and sockets are already created in SETUP so
it's actually wrong to require the presence of sinks in
_get_*_socket() functions.
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-02-14 10:41:02 +0100 Patricia Muscalu <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Update transport for multicast clients as well
If a multicast client requests different transport settings
than the existing one make sure that this new transport
configuruation is propagated to the multicast udp sink.
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-02-13 11:04:36 +0100 Patricia Muscalu <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
And not on unicast udp sinks
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-06-24 12:44:26 +0200 Tim-Philipp Müller <[email protected]>
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-thread-pool.c:
Update for g_type_class_add_private() deprecation in recent GLib
2018-06-24 12:45:49 +0200 Tim-Philipp Müller <[email protected]>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-stream.c:
Fix indentation
2018-06-22 23:17:08 +1000 Jan Schmidt <[email protected]>
* examples/Makefile.am:
* examples/test-video-disconnect.c:
examples: Add test-video-disconnect example
Simple example which cuts off all clients 10 seconds
after the first one connects.
2018-06-20 04:37:11 +0200 Mathieu Duponchelle <[email protected]>
* docs/libs/gst-rtsp-server-sections.txt:
* examples/test-auth-digest.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
rtsp-auth: Add support for parsing .htdigest files
Passwords are usually not stored in clear text, but instead
stored already hashed in a .htdigest file.
Add support for parsing such files, add API to allow setting
a custom realm in RTSPAuth, and update the digest example.
https://bugzilla.gnome.org/show_bug.cgi?id=796637
2018-06-19 14:53:02 +1000 Matthew Waters <[email protected]>
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
rtspclientsink: fix waiting for multiple streams
We were previously only ever waiting for a single stream to notify it's
blocked status through GstRTSPStreamBlocking. Actually count streams to
wait for.
Fixes rtspclientsink sending SDP's without out some of the input
streams.
https://bugzilla.gnome.org/show_bug.cgi?id=796624
2018-06-20 04:30:04 +0200 Mathieu Duponchelle <[email protected]>
* docs/libs/gst-rtsp-server-sections.txt:
docs: add missing auth methods
2018-06-20 00:10:18 +0200 Mathieu Duponchelle <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: only create funnel if it didn't exist already.
This precented using multiple protocols for the same stream.
https://bugzilla.gnome.org/show_bug.cgi?id=796634
2018-06-20 01:35:47 +0200 Mathieu Duponchelle <[email protected]>
* examples/meson.build:
meson: build auth-digest example
2018-06-05 08:44:44 +0200 Patricia Muscalu <[email protected]>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream-transport.c:
Get payloader stats only for the sending streams
Get/set payloader properties only for streams that actually
contain a payloader element.
https://bugzilla.gnome.org/show_bug.cgi?id=796523
2018-05-18 14:53:49 +0200 Edward Hervey <[email protected]>
* gst/rtsp-server/Makefile.am:
Makefile: Don't hardcode libtool for g-i build
Similar to the other commits in core/base/bad
2018-05-08 14:13:31 +0200 Johan Bjäreholt <[email protected]>
* gst/rtsp-server/rtsp-onvif-media-factory.h:
rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
https://bugzilla.gnome.org/show_bug.cgi?id=796229
2018-05-09 04:09:02 +1000 Jan Schmidt <[email protected]>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Don't deadlock in preroll on early close
If the connection is closed very early, the flushing
marker might not get set and rtspclientsink can get
deadlocked waiting for preroll forever.
https://bugzilla.gnome.org/show_bug.cgi?id=786961
2018-05-05 19:51:52 +0530 Nirbheek Chauhan <[email protected]>
* meson.build:
* meson_options.txt:
meson: Update option names to omit disable_ and with- prefixes
Also yield common options to the outer project (gst-build in our case)
so that they don't have to be set manually.
2018-04-25 11:00:32 +0100 Tim-Philipp Müller <[email protected]>
* meson.build:
meson: use -Wl,-Bsymbolic-functions where supported
Just like the autotools build.
2018-04-22 20:09:01 +0100 Tim-Philipp Müller <[email protected]>
* configure.ac:
* tests/check/Makefile.am:
configure: check for -good and -bad plugins only in uninstalled setup
Avoids confusing configure messages looking or a -good .pc file
that doesn't exist.
Also use plugindir variables that common macros set while at it.
https://bugzilla.gnome.org/show_bug.cgi?id=795466
2018-04-17 11:03:11 +0200 Joakim Johansson <[email protected]>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Fix session timeout
When streaming data over TCP then is not the keep-alive
functionality working.
The reason is that the function do_send_data have changed
to boolean but the code is still checking the received result
from send_func with GST_RTSP_OK.
The result is that a successful send_func will always lead to
that do_send_data is returning false and the keep-alive will
not be updated.
https://bugzilla.gnome.org/show_bug.cgi?id=795321
2018-04-02 22:49:35 +0200 Mathieu Duponchelle <[email protected]>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
Implement support for ULP Forward Error Correction
In this initial commit, interface is only exposed for RECORD,
further work will be needed in rtspsrc to support this for
PLAY.
https://bugzilla.gnome.org/show_bug.cgi?id=794911
2018-04-17 17:47:30 +0300 Sebastian Dröge <[email protected]>
* gst/rtsp-server/rtsp-onvif-media.c:
Revert "rtsp-server: Switch around sendonly/recvonly attributes"
This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
the opposite, just like the ONVIF standard.
Let's follow those RFCs as we're doing RTSP here, and add a property at
a later time if needed to switch to the SDP RFC behaviour.
https://bugzilla.gnome.org/show_bug.cgi?id=793964
2018-04-16 10:53:52 +0100 Tim-Philipp Müller <[email protected]>
* common:
Automatic update of common submodule
From 3fa2c9e to ed78bee
2018-04-04 10:06:06 +0300 Sebastian Dröge <[email protected]>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/rtspclientsink.c:
gst: Run everything through gst-indent again
2018-04-03 08:57:47 +0200 Branko Subasic <[email protected]>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
rtsp-media: query the position on active streams if media is complete
If the media is complete, i.e. one or more streams have been configured
with sinks, then we want to query the position on those streams only.
A query on an incomplete stream may return a position that originates from
an earlier preroll.
https://bugzilla.gnome.org/show_bug.cgi?id=794964
2018-04-02 12:35:04 +0100 Tim-Philipp Müller <[email protected]>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: make sure not to use freed string
Set transport string to NULL after freeing it, so that
at worst we get a NULL pointer if constructing a new
transport string fails (which shouldn't really fail here).
Also check return value of that, just in case.
CID 1433768.
2018-03-30 23:34:01 +0200 Mathieu Duponchelle <[email protected]>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: do not free string passed to take_header
2018-03-30 23:10:10 +0200 Mathieu Duponchelle <[email protected]>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: do not take lock in request_aux_receiver
Added it right before pushing the previous commit, it is
incorrect and deadlocks because this function gets called
from the join_bin thread, which already holds the lock,
that's the reason why request_aux_sender didn't take the
lock either.
2018-03-29 22:49:26 +0200 Mathieu Duponchelle <[email protected]>
* 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-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-server: add API to enable retransmission requests
"do-retransmission" was previously set when rtx-time != 0,
which made no sense as do-retransmission is used to enable
the sending of retransmission requests, where as rtx-time
is used by the peer to enable storing of buffers in order
to respond to retransmission requests.
rtsp-media now also provides a callback for the
request-aux-receiver signal.
https://bugzilla.gnome.org/show_bug.cgi?id=794822
2018-03-29 16:18:42 +0200 Mathieu Duponchelle <[email protected]>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: add rtx ssrc to mikey's crypto sessions
https://bugzilla.gnome.org/show_bug.cgi?id=794813
2018-03-29 16:15:45 +0200 Mathieu Duponchelle <[email protected]>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
This in order to be able to decrypt the RTCP backchannel
https://bugzilla.gnome.org/show_bug.cgi?id=794813
2018-03-29 16:12:26 +0200 Mathieu Duponchelle <[email protected]>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Send KeyMgmt header in ANNOUNCE response
When sending back an encrypted RTCP back channel, it is useful
for the client to know the encryption key.
https://bugzilla.gnome.org/show_bug.cgi?id=794813
2018-03-29 16:06:31 +0200 Mathieu Duponchelle <[email protected]>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: extract handle_keymgmt from rtsp-client
rtspclientsink will also need to parse KeyMgmt headers
sent by the server to decrypt the RTCP backchannel stream
https://bugzilla.gnome.org/show_bug.cgi?id=794813
2018-03-29 02:51:02 +0200 Mathieu Duponchelle <[email protected]>
* gst/rtsp-sink/gstrtspclientsink.c:
* tests/check/gst/rtspclientsink.c:
rtspclientsink: Fix client ports for the RTCP backchannel