Since commit 4d86f994, when setting an RTSP media both shared and
reusable, streaming cannot be restarted after the first time all the
clients disconnect. That happens because the sockets (unlike
addresses) of GstRTSPStream are not cleared in
gst_rtsp_stream_leave_bin, and on restart sockets and addresses are
not allocated in gst_rtsp_stream_allocate_udp_sockets, and then the
check in create_sender_part fails. Fix this by clearing sockets in
gst_rtsp_stream_leave_bin.
Fixesgstreamer/gst-rtsp-server#113
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6334>
While the suspend modes NONE and PAUSED provided a low startup latency
for connecting clients they did not ensure that streams started on
fresh data.
With this property we can maintain the low startup latency of those
suspend modes while also ensuring that a stream starts on a key unit.
Furthermore, by modifying the value of a new property,
ensure-keyunit-on-start-timeout, it is possible to accept a keyunit of
a certain age but discard it if too much time has passed and instead
force a new keyunit.
Fixes#2443
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4334>
Previously it was possible that a shared media was just in the process
of being unprepared because the last client disappeared, while another
client retrieved it from the cache and then tried to use it. Unless the
media was reusable this would've then failed unnecessarily.
To avoid this it is necessary to lock the media directly in
gst_rtsp_media_factory_construct() and return a locked media. After
locking the cached media it is necessary to check if the media was ever
unprepared or is actually reusable and based on that either reuse it or
create a new media.
This minimally changes the gst_rtsp_media_factory_construct() API to
always return a locked media, and adds a new
gst_rtsp_media_can_be_shared() function to check if a media can actually
be shared in practice.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
The idea is to give the application the possibility to adjust the error
code when responding to a request. For that purpose the pipeline's bus
messages are emitted to subscribers through a signal handle-message.
The subscribers can then check those messages for errors and adjust
the response error code by overriding the virtual method
adjust_error_code().
Fixes#1294
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2972>
Check back pressure of a stream transport before popping buffer from its backlog.
If the stream transport is not experiencing back pressure, the buffer can be popped from backlog and pushed to client.
Fixes:#1298
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2936>
The address/port is pre-defined by the caller of the function, so
retrying is only going to loop forever.
Ideally the multicast address should be checked after allocating but
this doesn't happen currently, so it's better to error out cleanly then
to loop forever trying the same address.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2975>
The documentation for several gst_*_writable_structure functions stated
that they would never return NULL, without making clear that the passed
object is required to be writable. This changes the wording in those
cases to make that requirement more clear.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1784>
When tunneling over HTTP, if connection on the second channel happens
before the control timer is created we may trigger an assert in
rtsp_ctrl_timeout_remove(). Avoid that by taking the priv->lock before
attaching the client thread to the context.
Fixes#1025
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1867>
Add gst_dep to gst_rtsp_server_deps, in the context of buildroot, this
will avoid the following build failure, because the correct girdir
location will be retrieved from gstreamer-1.0.pc:
/home/giuliobenetti/autobuild/run/instance-3/output-1/host/riscv32-buildroot-linux-gnu/sysroot/usr/bin/g-ir-compiler gst/rtsp-server/GstRtspServer-1.0.gir --output gst/rtsp-server/GstRtspServer-1.0.typelib --includedir=/usr/share/gir-1.0
Could not find GIR file 'Gst-1.0.gir'; check XDG_DATA_DIRS or use --includedir
error parsing file gst/rtsp-server/GstRtspServer-1.0.gir: Failed to parse included gir Gst-1.0
If the above error message is about missing .so libraries, then setting up GIR_EXTRA_LIBS_PATH in the .mk file should help.
Typically like this: PKG_MAKE_ENV += GIR_EXTRA_LIBS_PATH="$(@D)/.libs"
Fixes:
- http://autobuild.buildroot.org/results/04af6b22cfa0cffb6a3109a3b32b27137ad2e0b0
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1460>
Prior to this patch, we considered that a stream was blocking
whenever a pad probe was triggered for either the RTP pad or
the RTCP pad.
This led to situations where we subsequently unblocked and expected
to find a segment on the RTP pad, which was racy.
Instead, we now only consider that the stream is blocking when
the pad probe for the RTP pad has triggered with a blockable object
(buffer, buffer list, gap event).
The RTCP pad is simply blocked without affecting the state of the
stream otherwise.
Fixes#929
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1452>
Introduces a `libraries` variable that contains all libraries in a
list with the following format:
``` meson
libraries = [
[pkg_name, {
'lib': library_object
'gir': [ {full gir definition in a dict } ]
],
....
]
```
It therefore refactors the way we build the gir so that we can reuse the
same information to build them against 'gstreamer-full' in gst-build
when linking statically
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
Previously suspended medias immediately reached the UNPREPARED state
without going through the media's unprepare() vfunc. This didn't allow
the media subclass to do any additional cleanup, and for example the
shutdown-eos property of GstRTSPMedia was ignored.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1090>
handle_*_request() functions were all retrieving the session media from
the session by calling gst_rtsp_session_get_media () which is a transfer-none
call. If a session timeout happens at that time, the session media may get freed
making the pointer invalid..
Fixes#757
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1053>
