To prevent cases with prerolling when the inactive stream prerolls first
and the server proceeds without waiting for the active stream, we will
ignore GstRTSPStreamBlocking messages from incomplete streams. When
there are no complete streams (during DESCRIBE), we will listen to all
streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
Add another seek_one_active_stream test but with a demuxer. The demuxer
will flush both streams in opposed to the existing test which only
flushes the active stream. This will help exposing problems with the
prerolling process after a flushing seek.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
When play a media with both sender and receiver stream, like ONVIF
back channel audio in, gst_rtsp_media_get_rates call
gst_rtsp_stream_get_rates for each stream to set the rates. But
gst_rtsp_stream_get_rates return false for the receiver steam, which
lead a g_assert crash.
Instead to get rates on all streams, now just get rates on sender
streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/150>
ulpfec correction is obviously useless when receiving a stream
over TCP, and in TCP modes the rtp storage receives non
timestamped buffers, causing it to queue buffers indefinitely,
until the queue grows so large that sanity checks kick in and
warnings start to get emitted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/149>
rtsp_media_unsuspend() is called from handle_play_request()
before sending the play response. Unblocking the streams here
was causing data to be sent out before the client was ready
to handle it, with obvious side effects such as initial packets
getting discarded, causing decoding errors.
Instead we can simply let the media streams be unblocked when
the state of the media is set to PLAYING, which occurs after
sending the play response.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
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>
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>
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>
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 a7732a68e8
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>
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>
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>
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>
