After sending or retrieving data, gstrtspconnection resets the socket's
timeout to 0 (infinite). This could cause problems if sending and
receiving at the same time. For example, if RTCP data is sent from the
streaming thread while gstrtspsrc is already retrieving data.
With this patch, timeout is only reset to 0 if there is no other
thread using the socket.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1260>
When connecting to an RTSP server in tunnled mode (HTTP) the server
usually replies with a x-server header. This contains the address
of the intended streaming server. However some servers return an
"invalid" address. Here follows two examples when it might happen.
1. A server use Apache combined with a separate RTSP process to handle
Https request on port 443. In this case Apache handle TLS and
connects to the local RTSP server, which results in a local
address 127.0.0.1 or ::1 in the x-server reply. This address is
returned to the actual RTSP client in the x-server header.
The client will receive this address and try to connect to it
and fail.
2. The client use a ipv6 link local address with a specified scope id
fe80::aaaa:bbbb:cccc:dddd%eth0 and connects via Http on port 80.
The RTSP server receives the connection and returns the address
in the x-server header. The client will receive this address and
try to connect to it "as is" without the scope id and fail.
In the case of streaming data from RTSP servers like 1. and 2. it's
useful to have the option to simply ignore the x-server header reply
and continue using the original address.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1192>
Requires OpenGL 4.4 or EXT_buffer_storage
Current mesa exposes GL_ARB_buffer_storage when retrieving the relevant
functions returns no-ops and causes failures.
Improves throughput of uploads by roughly 30%-60% and download throughput by
roughly 10-30% across depending on the exact scenario and hardware.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1191>
There is currently no way for users to receive incoming events from
appsink while keeping them properly serialized with the buffers flow.
This can be especially useful when application is injecting custom
downstream events into the pipeline and needs to know when they reached
appsink.
Solving this by adding a new signal notifying about new incoming events
and a set of action signals and method to pull those events.
The API is actually pulling the samples and events all together as they
are actually fetched from the same queue.
Having a specific API to pull only events would have the side effect of
discarding samples (and pulling samples would discard events) making
this API not convenient for users.
Partially fix#247
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1046>
The problem is that EGLNativeWindowSurface and wl_egl_surface are the
same object underneath, so we must recreate both together. As an
optimization, the EGLNativeWindowSurface wrapper is only re-created
if the window_handle changed.
On Mesa, this would cause crash, which will be fixed by:
https://gitlab.freedesktop.org/mesa/mesa/-/merge_requests/11979
And will lead to proper errors in the future or on other GL stack. This
issue was encounter using a permanent GstGLDisplay after cycling one of
multiple independent pipelines through NULL state.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1230>
When the window is called, we properly destroy all surfaces, which effectively
will unmap that surface and should make it disapear on screen, but we also
destroy the wl_source, a GSource that is resposibble of dispatching and executing
messages to/from the Wayland server.
As a side effect, the server never gets the message and the surfaces are
"leaked" on the server. We fix this using wl_display_flush() before destroying
the wl_source.
Fixes#815
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1226>
Previously one of the branches did not check for the property value. To
avoid this in the future, check inside the QoS calculation function
instead.
As a side effect this now always prints the debug messages into the logs
when samples are dropped, which is useful information even without the
QoS messages.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1224>
If a buffer is dropped during resyncing on a discont because either its
end offset is already before the current output offset of the
aggregator or because it fully overlaps with the part of the current
output buffer that was already filled, then don't just assume that the
next buffer is going to start at exactly the expected offset. It might
still require some more dropping of samples.
This caused the input to be mixed with an offset to its actual position
in the output stream, causing additional latency and wrong
synchronization between the different input streams.
Instead consider each buffer after a discont as a discont until the
aggregator actually resynced and starts mixing samples from the input
again.
Also update the start output offset of a new input buffer if samples
have to be dropped at the beginning. Otherwise it might be mixed too
early into the output and overwrite part of the output buffer that
already took samples from this input into account.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/912
which is a regression introduced by https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1180/
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1224>
A video decoder can now receive subframes and start decoding
instead of waiting for the full frame to be complete.
Subframe support will reduce latency as described in the
video encoder base class.
A unit test illustrating this API is available in
tests/check/libs/videodecoder.c.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/454>
GstRTPHeaderExtension::write can map the RTP buffer for reading. If that
happens on a buffer that is already mapped WRITE-only by the payloader,
the payloader's mapping gets invalidated (GstRTPBuffer::map will point
to a different instance of GstMemory).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1173>
The finish() virtual function documentation state that "Sub-classes can refuse
to decode new data after." Though, it is very common to issue a non-flushing
seek after that event in gapless playback uses case. This fixes potential
stalls with code using segment seeks, by using drain() virtual funciton
instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1206>
