It is entirely possible for the cancellable to be cancelled (and freed)
in gst_rtsp_connection_flush() while there may be an ongoing read/write
operation.
Nothing prevents gst_rtsp_connection_flush() from waiting for the
outstanding read/writes.
This could lead to a crash like (where cancellable has been freed
within gst_rtsp_connection_flush()):
#0 0x00007ffff4351096 in g_output_stream_writev (stream=stream@entry=0x7fff30002950, vectors=vectors@entry=0x7ffe2c6afa80, n_vectors=n_vectors@entry=3, bytes_written=bytes_written@entry=0x7ffe2c6af950, cancellable=cancellable@entry=0x7fff300288a0, error=error@entry=0x7ffe2c6af958) at ../subprojects/glib/gio/goutputstream.c:377
#1 0x00007ffff44b2c38 in writev_bytes (stream=0x7fff30002950, vectors=vectors@entry=0x7ffe2c6afa80, n_vectors=n_vectors@entry=3, bytes_written=bytes_written@entry=0x7ffe2c6afb90, block=block@entry=1, cancellable=0x7fff300288a0) at ../subprojects/gst-plugins-base/gst-libs/gst/rtsp/gstrtspconnection.c:1320
#2 0x00007ffff44b583e in gst_rtsp_connection_send_messages_usec (conn=0x7fff30001370, messages=messages@entry=0x7ffe2c6afcc0, n_messages=n_messages@entry=1, timeout=timeout@entry=3000000) at ../subprojects/gst-plugins-base/gst-libs/gst/rtsp/gstrtspconnection.c:2056
#3 0x00007ffff44d2669 in gst_rtsp_client_sink_connection_send_messages (sink=0x7fffac0192c0, timeout=3000000, n_messages=1, messages=0x7ffe2c6afcc0, conninfo=0x7fffac019610) at ../subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c:1929
#4 gst_rtsp_client_sink_try_send (sink=sink@entry=0x7fffac0192c0, conninfo=conninfo@entry=0x7fffac019610, requests=requests@entry=0x7ffe2c6afcc0, n_requests=n_requests@entry=1, response=response@entry=0x0, code=code@entry=0x0) at ../subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c:2845
#5 0x00007ffff44d3077 in do_send_data (buffer=0x7fff38075c60, channel=<optimized out>, context=0x7fffac042640) at ../subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c:3896
#6 0x00007ffff4281cc6 in gst_rtsp_stream_transport_send_rtp (trans=trans@entry=0x7fff20061f80, buffer=<optimized out>) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream-transport.c:632
#7 0x00007ffff4278e9b in push_data (stream=0x7fff40019bf0, is_rtp=<optimized out>, buffer_list=0x0, buffer=<optimized out>, trans=0x7fff20061f80) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c:2586
#8 check_transport_backlog (stream=0x7fff40019bf0, trans=0x7fff20061f80) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c:2645
#9 0x00007ffff42793b3 in send_tcp_message (idx=<optimized out>, stream=0x7fff40019bf0) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c:2741
#10 send_func (stream=0x7fff40019bf0) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c:2776
#11 0x00007ffff7d59fad in g_thread_proxy (data=0x7fffbc062920) at ../subprojects/glib/glib/gthread.c:827
#12 0x00007ffff7a8ce2d in start_thread () from /lib64/libc.so.6
#13 0x00007ffff7b12620 in clone3 () from /lib64/libc.so.6
Fix by adding a cancellable lock and returning an extra reference used
across all read/write operations. gst_rtsp_connection_flush() can free
the in-use cancellable and it will no longer affect any in progress
read/write.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2799>
1.21.0.1 should not satisfy a check for 1.22.0.
If someone needs more control they should do a feature check for
the symbol in the headers or lib.
Based on a similar patch by Tim-Philipp Müller for libnice.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2501>
The caps negotiation should respect the selected method to the test pipeline below works properly.
gst-launch-1.0 videotestsrc ! video/x-raw,width=320,height=600 ! videoflip method=clockwise ! video/x-raw,width=600,height=320 ! fakesink
Signed-off-by: Adrian Fiergolski <adrian.fiergolski@fastree3d.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2803>
In the trick mode, driver may queue a valid buffer follow by an
empty buffer which has no valid data to indicate EOS.For the empty
buffer whose memory is multi-plane, need to resize it before
unreference it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2731>
Depending on device feature level, d3d11 runtime can support
ID3D11Fence which is equivalent to ID3D12Fence.
Waiting using fence has performance-wise benefit over pulling
ID3D11Query status. If ID3D11Fence is not supported by device,
then ID3D11Query will be used instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2790>
4x downscaling of chroma with co-sited chroma has never worked
it seems.
Fixes incorrect videotestsrc output and videoconvert conversions
to Y41B, YUV9, YVU9 and IYU9 with co-sited chroma.
e.g.
gst-launch-1.0 videotestsrc ! video/x-raw,format=Y41B,width=1280,height=720 ! \
videoconvert ! autovideosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2789>
It may happens that bitstream doesn't provided SPS in decoding order
(like in VPSSPSPPS_A_MainConcept_1 conformance test file).
To be sure that the decoder got the correct SPS parameters process
SPS just before start decoding the frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2575>
While possible defer computataion of pps and sps fields until
slice parsing since it may happens that bitstreams don't encoded
them in expected order.
A example weird ordered bitstreams is VPSSPSPPS_A_MainConcept_1
conformance test.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2575>
The function g_array_sized_new() leaves the len to 0, but the slice
implementation assumes it would be set to 4. Sending multiple slices is
not yet support for H.264 as no driver needed it yet, but if that code
was to be used it would have overflowed as the array would never grow as
multiple 0 by 2 always results in 0.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1079>
And also don't assert that there are no buffers queued up when handling
an EOS event. The pad's streaming thread might've already received a new
stream-start event and queued up a buffer in the meantime.
This still leaves a race condition where the srcpad task sees all pads
in EOS state and finishes the stream, while shortly afterwards a pad
might receive a stream-start event again, but this doesn't seem to be
solveable with the current aggregator design.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2769>
SMPTE 170M and 240M use the same RGB and white point coordinates
and therefore both primaries can be considered functionally
equivalent.
Also, some transfer functions have different name but equal
gamma functions. Adding another colorimetry compare function
to deal with thoes cases at once
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2765>
In file included from ../gstreamer1.0-plugins-good/1.20.3-r0/recipe-sysroot/usr/include/gstreamer-1.0/gst/gl/gstglfuncs.h:87,
from ../gst-plugins-good-1.20.3/ext/qt/qtglrenderer.cc:14:
../gstreamer1.0-plugins-good/1.20.3-r0/recipe-sysroot/usr/include/gstreamer-1.0/gst/gl/glprototypes/gstgl_compat.h:40:18: error: conflicting declaration 'typedef void* GLsync'
40 | typedef gpointer GLsync;
| ^~~~~~
In file included from ../gstreamer1.0-plugins-good/1.20.3-r0/recipe-sysroot/usr/include/QtGui/qopengl.h:127,
from ../gstreamer1.0-plugins-good/1.20.3-r0/recipe-sysroot/usr/include/QtQuick/qsggeometry.h:44,
from ../gstreamer1.0-plugins-good/1.20.3-r0/recipe-sysroot/usr/include/QtQuick/qsgnode.h:43,
from ../gstreamer1.0-plugins-good/1.20.3-r0/recipe-sysroot/usr/include/QtQuick/qsgrendererinterface.h:43,
from ../gstreamer1.0-plugins-good/1.20.3-r0/recipe-sysroot/usr/include/QtQuick/qquickwindow.h:44,
from ../gstreamer1.0-plugins-good/1.20.3-r0/recipe-sysroot/usr/include/QtQuick/QQuickWindow:1,
from ../gst-plugins-good-1.20.3/ext/qt/qtglrenderer.cc:6:
../gstreamer1.0-plugins-good/1.20.3-r0/recipe-sysroot/usr/include/QtGui/qopengles2ext.h:24:26: note: previous declaration as 'typedef struct __GLsync* GLsync'
24 | typedef struct __GLsync *GLsync;
| ^~~~~~
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2763>
These patches are taken from upstream, and they fix compile failures
with latest clang. These can be dropped when upgrading these wraps.
This is currently causing a warning because we do not require the
version of meson that ships with this feature: 0.63.0. The version has
not been bumped because older Meson versions gracefully ignore the
wrap field, this fix is optional and only needed on macOS, and 0.63.0
is a very new release with a bug that partially breaks this feature:
https://github.com/mesonbuild/meson/pull/10602
We can consider bumping the requirement once 0.63.1 is released.
Also switch from git to tarballs, no reason to use git here anymore.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2761>
We should move this functionality to gst-libs so that GstD3D11Converter
can be moved to gst-libs.
Another advantage is that applications can call our
HLSL compiler wrapper method without any worry about OS version
dependent system installed HLSL library.
Note that there are multiple HLSL compiler library versions
on Windows and system installed one would be OS version dependent.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2760>
We need GStreamer elements to do the bandwidth estimation as this way
they can also control the pacing of the transmission flow as specified
in the [GCC] algorithm for example.
Bandwidth estimator element are placed right before the "RTPSession" as
an "rtp-aux-sender" element. This way they can use the "Transport-wide
Congestion Control" RTCP feedback messages through the "RTPTwcc" custom
events that are sent by the rtpsession.
Applications are responsible to react to the bandwidth estimator element
and set the encoder target bitrate etc... which means that we can not
pass an estimator as an element factory, so a signal as been chosen
instead.
[GCC]: https://datatracker.ietf.org/doc/html/draft-ietf-rmcat-gcc-02
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2562>
Raw memory upload should always be the least preferred input
caps, only added by the raw memory uploader as the last thing
in the caps.
Caps negotiation should still choose raw data when it needs to,
and other upload methods that can accept raw data buffers will still do so.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2725>
The AVClass name of Animated PNG in FFmpeg 5.x is "(A)PNG"
and it will be converted to "-a-png" through
g_ascii_strdown() and g_strcanon(). But GLib disallow leading '-'
character for a GType name. Strip leading '-' to workaround it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2724>
There might be a sequence of event and buffer flow:
- Got stream-start/caps/segment events
- Got flush events
- And then buffers with a new segment event
In the above case, stream-start and caps event might not be reached to
peer proxysrc if peer proxysrc is not ready to receive them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1552>
gst_video_convert_scale_get_fixed_format() receives 'othercaps' from
basetransforms' fixate_caps() vmethod which explicitly mentions that
'`othercaps` may not be writable'.
The gst_caps_intersect() call just before may or may not produce new
caps. Particularly in cases like EMPTY or ANY caps on either of the
inputs, only a ref is taken and returned to the caller.
As a result, gst_video_convert_scale_fixate_format() may have attempted
to modify a non-writable caps structure.
Fix by adding a gst_caps_make_writable().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2709>
- ssl module requires an explicit TLS_SERVER role
- asyncio throws a deprecation warning when using
asyncio.get_event_loop(). Remove custom event loop handling entirely
- No need to keep the websocket server in a member variable, can use
a future to signal exit case along with the async with context manager
of websockets.serve()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2698>
