When a new segment event arrives, it immediately updates
the current stored segment, which was used for calculating
the running time of the current text buffer for every
passing video frame. This means a segment that arrives
after the text buffer might get used to (mis)calculate
the running times subsequently.
Instead, calculate and store the right running time
using the current segment when storing the buffer. Later
the stored segment can get freely updated.
This fixes the case where pieces of video and text streams
are seamlessly concatenated and fed through the text overlay.
Previously, it could lead to the current text buffer suddenly
have a massive running time and blocking all further input.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2802>
This can be important for instance when a container holds multiple
tracks with the same media type, with no indication (eg tags) of
which track is the default one.
In that case, players usually pick the first track by default.
This is especially useful when using smart editing with GES, as
it will result in the same ordering as the input file that was
used as a template.
For reference, this yields the same order as ffprobe.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1549>
when creating a profile from a discoverer info.
There is no justification for the existing code, and talking with
Thibault he cannot remember why the sort was in place.
On the other hand, this allows GES users to not have to implement
a callback for the select-tracks-for-object callback when using
it to trim a single clip, which the output profile was built from:
track elements will be placed in the appropriate track by default,
that is the one that will be connected to the matching profile.
For multi-clip timelines, the situation doesn't change, users will
still have to implement a callback and do the leg work of placing
track elements (if any) in a matching track (if any).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1549>
chroma-format, bit-depth-chroma, bit-depth-luma are all informative
fields set by the H265 and H265 parser upon receiving an SPS.
They shouldn't be constrained downstream of the parser, instead
if a user wants those to ultimately match certain values they
should do so by constraining a profile.
In this case however, we also always remove the profile constraint
in order to let encoders pick a suitable one as a function of the
raw input video format and their own capabilities.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1549>
With the 2.72 release, glib-networking developers have decided that
TLS certificate validation cannot be implemented correctly by them, so
they've deprecated it.
In a nutshell: a cert can have several validation errors, but there
are no guarantees that the TLS backend will return all those errors,
and things are made even more complicated by the fact that the list of
errors might refer to certs that are added for backwards-compat and
won't actually be used by the TLS library.
Our best option is to ignore the deprecation and pass the warning onto
users so they can make an appropriate security decision regarding
this.
We can't deprecate the tls-validation-flags property because it is
very useful when connecting to RTSP cameras that will never get
updates to fix certificate errors.
Relevant upstream merge requests / issues:
https://gitlab.gnome.org/GNOME/glib/-/merge_requests/2214https://gitlab.gnome.org/GNOME/glib-networking/-/issues/179https://gitlab.gnome.org/GNOME/glib-networking/-/merge_requests/193
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2494>
For formats which we don't have fast-path implementation, compositor
will convert it to common unpack formats (AYUV, ARGB, AYUV64 and ARGB64)
then blending will happen using the intermediate formats.
Finally blended image will be converted back to the selected output format
if required.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1486>
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>
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>
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>
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>
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>
There's no need to re-assign the return value of
g_string_append_*() functions and such to the variable
holding the GString. These return values are just for
convenience so function calls can be chained. The actual
GString pointer won't change, it's not a GList after all.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2685>
This reverts commit 6f9ae5d758.
The _transform_caps() function can't tell the difference
between the caller wanting to know the output caps
for the current method, or all possible output caps. If
it includes caps for all possible methods, glupload can
end up negotiating and sending the wrong output caps
downstream.
Partially reverts !2687Fixes#1310
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2699>
If no filter caps are provided with a caps query, always
generate a full set of all caps from all upload methods,
not just the configured one. This is needed to handle
renegotiation when dealing with raw sysmem caps - as the upload
method might accept raw sysmem caps, but only the raw data
uploader adds those to the caps query.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2687>
This reverts commit f3292dc156.
Only the raw data uploader should add sysmem caps to the
actual caps query, because we want them to be at the
lowest priority. If upstream does select to send raw
caps, then the correct upload method will still
be chosen because the accept_caps implementation
will accept them
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2687>
When checking if we need to reconfigure when uploading, check
specifically the output caps of the current method will
result in compatible/incompatible caps, not the full set
of output caps from all upload methods.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2687>
Fixes warnings like:
Received a structure string that contains '="0.5"'. Reading as a gdouble value, rather than a string value. This is undesired behaviour, and with GStreamer 1.22 onward, this will be interpreted as a string value instead because it is wrapped in '"' quotes. If you want to guarantee this value is read as a string, before this change, use '=(string)"0.5"' instead. If you want to read in a gdouble value, leave its value unquoted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2621>
Some encoders (e.g. Makito) have H265 field-based interlacing, but then
also specify an 1:2 pixel aspect ratio. That makes it kind-of work with
decoders that don't properly support field-based decoding, but makes us
end up with the wrong aspect ratio if we implement everything properly.
As a workaround, detect 1:2 pixel aspect ratio for field-based
interlacing, and check if making that 1:1 would make the new display
aspect ratio common. In that case, we override it with 1:1.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2577>
