The code for this is mostly lifted from audiobuffersplit, it
allows use cases such as keeping the buffers output by compositor
on one branch and audiomixer on another perfectly aligned, by
requiring the compositor to output a n/d frame rate, and setting
output-buffer-duration to d/n on the audiomixer.
The old output-buffer-duration property now simply maps to its
fractional counterpart, the last set property wins.
If the last WebVTT cue does not have an empty line after it, or if it
does not end with a newline at all, it does not get pushed out and it
won't be displayed.
gst_sub_parse_sink_event() already handles the issue for other subtitle
formats, enable handling it for GST_SUB_PARSE_FORMAT_VTT too.
While at it also add a test for this case.
Packed 10 bits per each R, G and B channel with MSB 2bits alpha channel.
This format is mapped to Windows' DXGI_FORMAT_R10G10B10A2_UNORM format which is
required for 10bits HDR rendering.
Note that this RGB10A2_LE format is R - B channel swapped version of BGR10A2_LE
... if subclass didn't update values. Note that the mastering-display-info
and content-light-level might be updated by user defined value (e.g., encoding option).
Introduce HDR signalling methods
* GstVideoMasteringDisplayInfo: Representing display color volume info.
Defined by SMPTE ST 2086
* GstVideoContentLightLevel: Representing content light level specified in
CEA-861.3, Appendix A.
Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/400
By using strtoul(), invalid values will get mapped to MAXULONG and we
would have to check errno. They won't get mapped to 0.
To solve this, use the signed g_ascii_strtoll(). This will map errors to
0 or G_MAXINT64 or G_MININT64, and the valid range for GstDateTime is >
0 and <= 9999 so we can directly check for this here.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/issues/384
As part of commit 808e7127, we prefixed the `GstWlWindow`'s `shell`
field with wl_, to differentiate it from the other types of shells a
Wayland compositor might support. However, this is apparently a struct
that we expose to our users, so changing it means we have an API break.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/592
Add the possible to limit the Content-Length
Define an appropriate request size limit and reject requests exceeding
the limit (413 Request Entity Too Large)
When the glupload element renegotiates the caps, set_caps will reset the
method_impl to NULL, but the method will be kept. transform_caps tries
to use the method_impl to transform the caps, because a method is set,
but will segfault.
Make rtspconnection a little more strict to RFC2326.
Make sure that CSeq is in every RTSP message and that CSeq is valid.
Also break the build_next loop if any parsing fails, By acting on
the builder->status code.
