The GST_VIDEO_BUFFER_FLAG_TOP_FIELD flag is a superset of
GST_VIDEO_BUFFER_FLAG_BOTTOM_FIELD as they are defined using other
flags. As a result we can't use GST_BUFFER_FLAG_IS_SET() to check for
those flags.
By setting the extension-ID for TWCC (Transport Wide Congestion Control),
the payloader will embed sequencenumbers as a RTP header-extension
according to https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01#section-2
The negotiation of this being enabled with downstream elements
is done with caps reflecting the way this is communicated using SDP.
With commit "basepayload: Expose onvif-no-rate-control property" the rtp
timestamp changed behaviour when rate control is disabled.
When disabling rate control, we must take care of the stream time to
avoid the timestamps to begin from zero again.
This simply implies not trying to "prepare" those buffers,
as mapping an empty buffer to a video frame does not make
much sense.
This also adds a simple test in compositor that performs
some trivial checking of the handling of gap events, the test
was not failing before, but an error was logged, this is
no longer the case.
Fixes#717
This validate that the base class properly save and return the flow
return value received when gst_rtp_base_depay_push/push_list() helper is
being used.
This is not set anywhere, and it's pretty clear the pipeline in
question has not been tested in a long time. Disable test with
a FIXME, test needs to be rewritten to not use real output devices.
overlaycomposition.c:276:5: warning: implicit declaration of function 'exit' [-Wimplicit-function-declaration]
overlaycomposition.c(263): warning C4090: 'initializing': different 'const' qualifiers
Previously this would've only set discont=TRUE and then for all future
buffers simply returned immediately.
Instead we also need to
a) drain previous input until its buffer time
b) update next_ts and base_ts accordingly for the gap
c) actually store the new buffer after the gap so it can be used in
the future and so the old buffer before the gap is gone
Also update the unit test accordingly so that it actually tests for this
behaviour. Previously it only tested that after the gap we got no output
at all.
By adding this field, buffer producers can now explicitly set the exact
geometry of planes, allowing users to easily know the padded size and
height of each plane.
GstVideoMeta is always heap allocated by GStreamer itself so we can
safely extend it.
When using gst_video_info_align() user had no easy way to retrieve the
padded size and height of each plane.
This can easily be implemented in fill_planes() as it's already called
in align() with the padded height.
Ideally we'd add a plane_size field to GstVideoInfo but the remaining
padding is too small so that would be an ABI break.
Fix#618
When checking the behaviour of live seeking on audiomixer or
adder we don't *really* need real audio devices. audiotestsrc
in live mode is enough to test the behaviour of those elements.
Also avoids people repeatedly wasting hours trying to figure out
whether that failing behaviour is due to their code or not.
This is done by reusing `gst_gl_memory_setup_buffer` avoiding to
duplicate code.
Without a VideoMeta, mapping those buffers lead to GstBuffer mapping the
buffer in system memory even when specifying the GL flags (through the
buffer merging mechanism) making the result totally broken.
Make more flexible. There is an extra
gethostbyname2_r@@GLIBC_2.2.5 (getXXbyYY_r.c:217)
in the trace on the build bots (F30).
Fixes the -base and -good valgrind jobs on the 1.16 branch CI.
The extmap attribute allows mapping RTP extension header IDs to
well-known RTP extension header specifications. See RFC8285 for details.
We store the extmap attribute either as string in the caps
extmap-X=extensionname
where X is the integer extension header ID, or as 3-tuple of strings
extmap-X=<direction,extensionname,extensionattributes>
where direction or extensionattributes are allowed to be the empty
string.
Both formats are allowed because usually only the extension name is
given and it's much simpler to handle in caps.
Add max-reorder property to make the old hard coded reordering limit of
100 configurable. It's particularly useful in some scenarios to set
max-reorder=0 to disable the behavior that the depayloader will drop
packets.
Note that although the default value is 100, the default limit has
increased with one because of the changed if-test. This was done to
allow the max-reorder value to be more intuitive. See tests.
Continuation of 4fd7a2c783
We check the availability of the high precision floats in GLSL shaders
which involves an OpenGL call and thus is required to be executed on the
OpenGL thread.
The tests were not respecting that and could fail on more strict
drivers.
Tests update for 675415bf2e
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/590
We check the availability of the high precision floats in GLSL shaders
which involves an OpenGL call and thus is required to be executed on the
OpenGL thread.
The tests were not respecting that and could fail on more strict
drivers.
Tests update for 675415bf2e
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/590
valgrind gets confused with the following piece of code:
var37.i = ORC_CLAMP_SL((orc_int64)var33.i + (orc_int64)var34.i);
Where all variables are orc_int32
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.
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)
It's invalid to have a 'interlace-mode=alternate' without the Interlaced caps
feature as well.
Modify gst_video_info_from_caps() to reject such case so we can easily
spot them in bugged elements.
This test takes a long time. It tests ca. 8900 conversion
combinations, and then it also runs each conversion for
at least 100ms in order to come up with some kind of benchmark.
Remove the benchmarking from the unit test, we have a separate
benchmarking tool for that now.
Also split the conversions into groups and run those as
separate checks, which allows better parallelisation at
the runner level (normal runs and when using valgrind).
Before a gap event is pushed downstream a segment event must be pushed
since the gap event can cause packet concealment downstream and hence
data flow. Since concealment before receiving any data packets usually
doesn't make any sense, the gap event is not sent downstream.
Alternatively one could generate a default caps and segment event, but
no need to complicate things until it's proven necessary.
https://bugzilla.gnome.org/show_bug.cgi?id=773104https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/301
We're creating buffers with one sample here for some reason. The
actual value of the segment stop is irrelevant for what we're testing
here, so lower it to 10ms so that we create fewer buffers which speeds
things up on slow machines and in valgrind.
../subprojects/gst-plugins-base/tests/check/elements/audiorate.c(192): warning C4047
Meaningful validation at that point seems to checking output GstAudioFormat
of gst_audio_format_from_string()
