When caps changes while streaming, the new caps was getting processed
immediately in videoaggregator, but the next buffer in the queue that
corresponds to this new caps was not necessarily being used immediately,
which resulted sometimes in using an old buffer with new caps. Of course
there used to be a separate buffer_vinfo for mapping the buffer with its
own caps, but in compositor the GstVideoConverter was still using wrong
info and resulted in invalid reads and corrupt output.
This approach here is more safe. We delay using the new caps
until we actually select the next buffer in the queue for use.
This way we also eliminate the need for buffer_vinfo, since the
pad->info is always in sync with the format of the selected buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=780682
When there are more than 64 channels, we don't want to exceed the
bounds of the ordering_map buffer, and in these cases we don't want to
remap at all. Here we avoid doing that.
Based on a patch originally for plugins-good/interleave in
https://bugzilla.gnome.org/show_bug.cgi?id=780331
This duplicated property is no longer needed as there is now API to
allow bindings access GST_TYPE_ARRAY (see gst_util_get/set/object_array).
Additionnally, Python has proper overrides which will make this looks
like Python. A 2x2 matrix would be set this way:
element = matrix = Gst.ValueArray(Gst.ValueArray([1.0, -1.0]),
Gst.ValueArray([1.0, -1.0))
Notice that you need to "cast" each arrays to Gst.ValueArray, otherwise
there is an ambiguity between Gst.ValueArray and Gst.ValueList list type.
Fortunatly, Gst.ValueArray implements the Sequence interface, so it can
be indexed like normal python matrix.
Inserts AU delimeter by default if missing au delimeter from upstream.
This should be done only in case of byte-stream format.
Note that:
We have to compensate for the new bytes added for the AU, otherwise
insertion of PPS/SPS will use wrong offsets and overwrite wrong data.
Also mark the AU delimiter blob const, and use frame->out_buffer for
storing the output to keep baseparse assumptions valid.
Original-Patch-By: Michal Lazo <michal.lazo@mdragon.org>
Helped by Sebastian Dröge <sebastian@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=736213
Those are the rules:
In the SPS:
* if frame_mbs_only_flag=1 => all frame progressive
* if frame_mbs_only_flag=0 => field_pic_flag defines if each frame is
progressive or interlaced, thus the mode is 'mixed' in GStreamer
terms.
https://bugzilla.gnome.org/show_bug.cgi?id=779309
Doing lazy conversion of PCR values doesn't work right
when a PCR discont is encountered. Instead, convert PCR
values to the continuous timestamp domain as soon as we
encounter them and store that instead.
This element transforms a given number of input channels into a given number of
output channels according to a given transformation matrix. The matrix
coefficients must be between -1 and 1. In the auto mode, input/output channels
are automatically negotiated and the transformation matrix is a truncated or
zero-padded identity matrix.
https://bugzilla.gnome.org/show_bug.cgi?id=777376
See https://bugzilla.gnome.org/show_bug.cgi?id=773666
This would ideally be solved in baseparse but that requires further
thought at this point, and in the meantime it would be good to have
rawbaseparse not assert on this but handle it gracefully instead.
DVDs always have subpictures that start on an even Y
coordinate, but gstspu does more generic vobsubs these
days, so handle ones that start on an odd vertical position.
https://bugzilla.gnome.org/show_bug.cgi?id=777400
timecodestamper will post an element message which contains the current
timecode it just stamped. If a timecode was already found and not
replaced, it will still post it in a message.
https://bugzilla.gnome.org/show_bug.cgi?id=777048
... rather than when determining when to end the frame.
The opportunity to do so might not come when forced to drain,
and it seems nicer anyway to do so at parse wrapup time.
This happens if we had no CAPS event yet but e.g. got an EOS event. We
would then try to output a 0-sized buffer, but getting that from the
adapter will give an assertion, return NULL and then crash.
If they were not ported after 4+ years it seems unlikely that anybody is
ever going to need them again. They're still in the GIT history if
needed.
https://bugzilla.gnome.org/show_bug.cgi?id=774530
Compositor does not support it currently and it needs special support
for handling this correctly, and is rather non-trivial to implement for
all formats.
For frame->buffer, baseparse is doing that automatically for us. For
frame->output_buffer it doesn't and assumes that the subclass is already
doing that. Consistency!
This is useful e.g. if audio buffers should be exactly the duration of a
video frame, or if a audio buffers should never be too large because of
latency constraints.
The element is taking a fractional buffer duration, to allow working
with e.g. 1001/30000 as output duration and it accumulates rounding
errors in the buffer durations and compensates for them by making some
buffers one sample larger than the others.
https://bugzilla.gnome.org/show_bug.cgi?id=774689
We will allocate a screen area of width*height*bpp bytes, however this
calculation can easily overflow if too high width or height are given
inside the stream. Nonetheless we would just assume that enough memory
was allocated, try to fill it and overwrite as much memory as wanted.
Also allocate the screen area filled with zeroes to ensure that we start
with full-black and not any random (or not so random) data.
https://scarybeastsecurity.blogspot.gr/2016/11/0day-poc-risky-design-decisions-in.html
Ideally we should just remove this plugin in favour of the one in
gst-libav, which generally seems to be of better code quality.
https://bugzilla.gnome.org/show_bug.cgi?id=774533
Type cast has higher precedence than bitwise shift, so the third
argument will truncate to 8 bits and then shift right by 8 bits
resulting in constant zero.
https://bugzilla.gnome.org/show_bug.cgi?id=774293
Consistently use GST_ROUND_UP_4(width) as stride for
bayer buffers. Bayer data will usually come in widths
that are multiples of 4 anyway, so hopefully this
should not have any adverse impact on anyone in
practice.
Before, bayer2rgb required input buffers to are sized
accordingly, but then didn't actually round up when
calculating row offsets. rgb2bayer didn't use a rounded
stride nor buffer size.
https://bugzilla.gnome.org/show_bug.cgi?id=752014
rawvideoparse wouldn't error out on not-negotiated,
but would just keep on going, because it didn't pass
the flow return value back to the parent class and
thus upstream, so the source wouldnt' stop streaming.
