This is not actually required (anymore?). Source pad caps can be
negotiated at any time regardless of any configured (or existing) sink
pads and videoaggregator comes up with some fixated caps based on the
downstream caps.
Subclasses can override this behaviour as needed by overriding
update_src_caps().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/793>
`gst_gl_memory_read_pixels` reads pixels from `GLMemory` into the
pointer, effectively writing to it. This is opposite from
`gst_gl_memory_texsubimage` which reads texture data from `read_pointer`
into `GLMemory`.
Both cases are clarified by changing `read_pointer` to `write_pointer`,
and explaining what `gst_gl_memory_texsubimage` does in addition to
referring back to `gst_gl_memory_read_pixels`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/806>
These then don't require going through the generic code path via AYUV64
first but can be converted directly.
This speeds up processing of
videotestsrc ! v210 ! videoconvert ! other_format ! fakesink
by a factor of 1.55 for I420/YV12 and 1.40 for the other destination
formats and reduces memory pressure considerably.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/775>
The type is called GstVideoTransferFunction so the function names should
match, otherwise gobject-introspection is keeping the functions as
global functions instead of methods on the type.
The same mistake was also made in lots of other APIs over the years, but
here we can at least fix it for 1.18 still.
Thanks to Marijn Suijten for noticing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/807>
For audio we copy metas that have no tags at all, or that only have the
"audio" and/or "audio-channels" tag. Audio codecs don't change the
audio aspect of the stream and in almost all cases don't change the
number of channels. They might however change the sample rate (e.g.
Opus). Subclasses that change the number of channels will have to
override ::transform_meta() accordingly.
For video we copy metas that have no tags at all, or that only have the
"video" and/or "video-size" and/or "video-orientation" tag. Video codecs
don't change the "video" aspect of the stream and in almost all cases
don't change the resolution or orientation. Subclasses that rescale or
change the orientation will have to override ::transform_meta()
accordingly.
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/576#note_610581
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/801>
Some languages have an ISO 639-2 representation but no 639-1
representation, for example where "eng" has a two-letter
equivalent in "en", "enm" doesn't have one.
Discarding those languages from our static table caused functions
such as gst_tag_get_language_code_iso_639_2T() or
gst_tag_get_language_code_iso_639_2B() to return NULL for
valid language codes such as "enm", potentially causing users
of these API such as mpegtsmux to discard language code tags
as invalid.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/785>
```
../subprojects/gst-plugins-base/ext/alsa/gstalsamidisrc.c:201:54: error: converting the result of '<<' to a boolean always evaluates to false [-Werror,-Wtautological-constant-compare]
snd_seq_ev_schedule_real (&ev, alsamidisrc->queue, SND_SEQ_TIME_MODE_ABS,
^
/usr/include/alsa/seq_event.h:215:34: note: expanded from macro 'SND_SEQ_TIME_MODE_ABS'
```
The ALSA API expects 0 or 1 here and will then add the flags accordingly,
and that's also what other code using this API does.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/777>
When a pad has alpha != 1.0 it means that the resulting frames will
contain alpha and thus can't fully obscure with a lower zorder.
Also simplifies the other checks as blending with an OVER or on a
transparent is not a no-op as previously assumed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/764>
Call gst_aggregator_selected_samples() after filling the queues
(but before preparing frames).
Implement GstAggregator.peek_next_sample.
Add an example that demonstrates usage of the new API in combination
with the existing buffer-consumed signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/728>
5 seconds might not be enough value for timeout in case an application
is running on a device with very limited computing power.
Note that ANGLE uses 10 seconds timeout value. So even if a timeout
happens here, it's also ANGLE's timeout condition as well
(meaning that bad things will happen either way)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/769>
It was not working properly and the implementation of the smartencoder
element was weird. This introduce a number of changes (which are all
in one single commit because they basically all work together and lead
to basically reimplementing the element):
* Make smartencoder a bin so that the reencoding chain of elements are
inside of it instead of not having any parent. Those elements were not
be visible when dumping the pipeline which was very confusing.
* Make encodebin create the right encoder with a capsfilter (and parser)
to properly enforce the format specified by the user, and so that the
encoder properties specified in the encoding profile are respected.
* Use `decodebin` to do the decoding instead of selecting a decoder
ourself and not plug any parser etc...
* Ensure that negotiated format in the sinkpad of smart encoder is fixed
through time when the user requested a non dynamic output
* Add a parser at the beginning of the smart encoder
* Handle errors when reencoding
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/751>
