Occasionally, a clean build would fail like this:
In file included from ../subprojects/gst-plugins-base/tests/examples/gl/gtk/gstgtk.c:24:
../subprojects/gst-plugins-base/gst-libs/gst/gl/gl.h:25:10: fatal error: gst/gl/gl-enumtypes.h: No such file or directory
25 | #include <gst/gl/gl-enumtypes.h>
| ^~~~~~~~~~~~~~~~~~~~~~~
Add the missing dependency so that the headers are generated beforehand.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/855>
This is explicitly specified as valid in the RFC, where the
syntax for both parameters is:
";" "client_port" "=" port [ "-" port ]
";" "server_port" "=" port [ "-" port ]
This is useful for applications where RTCP is either not necessary
or not possible.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/846>
When user is passing the actual interlace-mode when calling
gst_video_decoder_set_interlaced_output_state() it should not be
overidden by the input interlace-mode.
Needed to fix#825 as we want to keep interlace-mode=interleaved from
parsers and have the OMX decoder producing interlace-mode=alternate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/852>
Increases the throughput of compositing by using more CPU cycles across
multiple threads. Simple cases (the output contains one pixel from at
most one input) can have up to a 70% increase in throughput. Not so
simple cases are limited by the region with the most number of
composite operations.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/755>
Rename remaining `gst_video_color_transfer_{encode,decode}` functions on
the `GstVideoTransferFunction` enumeration to
`gst_video_transfer_function_{encode,decode}` permitting
gobject-introspection to turn these into associated functions and place
them under the respective `<enumeration>` block in gir XML files.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/805>
gst_pad_get_current_caps() may be wrong when there is a renegotiation in
progress for the pad and we have not yet received or selected the buffer
with different caps yet.
Fix by storing the caps through in a similar way to the existing code
for buffer/video-info selection machinery.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/813>
This allows subclasses that notice missing reference frames to request a
new sync point to allow seamless decoding again. While doing so the
subclass can also signal whether it wants a) all following input frames
until the sync point to be discarded or b) all output frames until the
sync point to be marked as corrupt.
Sending of force-keyunit events for this can be throttled by the
application via the "min-force-keyunit-interval" property.
This replaces custom behaviour for the same in various decoders, for
example openh264dec.
Based on patches by Haakon Sporsheim <haakon@pexip.com> and
Stian Selnes <stian@pexip.com>.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/730>
If the first frame(s) at the very beginning or after a flush are not a
sync point then the base class would discard them before passing them to
the subclass.
This also fixes the previously broken distance_from_sync handling: it
was never reset at sync points.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/730>
This can be used by applications to configure decoders so that corrupted
frames are directly discarded instead of being forwarded inside the
pipeline. It is a replacement for the "output-corrupt" property of the
ffmpeg decoders.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/730>
This can be used by subclasses to mark output frames as known to be
corrupted, for example if reference frames were missing. ffmpeg's
decoders can signal this.
In addition this flag is propagated downstream if the input frame had it
set.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/730>
Don't run the harness in live mode, or otherwise it would output frames
already in the very beginning before a buffer was provided to it due to
timeout.
Also send EOS/a second buffer before pulling a buffer as videoaggregator
has one frame of latency.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/812>
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>
