The goal is to be able to get back the original buffer
after performing analysis on a transformed version. Then put the
various GstMeta back on the original buffer.
An example pipeline would be
.. ! originalbuffersave ! videoscale ! analysis ! originalbufferestore ! draw_overlay ! sink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1428>
GST_PLUGIN_FEATURE_RANK=rtspsrc2:1 gst-play-1.0 [URI]
Features:
* Live streaming N audio and N video
- With RTCP-based A/V sync
* Lower transports: TCP, UDP, UDP-Multicast
* RTP, RTCP SR, RTCP RR
* OPTIONS DESCRIBE SETUP PLAY TEARDOWN
* Custom UDP socket management, does not use udpsrc/udpsink
* Supports both rtpbin and the rtpbin2 rust rewrite
- Set USE_RTPBIN2=1 to use rtpbin2 (needs other MRs)
* Properties:
- protocols selection and priority (NEW!)
- location supports rtsp[ut]://
- port-start instead of port-range
Co-Authored-by: Tim-Philipp Müller <tim@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
Only the examples of the fallbackswitch, livesync, and togglerecord
plugins require the gtk, gio, and gst-plugin-gtk4 features. The plugins
themselves don't actually have a dependency on GTK.
Only add the features (and examples) if the examples are actually
enabled to allow building these plugins without the GTK dependency.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1438>
This fixes most, but not all, of the build errors in Windows when using
static libraries.
The ones remaining are:
- redirection of gstreamer-1.0 towards gstreamer-full-1.0
- Cairo not exporting the C++ stdlib requirement when built statically
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1307>
Previously, there was no check performed on features of plugins if these
specify GStreamer plugins. This commit adds that, and ensures that the
plugins and pkg-config targets are skipped if no outputs are to be
generated (this is already done for examples).
Closes#369
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1303>
Previously, there was no check performed on features of plugins if these
specify GStreamer plugins. This commit adds that, and ensures that the
plugins and pkg-config targets are skipped if no outputs are to be
generated (this is already done for examples).
Closes#369
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1303>
This new plugin exposes two elements, intersink and intersrc. These act
as wormholes for data in the same process and can be used to forward
data from one pipeline to another.
The implementation makes use of gstreamer-utils' StreamProducer, and
supports dynamically adding and removing consumers, before and after
producers, and changing producer names while PLAYING, both on the sink
and the src.
This initial implementation comes with a small demo, and a few tests.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1257>
Required renaming threadshare/benchmark to threadshare/ts-benchmark
because 'benchmark' as a target name is reserved for meson's
`benchmark` target.
Disabled by default because cargo decides that it has to rebuild
everything, and is really slow because of that.
Also required adding --features for setting features required by the
examples.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1028>
Required a slight rework of the build file, and how options are passed
to cargo_wrapper.py
Necessitated a bump of the required gstreamer version to 1.20, which
should be fine for the meson build since its primary function is to be
built as part of the gstreamer monorepo build to get a dev env.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1028>
It attempts to produce a (nearly) gapless live stream by synchronizing
its output to the running time and forwarding the next input buffer if
its start is (nearly) flush with the end of the last output buffer.
If the input buffer is missing or too far in the future, it duplicates
the last output buffer with adjusted timestamps. If it is operating on a
raw audio stream, it will fill duplicate buffers with silence.
If an input buffer arrives too late, it is thrown away. If the last
input buffer was accepted too long ago (according to `late-threshold`),
a late input buffer is accepted anyway, but immediately considered a
duplicate. Due to the silence-filling, this has no effect on audio, but
video gets a "slideshow" effect instead of freezing completely.
The "many-repeats" property will be notified when this element has
recently duplicated a lot of buffers or recovered from such a state.
Co-authored-by: Vivia Nikolaidou <vivia@ahiru.eu>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/708>
