This MR provides a transform element that leverage ONNX runtime
to run AI inference on a broad range of neural network toolkits, running
on either CPU or GPU. ONNX supports 16 different providers at the
moment, so with ONNX we immediately get support for Nvidia, AMD, Xilinx
and many others.
For the first release, this plugin adds a gstonnxobjectdetector element to
detect objects in video frames. Meta data generated by the model is
attached to the video buffer as a custom GstObjectDetectorMeta meta.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1997>
Useful when having a service that runs a GStreamer pipeline
or application in Google Cloud to avoid storing the inputs
and outputs in the running container or service. For example
when analyzing a video from a Google Cloud Storage bucket
and extracting images or converting the video and then uploading
the results into another Google Cloud Storage bucket.
- gssrc allows to read from a file located in Google Cloud
Storage and it supports seeking.
- gssink allows to write to a file located in Google Cloud
Storage. There are 2 modes, one similar to multifilesink and
the other similar to filesink.
Example:
gst-launch-1.0 gssrc location=gs://mybucket/videos/sample.mp4 ! decodebin ! glimagesink
gst-launch-1.0 playbin uri=gs://mybucket/videos/sample.mp4
gst-launch-1.0 videotestsrc num-buffers=5 ! pngenc ! gssink object-name="img/img%05d.png" bucket-name="mybucket" next-file=buffer
gst-launch-1.0 filesrc location=sample.mp4 ! gssink object-name="videos/video.mp4" bucket-name="mybucket" next-file=none
When running locally simply set GOOGLE_APPLICATION_CREDENTIALS. But
when running in Google Cloud Run or Google Cloud Engine, just set the
"service-account-email" property on each element.
Closes#1264
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1369>
LDAC is an audio coding technology developed by Sony that enables the
transmission of High-Resolution (Hi-Res) audio contents over Bluetooth.
Currently Adaptive Bit Rate (ABR) as supported by libldac encoder is not
implemented.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1621>
The SVT-HEVC (Scalable Video Technology[0] for HEVC) Encoder is an
open source video coding technology[1] that is highly optimized for
Intel Xeon Scalable processors and Intel Xeon D processors.
[0] https://01.org/svt
[1] https://github.com/OpenVisualCloud/SVT-HEVC
This patch introduces the bootstrap code from the AVTP plugin (plugin
definition and init) as well as the build system files. Upcoming patches
will introduce payloaders, source and sink elements provided by the AVTP
plugin. These elements can be utilized by a GStreamer pipeline to
implement TSN audio/video applications.
Regarding the plugin build system files, both autotools and meson files
are introduced. The AVTP plugin is landed in ext/ since it has an
external dependency on libavtp, an opensource AVTP packetization
library. For further information about libavtp check [1].
[1] https://github.com/AVnu/libavtp
The wpe element is used to produce a video texture representing a web page
rendered off-screen by WPE. This element can be used to overlay HTML on top of
another video stream for instance.
We now have options for all plugins, so we will just disable these in
the cerbero recipe instead. These require external deps, so they won't
affect gst-build either.
SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/
The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer. In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.
The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.
With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=792523
SRT[0] is an open source transport technology[1] that optimizes
streaming performance across unpredictable networks.
Although SRT is based on UDP, it works like connection-oriented
protocol. However, it doesn't mean that the SRT server or client
is necessarily to link to a receiver or a sender so, here, the
pairs of source and sink elements are introduced.
- srtserversink: SRT server to feed SRT stream
- srtclientsrc: SRT client to get SRT stream from srtserversink
- srtclientsink: SRT client to send SRT stream
- srtserversrc: SRT server to listen from srtclientsink
[0] https://github.com/Haivision/srt
[1] http://www.srtalliance.org/https://bugzilla.gnome.org/show_bug.cgi?id=785730
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
This was used by MSN messenger in prehistoric times, it's safe
to say no one needs or wants this any more these days. For
decoding old recordings there's still a decoder in ffmpeg.
https://bugzilla.gnome.org/show_bug.cgi?id=597616
