* Library versioning should not be used for plugins since it will add
-{version}.dll suffix (and versioned libraries on Linux with symlink).
Then the library file name and plugin init function name mismatch
will result in blacklisted plugin.
* Don't define BUILDING_GST_CODECS, makes no sense
* Don't define G_LOG_DOMAIN, which should be used only for libraries,
not plugins
* Depends on gstcodecparsers libary, not gstcodecs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5249>
This page has been only sporadically updated for a decade, and it is
unlikely to be updated properly anytime soon. Update the top half, and
add a note about the tutorial section being out of date.
The trigger for this was a question on the mailing list about Windows
11 support, since it's not listed in the supported platforms list.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5239>
when playing some codec such as matroska with vp9 codec,
demuxer will save information like video_mastering_display_info
and video_content_light_level in caps that decoder need,
v4l2videodecoder can use it by calling V4L2_CTRL_CLASS_COLORIMETRY
ioctl.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4403>
If decoder notify a source change event when the capture format is
changed, not the resolution changed.
then gst_v4l2_object_acquire_format will retuen false due to
unsupported format.
we need to clear the format lists in the source change flow,
and reenumerate format list
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5218>
This elements pass RTP packets along unchanged and appear as a RTP
payloader element.
This is useful, for example when using the gstreamer-rtsp-server
library, in the case where you are receiving RTP packets from a
different source and want to serve them over RTSP. Since the
gst-rtsp-server library expect the element marked as payX to be a RTP
payloader element and assumes certain properties are available.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5204>
Imported dmabuf are not being duped, so they should never be closed. Instead,
we ensure their live time by having strong reference on their original
buffer. This should fix potential flickering due to dmabuf being closed
too early.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5101>
Now that we can split GStreamer buffers over multiple v4l2 buffer, we may
endup waiting for these buffers to be processed. Avoid waiting for any of
the parts being processed. As a side effect, the pool will now try to
grow if the number of buffers is not sufficient, and will fail
otherwise.
This fixes a hang if the very first frame did not fit. In this case, the
driver will retrain that buffer until the capture is setup, but
GStreamer won't setup the capture until process() function have
returned.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5100
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5143>
Fix warnings from bindings changes in various plugin
examples
Fix the python mixer plugin by ensuring that PIL
is not holding a reference to mapped GstBuffer memory.
Port the filesrc example from old_examples
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5187>
The hack enforcing strictly increasing timestamps was, according to the
code comments, because librtmp was confused with backwards timestamps.
rtmp2sink is not using librtmp as rtmpsink did, so this is no longer
required.
Also changing the timestamps is causing audio glitches when streaming to
Youtube.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5212>
Add gst_audio_ring_buffer_set_errored() that will mark the
ringbuffer as errored only if it is currently started or paused,
so gst_audio_ringbuffer_stop() can be sure that the error
state means that the ringbuffer was started and needs stop called.
Fixes a crash with osxaudiosrc if the source element posts
an error, because the ringbuffer would not get stopped and CoreAudio
would continue trying to do callbacks.
Also, anywhere that modifies the ringbuffer state, make sure to
use atomic operations, to guarantee their visibility
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5205>
Section 3.4 in RFC8835 states that if a WebRTC endpoint uses an HTTP
proxy to access the Internet it MUST include the "ALPN" header. This
commit adds this header.
By default the ALPN used when connecting to the TURN/TCP server via a
proxy is set to "webrtc". It can be changed by adding an alpn url
option for the http-proxy. For example:
http://user:pass@my.http.proxy.com:8080?alpn=c-webrtc
This will add the header "ALPN: c-webrtc" to the HTTP proxy CONNECT
request.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4212>
