Age | Commit message (Collapse) | Author |
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We were setting audio and video caps by default even when the user
might have requested only video or audio. This would then result
in a `Could not reuse transceiver` error from the webrtcbin.
Fix this by allowing the user to specify audio or video caps as
None. This allows us to maintain the earlier behaviour for backward
compatibility while allowing the user to not request audio or video
as need be.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1433>
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These are automatically implemented.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/483
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
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The JanusVRWebRTCSink is a new plugin that integrates with the Video
Room plugin of the Janus Gateway, which simplifies WebRTC communication.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1362>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1381>
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Easier to read that way.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1430>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1423>
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They fail state changes, which cases `GstHarness` to abort.
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Disabling while we figure out why it's failing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1429>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
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This reverts commit b128d127c25e42f4908a096cad6829c6f60962b9.
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1421>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1421>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1421>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
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If a peer connects via TCP and never initiates TLS, then the server
will get stuck in the accept loop. Spawn a task when accepting a TLS
connection, and timeout if it doesn't complete in 5 seconds.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
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from_glib_full() was not in scope.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1417>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1389>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1415>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1414>
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See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1413>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1413>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1413>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
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Makes sure we can send out data even if the pipeline shutdown in error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
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Lets us connect when we perform a flush
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
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When streaming small amounts of data, using awss3sink might not be a
good idea, as we need to accumulate at least 5 MB of data for a
multipart upload (or we flush on EOS).
The alternative, while inefficient, is to do a complete PutObject of
_all_ the data periodically so as to not lose data in case of a pipeline
failure. This element makes a start on this idea by doing a PutObject
for every buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1409>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1409>
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This manifests in a gst-launch-1.0 pipeline using ts-udpsrc, since
notification of used-socket results in the property being read by the
application, and the settings lock causes a deadlock.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1408>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1408>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1407>
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Version 0.61 was yanked from crates.io.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1407>
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Keep dash-mpd at 0.14.5 and xattr at 1.0.1 because otherwise compilation
fails, see:
- https://github.com/Stebalien/xattr/issues/44
- https://github.com/bytecodealliance/rustix/issues/945
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1403>
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Changes from https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1355
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1403>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1400>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1400>
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webrtcbin will refuse pad requests for all sorts of reasons, and should
be logging an error when doing so, simply post an error message and let
the application deal with it, the reason for the refusal should
hopefully be available in the logs to the user.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1399>
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