gst-examples merge requestshttps://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests2022-06-27T16:11:21Zhttps://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/24WIP: Merge UWP example application2022-06-27T16:11:21ZSeungha Yangseungha@centricular.comWIP: Merge UWP example applicationMerge GStreamer UWP example application available at https://gitlab.freedesktop.org/seungha.yang/gst-uwp-example
- [ ] Check license decl.
- [ ] @nirbheek's `gst-dll-tools` dependency
- [ ] Preserve commit history with merge commit or s...Merge GStreamer UWP example application available at https://gitlab.freedesktop.org/seungha.yang/gst-uwp-example
- [ ] Check license decl.
- [ ] @nirbheek's `gst-dll-tools` dependency
- [ ] Preserve commit history with merge commit or squash commits ?https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/43webrtc/signaling: Correct the condition when calling a busy peer2022-05-20T12:53:15Zbyran77webrtc/signaling: Correct the condition when calling a busy peerHi,
Thanks for such great examples for reference and study!
I thought there exists a little mistake in webrtc signaling server. When a session request is coming, `ERROR` occurs when the callee is busy. But `peer_status` is the status o...Hi,
Thanks for such great examples for reference and study!
I thought there exists a little mistake in webrtc signaling server. When a session request is coming, `ERROR` occurs when the callee is busy. But `peer_status` is the status of the caller, which is of course `None` when calling someone, while `self.peers[callee_id][2]` is that of the callee.https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/13gst-play: use novel signal-adapter (requires gstplayer lib patch from gst-plu...2022-05-20T12:43:55ZStephan Hessegst-play: use novel signal-adapter (requires gstplayer lib patch from gst-plugins-bad MR #35)gst-play: use novel signal-adapter (requires gstplayer lib patch from https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/35)
@slomogst-play: use novel signal-adapter (requires gstplayer lib patch from https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/35)
@slomohttps://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/30webrtc-rust: allow setting a port range2021-09-24T21:37:16ZMathieu Duponchellewebrtc-rust: allow setting a port rangeThis is useful as an example of how to access and interface with
the ICE agent properly.
Marked as WIP because of the local dependency on the libnice bindings, see https://github.com/Johni0702/rust-libnice/pull/4This is useful as an example of how to access and interface with
the ICE agent properly.
Marked as WIP because of the local dependency on the libnice bindings, see https://github.com/Johni0702/rust-libnice/pull/4https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/32sendrecv/gst: Fix python webrtc sample2021-02-25T21:19:37ZSanchayan Maitysanchayan@sanchayanmaity.netsendrecv/gst: Fix python webrtc sampleCurrently running this Python webrtc example gives several warnings and
eventually stops working after some time.
GstStructure is not subscriptable and GstCaps does not have a 'len'.
Also, the method adding the element to the pipeline b...Currently running this Python webrtc example gives several warnings and
eventually stops working after some time.
GstStructure is not subscriptable and GstCaps does not have a 'len'.
Also, the method adding the element to the pipeline bin only takes one
argument which is the element to be added.
This fixes all the above warnings/errors.