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Author SHA1 Message Date
François Laignel
875e01e90a Use gst_element_request_pad_simple... 
Instead of the deprecated gst_element_get_request_pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/35>
 2021-05-05 06:17:06 +00:00
Olivier Crête
c89dccbf4e validate README: Document paths for gst-examples 
As the webrtc demos have now been merged, change the paths for
easier copy-pasting.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/38>
 2021-04-30 17:20:18 -04:00
Nirbheek Chauhan
2c3d78c9a6 webrtc/signalling: Document cert exception needed for browsers 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-examples/-/issues/28

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/34>
 2021-03-16 19:41:27 +05:30
Philippe Normand
e0c77b75cd gtk-play: Port to GstPlay 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/33>
 2021-03-09 21:36:27 +00:00
Philippe Normand
840fcf43f5 gst-play: Port to GstPlay 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/33>
 2021-03-07 17:59:35 +00:00
Stephan Hesse
50e61f52ed gst-play.c: update to signal-adapter constructor change 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/33>
 2021-03-07 17:59:35 +00:00
Stephan Hesse
d5a183cc2f gst-play: use novel signal-adapter (requires gstplayer lib patch from https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/35) 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/33>
 2021-03-07 17:59:35 +00:00
Nirbheek Chauhan
f8cbae9d6e sendrecv: Implement remote-offerer option for JS example 
Now you can check the "Remote offerer" checkbox in the JS example to
force the peer to send the SDP offer. This involved implementing
support for receiving the OFFER_REQUEST message in the C example.

As a side-effect of this, the C example will no longer send
OFFER_REQUEST automatically when the --our-id option is passed. It
will only do so when the --remote-offerer option is explicitly passed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/31>
 2021-02-10 16:23:40 +05:30
Nirbheek Chauhan
28aa23dc20 sendrecv/gst: Some misc whitespace fixes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/31>
 2021-02-10 16:23:40 +05:30
Nirbheek Chauhan
2892a8b206 sendrecv/js: Implement state handling for Connect button 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/31>
 2021-02-10 16:21:34 +05:30
Nirbheek Chauhan
eb89cd01ba webrtc: Document OFFER_REQUEST in the protocol doc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/31>
 2021-02-10 16:21:32 +05:30
Nirbheek Chauhan
ea3c0e8766 sendrecv/js: Handle OFFER_REQUEST as part of the switch 
This is clearer, and also stricter w.r.t. what sort of messages we
accept.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/31>
 2021-02-10 16:21:30 +05:30
Nirbheek Chauhan
a508bc243d sendrecv/gst: Don't need to allocate to send OFFER_REQUEST 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/31>
 2021-02-10 16:21:24 +05:30
Seungha Yang
767f46b1a0 webrtc: sendonly: Add support for Windows 
Add meson build script and use mfvideosrc element in case of Windows

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/29>
 2020-12-10 20:18:30 +09:00
Seungha Yang
85aeda42fe sendrecv/js: Add an UI for connecting to specified peer id 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/28>
 2020-12-07 20:40:50 +09:00
Seungha Yang
9e83c09be6 sendrecv/js: Convert taps to spaces 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/28>
 2020-12-07 20:40:50 +09:00
Seungha Yang
753f14f5de sendrecv: Add an option for example to be able to accept connection request from peer 
Add "our-id" option to specify id to be used for registering to
signalling server and wait connection request from peer

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/28>
 2020-12-07 20:40:41 +09:00
Emmanuel Gil Peyrot
20bc59f1ff rust: Regenerate Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/27>
 2020-11-23 15:29:44 +01:00
Emmanuel Gil Peyrot
3710c81432 rust: Bump async-tungstenite 
This removes the pin-project 0.4 dependency to use 1.0 instead like the
rest of the code.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/27>
 2020-11-23 15:28:28 +01:00
Olivier Crête
4e141f1076 webrtc sendonly: Add priority to example 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
 2020-10-08 16:16:36 -04:00
Olivier Crête
992cb3c5f4 webrtc sendonly: Add videoscale to avoid webcam compat issues 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
 2020-10-08 16:16:19 -04:00
Olivier Crête
4d06428001 webrtc sendonly: Exit on bus errors 
Catch bus errors and cleanly error out

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
 2020-10-08 16:16:16 -04:00
Nirbheek Chauhan
43f8275ca9 playback: Remove libvisual plugin from iOS GstPlayer example 
We won't be building the plugin in Cerbero anymore, so remove it from
the iOS example too. See:
https://gitlab.freedesktop.org/gstreamer/cerbero/-/merge_requests/605

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/26>
 2020-09-19 11:45:30 +05:30
Tim-Philipp Müller
1f66cda890 Back to development 2020-09-08 16:59:14 +01:00
Tim-Philipp Müller
009290dc87 Release 1.18.0 2020-09-08 00:10:23 +01:00
Tim-Philipp Müller
899cd55b5f Release 1.17.90 2020-08-20 16:16:55 +01:00
Matthew Waters
09195ebe86 webrtc/android: add decodebin/autoaudiosink to plugin list 
Otherwise the app fails to run

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/25>
 2020-08-19 20:42:16 +10:00
Matthew Waters
8b4d156712 webrtc/android: initialize the debug category 
Fixes possible critical/crash on startup

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/25>
 2020-08-19 20:42:16 +10:00
Matthew Waters
101d9965e5 webrtc/android: use a better name for the output apk 
Instead of a generic app-debug.apk

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/25>
 2020-08-19 20:01:56 +10:00
Matthew Waters
a7daeb14c3 webrtc/android: explicitly link to iconv 
As is now required

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/25>
 2020-08-19 20:01:56 +10:00
Matthew Waters
a7d0e6051c webrtc/android: use the openssl Gio module 
That's what is shipped upstream now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/25>
 2020-08-19 20:01:56 +10:00
Matthew Waters
d1b81046a4 webrtc/android: add missing gradle-wrapper jar 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/25>
 2020-08-19 20:01:56 +10:00
Carl Karsten
e1de93cf40 Update README.md 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/23>
 2020-08-09 20:06:54 +00:00
Sebastian Dröge
bbed24d919 webrtc: Change H264 examples to use aggregate-mode=zero-latency for best compatibility 
The default changed back to none because it broke existing code.

See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/749

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/22>
 2020-08-05 10:47:55 +03:00
Sebastian Dröge
6378337a0e sendrecv/Rust: Only set pipeline to Playing after connecting to the signals 
Might miss some signal emissions otherwise, especially the
on-negotiation-needed signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/21>
 2020-07-31 12:03:46 +03:00
Sebastian Dröge
3492c81fcf Update Rust examples to latest bindings versions 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/21>
 2020-07-31 11:59:58 +03:00
Seungha Yang
61d200a957 Port to gst_print* family 
g_print* would print broken string on Windows
See also https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/258

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/20>
 2020-07-27 16:28:33 +09:00
Tim-Philipp Müller
38d6a5873a Back to development 2020-07-03 02:04:21 +01:00
Tim-Philipp Müller
a8510e63d1 Release 1.17.2 2020-07-03 00:37:47 +01:00
Philippe Normand
234dff8dbb webrtc: Add Janus video-room example 
This Rust crate provides a program able to connect to a Janus instance using
WebSockets and send a live video stream to the videoroom plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/15>
 2020-06-29 14:08:51 +01:00
Matthew Waters
f5d9471639 webrtc/test: check if selenium is available before attempting to add tests 
Fixes the following error

File "/builds/vivia/gst-plugins-bad/gst-build/build/../subprojects/gst-examples/webrtc/check/basic.py", line 5, in <module>
     from selenium import webdriver

ModuleNotFoundError: No module named 'selenium'
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/17>
 2020-06-25 22:11:33 +10:00
Matthew Waters
204945b902 webrtc: indent sources 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/16>
 2020-06-25 18:36:22 +10:00
Matthew Waters
e1c3dad258 webrtc: update for move to gst-examples 
- Integrate with the build system.
- Some README updates.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/16>
 2020-06-25 18:36:22 +10:00
Matthew Waters
a88e90fa9e Move gstwebrtc-demos into gst-examples 
Original repository location: https://github.com/centricular/gstwebrtc-demos
 2020-06-25 18:36:22 +10:00
Nirbheek Chauhan
d44b2316fa sendonly: Don't assume we're building on UNIX 
Fixes https://github.com/centricular/gstwebrtc-demos/issues/203
 2020-06-25 18:36:18 +10:00
Tim-Philipp Müller
01882c92d1 Back to development 2020-06-20 00:28:41 +01:00
Tim-Philipp Müller
5f8bf174e8 Release 1.17.1 2020-06-19 19:28:16 +01:00
Nirbheek Chauhan
751d06af6f signalling: Fix simple-server script name in Dockerfile 
Fixes https://github.com/centricular/gstwebrtc-demos/issues/202
 2020-06-18 23:34:48 +10:00
Corey Cole
17f84bfd81 fix: python webrtc_sendrecv.py typo 2020-06-18 23:34:48 +10:00
Nirbheek Chauhan
0776def18c simple_server: asyncio TimeoutError has moved 
We didn't notice this because the logging was broken.
 2020-06-18 23:34:48 +10:00