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Author SHA1 Message Date
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
Nirbheek Chauhan
77ae10ab66 simple_server: Restart when the certificate changes 
Reload the SSL context and restart the server if the certificate
changes. Without this, new connections will continue to use the old
expired certificate.
 2020-06-18 23:34:48 +10:00
Nirbheek Chauhan
4761396d87 simple_server: Abstract out ssl context generation 2020-06-18 23:34:48 +10:00
Nirbheek Chauhan
7b96b06752 simple_server: Make the server class loop-aware 
First step in making the class able to manage its own state.
 2020-06-18 23:34:48 +10:00
Nirbheek Chauhan
b8c1bd1fa3 simple_server: Fix init of websockets log handler 
This has changed since the original code was written:
https://websockets.readthedocs.io/en/stable/cheatsheet.html#debugging
 2020-06-18 23:34:48 +10:00
Nirbheek Chauhan
78df1ca74c simple_server: Correctly pass health option 
It was completely ignored. Also don't de-serialize options. Just parse
them directly in `__init__`. Less error-prone.
 2020-06-18 23:34:48 +10:00
Sebastian Dröge
180e1ce24c Update dependencies of Rust demos 2020-06-18 23:34:48 +10:00
Philippe Normand
c0f303eacf janus: Remove unused parameters and refactor 2020-05-14 11:04:37 +01:00
Matthew Waters
219415dbf6 add vulkan example for android 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/14>
 2020-05-12 16:24:38 +10:00
Jan Schmidt
255fef3896 webrtc-recvonly-h264: Add a recvonly standalone example. 
This example sets up a recvonly H.264 transceiver and receives
H.264 from a peer, while sending bi-directional Opus audio.
 2020-05-09 19:13:52 +10:00
Jan Schmidt
8da8375986 sendonly: Fix transceivers leak. 
Make sure to unref the transceivers array after use.
 2020-05-09 19:13:52 +10:00