Commit graph

26 commits

Author SHA1 Message Date
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
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
Sebastian Dröge
180e1ce24c Update dependencies of Rust demos 2020-06-18 23:34:48 +10:00
Costa Shulyupin
8c4345da7d android, mp-webrtc-sendrecv, sendonly: cleanup
webrtc-unidirectional-h264.c: removed empty lines

android: removed unused var
2020-04-16 17:34:11 +02:00
Costa Shulyupin
804c0c2f5e gst-indent 2020-04-14 14:40:37 +03:00
Sebastian Dröge
699b830213 Update Rust examples to async-tungstenite 0.4 2020-02-01 15:21:08 +02:00
Sebastian Dröge
42c6eac7f1 Update dependencies of Rust examples and simplify slightly 2020-01-23 08:36:21 +02:00
Sebastian Dröge
d995a00774 Update Rust examples to async-tungstenite 0.3 2020-01-05 11:41:31 +02:00
Sebastian Dröge
f5e4df464f Update Rust demos to gstreamer 0.15 bindings release 2019-12-19 01:04:01 +02:00
Sebastian Dröge
5e18b460b3 multiparty/rust: Add Rust version of multiparty demo
Different to the C version this also mixes all participants into a grid
with videomixer.
2019-11-29 20:49:46 +01:00
Seungha Yang
60dbf27896 Add meson build script
make build easy with meson
2019-07-02 14:40:36 +01:00
Bernhard Jung
62469f1155 unref sinkpad also in mp version 2019-07-01 13:21:20 +03:00
Bernhard Jung
92050d6a59 do no use gst_element_link but gst_pad_link in pad-added callbacks to prevent situations where
on multiple incoming streams they might not get linked correctly and leave a stream unconnected
2019-07-01 13:21:20 +03:00
Jason Sun
92bce589d8 Improve building documentation
- Add apt-get install lines for Ubuntu 18.04
- add gstreamer-webrtc-1.0 and gstreamer-sdp-1.0 to CFLAGS
- make the CLAGS match LIBS in Makefile dependencies
2018-11-22 05:23:15 +00:00
Jan Alexander Steffens (heftig)
fd1d53b04a on_server_message: Do not unref message GBytes
We don't own the reference. Since GLib 2.58, the g_bytes_unref that
follows the signal emission in libsoup loudly complains about the
attempt to underflow the refcount.
2018-09-21 13:12:43 +00:00
Eloi Bail
d6741c1f80 mp-webrtc-sendrecv.c: add missing comma in the list of package required
A comma is missing in the list of package required. Thus the package
'srtprtpmanager' is checked instead of packages srtp and rtpmanager.
2018-04-03 15:04:57 +00:00
Nirbheek Chauhan
82314cabbb Don't use strict ssl certificate checking for localhost
When using localhost signalling servers, we don't want to use
strict ssl because it's probably using a self-signed certificate
and there's no need to do certificate checking over localhost anyway.
2018-03-31 10:27:05 +05:30
Nirbheek Chauhan
0e1be2a63f Add Makefiles for all C demos 2018-03-23 19:00:37 +05:30
Nirbheek Chauhan
2d2bc0fe0e Fix compiler warnings in all C demos 2018-03-23 19:00:37 +05:30
Nirbheek Chauhan
55e86469d9 Check for all necessary plugins at startup
People seem to be having problems ensuring that they have all the
right plugins built, so make it a bit easier for them.
2018-03-10 01:54:48 +05:30
Nirbheek Chauhan
fa2adc717b Fix crash on Windows by delimiting option entries with NULL
Also use more verbose forms of g_assert which print values on failure
2018-03-08 20:10:55 +05:30
Mathieu Duponchelle
e5c5767298 Update to new promise API 2017-11-22 22:28:55 +10:00
Nirbheek Chauhan
0c5e799952 multiparty sendrecv: Add a queue before the audio sink
Missed this, fixes the bug where removing a peer causes the pipeline to
get stuck. However, when peers leave, there is still a chance that the
pipeline will get stuck.
2017-10-30 13:24:21 +05:30
Nirbheek Chauhan
9b1a0e5389 WIP: Add a new multiparty sendrecv gstreamer demo
You can join a room and an audio-only call will be started with all
peers in that room. Currently uses audiotestsrc only.

BUG: With >2 peers in a call, if a peer leaves, the pipeline stops
     outputting data from the remaining peers to the (audio) sink.

TODO: JS code to allow a browser to join the call
TODO: Cleanup pipeline when a peer leaves
TODO: Add ICE servers to allow calls over the Internet
TODO: Perhaps setup a TURN server as well
2017-10-30 09:14:29 +05:30