Sebastian Dröge
fc5bad5f75
examples: webrtc: rust: Fix a couple of minor clippy warnings
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3928 >
2023-02-10 11:43:00 +00:00
Sebastian Dröge
28ab612a88
examples: webrtc: rust: Update to gstreamer-rs 0.20
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3928 >
2023-02-10 11:43:00 +00:00
Nirbheek Chauhan
033a71e405
webrtc examples: Use webrtc.gstreamer.net
...
Actually just a CNAME to webrtc.nirbheek.in for now, but it allows
replacement / hosting without my involvement, so reduces the bus
factor.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3802 >
2023-02-04 13:37:02 +00:00
Tim-Philipp Müller
06e9d78ade
gst-examples: drop use of GSlice allocator
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784 >
2023-02-03 17:48:09 +00:00
Matthew Waters
b134433e0b
examples/webrtc-sendrecv: add some dot file dumps on async-done and error messages
...
Just as a helpful thing if debugging is needed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3823 >
2023-01-30 05:22:59 +00:00
Nirbheek Chauhan
32e8ff4e2a
webrtc_sendrecv.py: Fix PEP8 warnings in CI lint
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3742 >
2023-01-25 16:53:17 +00:00
Nirbheek Chauhan
6a83602601
webrtc_sendrecv.py: Handle LATENCY messages
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3742 >
2023-01-25 16:53:17 +00:00
Nirbheek Chauhan
5500c228f6
webrtc_sendrecv.py: Add bus message handling
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3742 >
2023-01-25 16:53:17 +00:00
Nirbheek Chauhan
9b2404e76d
webrtc_sendrecv.py: Add support for using H264 encoding
...
Currently only works when we are creating the offer or the offer only
contains H264.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3742 >
2023-01-25 16:53:17 +00:00
Nirbheek Chauhan
6f99faa080
webrtc_sendrecv.py: Use sine wave for audio instead of red-noise
...
Makes it easier to notice when there's packet loss or other audio
distortion.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3742 >
2023-01-25 16:53:17 +00:00
Tim-Philipp Müller
41c69372b5
Back to development
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3775 >
2023-01-23 23:04:53 +00:00
Tim-Philipp Müller
f13c65d977
Release 1.22.0
2023-01-23 19:41:07 +00:00
Sebastian Dröge
4e86c77270
examples: webrtc: rust: Update dependencies
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758 >
2023-01-20 11:36:57 +00:00
Sebastian Dröge
f45136827b
examples: webrtc: multiparty-sendrecv: rust: Remove unnecessary macro recursion limit annotation
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758 >
2023-01-20 11:36:57 +00:00
Sebastian Dröge
bf4a3c89cd
examples: webrtc: sendrecv: rust: Implement OFFER_REQUEST
handling
...
Allow requesting an offer from the peer if we're joining a call with a
peer, and allow the peer to request an offer from us if waiting for an
incoming call.
This implements all 4 variants the protocol allows for.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758 >
2023-01-20 11:36:57 +00:00
Sebastian Dröge
638465908e
examples: webrtc: sendrecv: rust: Allow providing our ID via the commandline
...
Otherwise it continues to use a random ID as before.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758 >
2023-01-20 11:36:57 +00:00
Sebastian Dröge
541c637910
examples: webrtc: sendrecv: rust: Implement TWCC support in both directions
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758 >
2023-01-20 11:36:57 +00:00
Sebastian Dröge
6541dccaea
examples: webrtc: rust: Set keyframe-max-dist=2000 and picture-id-mode=15-bit for VP8 and perfect-timestamps=true for audio
...
This makes it in sync with the C sendrecv and generally behaves better.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758 >
2023-01-20 11:36:57 +00:00
Sebastian Dröge
083b9f2a6e
examples: webrtc: sendrecv: rust: Use the correct payload types if the remote is the offerer
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758 >
2023-01-20 11:36:57 +00:00
Sebastian Dröge
ac1d10f80c
gst-examples: Update Rust dependencies
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3750 >
2023-01-19 10:40:32 +02:00
Tim-Philipp Müller
a9ec35b1ca
Release 1.21.90
2023-01-13 19:08:48 +00:00
Sebastian Dröge
085e6c036a
android: Update minimum SDK version to Android 21
...
Otherwise we can't bump the minimum version of the cerbero build without
it breaking linking of the applications.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3717 >
2023-01-12 20:11:14 +00:00
Olivier Crête
b7c0e8bc84
webrtc examples: Force regular non-MULTIOPUS
...
Using MULTIOPUS breaks with most browsers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3675 >
2023-01-04 12:02:25 +00:00
Olivier Crête
c7bc6bc064
webrtc-unidirectional: Avoid critical
...
Don't unref the parameter passed to a signal, it's always owned by
the caller. Fixes a GLib critical.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3675 >
2023-01-04 12:02:25 +00:00
Sebastian Dröge
c739fcbe41
examples: webrtc: Add handling of the LATENCY messages to the Rust examples
...
Without this the configured latency on the pipeline will be wrong.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609 >
2022-12-20 13:10:27 +02:00
Sebastian Dröge
284d22437e
examples: webrtc: Update dependencies
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609 >
2022-12-20 13:06:43 +02:00
Sebastian Dröge
ec6290d63f
examples: webrtc: Remove the bus watch at the end
...
Otherwise a file descriptor will be leaked.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609 >
2022-12-20 13:03:44 +02:00
Sebastian Dröge
1f4f338d85
examples: webrtc: Add handling of the LATENCY messages to the C examples
...
Without this the configured latency on the pipeline will be wrong.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609 >
2022-12-20 13:03:15 +02:00
Sebastian Dröge
d10981f7b9
examples: webrtc: Add bus handling to the Android and C sendrecv examples
...
Without a bus, messages will just pile up and errors are not handled at
all. Also without handling the LATENCY messages the latency configured
on the pipeline will be wrong.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609 >
2022-12-20 13:02:08 +02:00
Seungmin Kim
0db1ff532d
Change GstSdp.sdp_message_parse_buffer to GstSdp.SDPMessage.new_from_text in examples
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3477 >
2022-12-16 10:40:41 +00:00
Nirbheek Chauhan
7fd8e4001c
webrtc/signalling: Give a helpful error when starting a double-session
...
If the peer is already in a session and tries to start a new one, give
them a helpful error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2460 >
2022-12-12 15:08:23 +00:00
byran77
1e5abde7b1
gst-examples: webrtc: signalling: simple-server Fix condition when calling a busy peer
...
When a session request is coming in, 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.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2460 >
2022-12-12 15:08:23 +00:00
Guillaume Desmottes
cbab7ffefb
examples: webrtc: fix unidirectional pipeline
...
'autoaudiosrc' does not have a 'is-live' property.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3550 >
2022-12-09 13:49:44 +01:00
Guillaume Desmottes
ebfbdf9076
examples: webrtc: fix plugins check
...
`videoconvert` and `videoscale` are now part of the `videoconvertscale`
plugin, see d11f13f476
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3529 >
2022-12-05 17:04:57 +00:00
Tim-Philipp Müller
1f65d7cc5c
Back to development
2022-12-05 02:29:08 +00:00
Tim-Philipp Müller
fd6a3948c6
Release 1.21.3
2022-12-05 01:28:21 +00:00
Jan Schmidt
8177588250
examples/sendrecv: Remove extra unref of webrtcbin
...
The code now constructs webrtcbin with a floating ref and then
gives it to the pipeline. The extra unref is one too many.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3436 >
2022-11-19 19:51:54 +11:00
Jan Schmidt
f2ae481a69
examples/webrtc: Configure payload types
...
MR 2398 broke the webrtc sendrecv example
by not configuring the payload types, so both audio and video streams
get sent on payload 96.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3434 >
2022-11-19 13:12:58 +11:00
Tim-Philipp Müller
db450689db
Back to development
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3358 >
2022-11-08 02:08:08 +00:00
Tim-Philipp Müller
3e29ac35c4
Release 1.21.2
2022-11-07 23:54:03 +00:00
Nicolas Dufresne
4fb9f2a2b4
meson: Fix path for webrtc validate tests
...
This fixes a crash when trying to run gst-validate-launcher from inside
the meson devenv. The error was:
ModuleNotFoundError: No module named 'observer'
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3273 >
2022-10-26 18:16:25 +00:00
Patrick Griffis
2a59e8af97
webrtc: Fix double free in webrtc-recvonly-h264 demo
...
The "message" signal does not transfer ownership of the GBytes passed
to it so calling g_bytes_unref() on it is incorrect.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3257 >
2022-10-24 22:16:44 +00:00
Sebastian Dröge
7193a601b3
examples: webrtc: Update to gstreamer-rs 0.19 release
...
Also update the macOS workaround for gstreamer-gl requiring a
`NSRunLoop` / `NSApp` on the main thread, and update from strucopt to
clap 4.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3255 >
2022-10-24 11:50:09 +00:00
Patrick Griffis
d0e2b31470
webrtc: Fix critical in webrtc-recvonly-h264 example
...
This signal only takes 2 properties yet a third was passed.
This would cause a critical in GLib.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3252 >
2022-10-23 22:51:28 +00:00
Sam Van Den Berge
094b251901
examples: webrtc: mp-sendrecv: add bus handler
...
Without this bus handler, messages posted to the bus will keep a ref to
their source elements, preventing them from being disposed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3219 >
2022-10-19 00:51:44 +00:00
Sam Van Den Berge
93ed51cbb2
examples: webrtc: mp-sendrecv: set element states to NULL after removing them from pipeline
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3219 >
2022-10-19 00:51:44 +00:00
Sam Van Den Berge
17c111d2b9
examples: webrtc: mp-sendrecv: remove wrong gst_object_unrefs
...
In !2958 some gst_object_unrefs were added. However these two don't
belong there because ownership is transfered due to the gst_bin_add_many
call a bit above.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3219 >
2022-10-19 00:51:44 +00:00
Sebastian Dröge
64c376b5b2
webrtc: Add/fix various annotations
...
And mark string parameters as const.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194 >
2022-10-18 08:56:58 +00:00
Matthew Waters
d586c2cc28
examples/webrtc: don't use factory_make_full() for enums
...
They are not currently translated into their respective enum values and
will produce an error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3210 >
2022-10-18 01:30:37 +00:00
Sam Van Den Berge
07d8e53aac
examples/webrtc/signalling: Fix compatibility with Python 3.10
...
Fix asyncio throwing a deprecation warning when using
asyncio.get_event_loop().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3196 >
2022-10-17 11:46:51 +02:00