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115 commits

Author SHA1 Message Date
Tim-Philipp Müller
79cdbc37d5 Back to development 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5826>
 2023-12-18 13:52:12 +00:00
Tim-Philipp Müller
4af14db10e Release 1.22.8 2023-12-18 12:09:37 +00:00
Tim-Philipp Müller
b1067c023d Back to development 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5650>
 2023-11-13 14:57:09 +00:00
Tim-Philipp Müller
4d13eddc8b Release 1.22.7 2023-11-13 11:04:22 +00:00
Tim-Philipp Müller
0cccb1c9a6 Back to development 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5371>
 2023-09-20 19:41:00 +01:00
Tim-Philipp Müller
ddb4fbe431 Release 1.22.6 2023-09-20 18:10:57 +01:00
Tim-Philipp Müller
60120003c0 Back to development 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5082>
 2023-07-20 16:57:47 +01:00
Tim-Philipp Müller
bf6ce1d64a Release 1.22.5 2023-07-20 15:22:48 +01:00
Tim-Philipp Müller
adafbe4fad Back to development 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4907>
 2023-06-20 19:10:38 +01:00
Tim-Philipp Müller
064711d8b3 Release 1.22.4 2023-06-20 17:42:25 +01:00
Tim-Philipp Müller
9994bbbd4c Back to development 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4679>
 2023-05-19 12:36:19 +01:00
Tim-Philipp Müller
ecd471f5ea Release 1.22.3 2023-05-19 09:23:19 +01:00
Tim-Philipp Müller
d838d8dd1b Back to development 2023-04-12 00:31:17 +01:00
Tim-Philipp Müller
a8f569e801 Release 1.22.2 2023-04-11 17:29:28 +01:00
Tim-Philipp Müller
3acf83be50 Back to development 2023-03-04 16:13:04 +00:00
Tim-Philipp Müller
3ab8a0bc3e Release 1.22.1 2023-03-04 13:42:32 +00:00
Nirbheek Chauhan
f6e672f27f 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/3892>
 2023-02-04 16:01:08 +01:00
Nirbheek Chauhan
26ee3d83fb webrtc_sendrecv.py: Fix PEP8 warnings in CI lint 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3816>
 2023-01-28 03:05:20 +00:00
Nirbheek Chauhan
dff9f5151b webrtc_sendrecv.py: Handle LATENCY messages 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3816>
 2023-01-28 03:05:20 +00:00
Nirbheek Chauhan
361f0f406b webrtc_sendrecv.py: Add bus message handling 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3816>
 2023-01-28 03:05:20 +00:00
Nirbheek Chauhan
78c928eefe 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/3816>
 2023-01-28 03:05:20 +00:00
Nirbheek Chauhan
6179d5ef61 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/3816>
 2023-01-28 03:05:19 +00:00
Tim-Philipp Müller
e87857a210 Back to development 2023-01-25 16:46:42 +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