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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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'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
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`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
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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
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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
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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
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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
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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