Tim-Philipp Müller
5630be477f
Back to development after 1.24.9
2024-10-30 20:40:12 +00:00
Tim-Philipp Müller
b309f90bfd
Release 1.24.9
2024-10-30 20:33:30 +00:00
Tim-Philipp Müller
29b77fb2df
Back to development after 1.24.8
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7549 >
2024-09-19 12:12:53 +02:00
Tim-Philipp Müller
c396e7da4c
Release 1.24.8
2024-09-19 12:01:21 +02:00
Tim-Philipp Müller
29b51fb13d
Back to development after 1.24.7
2024-08-21 12:33:28 +01:00
Tim-Philipp Müller
0f1e984e41
Release 1.24.7
2024-08-21 12:25:15 +01:00
Tim-Philipp Müller
ad0cc551e1
Back to development after 1.24.6
2024-07-29 16:48:02 +01:00
Tim-Philipp Müller
8d175ea255
Release 1.24.6
2024-07-29 16:41:37 +01:00
Tim-Philipp Müller
a58953cbf6
Back to development after 1.24.5
2024-06-20 13:02:19 +01:00
Tim-Philipp Müller
3c66f10e21
Release 1.24.5
2024-06-20 12:54:15 +01:00
Tim-Philipp Müller
03cfca1033
Back to development after 1.24.4
2024-05-29 13:51:27 +03:00
Tim-Philipp Müller
9137f539a0
Release 1.24.4
2024-05-29 13:44:50 +03:00
Tim-Philipp Müller
204ba57e81
Back to development after 1.24.3
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6772 >
2024-04-30 00:36:59 +01:00
Tim-Philipp Müller
da69285863
Release 1.24.3
2024-04-30 00:15:23 +01:00
Tim-Philipp Müller
934af761fd
Back to development after 1.24.2
2024-04-10 00:15:42 +01:00
Tim-Philipp Müller
2d82731515
Release 1.24.2
2024-04-09 21:48:55 +01:00
Guillaume Desmottes
5fcc15514f
examples: set perfect-timestamp=true on opusenc
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Fix audio streaming on Chrome, see https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1524
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6523 >
2024-04-03 11:35:08 +00:00
Tim-Philipp Müller
175d116f94
Back to development
2024-03-22 01:38:06 +01:00
Tim-Philipp Müller
0d0a1d9d16
Release 1.24.1
2024-03-21 21:47:53 +01:00
Tim-Philipp Müller
2c7bb61580
Back to development
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6264 >
2024-03-05 14:00:10 +00:00
Tim-Philipp Müller
b125253cad
Release 1.24.0
2024-03-04 23:59:25 +00:00
Edward Hervey
a3980f4838
docs: Use Discourse and Matrix as prefered communication channels
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Part of: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6220
2024-02-27 09:35:47 +01:00
Tim-Philipp Müller
d474de8ff0
Release 1.23.90
2024-02-23 18:20:11 +00:00
Tim-Philipp Müller
88412ef100
Back to development
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6126 >
2024-02-15 16:38:53 +00:00
Tim-Philipp Müller
88751d4110
Release 1.23.2
2024-02-15 15:37:17 +00:00
Sebastian Dröge
0871d1edc4
examples: webrtc: Update dependencies in Rust examples
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6078 >
2024-02-09 09:35:10 +00:00
Tim-Philipp Müller
2111d6f015
Back to development
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6066 >
2024-02-06 18:29:31 +00:00
Tim-Philipp Müller
9255e397f0
Release 1.23.1
2024-02-06 16:43:27 +00:00
Eva Pace
e5194d4c45
examples: webrtc: update sendrecv dependencies
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5809 >
2023-12-14 21:14:48 +00:00
Eva Pace
c43c550c06
examples: webrtc: update multiparty-sendrecv dependencies
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5809 >
2023-12-14 21:14:48 +00:00
Eva Pace
009e286375
examples: webrtc: ignore rust target folders
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5809 >
2023-12-14 21:14:48 +00:00
Eva Pace
914d46c02f
examples: webrtc: update janus dependencies
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5809 >
2023-12-14 21:14:48 +00:00
Philippe Normand
66373721d5
gstplay: Add a minimal documentation header
...
Also mentioning the need to set the bus to flushing state before disposing the
player in order to avoid reference cycles.
Fixes #3107
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5631 >
2023-11-11 10:48:27 +00:00
Jeff Wilson
5c8fff0807
examples: webrtc: Actually create the custom ICE agent
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5568 >
2023-10-30 19:58:59 +00:00
Nirbheek Chauhan
62e33e04ea
webrtc_sendrecv.py: Allow using a camera instead of test sources
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5504 >
2023-10-19 05:47:03 +00:00
Eva Pace
003e419ff5
examples: webrtc: rust: i64 -> u64 for session and handle ids
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5307 >
2023-09-11 06:21:32 +00:00
Sebastian Dröge
ae28e1035e
examples: webrtc: rust: Update to gstreamer-rs 0.21
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5181 >
2023-08-14 09:06:08 +00:00
Matthew Waters
ce81b81d3f
examples: update ios deplyoment target to 12.0
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5111 >
2023-07-27 13:05:37 +00:00
Nirbheek Chauhan
639f8a24ae
webrtc/js: Support renegotiation during a call correctly
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When a video track is muted, hide the video element to differentiate
it from a track that is stuck because we stopped receiving RTP data.
Show it again when it is unmuted.
When a video track is removed, remove the video element. It will be
re-added on renegotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5045 >
2023-07-19 13:01:49 +00:00
Nirbheek Chauhan
57b6c743ef
webrtc/js: Remove obsolete mozilla stun server
...
Mozilla's public stun server is gone. Remove it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5045 >
2023-07-19 13:01:49 +00:00
Nirbheek Chauhan
80603746af
webrtc/js: Support pressing "enter" to connect
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I press "enter" every time which doesn't work and then I click
"Connect", so let's fix that.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5045 >
2023-07-19 13:01:49 +00:00
Tim-Philipp Müller
19502f5c1a
gst-examples: prepare for removal of kate plugin from cerbero
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See https://gitlab.freedesktop.org/gstreamer/cerbero/-/merge_requests/1114
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4769 >
2023-06-05 06:45:54 +00:00
Matthew Waters
c46805cb0d
examples/webrtc/android: fix build
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Was missing a GstBus *bus; local variable
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4747 >
2023-06-03 23:21:35 +00:00
Matthew Waters
63b6071a4a
examples/webrtc/android: update for videoconvertscale addition
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4747 >
2023-06-03 23:21:34 +00:00
Matthew Waters
5889059cff
examples/android: specify the exact NDK (r25c) version to use
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4747 >
2023-06-03 23:21:34 +00:00
Stéphane Cerveau
dd17beb681
gstreamer-full: add full static support
...
Allow a project to use gstreamer-full as a static library
and link to create a binary without dependencies.
Introduce the option 'gst-full-target-type' to
select the build type, dynamic(default) or static.
In gstreamer-full/static build configuration gstreamer (gst.c)
needs the symbol gst_init_static_plugins which is defined
in gstreamer-full.
All the tests and examples are linking with gstreamer but the
symbol gst_init_static_plugins is only defined in the gstreamer-full
library. gstreamer-full can not be built first as it needs to know what plugins
will be built.
One option would be to build all the examples and tests after
gstreamer-full as the tools.
Disable tools build in subprojects too as it will be built at the end of
build process.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4128 >
2023-05-31 15:17:11 +00:00
Nirbheek Chauhan
aa1fa50129
webrtc_sendrecv.py: Add AV1 support when creating the offer
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Requires svtav1enc at present for simplicity.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4644 >
2023-05-17 16:20:36 +00:00
Nirbheek Chauhan
61e536b546
webrtc_sendrecv.py: Fix warnings about gi version
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4644 >
2023-05-17 16:20:36 +00:00
François Laignel
1abc8aa733
examples: webrtc/janus/rust: add mandatory ws HTTP request headers
...
Trying to run the `janus` Rust `gst-example`, `tungstenite` reports:
> Missing, duplicated or incorrect header sec-websocket-key
Indeed, all mandatory headers from the following list are missing
(code from `tungstenite:🤝 :client::generate_request`):
```rust
const WEBSOCKET_HEADERS: [&str; 5] =
["Host", "Connection", "Upgrade", "Sec-WebSocket-Version", KEY_HEADERNAME];
```
These headers are mandatory for the websocket handshake. This feature is
selected by async-tungstenite.
Prior to this commit, the HTTP request was created with the header
"Sec-WebSocket-Protocol" only. Delegating the request creation to tungstenite
adds the missing headers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4240 >
2023-03-22 09:48:28 +00:00
Tim-Philipp Müller
9e1a33334b
examples: iOS: GstPlay: update for pending ivorbisdec plugin removal
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See https://gitlab.freedesktop.org/gstreamer/cerbero/-/merge_requests/1103
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4075 >
2023-02-27 17:40:43 +00:00