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153 commits

Author SHA1 Message Date
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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
Philippe Normand
906b90287c webrtcbin: Relay add-ice-candidate errors from Ice implementation to Application 
The `add_candidate` vfunc of the GstWebRTCICE interface gained a GstPromise
argument, which is an ABI break. We're not aware of any external user of this
interface yet so we think it's OK.

This change is useful in cases where the application needs to bubble up errors
from the underlying ICE agent, for instance when the agent was given an invalid
ICE candidate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3960>
 2023-02-27 09:09:47 +00:00
Thibault Saunier
0f577533e6 examples: Add an option to disable tests 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3930>
 2023-02-10 12:59:55 +00:00