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136 commits

Author SHA1 Message Date
Tim-Philipp Müller
756064b9c3 Back to development 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6261>
 2024-03-05 12:58:57 +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
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