Commit graph

102 commits

Author SHA1 Message Date
Sebastian Dröge
542fa9be89 examples: rust: Update various dependencies
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6999>
2024-06-06 09:44:59 +03:00
Jan Schmidt
362a73218b webrtc.js: Fix a small typo
Fix a typo caused by referencing the event struct after
it's already been destructured

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6547>
2024-04-07 21:50:17 +00:00
Guillaume Desmottes
ed54734825 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/6512>
2024-04-02 22:08:31 +00:00
L. E. Segovia
31baaf8906 Android tutorials: add outputs to gitignore
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6419>
2024-03-27 08:20:27 +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
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
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
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
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
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
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
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
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
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