Commit graph

117 commits

Author SHA1 Message Date
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
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
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
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
Sebastian Dröge
d10981f7b9 examples: webrtc: Add bus handling to the Android and C sendrecv examples
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
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
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
'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
`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