mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-06-30 03:40:41 +00:00
Code Issues 1 Projects Releases Wiki Activity
118023 commits 44 branches 1505 tags 414 MiB
Commit graph

120 commits

Author SHA1 Message Date
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
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