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105 commits

Author SHA1 Message Date
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
Jan Schmidt 8177588250 examples/sendrecv: Remove extra unref of webrtcbin 
The code now constructs webrtcbin with a floating ref and then
gives it to the pipeline. The extra unref is one too many.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3436>
 2022-11-19 19:51:54 +11:00
Jan Schmidt f2ae481a69 examples/webrtc: Configure payload types 
MR 2398 broke the webrtc sendrecv example
by not configuring the payload types, so both audio and video streams
get sent on payload 96.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3434>
 2022-11-19 13:12:58 +11:00
Tim-Philipp Müller db450689db Back to development 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3358>
 2022-11-08 02:08:08 +00:00
Tim-Philipp Müller 3e29ac35c4 Release 1.21.2 2022-11-07 23:54:03 +00:00
Nicolas Dufresne 4fb9f2a2b4 meson: Fix path for webrtc validate tests 
This fixes a crash when trying to run gst-validate-launcher from inside
the meson devenv. The error was:

  ModuleNotFoundError: No module named 'observer'

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3273>
 2022-10-26 18:16:25 +00:00
Patrick Griffis 2a59e8af97 webrtc: Fix double free in webrtc-recvonly-h264 demo 
The "message" signal does not transfer ownership of the GBytes passed
to it so calling g_bytes_unref() on it is incorrect.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3257>
 2022-10-24 22:16:44 +00:00
Sebastian Dröge 7193a601b3 examples: webrtc: Update to gstreamer-rs 0.19 release 
Also update the macOS workaround for gstreamer-gl requiring a
`NSRunLoop` / `NSApp` on the main thread, and update from strucopt to
clap 4.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3255>
 2022-10-24 11:50:09 +00:00
Patrick Griffis d0e2b31470 webrtc: Fix critical in webrtc-recvonly-h264 example 
This signal only takes 2 properties yet a third was passed.
This would cause a critical in GLib.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3252>
 2022-10-23 22:51:28 +00:00
Sam Van Den Berge 094b251901 examples: webrtc: mp-sendrecv: add bus handler 
Without this bus handler, messages posted to the bus will keep a ref to
their source elements, preventing them from being disposed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3219>
 2022-10-19 00:51:44 +00:00
Sam Van Den Berge 93ed51cbb2 examples: webrtc: mp-sendrecv: set element states to NULL after removing them from pipeline 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3219>
 2022-10-19 00:51:44 +00:00
Sam Van Den Berge 17c111d2b9 examples: webrtc: mp-sendrecv: remove wrong gst_object_unrefs 
In !2958 some gst_object_unrefs were added. However these two don't
belong there because ownership is transfered due to the gst_bin_add_many
call a bit above.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3219>
 2022-10-19 00:51:44 +00:00
Sebastian Dröge 64c376b5b2 webrtc: Add/fix various annotations 
And mark string parameters as const.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
 2022-10-18 08:56:58 +00:00