Commit graph

309 commits

Author SHA1 Message Date
Mathieu Duponchelle
b3ace3678b webrtcsink: fix naming of error dot files for discovery pipelines
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1843>
2024-10-03 14:35:45 +00:00
Guillaume Desmottes
d9e8f4054c webrtc: allow PAR change in webrtcsink input caps
We are already allowing resolution changes which can lead to change in
pixel-aspect-ratio.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1830>
2024-09-30 14:40:48 +02:00
Sebastian Dröge
dcb072ee23 webrtc: livekit: Set connection earlier during setup
Otherwise it's not available yet when handling the initial participants
that are already in the session when joining.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1794>
2024-09-30 13:04:24 +03:00
Sebastian Dröge
cd2b641321 livekitwebrtcsrc: Add API for disabling/enabling a track
A disabled track is still negotiated but no data is sent for it
temporarily until it is enabled again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1794>
2024-09-30 13:04:24 +03:00
Sebastian Dröge
27dc76826e livekitwebrtcsrc: Add pad properties for various LiveKit participant / track metadata
The content of the TrackInfo and ParticipantInfo structs is exposed as
gst::Structure.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1794>
2024-09-30 13:04:24 +03:00
Mathieu Duponchelle
87c6719e1d webrtcsink: add define-encoder-bitrates signal
When congestion control is used for a session with multiple encoders,
the default implementation simply divides the overall bitrate equally
between encoders.

This is not always desirable, and this patch exposes a new signal
that users can register to, with two arguments:

* The overall bitrate to allocate
* A structure with an encoder.stream_name -> bitrate mapping

Handlers should return a similar structure with a custom mapping.

An example is also provided.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1792>
2024-09-25 15:19:44 +00:00
François Laignel
f532d523b2 webrtcsink: fix RFC7273 attributes
RFC7273 related attributes are set in the SDP offer by passing them via the
transceiver `codec-preferences` signal. These attributes are intended to be set
at the media level so they must be prefixed by `a-` in the `Caps` argument to
the signal. Otherwise they end up under `a=fmtp`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1810>
2024-09-25 09:30:48 +00:00
Mathieu Duponchelle
5c66d8c107 webrtcsrc: ensure source pad has msid when added
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1800>
2024-09-24 14:50:30 +00:00
Mathieu Duponchelle
f70482d9bc webrtcsrc: fix default msid property value
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1800>
2024-09-24 14:50:30 +00:00
Mathieu Duponchelle
a85b0cb72e webrtcsrc: expose MSID property on source pad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1789>
2024-09-20 09:31:57 +03:00
Mathieu Duponchelle
6a23ae168f webrtcsink: implement mechanism to forward metas over control channel
It may be desirable for the frontend to receive ancillary information
over the control channel.

Such information includes but is not limited to time code metas, support
for other metas (eg custom meta) might be implemented in the future, as
well as downstream events.

This patch implements a new info message, probes buffers that arrive at
nicesink to look up timecode metas and potentially forwards them to the
consumer when the `forward-metas` property is set appropriately.

Internally, a "dye" meta is used to trace the media identifier the
packet we are about to send over relates to, as rtpfunnel bundles all
packets together.

The example frontend code also gets a minor update and now logs info
messages to the console.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1749>
2024-09-19 08:41:47 +00:00
Mathieu Duponchelle
db026ad535 gstwebrtc-api: expose API on consumer-session for munging stereo
We cannot do that by default as this is technically non-compliant,
so we need to expose API to let the user opt into it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1754>
2024-09-19 07:37:23 +00:00
Sebastian Dröge
c505d9a418 Update to async-tungstenite 0.28
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1772>
2024-09-10 09:19:18 +03:00
Arun Raghavan
e72db57179 webrtc: Fix whipclientsink name in README
The element name was changed, but the documentation wasn't updated to
match.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1764>
2024-09-03 16:44:19 -04:00
Mathieu Duponchelle
2f9bb62b6b gstwebrtc-api: create control data channel when offering
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1755>
2024-08-27 07:52:12 +02:00
Mathieu Duponchelle
4cf93ccbdb net/webrtc: Add missing npm command to README
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/589

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1746>
2024-08-22 15:46:28 +02:00
Jerome Colle
dee0e32dde webrtcsink: add nvv4l2av1enc support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1735>
2024-08-22 06:41:52 +00:00
Mathieu Duponchelle
8ad882bed5 gstwebrtc-api: address issues raised by mix matrix support
1c48d7065d was mistakenly merged too
early, and there were concerns about the implementation and API design:

The fact that the frontend had to expose a text area specifically for
sending over a mix matrix, and had to manually edit in floats into the
stringified JSON was suboptimal.

Said text area was always present even when remote control was not
enabled.

The sendControlRequest API was made more complex than needed by
accepting an optional stringifier callback.

This patch addresses all those concerns:

The deserialization code in webrtcsink is now made more clever and
robust by first having it pick a numerical type to coerce to when
deserializing arrays with numbers, then making sure it doesn't allow
mixed types in arrays (or arrays of arrays as those too must share
the same inner value type).

The frontend side simply sends over strings wrapped with a request
message envelope to the backend.

The request text area is only shown when remote control is enabled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1725>
2024-08-22 05:54:46 +00:00
Mathieu Duponchelle
5dc2d56c0e webrtcsink: store mids per-session instead of globally
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1730>
2024-08-21 21:20:40 +00:00
Mathieu Duponchelle
16ee51621e webrtcsink: fix segment format mismatch with remote offer
webrtcsink was starting the negotiation process on Ready and concurrently
moving the consumer pipeline to Playing, but when answering the remote
description was set so fast that input streams were connected (and the time
format set on appsrc) before the state change to Paused had completed.

This meant gst_base_src_start was happening after that and setting the format
back to bytes, the time segment that was next coming in then caused:

basesrc gstbasesrc.c:4255:gst_base_src_push_segment:<video_0> segment format mismatched, ignore

And the consumer pipeline errored out.

The same issue existed in theory when webrtcsink was creating the offer,
but was much harder to trigger as it required that the remote answer
came in before the state change to Paused had completed.

This commit fixes the issue by simply waiting for the state to have
changed to Paused before negotiating.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1730>
2024-08-21 21:20:40 +00:00
Mathieu Duponchelle
1c48d7065d gstwebrtc-api example: add support for requesting mix matrix
This is one example of how a consumer might send over custom upstream
event requests to the producer.

As webrtcsink will deserialize numbers in priority as integers, we need
a custom stringifying function to ensure members of the matrix array are
indeed serialized with the floating point.

An optional stringifier parameter is thus added to the
sendControlRequest API.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1711>
2024-08-15 15:42:04 +00:00
Mathieu Duponchelle
01e28ddfe2 webrtcsink: implement generic data channel control mechanism ..
.. and deprecate data channel navigation in favor of it.

A new property, "enable-data-channel-control" is exposed, when set to
TRUE a control data channel is offered, over which can be sent typed
upstream events.

This means further upstream events will be usable, for now only
navigation and custom upstream events are handled.

In addition, send response messages to notify the consumer of whether
its requests have been handled.

In the future this can also be extended to allow the consumer to send
queries, or seek events ..

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1711>
2024-08-15 15:42:04 +00:00
Mathieu Duponchelle
0a6963f7ce gstwebrtc-api: example: use http by default
That way the webpage connects with ws:/ to the signaller.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/589
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1704>
2024-08-14 14:10:04 +00:00
Sebastian Dröge
bc930122ba webrtcsrc: Make sure to always call end_session() without the state lock
This was already done in another place for the same reason: preventing a
deadlock. It's probably not correct as hinted by the FIXME comment but
better than deadlocking at least.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1701>
2024-08-13 06:04:09 +00:00
Mathieu Duponchelle
0da1c8e9c9 webrtcsink: fix assertions when finalizing
Dumping the pipeline on state changes from an async bus handler was
triggering criticals.

Instead, dump from the sync handler.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1706>
2024-08-12 09:13:06 +02:00
Mathieu Duponchelle
9080c90120 net/webrtc: add support for answering to webrtcsink
Support was added to the base class when the AWS KVS signaller was
implemented, but the default signaller still only supported the case
where the producer was creating the offer.

Also extend the javascript API

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1702>
2024-08-09 14:02:48 +02:00
Mathieu Duponchelle
a9ff9615ff net/webrtc: correct signaller debug category
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1702>
2024-08-08 18:28:43 +02:00
Mathieu Duponchelle
64f0b76f71 webrtc: update README with section on embedded signalling / web services
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1671>
2024-08-08 16:40:46 +02:00
Mathieu Duponchelle
9455e09d9f webrtcsink: expose properties for running web server
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1671>
2024-08-08 16:40:46 +02:00
Mathieu Duponchelle
b709c56478 webrtcsink: expose properties for running signalling server
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1671>
2024-08-07 19:55:00 +02:00
Sebastian Dröge
6c04b59454 webrtcsrc: Don't hold the state lock while removing sessions
Removing a session can drop its bin and during release of the bin its
pads are removed, but the pad-removed handler is also taking the state
lock.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1695>
2024-08-07 09:35:15 +00:00
Sebastian Dröge
b83b6031e5 Update etherparse and async-tungstenite dependencies
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1691>
2024-08-06 09:00:32 +03:00
Dave Lucia
3a949db720 net/webrtc: Fix turn-servers nick: user -> use
Noticed this typo

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1690>
2024-08-05 12:38:51 -04:00
Sebastian Dröge
fa060b9fa0 Fix various 1.80 clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1688>
2024-08-05 14:14:17 +03:00
Mathieu Duponchelle
86039dd5c1 webrtc-api example: do not rely on webpack / npm proxying websocket
Instead simply use the desired address directly from the reference
example, this makes it work out of the box without placing expectations
on the web server.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1674>
2024-07-30 16:29:54 +00:00
Loïc Le Page
5a1d12419f gstwebrtc-api: always include index file in dist for convenience
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1670>
2024-07-17 08:27:31 +00:00
François Laignel
34b791ff5e webrtc: add raw payload support
This commit adds support for raw payloads such as L24 audio to `webrtcsink` &
`webrtcsrc`.

Most changes take place within the `Codec` helper structure:

* A `Codec` can now advertise a depayloader. This also ensures that a format
  not only can be decoded when necessary, but it can also be depayloaded in the
  first place.
* It is possible to declare raw `Codec`s, meaning that their caps are compatible
  with a payloader and a depayloader without the need for an encoder and decoder.
* Previous accessor `has_decoder` was renamed as `can_be_received` to account
  for codecs which can be handled by an available depayloader with or without
  the need for a decoder.
* New codecs were added for the following formats:
  * L24, L16, L8 audio.
  * RAW video.

The `webrtc-precise-sync` examples were updated to demonstrate streaming of raw
audio or video.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1501>
2024-07-16 19:32:02 +00:00
François Laignel
6e9855c36b webrtcsink: fix property types for rav1enc
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/572
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1667>
2024-07-12 18:59:20 +02:00
Sebastian Dröge
a8ccfe49d9 webrtc: Require livekit-protocol < 0.3.4 due to uncoordinated breaking changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1663>
2024-07-11 20:00:24 +03:00
Sebastian Dröge
98b28d69ce Update for new debug log macro syntax
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1658>
2024-07-08 11:25:23 +03:00
leonardo salvatore
f303992e0c webrtcsink: initial support for vpuenc_h264 encoder for imx8mp, default values set to cover a common streaming scenario
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1639>
2024-07-01 07:34:04 +00:00
Sebastian Dröge
47d62b6d78 Update for new clone/closure macro syntax
Also fix various weak/strong references in the webrtc plugin, and make
sure to pass the object to debug log functions in every place.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1625>
2024-06-21 11:54:58 +03:00
Sebastian Dröge
9b323a6519 Use Option::is_some_and(...) instead of Option::map_or(false, ...)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1630>
2024-06-19 13:03:37 +00:00
Sebastian Dröge
23d998a1db Slightly improve code making use of element factories retrieved from an element
We can use `is_some_and(...)` instead of `map_or(false, ...)`.

Also in a few places the factory was retrieved multiple times, one time
with unwrapping and another time with handling the `None` case
correctly. Instead of unwrapping, move code to handle the `None` case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1630>
2024-06-19 13:03:37 +00:00
Sebastian Dröge
ba70bb1154 deny: Add override for older tungstenite
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1603>
2024-06-06 10:34:12 +00:00
Sebastian Dröge
85c38107cf webrtc: Update to async-tungstenite 0.26
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1603>
2024-06-06 10:34:12 +00:00
Francisco Javier Velázquez-García
8fc652f208 webrtcsink: Refactor value retrieval to avoid lock poisoning
When setting an incorrect property name in settings,
start_stream_discovery_if_needed would panic because it attempts to
unwrap a poisoned lock for settings.

This refactor avoids that situation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1594>
2024-05-31 08:10:23 +00:00
Francisco Javier Velázquez-García
568e8533fa webrtcsink: Fix typo in property name for av1enc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1594>
2024-05-31 08:10:23 +00:00
Arun Raghavan
04e9e5284c webrtc: signaller: A couple of minor doc fixups
The expectation is `Returns:`, not `Return:`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1525>
2024-05-30 22:16:46 +03:00
Arun Raghavan
1c54c77840 webrtcsink: Add a mechanism for SDP munging
Unfortunately, server implementations might have odd SDP-related quirks,
so let's allow clients a way to work around these oddities themselves.
For now, this means that a client can fix up the H.264 profile-level-id
as required by Twitch (whose media pipeline is more permissive than the
WHIP implementation).

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/516
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1525>
2024-05-30 22:16:46 +03:00