Commit graph

219 commits

Author SHA1 Message Date
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
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
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
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
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
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
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
Taruntej Kanakamalla
712d4757c3 net/webrtc/whip_signaller: multiple client support in the server
- generate a new session id for every new client
use the session id in the resource url

- remove the producer-peer-id property in the WhipServer signaler as it
is redundant to have producer id in a session having only one producer

- read the 'producer-peer-id' property on the signaller conditionally
if it exists else use the session id as producer id

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
2024-05-29 21:03:27 +00:00
Taruntej Kanakamalla
de726ca8d2 net/webrtc: multi producer support in webrtcsrc
- Add a new structure Session
  - manage each producer using a session
  - avoid send EOS when a session terminates, instead keep running
    waiting for any new producer to connect

- Maintain a bin element per session
  - each session bin encapsulates webrtcbin and the decoder if needed
    as well as the parser and filter if requested by the application
    (through request-encoded-filter)
  - this will be helpful to cleanup the session's respective elements
    when the corresponding producer terminates the session

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
2024-05-29 21:03:27 +00:00
cdelguercio
c99cabfbc5 webrtcsink: Add VP9 parser after the encoder for VP9 too
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
2024-05-23 10:16:59 +03:00
cdelguercio
f5a7de9dc3 webrtcsink: Support av1 via nvav1enc, av1enc, and rav1enc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
2024-05-23 10:16:59 +03:00
Robert Ayrapetyan
bac5845be1 webrtc: add support for insecure tls connections
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1553>
2024-05-16 19:34:57 +00:00
Sebastian Dröge
e8e173d0d0 webrtc: Update Signallable interface to new interface definition API
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1570>
2024-05-10 14:13:55 +03:00
Mathieu Duponchelle
8861fc493b webrtcsink: improve error when no discovery pipeline runs
If for instance no encoder was found or the RTP plugin was missing,
it is possible that no discovery pipeline will run for a given stream.

Provide a more helpful error message for that case.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/534
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1560>
2024-05-06 11:39:48 +00:00
Sebastian Dröge
be3ae583bc Fix new Rust 1.78 clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1559>
2024-05-02 18:36:23 +03:00
François Laignel
542030fd82 webrtcsink: don't panic if input CAPS are not supported
If a user constrained the supported CAPS, for instance using `video-caps`:

```shell
gst-launch-1.0 videotestsrc ! video/x-raw,format=I420 ! x264enc \
    ! webrtcsink video-caps=video/x-vp8
```

... a panic would occur which was internally caught without the user being
informed except for the following message which was written to stderr:

> thread 'tokio-runtime-worker' panicked at net/webrtc/src/webrtcsink/imp.rs:3533:22:
>   expected audio or video raw caps: video/x-h264, [...] <br>
> note: run with `RUST_BACKTRACE=1` environment variable to display a backtrace

The pipeline kept running.

This commit converts the panic into an `Error` which bubbles up as an element
`StreamError::CodecNotFound` which can be handled by the application.
With the above `gst-launch`, this terminates the pipeline with:

> [...] ERROR  webrtcsink net/webrtc/src/webrtcsink/imp.rs:3771:gstrswebrtc::
>   webrtcsink:👿:BaseWebRTCSink::start_stream_discovery_if_needed::{{closure}}:<webrtcsink0>
> Error running discovery: Unsupported caps: video/x-h264, [...] <br>
> ERROR: from element /GstPipeline:pipeline0/GstWebRTCSink:webrtcsink0:
>   There is no codec present that can handle the stream's type. <br>
> Additional debug info: <br>
> net/webrtc/src/webrtcsink/imp.rs(3772): gstrswebrtc::webrtcsink:👿:BaseWebRTCSink::
> start_stream_discovery_if_needed::{{closure}} (): /GstPipeline:pipeline0/GstWebRTCSink:webrtcsink0:
> Failed to look up output caps: Unsupported caps: video/x-h264, [...] <br>
> Execution ended after 0:00:00.055716661 <br>
> Setting pipeline to NULL ...

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1540>
2024-04-14 23:09:09 +02:00
François Laignel
168af88eda webrtc: add features for specific signallers
When swapping between several development branches, compilation times can be
frustrating. This commit proposes adding features to control which signaller
to include when building the webrtc plugin. By default, all signallers are
included, just like before.

Compiling the `webrtc-precise-sync` examples with `--no-default-features`
reduces compilation to 267 crates instead of 429 when all signallers are
compiled in.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1539>
2024-04-12 19:10:42 +02:00
François Laignel
83d70d3471 webrtc: add RFC 7273 support
This commit implements [RFC 7273] (NTP & PTP clock signalling & synchronization)
for `webrtcsink` by adding the "ts-refclk" & "mediaclk" SDP media attributes to
identify the clock. These attributes are handled by `rtpjitterbuffer` on the
consumer side. They MUST be part of the SDP offer.

When used with an NTP or PTP clock, "mediaclk" indicates the RTP offset at the
clock's origin. Because the payloaders are not instantiated when the offer is
sent to the consumer, the RTP offset is set to 0 and the payloader
`timstamp-offset`s are set accordingly when they are created.

The `webrtc-precise-sync` examples were updated to be able to start with an NTP
(default), a PTP or the system clock (on the receiver only). The rtp jitter
buffer will synchronize with the clock signalled in the SDP offer provided the
sender is started with `--do-clock-signalling` & the receiver with
`--expect-clock-signalling`.

[RFC 7273]: https://datatracker.ietf.org/doc/html/rfc7273

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1500>
2024-04-12 14:18:09 +02:00
François Laignel
2ad452ee89 webrtcsink: don't panic with bitrate handling unsupported encoders
When an encoder was not supported by the `VideoEncoder` `bitrate` accessors, an
`unimplemented` panic would occur which would poison `state` & `settings`
`Mutex`s resulting in other threads panicking, notably entering `end_session()`,
which lead to many failures in `BinImplExt::parent_remove_element()` until a
segmentation fault ended the process. This was observed using `vaapivp9enc`.

This commit logs a warning if an encoder isn't supported by the `bitrate`
accessors and silently by-passes `bitrate`-related operations when unsupported.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1534>
2024-04-09 15:48:59 +00:00
Taruntej Kanakamalla
f4b086738b webrtcsrc: change the producer-id type for request-encoded-filter
With https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1477
the producer id used while emitting the request-encoded-filter
can be a None if the msid of the webrtcbin's pad is None.
This might not affect the signal handler written in C but
can panic in an existing Rust application with signal
handler which can only handle valid String type as its param
for the producer id.

So change the param type to Option<String> in the signal builder
for request-encoded-fiter signal

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1528>
2024-04-09 06:01:15 +00:00
Taruntej Kanakamalla
70adedb142 net/webrtc: fix inconsistencies in documentation of object names
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1529>
2024-04-05 14:10:35 +00:00
Guillaume Desmottes
b5cbc47cf7 web: webrtcsink: improve panic message on unexpected caps during discovery
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1524>
2024-04-02 14:25:58 +02:00
Guillaume Desmottes
35b84d219f webrtc: webrtcsink: set perfect-timestamp=true on audio encoders
Chrome audio decoder doesn't cope well with not perfect ts, generating
noises in the audio.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1524>
2024-04-02 14:25:51 +02:00
François Laignel
9f27bde36a aws: use fixed BehaviorVersion
Quoting [`BehaviorVersion` documentation]:

> Over time, new best-practice behaviors are introduced. However, these
> behaviors might not be backwards compatible. For example, a change which
> introduces new default timeouts or a new retry-mode for all operations might
> be the ideal behavior but could break existing applications.

This commit uses `BehaviorVersion::v2023_11_09()`, which is the latest
major version at the moment. When a new major version is released, the method
will be deprecated, which will warn us of the new version and let us decide
when to upgrade, after any changes if required. This is safer that using
`latest()` which would silently use a different major version, possibly
breaking existing code.

[`BehaviorVersion` documentation]: https://docs.rs/aws-config/1.1.8/aws_config/struct.BehaviorVersion.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1520>
2024-03-26 17:44:16 +01:00
Philippe Normand
be12c0a5f7 Fix clippy warnings after upgrade to Rust 1.77
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1512>
2024-03-21 17:33:32 +00:00
François Laignel
c5e7e76e4d webrtcsrc: add do-retransmission property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1509>
2024-03-21 07:25:30 +00:00
François Laignel
5476e3d759 webrtcsink: prevent video-info error log for audio streams
The following error is logged when `webrtcsink` is feeded with an audio stream:

> ERROR video-info video-info.c:540:gst_video_info_from_caps:
>       wrong name 'audio/x-raw', expected video/ or image/

This commit bypasses `VideoInfo::from_caps` for audio streams.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1511>
2024-03-20 19:57:45 +01:00
Guillaume Desmottes
96337d5234 webrtc: allow resolution and framerate input changes
Some changes do not require a WebRTC renegotiation so we can allow
those.

Fix #515

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1498>
2024-03-18 14:52:01 +01:00
Guillaume Desmottes
8f997ea4e3 webrtc: janus: handle 'hangup' messages from Janus
Fix error about this message not being handled:

{
   "janus": "hangup",
   "session_id": 4758817463851315,
   "sender": 4126342934227009,
   "reason": "Close PC"
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1481>
2024-03-13 10:14:38 +00:00
Guillaume Desmottes
992f8d9a5d webrtc: janus: handle 'destroyed' messages from Janus
Fix this error when the room is destroyed:

ERROR   webrtc-janusvr-signaller imp.rs:413:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignallerU64@0x55b166a3fe40> Unknown message from server: {
   "janus": "event",
   "session_id": 6667171862739941,
   "sender": 1964690595468240,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "destroyed",
         "room": 8320333573294267
      }
   }
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1481>
2024-03-13 10:14:38 +00:00
Guillaume Desmottes
9c6a39d692 webrtc: janus: handle (stopped-)talking events
Expose those events using a signal.

Fix those errors when joining a Janus room configured with
'audiolevel_event: true'.

ERROR   webrtc-janusvr-signaller imp.rs:408:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignaller@0x560cf2a55100> Unknown message from server: {
   "janus": "event",
   "session_id": 2384862538500481,
   "sender": 1867822625190966,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "talking",
         "room": 7564250471742314,
         "id": 6815475717947398,
         "mindex": 0,
         "mid": "0",
         "audio-level-dBov-avg": 37.939998626708984
      }
   }
}
ERROR   webrtc-janusvr-signaller imp.rs:408:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignaller@0x560cf2a55100> Unknown message from server: {
   "janus": "event",
   "session_id": 2384862538500481,
   "sender": 1867822625190966,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "stopped-talking",
         "room": 7564250471742314,
         "id": 6815475717947398,
         "mindex": 0,
         "mid": "0",
         "audio-level-dBov-avg": 40.400001525878906
      }
   }
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1481>
2024-03-13 10:14:38 +00:00