Commit Graph

3105 Commits

Author SHA1 Message Date
Matthew Waters e1cd52178e transcriberbin: also support 608 inside 708
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2024-03-28 13:46:28 +11:00
Matthew Waters 55b4de779c tttocea708: add support for writing 608 compatibility bytes
608 compatibility bytes are generated using the same functionality as
tttocea608.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2024-03-28 13:46:28 +11:00
Matthew Waters 9db4290d2d tttocea608: move functionality to a separate object
Will be used by tttocea708 later.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2024-03-28 13:46:28 +11:00
Matthew Waters df30d2fbd3 transcriberbin: add support for generating cea708 captions
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2024-03-28 13:46:28 +11:00
Matthew Waters b0cf7e5c81 cea708mux: add element muxing multiple 708 caption services together
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2024-03-28 13:46:28 +11:00
Matthew Waters 756abbf807 tttocea708: add element converting from text to cea708 captions
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2024-03-28 13:46:28 +11:00
Martin Nordholts 5d7e068a8b rtpgccbwe: Add increasing_duration and counter to existing gst::log!()
Add `self.increasing_duration` and `self.increasing_counter`
to logs to provide more details of why `overuse_filter()`
determines overuse of network.

To get access to the latest values of those fields we need
to move down the log call. But that is fine, since no other
logged data is modified between the old and new location of
`gst::log!()`.

We do not bother logging `self.last_overuse_estimate` since
that is simply the previously logged value of `estimate`. We
must put the log call before we write the latest value to it
though, in case we want to log it in the future.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1522>
2024-03-27 15:08:23 +00:00
François Laignel a870d60621 aws: improve error message logs
The `Display` and `Debug` trait for the AWS error messages are not very useful.

- `Display` only prints the high level error, e.g.: "service error".
- `Debug` prints all the fields in the error stack, resulting in hard to read
  messages with redudant or unnecessary information. E.g.:

> ServiceError(ServiceError { source: BadRequestException(BadRequestException {
> message: Some("1 validation error detected: Value 'test' at 'languageCode'
> failed to satisfy constraint: Member must satisfy enum value set: [ar-AE,
> zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH,
> de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US,
> fr-CA, en-GB]"), meta: ErrorMetadata { code: Some("BadRequestException"),
> message: Some("1 validation error detected: Value 'test' at 'languageCode'
> failed to satisfy constraint: Member must satisfy enum value set: [ar-AE,
> zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH,
> de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US,
> fr-CA, en-GB]"), extras: Some({"aws_request_id": "1b8bbafd-5b71-4ba5-8676-28432381e6a9"}) } }),
> raw: Response { status: StatusCode(400), headers: Headers { headers:
> {"x-amzn-requestid": HeaderValue { _private: H0("1b8bbafd-5b71-4ba5-8676-28432381e6a9") },
> "x-amzn-errortype": HeaderValue { _private:
> H0("BadRequestException:http://internal.amazon.com/coral/com.amazonaws.transcribe.streaming/") },
> "date": HeaderValue { _private: H0("Tue, 26 Mar 2024 17:41:31 GMT") },
> "content-type": HeaderValue { _private: H0("application/x-amz-json-1.1") },
> "content-length": HeaderValue { _private: H0("315") }} }, body: SdkBody {
> inner: Once(Some(b"{\"Message\":\"1 validation error detected: Value 'test'
> at 'languageCode' failed to satisfy constraint: Member must satisfy enum value
> set: [ar-AE, zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT,
> es-ES, th-TH, de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP,
> ca-ES, es-US, fr-CA, en-GB]\"}")), retryable: true }, extensions: Extensions {
> extensions_02x: Extensions, extensions_1x: Extensions } } })

This commit adopts the most informative and concise solution I could come up
with to log AWS errors. With the above error case, this results in:

> service error: Error { code: "BadRequestException", message: "1 validation
> error detected: Value 'test' at 'languageCode' failed to satisfy constraint:
> Member must satisfy enum value set: [ar-AE, zh-HK, en-US, ar-SA, zh-CN, fi-FI,
> pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH, de-DE, it-IT, fr-FR, ko-KR, hi-IN,
> en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US, fr-CA, en-GB]",
> aws_request_id: "a40a32a8-7b0b-4228-a348-f8502087a9f0" }

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1521>
2024-03-26 20:05:32 +01:00
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
Matthew Waters e868f81189 gopbuffer: implement element buffering of an entire GOP
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1349>
2024-03-26 15:29:48 +11:00
Sebastian Dröge bac2e02160 deny: Add overrides for duplicates hyper / reqwest dependencies 2024-03-24 11:30:30 +02:00
Nirbheek Chauhan ae7c68dbf8 ci: Add a job to trigger a cerbero build, similar to the monorepo
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1513>
2024-03-23 23:02:27 +00:00
Sebastian Dröge 0b11209674 Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1510>
2024-03-23 14:33:07 +02:00
Sebastian Dröge f97150aa58 reqwest: Update to reqwest 0.12
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1510>
2024-03-23 14:30:31 +02:00
Philippe Normand 7e1ab086de dav1d: Require dav1d-rs 0.10
This version depends on libdav1d >= 1.3.0. Older versions are no longer
supported, due to an ABI/API break introduced in 1.3.0.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1512>
2024-03-21 17:33:32 +00:00
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
Sebastian Dröge 317f46ad97 Update CHANGELOG.md for 0.12.3 2024-03-21 18:55:29 +02:00
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
Sebastian Dröge 6556d31ab8 livesync: Ignore another racy test
Same problem as https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/328
2024-03-21 09:27:09 +02:00
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
François Laignel cc7b7d508d rtp: gccbwe: don't break downstream assumptions pushing buffer lists
Some elements in the RTP stack assume all buffers in a `gst::BufferList`
correspond to the same timestamp. See in [`rtpsession`] for instance.
This also had the effect that `rtpsession` did not create correct RTCP as it
only saw some of the SSRCs in the stream.

`rtpgccbwe` formed a packet group by gathering buffers in a `gst::BufferList`,
regardless of whether they corresponded to the same timestamp, which broke
synchronization under certain circonstances.

This commit makes `rtpgccbwe` push the buffers as they were received: one by one.

[`rtpsession`]: bc858976db/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpsession.c (L2462)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1502>
2024-03-20 18:19:14 +00:00
Sebastian Dröge 2b9272c7eb fmp4mux: Move away from deprecated chrono function
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
2024-03-20 15:37:18 +02:00
Sebastian Dröge cca3ebf520 rtp: Switch from chrono to time
Which allows to simplify quite a bit of code and avoids us having to
handle some API deprecations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
2024-03-20 15:05:39 +02:00
Sebastian Dröge 428f670753 version-helper: Use non-deprecated type alias from toml_edit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
2024-03-19 18:16:42 +02:00
Sebastian Dröge fadb7d0a26 deny: Add override for heck 0.4
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
2024-03-19 17:52:32 +02:00
Sebastian Dröge 2a88e29454 originalbufferstore: Update for `VideoMetaTransform` -> `VideoMetaTransformScale` rename
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
2024-03-19 17:51:41 +02:00
Sebastian Dröge bfff0f7d87 Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
2024-03-19 17:50:32 +02:00
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
Tim-Philipp Müller eb49459937 rtp: m2pt: add some unit tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
2024-03-16 10:07:37 +00:00
Tim-Philipp Müller ce3960f37f rtp: Add MPEG-TS RTP payloader
Pushes out pending TS packets on EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
2024-03-16 10:07:37 +00:00
Tim-Philipp Müller 9f07ec35e6 rtp: Add MPEG-TS RTP depayloader
Can handle different packet sizes, also see:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1310

Has clock-rate=90000 as spec prescribes, see:
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/691

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
2024-03-16 10:07:37 +00:00
Mathieu Duponchelle f4366f8b2e gstregex: add support for switches exposed by RegexBuilder
The builder allows for instance for switching off case-sensitiveness for
the entire pattern, instead of having to do so inline with `(?i)`.

All the options exposed by the builder at
<https://docs.rs/regex/latest/regex/struct.RegexBuilder.html> can now be
passed as fields of invidual commands, snake-cased.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1497>
2024-03-15 17:41:39 +00:00
Guillaume Desmottes 523a46b4f5 gtk4: scale texture position
Fix regression in 0.12 introduced by 3423d05f77

Code from Ivan Molodetskikh suggested on Matrix.

Fix #519

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1499>
2024-03-15 13:43:32 +01:00
Nirbheek Chauhan 6f8fc5f178 meson: Disable docs completely when the option is disabled
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1496>
2024-03-14 15:30:17 +05:30
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
Guillaume Desmottes b29a739fb2 uriplaylistbin: disable racy test
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/514

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1494>
2024-03-12 16:57:22 +01:00
Guillaume Desmottes 1dea8f60a8 threadshare: disable racy tests
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/250

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1494>
2024-03-12 16:54:21 +01:00
Guillaume Desmottes 2629719b4e livesync: disable racy tests
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/328
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/357

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1494>
2024-03-12 16:32:47 +01:00
Guillaume Desmottes 9e6e8c618e togglerecord: disable racy test_two_stream_close_open_nonlivein_liveout test
See https://gitlab.freedesktop.org/gdesmott/gst-plugins-rs/-/jobs/56183085

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1494>
2024-03-12 16:21:52 +01:00
François Laignel 995f64513d Update Cargo.lock to use latest gstreamer-rs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1491>
2024-03-11 14:42:36 +01:00
François Laignel 5b01e43a12 webrtc: update further to WebRTCSessionDescription sdp accessor changes
See: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1406
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1491>
2024-03-11 13:39:19 +01:00
Guillaume Desmottes 03abb5c681 spotify: document how to use with non Facebook accounts
See discussion on #203.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1490>
2024-03-11 09:46:40 +01:00
Zhao, Gang 7a46377627 rtp: tests: Simplify loop
All buffers can be added in 100 outer loops. Add buffer less than 200 in the last (i = 99) loop.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1489>
2024-03-10 16:47:30 +08:00
Olivier Crête 15e7a63e7b originalbuffer: Pair of elements to keep and restore original buffer
The goal is to be able to get back the original buffer
after performing analysis on a transformed version. Then put the
various GstMeta back on the original buffer.

An example pipeline would be
.. ! originalbuffersave ! videoscale ! analysis ! originalbufferestore ! draw_overlay ! sink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1428>
2024-03-08 15:15:13 -05:00
Guillaume Desmottes 612f863ee9 webrtc: janusvrwebrtcsink: add 'use-string-ids' property
Instead of exposing all ids properties as strings, we now have two
signaller implementations exposing those properties using their actual
type. This API is more natural and save the element and application
conversions when using numerical ids (Janus's default).

I also removed the 'joined-id' property as it's actually the same id as
'feed-id'. I think it would be better to have a 'janus-state' property or
something like that for applications willing to know when the room has
been joined.
This id is also no longer generated by the element by default, as Janus
will take care of generating one if not provided.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1486>
2024-03-07 09:34:58 +01:00
Seungha Yang 237f22d131 sccparse: Ignore invalid timecode during seek as well
sccparse holds last timecode in order to ignore invalid timecode
and fallback to the previous timecode. That should happen
when sccparse is handling seek event too. Otherwise single invalid
timecode before the target seek position will cause flow error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1485>
2024-03-06 11:12:04 +00:00
Sebastian Dröge 2839e0078b rtp: Port RTP AV1 payloader/depayloader to new base classes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1472>
2024-03-06 09:40:35 +00:00
Jordan Yelloz 0414f468c6 livekit_signaller: Added missing getter for excluded-producer-peer-ids
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1484>
2024-03-04 10:08:11 -07:00