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Author SHA1 Message Date
Mathieu Duponchelle
17d7997137 transcriberbin: add support for consuming secondary audio streams 
In some situations, a translated alternate audio stream for a content
might be available.

Instead of going through transcription and translation of the original
audio stream, it may be preferrable for accuracy purposes to simply
transcribe the secondary audio stream.

This MR adds support for doing just that:

* Secondary audio sink pads can be requested as "sink_audio_%u"

* Sometimes audio source pads are added at that point to pass through
  the audio, as "src_audio_%u"

* The main transcription bin now contains per-input stream transcription
  bins. Those can be individually controlled through properties on the
  sink pads, for instance translation-languages can be dynamically set
  per audio stream

* Some properties that originally existed on the main element still
  remain, but are now simply mapped to the always audio sink pad

* Releasing of secondary sink pads is nominally implemented, but not
  tested in states other than NULL

An example launch line for this would be:

```
$ gst-launch-1.0 transcriberbin name=transcriberbin latency=8000 accumulate-time=0 \
      cc-caps="closedcaption/x-cea-708, format=cc_data" sink_audio_0::language-code="es-US" \
      sink_audio_0::translation-languages="languages, transcript=cc3"
    uridecodebin uri=file:///home/meh/Music/chaplin.mkv name=d
      d. ! videoconvert ! transcriberbin.sink_video
      d. ! clocksync ! audioconvert ! transcriberbin.sink_audio
      transcriberbin.src_video ! cea608overlay field=1 ! videoconvert ! autovideosink \
      transcriberbin.src_audio ! audioconvert ! fakesink \
    uridecodebin uri=file:///home/meh/Music/chaplin-spanish.webm name=d2 \
      d2. ! audioconvert ! transcriberbin.sink_audio_0 \
      transcriberbin.src_audio_0 ! fakesink
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1546>
 2024-04-25 11:56:01 +02:00
Sebastian Dröge
66030f36ad tracers: Add a pad push durations tracer 
This tracer measures the time it takes for a buffer/buffer list push to return.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1506>
 2024-04-17 16:20:43 +03:00
Seungha Yang
b3d3895ae7 cea608overlay: Fix black-background setting 
Apply the property to newly created renderer

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1542>
 2024-04-15 15:38:31 +00:00
Sebastian Dröge
d6a855ff1b rtp: Add VP8/9 RTP payloader/depayloader 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1487>
 2024-04-15 14:03:56 +00:00
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
3fc38be5c4 webrtc: add missing tokio feature for precise sync examples 
Clippy caught the missing feature `signal` which is used by the WebRTC precise
synchronization examples. When running `cargo` `check`, `build` or `clippy`
without `no-default-dependencies`, this feature was already present due to
dependents crates.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1541>
 2024-04-14 16:50:33 +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
Guillaume Desmottes
596a9177ce uriplaylistbin: disable racy test 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1537>
 2024-04-12 10:17:40 +00:00
Philippe Normand
2341ee6935 dav1d: Set colorimetry parameters on src pad caps 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1514>
 2024-04-12 09:14:34 +00:00
Guillaume Desmottes
61c9cbdc8f uriplaylistbin: allow to change 'iterations' property while playing 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1492>
 2024-04-11 11:13:20 +02:00
Guillaume Desmottes
00b56ca845 uriplaylistbin: stop using an iterator to manage the playlist 
Will make it easier to update the playlist while playing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1492>
 2024-04-11 10:48:50 +02:00
François Laignel
42158cbcb0 gccbwe: don't log an error when handling a buffer list while stopping 
When `webrtcsink` was stopped, `gccbwe` could log an error if it was handling a
buffer list. This commit logs an error only if `push_list()` returned an error
other than `Flushing`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1535>
 2024-04-11 01:29:53 +00:00
Matthew Waters
4dcc44687a cea608overlay: move Send impl lower in the stack 
Try to avoid hiding another non-Send object in the State struct.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1519>
 2024-04-10 06:55:34 +00:00
Matthew Waters
fbce73f6fc closedcaption: implement cea708overlay element 
Can overlay any single CEA-708 service or any single CEA-608 channel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1519>
 2024-04-10 06:55:34 +00:00
Matthew Waters
f0c38621c1 cea608overlay: also print bytes that failed to decode 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1519>
 2024-04-10 06:55:34 +00:00
Sanchayan Maity
a3e30b499f aws: Introduce a property to use path-style addressing 
AWS SDK switched to virtual addressing as default instead of path
style earlier. While MinIO supports virtual host style requests,
path style requests are the default.

Introduce a property to allow the use of path style addressing if
required.

For more information, see
https://github.com/minio/minio/blob/master/docs/config/README.md#domain
https://docs.aws.amazon.com/AmazonS3/latest/userguide/VirtualHosting.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1527>
 2024-04-10 00:23:22 +00:00
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
Simonas Kazlauskas
5d939498f1 mp4/fmp4: support flac inside the iso (f)mp4 container 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1401>
 2024-04-09 14:37:05 +03:00
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
Tim-Philipp Müller
6b30266145 ci: tag linter and sanity check jobs as a "placeholder" jobs 
They hardly use any resources and almost finish immediately.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1533>
 2024-04-08 23:42:23 +00:00
Tim-Philipp Müller
c8180e714e ci: make sure version Cargo.toml matches version in meson.build 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1532>
 2024-04-08 14:46:00 +01:00
Sebastian Dröge
0b356ee203 deny: Update 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1530>
 2024-04-06 11:12:16 +03:00
Sebastian Dröge
c2ebb3083a Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1530>
 2024-04-06 11:12:16 +03:00
Sebastian Dröge
921938fd20 fmp4mux: Require gstreamer-pbutils 1.20 for the examples 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1530>
 2024-04-06 11:10:58 +03:00
Sebastian Dröge
fab246f82e webrtchttp: Update to reqwest 0.12 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1530>
 2024-04-06 11:07:16 +03:00
Sebastian Dröge
7757e06e36 onvifmetadataparse: Reset state in PAUSED->READY after pad deactivation 
Otherwise the clock id will simply be overridden instead of unscheduling
it, and if the streaming thread of the source pad currently waits on it
then it will wait potentially for a very long time and deactivating the
pad would wait for that to happen.

Also unschedule the clock id on `Drop` of the state to be one the safe
side and not simply forget about it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1526>
 2024-04-05 15:19:37 +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
Matthew Waters
7f6929b98d closedcaption: remove libcaption code entirely 
It is now unused.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1517>
 2024-04-05 19:29:24 +11:00
Matthew Waters
2575013faa cea608tott: use our own CEA-608 frame handling instead of libcaption 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1517>
 2024-04-05 19:29:24 +11:00
Matthew Waters
d8fe1c64f1 cea608overlay: use or own CEA-608 caption frame handling instead of libcaption 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1517>
 2024-04-05 19:29:24 +11:00
Matthew Waters
fea85ff9c8 closedcaption: use cea608-types for parsing 608 captions instead of libcaption 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1517>
 2024-04-05 19:29:24 +11:00
François Laignel
cc43935036 webrtc: add precise synchronization example 
This example demonstrates a sender / receiver setup which ensures precise
synchronization of multiple streams in a single session.

[RFC 6051]-style rapid synchronization of RTP streams is available as an option.
See the [Instantaneous RTP synchronization...] blog post for details about this
mode and an example based on RTSP instead of WebRTC.

[RFC 6051]: https://datatracker.ietf.org/doc/html/rfc6051
[Instantaneous RTP synchronization...]: https://coaxion.net/blog/2022/05/instantaneous-rtp-synchronization-retrieval-of-absolute-sender-clock-times-with-gstreamer/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1463>
 2024-04-03 19:10:40 +02:00
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
Sebastian Dröge
0aabbb3186 fmp4: Update to dash-mpd 0.16 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1523>
 2024-03-31 09:36:53 +03:00
Sebastian Dröge
4dd6b102c4 Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1523>
 2024-03-31 09:35:46 +03:00
Sebastian Dröge
0dd03da91f ci: Ignore env_logger for cargo-outdated 
It requires Rust >= 1.71.
 2024-03-29 11:03:04 +02:00
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