Commit graph

3164 commits

Author SHA1 Message Date
Sanchayan Maity
953f6a3fd7 net: Add QUIC source and sink
To test, run receiver as

```bash
gst-launch-1.0 -v -e quicsrc caps=audio/x-opus use-datagram=true ! opusparse ! opusdec ! audio/x-raw,format=S16LE,rate=48000,channels=2,layout=interleaved ! audioconvert ! autoaudiosink
```

run sender as

```bash
gst-launch-1.0 -v -e audiotestsrc num-buffers=512 ! audio/x-raw,format=S16LE,rate=48000,channels=2,layout=interleaved ! opusenc ! quicsink use-datagram=true
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 18:01:49 +05:30
Robert Mader
8e675de690 gtk4paintablesink: Add some documentation
And sync with `README.md` in order to make the environment variables
`GST_GTK4_WINDOW` and `GST_GTK4_WINDOW_FULLSCREEN` discoverable - and
because it's generally useful.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1555>
2024-04-30 09:59:49 +03:00
Robert Mader
4326c3bfce gtk4paintablesink: Also create window for gst-play
So it can be easily tested with
```
gst-play-1.0 --videosink=gtk4paintablesink ...
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1555>
2024-04-29 22:44:56 +02:00
Robert Mader
47b788d44b gtk4paintablesink: Add env var to fullscreen window
For testing purposes with e.g. gst-launch.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1555>
2024-04-29 20:44:05 +00:00
François Laignel
16b0a4d762 rtp: add mp4gpay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-29 13:33:42 +00:00
François Laignel
b588ee59bc rtp: add mp4gdepay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-29 13:33:42 +00:00
François Laignel
5466cafc24 rtp: add mp4apay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-29 13:33:42 +00:00
François Laignel
812fe0a9bd rtp: add mp4adepay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-29 13:33:42 +00:00
Sebastian Dröge
c55c4ca42a Update CHANGELOG.md for 0.12.5 2024-04-29 13:49:17 +03:00
Sebastian Dröge
83bd7be92a deny: Remove syn override 2024-04-29 11:54:38 +03:00
Sebastian Dröge
70397a9f05 Update CHANGELOG.md for 0.12.4 2024-04-29 11:46:19 +03:00
Maksym Khomenko
a87eaa4b79 hrtfrender: use bitmask, not int, to prevent a capsnego failure
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1549>
2024-04-26 20:24:19 +00:00
Philippe Normand
88cbc93338 dav1ddec: Negotiate bt709 colorimetry when values from seq header are unspecified
With unknown range colorimetry validation would fail in video-info. As our
decoder outputs only YUV formats Bt709 should be a reasonable default.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1548>
2024-04-26 19:35:41 +00:00
Sebastian Dröge
927c3fcdb6 gtk4paintablesink: Update README.md with all the new features
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1547>
2024-04-26 12:29:10 +03:00
Sebastian Dröge
5803904deb gtk4paintablesink: meson: Add auto-detection of GTK4 versions and dmabuf feature
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1547>
2024-04-26 12:29:10 +03:00
Sebastian Dröge
c95e07a897 gtk4paintablesink: Improve scaling logic
If force-aspect-ratio=false then make sure to fully fill the given
width/height with the video frame and avoid rounding errors. This makes
sure that the video is rendered in the exact position selected by the
caller and that graphics offloading is going to work more likely.

In other cases and for all overlays, make sure that the calculated
positions are staying inside (0, 0, width, height) as rendering outside
is not allowed by GTK.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1547>
2024-04-26 12:29:10 +03:00
Sebastian Dröge
b42bd3d026 gtk4paintablesink: Add force-aspect-ratio property like in other video sinks
Unlike in other sinks this defaults to false as generally every user of
GDK paintables already ensures that the aspect ratio is kept and the
paintable is layed out in the most optimal way based on the context.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1547>
2024-04-26 12:29:10 +03:00
Sebastian Dröge
3dd800ac77 gtk4paintablesink: Implement child proxy interface
This allows setting properties on the paintable from gst-launch-1.0.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1547>
2024-04-26 12:29:10 +03:00
Sebastian Dröge
c92462b240 gtk4: Implement support for directly importing dmabufs
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/441

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1547>
2024-04-26 12:29:10 +03:00
Sebastian Dröge
7573caa8e9 rtpgccbwe: Move away from deprecated time::Instant to std::time::Instant
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1554>
2024-04-25 15:37:28 +03:00
Sebastian Dröge
c12585377c Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1554>
2024-04-25 14:46:45 +03:00
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