Commit graph

3674 commits

Author SHA1 Message Date
Mathieu Duponchelle
79657e5671 transcriberbin: fix inspect with missing elements
Relax the dependency on `awstranscriber` by still building the initial
state when it is absent, this also means an alternative transcriber can
be linked even when `awstranscriber` was not available during
construction.

Also fix property getter / setters to avoid unwrapping the pad state,
and bubble up channel bin construction errors instead of unwrapping (eg
when textwrap was not available).

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/584
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1679>
2024-07-29 08:38:36 +00:00
Sebastian Dröge
380448587b gtk4: Enable GtkGraphicsOffload::black-background property when building with GTK 4.16
This allows offloading in more situations.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/576

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1673>
2024-07-18 12:28:20 +03:00
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
Sebastian Dröge
9c84748fc3 gopbuffer: Use workspace dependency for gst-plugin-version-helper 2024-07-16 19:13:49 +03:00
Sebastian Dröge
d4d02d70a8 rtp: Require bitstream-io < 2.4.0
Version 2.4.0 contains a breaking change that it shouldn't, and updating
to 2.4.0 requires a newer Rust version.

See https://github.com/tuffy/bitstream-io/issues/22
2024-07-16 19:13:49 +03:00
Sebastian Dröge
d20ffd5d39 Update CHANGELOG.md for 0.13.0 2024-07-16 19:13:49 +03:00
sergey radionov
fdfa3a33d9 fmp4mux: added image orientation tag support
fix #565

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1669>
2024-07-16 18:49:07 +07:00
Taruntej Kanakamalla
3a8462367e threadshare: udpsrc: add buffer-size property
Use buffer-size to set the receive buffer size
on the socket

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1636>
2024-07-15 12:13:41 +00:00
Taruntej Kanakamalla
276ec91cb2 threadshare: udpsrc: add loop property to set multicast loopback
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1636>
2024-07-15 12:13:41 +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
François Laignel
000c486568 rav1enc: document bitrate property unit
See:

e34e772e47/src/rate.rs (L365)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1667>
2024-07-12 18:59:17 +02:00
Sanchayan Maity
12be9a24a6 net/quinn: Fix generation of self signed certificate
The certificate chain was incorrectly being passed the private key instead
of certificate. With rustls 0.23.11 version, this error was being caught
and reported. As stated in the 0.23.11 release, it has a new feature

"API for determining whether a CertifiedKey's certificate and private key
matches: keys_match(). This is called from existing fallible functions
that accept a private key and certificate (for example, with_single_cert())
so these functions now detect this misconfiguration."

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1666>
2024-07-12 12:26:54 +05:30
Sebastian Dröge
797dd3f3ca Update version to 0.14.0-alpha.1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1663>
2024-07-11 20:00:24 +03:00
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
73fa904a7b Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1663>
2024-07-11 20:00:24 +03:00
Robert Mader
c7ef8e8185 gtk4: Use scale instead of rotate where possible
In order to make it easier for GTK4 to figure out that the resulting
operation is 2D and - crucially - can get offloaded to Wayland.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1662>
2024-07-10 10:58:20 +02:00
Robert Mader
2238db2005 gtk4: Support RGBx formats in SW paths
GTK4 has matching enums and thus should handle them fine. Further more
it should allow renderers to reduce memory bandwidth by applying
occlusion culling.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1660>
2024-07-09 16:53:01 +02:00
Sebastian Dröge
3609411801 gtk4: Invalidate paintable size if changing because of orientation changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1659>
2024-07-08 14:49:43 +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
Sebastian Dröge
f88f5b03c4 Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1658>
2024-07-08 10:58:14 +03:00
Sebastian Dröge
4123b5d1a1 mpegtslive: Update for gst::Clock::set_calibration() API changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1658>
2024-07-08 09:59:06 +03:00
Sebastian Dröge
8522c8a445 gtk4: Add support for rotations / flipping
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/284

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1590>
2024-07-07 07:43:49 +00:00
Sanchayan Maity
2fe852166e aws/s3hlssink: Do not call abort before finishing uploads
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1653>
2024-07-06 14:44:08 +00:00
Sebastian Dröge
6e974cf4b9 gtk4: Document paintable properties correctly
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1655>
2024-07-06 11:36:55 +00:00
Sebastian Dröge
195c089f18 gtk4: Declare correct default value for force-aspect-ratio property
It's defaulting to false as generally keeping the aspect ratio is the
job of the widget layout and not the paintable.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1654>
2024-07-06 13:41:44 +03:00
Artem Martus
ac0e24b2bd tutorial-1: Fix broken links for struct references
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1652>
2024-07-04 15:01:22 +00:00
Sebastian Dröge
4ab8d92f28 mpegtslivesrc: Don't skip the first MPEG-TS packet
If every buffer contains only a single MPEG-TS packet we would otherwise
skip over everything and would never observe a PCR.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1651>
2024-07-04 17:01:43 +03:00
Sebastian Dröge
c701aa6f84 audioloudnorm: Fix limiter buffer index wraparound off-by-one for the last buffer
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1649>
2024-07-02 19:31:11 +03:00
Sebastian Dröge
bd2a039c8d livesync: Use the actual output buffer duration of gap filler buffers
Otherwise the following can happen:

  - 25fps stream
  - buffer with PTS 0ms, duration 20ms arrives, is output
  - buffer with PTS 40ms, duration 20ms arrives
  - is considered early because 20ms < 40ms
  - filler buffer with PTS 20ms and 40ms duration is output
  - buffer with PTS 40ms is output

After this change no filler would be inserted because the gap is smaller
than the duration of a filler buffer.

Also, previously the 40ms duration would be used if a filler was
previously output because in that case the cached output buffer duration
would've already been patched from 20ms to 40ms.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1647>
2024-07-02 17:15:58 +03:00
Philippe Normand
eee93aea52 rtp2: Fix typo on auto-header-extension property name
The rtp (de)pay elements use auto-header-extension so the new elements should do
the same.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1646>
2024-07-02 09:35:39 +01:00
Edward Hervey
95ae67752f net: New mpegtslive element
This element allows wrapping an existing live "mpeg-ts source" (udpsrc,
srtsrc,...) and providing a clock based on the actual PCR of the stream.

Combined with `tsdemux ignore-pcr=True` downstream of it, this allows playing
back the content at the same rate as the (remote) provider **and** not modify
the original timestamps.

Co-authored-by: Sebastian Dröge <slomo@coaxion.net>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1640>
2024-07-01 15:29:22 +02:00
Mathieu Duponchelle
0ef886ea16 transcriberbin: fix internal ghost pad name regression
As part of https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1593
source pad names on inner transcription bins were appended a suffix, but
other pieces of the code were not updated to account for that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1645>
2024-07-01 11:47:39 +02:00
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
Jordan Petridіs
718e757669 video/gtk4: Dehardcode module name in the Flatpak example in the readme 2024-06-29 15:56:06 +00:00
Mathieu Duponchelle
f0df6874d8 transcriberbin: fix property proxying
As part of https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1546
the element started implementing the GstChildProxy interface in order to
expose properties on its sink pads, but the implementation was
incorrect and broke proxying to children elements.

In addition, an intermediary bin was introduced with no name, making it
hard to set the properties of the inner elements through the child
proxy interface, it is now named according to the name of the pad it
corresponds to.

Finally, the default transcriber is back to being named "transcriber".

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1642>
2024-06-28 14:24:08 +00:00
Sebastian Dröge
960529d90d livesync: Add sync property for allowing to output buffers as soon as they arrive
By default livesync will wait for each buffer on the clock. If sync is
set to false, it will output buffers immediately once they're available
and only waits on the clock for outputting gap filler buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1635>
2024-06-26 16:21:42 +00:00
Sebastian Dröge
bbf131086a livesync: Synchronize on the first buffer too
Previously the first buffer would be output immediately and
synchronization would only happen from the second buffer onwards.
This would mean that the first buffer would potentially be output too
early.

Instead, if there is no known output timestamp yet but a buffer with a
timestamp, first of all take its start as the initial output timestamp
and synchronize on that buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1635>
2024-06-26 16:21:42 +00:00
Sebastian Dröge
7caf6b2073 livesync: Use let-else in a few more places
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1635>
2024-06-26 16:21:41 +00:00
Sebastian Dröge
505fab2e1c livesync: Allow queueing up to latency buffers
This was already reported by the latency query, and not doing this would
require to always put a queue before livesync.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1635>
2024-06-26 16:21:41 +00:00
Guillaume Desmottes
a10577b42c aws: log error if sink failed to start
I find it confusing that the element was failing without reporting any
error in its logs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1638>
2024-06-26 11:22:54 +02:00
Guillaume Desmottes
0ecbd3f953 aws: use DisplayErrorContext when displaying SDK errors
As suggested in the aws crate documentation, wrap SDK errors with
DisplayErrorContext so their Display implementation outputs the full
context.

Improve error display from "dispatch failure" to

"dispatch failure: io error: error trying to connect: dns error: failed
to lookup address information: Name or service not known: dns error:
failed to lookup address information: Name or service not known: failed
to lookup address information: Name or service not known
(DispatchFailure(DispatchFailure { source: ConnectorError { kind: Io,
source: hyper::Error(Connect, ConnectError(\"dns error\", Custom { kind:
Uncategorized, error: \"failed to lookup address information: Name or
service not known\" })), connection: Unknown } }))"

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1638>
2024-06-26 10:47:10 +02:00
Guillaume Desmottes
3b7b2cd37b aws: rely on WaitError Display implementation
The Display implementation of WaitError already displays the underlying
SDK error and the metadata, so can just use that.

Will also be used to provide more context in the next patch.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1638>
2024-06-26 10:46:46 +02:00
Sanchayan Maity
0bd98e2c34 net/quinn: Allow dropping buffers when buffer size exceeds maximum datagram size
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-25 20:15:40 +05:30
Sanchayan Maity
e00ebca63f net/quinn: Add stats property for connection statistics
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-25 20:15:40 +05:30
Sanchayan Maity
2b35f009fb net/quinn: Update quinn to 0.11.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-25 20:15:40 +05:30
Sanchayan Maity
cf7172248c net/quinn: Allow setting some parameters from TransportConfig
As of now, we expose the below four properties from `TransportConfig`.
- Initial MTU
- Minimum MTU
- Datagram receive buffer size
- Datagram send buffer size

Maximum UDP payload size from `EndpointConfig` and upper bound from
`MtuDiscoveryConfig` are also exposed as properties.

See the below documentation for further details.
- https://docs.rs/quinn/latest/quinn/struct.TransportConfig.html
- https://docs.rs/quinn/latest/quinn/struct.MtuDiscoveryConfig.html
- https://docs.rs/quinn/latest/quinn/struct.EndpointConfig.html

While at it, also clean up passing function parameters to the functions
in utils.rs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-25 20:15:40 +05:30
Sanchayan Maity
bc5ed023e4 net/quinn: Improve datagram handling
We now check if the peer actually supports Datagram and refusing to
proceed if it does not. Since the datagram size can actually change
over the lifetime of a connection according to variation in path MTU
estimate, also check buffer size before trying to send.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-25 20:15:40 +05:30
Sebastian Dröge
dbad98132f deny: Add another override for livekit 2024-06-25 10:58:56 +03:00
Matthew Waters
39b61195ad rtprecv: ensure that stopping the rtp src task does not critical
When pad a released, then we were removing the pad from an internal
list. If the pad was not already deactivated, the deactiviation would
attempt to look for the pad in that list and panic if it was not there.

Fix by delaying removal of the pad from the list until after pad
deactivation occurs.

Also includes test.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
2024-06-24 13:13:28 +00:00