Commit graph

3352 commits

Author SHA1 Message Date
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
Matthew Waters
10a31a397e rtp/recv: support pushing buffer lists from the jitterbuffer
Multiple concurrent buffers produced by the jitterbuffer will be
combined into a single buffer list which will be sent downstream.

Events or queries that interrupt the buffer flow will cause a split in
the output buffer list.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
2024-06-24 13:13:28 +00:00
Matthew Waters
d036abb7d2 rtp/recv: support buffers lists on rtp sink pad
In one case, improves throughput by 25% when buffer lists are used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
2024-06-24 13:13:28 +00:00
Matthew Waters
df4a4fb2ef rtp/send: support receiving buffer lists
Can reduce processing overhead if many buffers are pushed concurrently.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
2024-06-24 13:13:28 +00:00
Matthew Waters
2d1f556794 rtp/session: guard against a busy wait with no members
If the number of members is 0, then the calculated time to the next rtcp
wakup would be 'now' and could result in a busy loop in the rtcp
processing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
2024-06-24 13:13:28 +00:00
Matthew Waters
84a9f9c61f rtp/source: use extended sequence number helper
Instead of rolling our own

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
2024-06-24 13:13:28 +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
90e926def4 deny: Add override for older system-deps version
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1625>
2024-06-21 11:07:22 +03:00
Nirbheek Chauhan
345edeb947 meson: Printing a list is only available with meson >=1.3
Fixes https://gitlab.freedesktop.org/tpm/gstreamer/-/jobs/60108579

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1632>
2024-06-20 14:54:11 +05:30
Sebastian Dröge
a7764ff033 Update CHANGELOG.md for 0.12.7
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1631>
2024-06-19 21:03:33 +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
Arun Raghavan
8f96509f03 aws: s3: Enable tests again
We lost the environment variable checks during the addition of the
putobjectsink tests, which caused failures on MR branches.

It would be nicer to use some other mechanism to validate the tests can
run, so we don't count on only the environmnent, but for now this will
have to do.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1629>
2024-06-18 11:58:43 -04:00
Sebastian Dröge
59bada0a9f deny: Add another override for hermit-abi
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1622>
2024-06-18 10:27:27 +03:00
Sebastian Dröge
743ab29ba8 Update Cargo.lock and MSRV to 1.71
cea608-types requires that now because it updated the env_logger
dependency. As a result, we can also update it here now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1622>
2024-06-18 10:27:27 +03:00
Sebastian Dröge
970d1c9afd Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1622>
2024-06-18 10:27:27 +03:00
Sebastian Dröge
5aedcab32f Revert "aws: s3: Re-enable tests"
This reverts commit b4b56eb282.
The tests are still failing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
2024-06-18 08:50:07 +03:00
Sebastian Dröge
4677948a82 rtp: av1pay: Derive Default trait for the state instead of manual implementation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
2024-06-18 08:07:24 +03:00
Sebastian Dröge
d357a63bf9 rtp: av1pay: Correctly use N flag for marking keyframes
The "first packet of a coded video sequence" means that this should be
the first packet of a keyframe that comes together with a sequence
header, not the first packet of a new frame.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/558

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
2024-06-18 08:06:59 +03:00
Sebastian Dröge
5cd9e34265 rtp: av1pay: Correctly skip over ignored OBUs
The reader is already after the header at this point so only the OBU
content has to be skipped.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
2024-06-18 08:06:59 +03:00
Sebastian Dröge
bbe38b9599 rtp: av1: Drop padding OBUs too like Chrome does
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
2024-06-18 08:06:59 +03:00
Arun Raghavan
b4b56eb282 aws: s3: Re-enable tests
These seem to have stopped working due to bad/rotated creds. Should work
fine now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1623>
2024-06-17 06:08:18 -04:00
Sebastian Dröge
343680ffea rtp: av1depay: Don't return an error if parsing a packet fails
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1612>
2024-06-14 13:13:21 +00:00