This was defined one bracket above, which was causing the
gst-gl codepath below to also be disabled when there was
no dmabuf feature enabled.
This was also resulting in the following warning as
we were never creating the MappedFrame::GL vartiant due to this
```
warning: unused variable: `wrapped_context`
--> video/gtk4/src/sink/frame.rs:541:85
|
541 | ...", feature = "gst-gl"))] wrapped_context: Option<
| ^^^^^^^^^^^^^^^ help: if this is intentional, prefix it with an underscore: `_wrapped_context`
|
= note: `#[warn(unused_variables)]` on by default
warning: variant `GL` is never constructed
--> video/gtk4/src/sink/frame.rs:80:5
|
74 | enum MappedFrame {
| ----------- variant in this enum
...
```
Move the cfg to the appropriate place where it encaplsulates only
the dmabuf related code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1682>
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>
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>
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>
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>
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>
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>
