When the transcriber is used in a live situation, it can be useful
to save a transcript for editing after the fact when producing a
VOD.
Each source pad now gets an "unsynced_" pendant. That unsynced pad
is pushed to from the context of the "live" source pad task. Flow
returns from the unsynced pads are ignored, we simply check the
last flow return before attempting to push the next transcript.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1915>
similar to the non threadshare counterparts, the ts-udpsink can accept
only one multicast interface and the ts-udpsrc can accept a list of
interfaces to be listening on for the multicast.
Use the getifaddrs crate to get the available network interfaces and filter
the desired interfaces from the available interfaces
Reuse a custom api written for PTP helper to join and leave multicast group
for IPv4 based addresses. Continue to use the UdpSocket crate's _multicast_v6
to join/leave an IPv6 multicast group
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1420>
We want to enable passthrough internally, and only notify that
internally it has been enabled once the transcriber has been unlinked.
This way applications connected to the notify handler can synchronously
update the properties and attempt to disable passthrough again.
Doing so properly requires a refactoring of the transition to the
passthrough state, with the currently set passthrough mode maintained
separately from the target passthrough state.
This commit also finishes the work left incomplete in
17d7997137 by moving the passthrough
property to the sink pad class, making each transcriber passthrough
state independent from the others.
Also adds an example to demonstrate the behavior
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1856>
The event-types property defaults to Eos. Setting an
array of additional, serialized event types results in
calling the producer.set_forward_events with those types
so that the events will be forwarded to any consumers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1875>
When congestion control is used for a session with multiple encoders,
the default implementation simply divides the overall bitrate equally
between encoders.
This is not always desirable, and this patch exposes a new signal
that users can register to, with two arguments:
* The overall bitrate to allocate
* A structure with an encoder.stream_name -> bitrate mapping
Handlers should return a similar structure with a custom mapping.
An example is also provided.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1792>
It may be desirable for the frontend to receive ancillary information
over the control channel.
Such information includes but is not limited to time code metas, support
for other metas (eg custom meta) might be implemented in the future, as
well as downstream events.
This patch implements a new info message, probes buffers that arrive at
nicesink to look up timecode metas and potentially forwards them to the
consumer when the `forward-metas` property is set appropriately.
Internally, a "dye" meta is used to trace the media identifier the
packet we are about to send over relates to, as rtpfunnel bundles all
packets together.
The example frontend code also gets a minor update and now logs info
messages to the console.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1749>
Allows you to switch output between folders without having to state change to READY to close the current playlist.
Closes the current playlist immediately and starts a new one at the currently set location.
Should be used after changing the relevant location properties.
Makes use of the send-headers signal in cmafmux.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1692>
This is one example of how a consumer might send over custom upstream
event requests to the producer.
As webrtcsink will deserialize numbers in priority as integers, we need
a custom stringifying function to ensure members of the matrix array are
indeed serialized with the floating point.
An optional stringifier parameter is thus added to the
sendControlRequest API.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1711>
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>
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>
