Add a new PortConfig parameter to configure ports of elements that
are marked with the SPA_NODE_FLAG_*_PORT_CONFIG. This is used to
configure the operation of the audioconver/audioadapter nodes and
how it should convert the internal format. We want to use the
Profile parameter only for cases where there is an enumeration of
values, like with device configuration.
Add unit tests for audioconvert and adapter to check if they handle
PortConfig correctly.
Make the media session use the PortConfig to dynamically configure
the device nodes.
Remove audio-dsp, it is not used anymore and can/should be implemented
with a simple audioconvert spa node now and some PortConfig.
Use an adapter instead of the client-stream. This means we run
the audioconverters and resamplers in the client instead of the
pipewire daemon. It also allows us to implement the audio mixing
correctly in the capture client.
The only pending piece is that we now wake up the client with the
period of the server. Maybe we can later optimize that and
accumulate/split buffers before waking the client.
This probably needs fixing with video..
An adapter is like an audio-dsp node and client-stream combined.
It allows tighter control with the device (for rate control and
variable buffer sizes), software volume.
The idea is to also implement the client-stream with this
eventually.
Make the code to export objects more generic. Make it possible for
modules to register a type to export.
Make the client-node also able to export plain spa_nodes.
Let the remote signal the global of the exported object if any. We can
then remote the (unused) remote_id from the proxy.
Rename the flatpak module to access module. The access module should
either let the client connect or mask the client busy while the
permissions are being configured. It is then up to the session manager
to collect the right permissions of the objects and configure those
in the client.
Let the media session monitor the clients and configure the permissions.
Make it possible to assign an arbitary node as the port mixer.
Also remove dynamically added ports.
Improve negotiation and allocation on the mixer ports
Add some more SSE optimisations
Move float mixer from the audio dsp to the port
Remove pw_node_get_free_port() and do things more explicitly.
Handle mixer ports in client-node
A client stream is a more specialized way to send 1 stream to pipewire.
On the client side and receiver side it can do conversion and the
buffer size of the client can be choosen arbitrarily.
Add an audio dsp module that adds an interleaver for each audio sink
and only allows 1 buffer size and format on the ports. The idea is that
dsp (pro-audio) nodes can be inserted in this part of the pipeline.
Make the protocol client connect call async with a callback when it
completes.
Move the connect methods into separate files, add an empty connect
method that will use the screencast portal to get a pipewire fd.
Use the remote intention to get the connect method.
Add some better error reporting.
Make a link factory and use create-object to make links. That way
we can have different kinds of links based on the factory and we
can also hide the factory when link creation should be blocked.
