Use the channelVolume property to update the individual channel
volumes.
Update property values to match pulseaudio.
Handle the case where the number of channels changes. In pulse this
can normally never happen. Emit a remove and add of the node when
this happens as to not confuse pulse clients.
Don't free the buffers when the mix is released, this will happen
when nothing is using them anymore with port_use_buffers later.
Mark a mixer as having buffers when the output port has buffers.
Don't set buffers on a mixer port that already has buffers.
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.
This allows picking F32LE as the default format on links that have
no restriction and it avoids failing negotiation when the restricted
end cannot handle S16/F32/F32P
For instance this pipeline would previously fail:
audio-dsp mode=merge ! audio-dsp mode=convert ! alsa-sink
old negotiation: S16LE S24_32LE
new negotiation: F32LE S24_32LE
The link between the audio-dsp nodes has no restriction, so previously
it would negotiate S16LE, which would then fail to negotiate with alsa-sink
because fmtconvert does not know how to convert S16LE to S24_32LE directly.
With this change, the middle link negotiates to F32LE, which can be
converted to anything.
