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.
Pass some state to convert and channelmix functions. This makes it
possible to select per channel optimized convert functions but
also makes it possible to implement noise shaping later.
Pass the channelmix matrix and volume in the state.
Handle specialized 2 channel s16 -> f32 conversion
Add a sinc based resampler that, unlike speex, avoids memcpy and
works directly on the source data. It also allows for ssse3
optimizations and aligned loads. It will later switch to table
interpolation when doing variable rate.
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
Implement audioconvert as a complex element of fmtconver,
channelmix and resample.
Make copying resample just to test.
Plug the converter into pw_stream.
