Implement per channel volume on channelmix. Extend control on stream to
take an array of values when possible.
Remove name argument from pw_node_new and pw_device_new. We can pass
this as a property instead.
Improve properties on nodes to more closely match what pulseaudio does.
Don't let the monitor do too much with the udev properties but let the
session manager set the description and icon-names.
Remove some change_mask flags for things that don't change in
introspect. Use the flags to mark changes in -cli and -monitor.
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.
Remove the node buffers reply again. We don't need it. Instead add a
new method to the client-node to upload an array of buffer datas.
This method is called after the client has allocated buffer mem. It
will update the buffers on the server side with the client allocated
memory.
Wait for the async reply of use_buffers when doing alloc_buffers so
that we can get the updated buffer mem before we continue.
Let the link follow the states of the ports.
Add some error code to the port error states.
Add PW_STREAM_FLAG_ALLOC_BUFFERS flag to make the client alloc buffer
memory.
Remove the CAN_USE_BUFFERS flag, it is redundant. We can know this
because of the IO params and buffer params.
Add flags to the port_use_buffer call. We also want this call to
replace port_alloc_buffer. Together with a new result event we can
ask the node to (a)synchronously fill up the buffer data for us. This
is part of a plan to let remote nodes provide buffer data.
Add a tag field when creating a memmap so that we can do lookup on it.
This makes it easier to implement the tracking of mappings for io areas.
Remove custom io memory tracking and use the tags.
Add flags to spa_chunk to make data corrupted. The flags on the buffer
stay constant for the life of the buffer. Add flags to mark memory
readable and writable. Mark memory readonly in v4l2-source.
Pass the daemon activation area to the client in the transport event.
This never changes and need to be handled differently from the other
activation areas.
Use the right flags when importing memory.
Add the (desired) memory type to mempool_alloc.
improve some debug.
Track memory on buffers in the server and simplify some memory tracking
on the client.
Don't send internal memblock flags to client.
Add read/write flags to memblock.
Add a memory pool to manage blocks of memory. Make it possible
to allocate and import blocks.
Add add_mem and remove_mem to the core events to signal a client
of a block of memory. Remove the client-node add_mem.
Make a global pool for memory and a per client pool where we
import and share the memory we need with the client.
Use the new memory pool to track and map memory in clients.
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..
Use a new rate_match io area to exhange rate matching info between
sink/source and resampler.
Compensate for the rate match delay when scheduling timeouts.
Let the resampler notify the source of how many samples it needs to
produce the desired quantum. Make sure we keep an extra buffer in
the device to be able to make this possible.
Let the adapter directly call the slave node process function.
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.
