Just leave them be. Something else is supposed to changed the volumes
if needed. In the usual case where nothing changes, we don't want to
override the volumes.
Remove the soft mute/volume events, add a new method to get the soft
volume and use the volume_changed event to emit the changed soft and
monitor (real) volumes event for the node.
Make sure the monitor ports always uses the monitor volume, which is the
real volume unaffected by the mixer volumes.
This configures the soft and real volume on the sink/source in all
cases and makes the monitor port follow the real volume of the sink.
See #897
Implement monitor volumes in the merger. There are two volumes,
the channel volume and the monitor volume. The monitor volume
is always applied.
By default the monitor volume will now follow the main volume of
the node. This can be disabled with a monitor.channel-volumes
property.
See #674
We also need to remap channels for the splitter and merger.
Remember the port-config format and its channel layout. Internally,
we use a canonical channel layout which is simply all channels sorted
by id. Remap the channels accordingly.
Fixes#445
This is needed for example for Clang compiler which uses different
annotations than GCC. It will make WebRTC to happily use PipeWire
since the spa library is header-only and WebRTC defaults to use
Clang with -Wimplicit-fallthrough.
We only need as many ports and buffer data as the maximum number
of channels, which is 64.
Fix empty output size. We're only ever going to fill this with
float samples.
Use the DSP media subtype to describe DSP formats. DSP formats
don't include the rate, channels and channel position in the
format and must use the rate and duration from the position io. This
makes it possible to later change the samplerate dynamically without
having to renegotiate the graph.
The same goes for the video DSP format, which uses the io_video_size
from the io_position to get the size/stride. Set this up in the node
based on the defaults from the context.
Make it possible to define defaults in the daemon config file, such
as samplerate, quantum, video size and framerate. This is stored in
the context and used for the DSP formats.
This is more in line with wayland and it allows us to create new
interfaces in modules without having to add anything to the type
enum. It also removes some lookups to map type_id to readable
name in debug.
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 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.
