Sync is enabled when clients need time to move to a new location.
It's a bit like GStreamer preroll after a seek. Clients that need
time, increment the sync_total. Whenever a seek is done, the server
waits in the Starting state until the sync_pending is 0 (or timeout
later).
Improve atomic operations
Add an offset to apply to the clock time before we can compare to the
segment values. This way we can keep the segment start independent of the
clock values and we only need to adjust the offset when paused. It's
like the base_time in GStreamer to calculate the running time.
Move fields from the io_position to io_segment. The segment contains
the mapping between raw clock time and stream time in various
formats. We keep an array of pending segments available in the
io_position field so clients can anticipate changes.
Make looping a flag in the segment instead of a state.
Prepare for segment masters. These will be registered clients that
are responsible for updating parts of the extended segment info.
Add namespace to some defines.
Move some things around. Move the duration of the current cycle
to the clock. Also add the estimated next timeout to the clock.
Add a generic media specific counter to the clock.
Clean up the position_bar info. We can do with only a double beat
value and make the signature in floats.
Flesh out the io_position info. This has now the information needed
to convert a raw clock time into a stream time. It basically has
the same kind of features as GStreamer segments such as looping,
variable rate playback etc.. It also contains the state of the
timeline (paused/playing) and it can be used to update the position
and state from clients.
There is also extended information in the position field that
clients can update when they can.
Plugins basically only update the clock info they get (and use
the position info to check if they are slaved or not).
Before each cycle, check if there is a pending position update and
apply it.
Add flag to always assign a node to a driver. This makes sure that
even when the node is not linked to anything, it will still be
scheduled by an active driver. This is needed for JACK support.
clean_convert() removes the internally negotiated formats but
it does not set the format (or buffers) of the externally visible
ports. Therefore, don't clear the buffers_set flags.
Instead, clear the buffers_set flag when we explicitly reconfigure
the ports, when we also clear the format.
Also clear the port buffers when we set a NULL format.
Fixes#178
Makes a device with a source and sink that automatically
proxies all physical ports from jack. Jack then drives our
PipeWire pipeline from its own thread.
When we do invoke from the loop thread, first process the pending
invoke items before handling ours. This can be used to sync the
invoke queue before cleanup.
Keep the output port buffers in a special mix info. They should
stay there even when the mix is removed and should only be cleared
explicitly with a NULL Format or 0 use_buffers.
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.
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.
