Add a monitor.passthrough option. This will pass all latency information
directly between the port and its monitor ports.
This is interesting when the adapter (and audioconvert) is used with a
null-audio-sink that simply forwards the data to a real sink/souce. In
that case, we want the sink/source latency to be passed unmodified.
Set the monitor.passthrough on the pulseaudio null-sink because
a passthrough virtual sink is the most likely use case for this.
Add some monitor.passthrough default config where it makes sense.
Fixes#3888
Keep track of the valid ports and don't emit port info for
invalid ports. When a listener is added while the ports are being
created, it is possible that the ports are still NULL or invalid.
The rate we get from dlls can have a subsample precision. However,
the check for using process_copy is in sample precision. This means
that an adaptive stream will oscillate rather then lock into the
exact rate.
When starting the converter, calculate the initial size needed by
the resampler to fill one quantum.
This makes it possible to get the requested amount of samples before
the first process call is made.
Update the started and ready state after we suspend/pause the node so
that we don't complain if scheduling happens between setting the fields
and actually stopping the follower.
Also only complain when the scheduling happens when the node is not
ready. It is possible that the node is scheduled before we manage to set
the started field.
Move the driver and warned bits after the int field in the struct so
that they are placed in separate memory.
Otherwise, a write from the data thread might race with a write from the
main thread and leave the bits in the wrong state.
Don't blindly clear the format when EnumFormat changes. This will
just stop the node without renegotiating.
We should probably find a new best format, check if it changed and
then Stop/configure/Resume the follower with the new format.
This fixes a stall when a node is running and you change the allowed
codecs.
Don't try to allocate each time port buffers are set but only once
before we start procesing.
Also allocate enough temp buffers are there are ports. This saves us
quite a bit of memory in the normal case.
When we are operating in passthrough, let the Start command pass through
to the follower in all cases. Only do the negotiate/buffers when not in
passthrough.
This fixes a case where the buffers are cleared on the alsa node and it
Pauses but then never resumes in a Start because the node is already
started. The real problem is probably somewhere else (in PipeWire) but
for now this will improve things in passthrough.
Don't just forward the tag and latency events to the follower but let
the audioconvert aggregate and emit the updated tag/latency event
that is then configured on the follower.
When using the DSP mode of the audioconvert, this results in an
accumulated latency/tag from all the DSP ports instead of just
the last DSP port param update.
Put properties with media. prefix in tags in pw-cat.
