Add a new JACK sink/source pair that translates to a single JACK
client.
The JACK playback port appears as PipeWire source and is processed
directly, synchronously, through the complete pipewire graph into
the PipeWire sink that is then made available on the JACK capture
ports.
Because all this happens in the same JACK cycle with no delay, the
latency is 0. A jack_iodelay on the JACK server has exactly the same
latency as the jack_iodelay on the PipeWire side.
The PipeWire sink and source are forced into the same rate and
buffer_size as the JACK server and can't dynamically change.
This only supports Audio for now.
When we consumed all the buffer data, don't clear all the fds but only
those that were already consumed in the message. It is possible that we
already have fds for the next message and we don't want to discard
those.
Fixes some intermittend memory map errors.
Calculate the stats at the start of the new cycle. The results will be
about the previous cycle but this gives more accurate results because
we can also include awake and finish times of remote nodes.
Make sure not to change the status of the activation in the ready event
so that we don't overwrite the status of the last cycle yet.
This means we can always set the AWAKE and awake_time, the remote node
might update it when triggered but that's ok.
After processing we can update the FINISHED state for non-remote nodes,
the remote nodes will update it after they complete the process
function.
Pass the ready status to the client-node using the state array.
Don't just SPA_STATUS_HAVE_DATA on the server side but use the value
from the client.
This avoids some potential extra work when a driver sink pulls in data
with the NEED_DATA ready callback but then the server performs the
actions (tee) as if it were SPA_STATUS_HAVE_DATA.
Intercept the stream volume/mute and set it as the remove volume/mute.
Listen for remote volume/mute changes and set this as the local stream
volume. Make sure the adapter is using 1.0 software volume but reports
the real channelVolume of the remote stream.
Make the XOpenDiplay call failure print an info message instead of a
warning. We usually ignore this error in the config file. Add a
suggestion for how to fix this issue in the info log.
Fixes#2918
Actually convert the aec_args argument to something that the
echo-canceler will understand. Add the beamforming and mic-geometry
and target_direction properties.
Make the channels configurable for capture, source, playback and
sink streams. The playback and sink stream need to have the same number
of channels but the layout can be different.
Use init2 of the echo-cancel plugin when available to pass this info
around. The echo-canceler will then further adjust the settings to make
things work.
The raw biquad takes its values from the config and then updates the
control ports with them in activate. At runtime, the changes to the
ports are applied to the biquad.
Other biquads take values from the config ports in activate.
When the timing_port is 0, just don't send out an initial timing packet.
When we receive a timing packet, reply to the same address/port that the
timing packet was sent from.
Fixes#3133
We need to be able to fill up one quantum of data when the stream
requests it so suggest a quantum_limit buffer size. Normally, the
converter will further improve the buffer size parameters when
resampling is involved but this is a good default for when working in
passthrough mode.
We don't actually use the stream buffering features so we're happy with
just 1 or 2 buffers.
Make an option to allow link.passive properties and set it to false by
default.
This effectively ignores the link.passive properties from the session
manager, jack clients and pw-link when set. This is a good idea because
the logic for making passive links is better handled in the core.
We are not allowed to call pipewire methods from any other thread than
the main thread so use invoke to schedule a module unload from the
pulseaudio thread.
Fixes some infinite loops when the work-queue list gets corrupted.
Filling a buffer with zeros to produce silence is wrong for
unsigned sample formats and compressed sample formats. This is
silence with an offset. A silent stream with unsigned or
compressed samples mixed with another stream produces a clipped
stream. To hear the problem start QEMU with
qemu-system-x86_64 -accel kvm -smp 4 -m 4096 \
-audiodev pa,id=audio0,out.mixing-engine=off \
-machine pc,pcspk-audiodev=audio0 \
-device ich9-intel-hda -device hda-duplex,audiodev=audio0 \
-boot d -cdrom Fedora-Workstation-Live-x86_64-37-1.7.iso
on a host configured to use PipeWire and start audio playback on
the guest HDA device.
PA_SAMPLE_U8 is the only unsigned PulseAudio sample format.
There is no need to care about unsigned multi-byte formats.
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Emit both the bound_id and bound_props events from the protocol on
the core_resource.
Doing the dispatching of the bound_id/bound_props in the core to the
proxy doesn't handle the case where the client has a listener directly
on the core_resource.
Fixes#3109
this event extends the bound_id event and sends the global properties as
well.
This can be used to get the object.serial, for example.
It can also be used in the future to let the server generate unique
property values, like the node.name, and let the client know about the
new property value.
