@DEFAULT_MONITOR@ finds the default sink but returns is_monitor true.
Always lookup the device based on the name and index. This transform
the special device names like @DEFAULT_XXX@ to the default device.
Always use the found device name as the target. Make sure to set the
CAPTURE_SINK property when dealing with a monitor.
Fixes#3284
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.
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>
When a stream uses the FIX_ flags it should negotiate to the format of
the sink or source it connects to. To do this, look up the sink or
source and look at the format, use this as the allowed format for the
stream when the FIX flags are set.
Make it still possible to override with with properties. Use
audio.position to make it possible to set a channelmap override.
Use a different key than the usual one to select an audio format when we
are fixating a stream format to avoid confusion.
So pulse.fix.rate, pulse.fix.format, pulse.fix.channels are now used to
force a specific format when the stream has the FIX_ stream flags.
Add support for using custom format, rate or channels when the streams
asks to fix those parameters for us.
Some streams might expect to see S16LE when they connect to a sink in
S16LE when the FIX_FORMAT flag is set but this is not the case in
PipeWire because the audio DSP pipeline works in F32, and so F32 is
choosen. Make it possible to use a pulse.rule with a audio.format
property to control this.
Make a real debug context with a log function and move it to a new file.
This way we don't need to redefine a macro.
Make a new context for debugging to a log file. Make new functions to
debug to a log file.
Move the stringbuffer to string utils.
Integrate file/line/func and topics into the debug log.
We can remove some more things from the pipewire log_object function and
also add support for topics.
The pulse API uses either the node.name or object.serials so pass this
along in the TARGET_OBJECT instead of the NODE_TARGET now that
wireplumber handles this.
When we start the drain, we unpause the stream. When we conplete the
drain, we unpause again, which does nothing when the stream was already
unpaused. However, this leaves the drained state on the stream and so
the stream will never be able to play new data.
Trigger a new pw_stream_set_active() with the current stream state to
clear the drained state.
Fixes#2928
Add a new config section that can contain a set of commands to run after
starting.
There is only load-module available now but it can be used to remove the
dependency on pactl when starting the server.
Only enable pulse.idle.timeout for selected applications
(speech-dispatcher) and let other applications be. We can also
lower the minreq and min-quantum of speech-dispatcher now.
Fixes#2880
We don't need to pass the client to the module create and load
functions, they can work without a client.
The only place the client is used is to access the properties to make a
new connection to pipewire. This is also however not a good idea, we
should simply use the defaults used by the context or else a client
could set strange properties like remote.name etc for these internal
connections.
Also removing the dependency of the client will make it possible to load
modules from the startup script or other modules later.
When a client is not sending any data when it should be and causes an
underrun, mark it as idle and record the timestamp.
When a client is idle for pulse.idle.timeout seconds, set the stream
as inactive. When more data is received, set it back to active.
Add a pulse.idle.timeout option to set a global server default or
a per-stream value. Set the server default to 5 seconds. A value of 0
can be used to disable this feature.
With this change, badly behaving clients that are not sending any data
will be paused so that the sinks can suspend to save battery power.
Fixes#2839
Make a method to pause and resume a stream and keep track of the paused
state of the stream. Use this function instead of setting the stream
inactive/active so that we get nice logging for each state change.
When no maxlength is given, we use the MAXLENGTH value but we need to
round it DOWN (instead of up) because our buffer is only MAXLENGTH big.
Use CLAMP where we can.
When in capture mode, the maxlength exceeds MAXLENGTH, scale down the
fragsize instead.
Fixes noise in audacious when playing 6 channels sounds. float32 * 6
channels with the maximum buffer size would result in the ringbuffer
being overwritten.
Reducing the latency is just papering over the issue in #2702. The
real fix is to limit the blocksize to the fragsize like what is done
in 00a234daf2
Reducing the latency then also causes regressions like #2715 so don't
do that anymore.
Fixes#2715
Move the latency fraction calculation to fix_ functions so that the
new latency rate can be used when creating the streams.
Actually set the requested record attributes on the stream instead
of modifying the defaults.
See #2671
PulseAudio configures half of the fragsize as the source latency. It
also sends chunks as soon as they become available.
This means that we also need to configure the source with half of
the fragsize latency and send in chunks of fragsize/2. Keep this in
the unused (for record) minreq field.
