Make it possible for a aec-plugin to provide props/params
through module-echo-cancel and make it possible for a
aec-plugin to listen and react to these props/params getting updated.
Add method enum_props() that is used under setup_streams() to
get props from aec_plugin.
Add get_params() and set_params() for updating and getting
aec-plugin params under the input/output_param_changed calls
and getting params under setup_streams().
While we process the Pause, the node might sill emit some events or do
processing, which would give an error because we already removed the
node from the graph.
Instead, remove the node from the graph when the node state is updated
to the new state.
Also check the active state of the input node, not just if the node
was added to the graph because a node can become inactive and only
later be removed from the graph.
The graph cycle goes: driver timeout -> process output nodes -> process
driver node. Hence, driver should produce buffers in the timeout,
otherwise there's one quantum extra latency.
Make the bluez5 media/sco sources as drivers put a buffer to io before
indicating ready, and as follower do it in process. Also make checks if
io == NULL, and don't set io->status to HAVE_DATA unless there really is
a buffer ready.
Introduce `unregister_media_endpoint()` to unregister the specified
media endpoint object from dbus. Moreover, move the logic that decides
whether or not a particular codec should be registered for a given
direction into `endpoint_should_be_registered()` and use that in
both `(un)register_media_endpoint()`.
Do not use old delayed data in play_buffer when a new stream is started.
Copy silence to output until play_buffer has been filled with new data
instead.
Driver timeouts need to be started/stopped when we switch from follower
to driver or vice versa.
The BT sources fail to do this, so fix it. Sinks already do it right.
Bind to the settings metadata.
Add a property to control if a client will set a temporary or global
quantum whith jack_bufsize.
Make a match rule for jack_bufsize and force a global quantum change.
Fixes#1273
Add a target.delay.sec property to module-loopback that uses a
ringbuffer to further delay the signal to the requested value. This
also takes into account the graph delay to get an end-to-end delay.
Add a -d property to pw-loopback to control this.
Implement latency_msec on the pulse module with this new property so
that it behaves similar to pulseaudio.
...except in the build_session_managers job.
This decouples pipewire's CI from wireplumber's dependencies
and potential failures. The build_session_managers job is supposed
to catch session manager integration errors, not any other job.
Use separate timers for driving graph and for flushing, since they don't
have the same period.
Flushing is done based on the time positions of the next sample to be
written, so it will stay in sync with the graph. Because writing too
much data to SCO sockets generally only causes the device to skip ahead,
we don't need to handle the case where the writing has been lagging.
This fixes simultaneous playback to both ALSA and SCO sinks from the
same graph, with SCO as driver, which previously produces broken sound
(e.g. with pw-play --latency 512, linked to the two sinks) ALSA nodes
require regular driver intervals, which was not true previously.
Bigger buffer allows for more fluctuation in transmission rate without
sound glitches.
It doesn't matter much for latency, as under normal conditions we are
not producing data faster than the BT adapter can transmit, so the
buffer generally is almost always empty or full, and in the latter case
we have to reduce the bitrate.
Pulseaudio requires that we call pa_stream_write with a multiple
of frame_size bytes. Because our ringbuffer is a power of two, this
might cause problems at the edge of ringbuffer where a sample is
split between the end and beginning of the ringbuffer.
Avoid this by letting pulse allocate a buffer instead and memcpy
the requires samples into it.
Fixes multichannel output on module-pulse-tunnel.
Do not start the playback device until there is data to play. Otherwise
time consuming configuration of other nodes (such as setting hw params
of a capture device) may be done after playback has been started, which
may cause xrun.
Instead of calling the default implementation that does not implement
these methods, keep the last thread-utils around in a global and use
that instead.
These are fallback functions that are not implemented. It is expected
that the thread-utils is used from the context, which is a complete
implementation provided by module-rt.
The parsing functions expect float values in the default locale so use
the spa_dtoa function to generate such a float.
Fixes setting params with floating point values when the locale is not
the default locale.
