Signed-off-by: Dylan Aïssi <dylan.aissi@collabora.com>

We don't need to use the minreq field. The fragsize is only updated when
adjust_latency is used.

Only send 2/3 of the fragsize

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.

We should not generate a warning.

The io is set to NULL when the port becomes unnegotiated.

Instruct the session manager to save the volumes when set with the
ALSA ctrl API.

Make sure all jack nodes from the same pid are in the same group.

It is normally not a problem for different JACK clients to live in
different groups and be scheduled by different drivers with different
quantum and rate.

Some applications (like Carla), however, make multiple clients and
expect all of them to use the same quantum and rate. Enforce this by
adding all jack clients to the application jack group.

When the device is not running but has a format, close/open the
device to get all the available formats again. Do the same when
setting a format.

Otherwise, the configuration space of the device is restricted to the
current negotiated format and we can't query the other possibilities
or change it.

Fixes #2625

See #1707

When the pulse context or stream is in error, destroy the module.

The sign of the error was wrong, making the rate controller adjust in
the wrong direction and make things worse.

See #2548

Rate changes can happen very often when a stream is doing rate control,
so don't emit the changes every time.

could_use_rtkit -> can_use_rtkit.
Only warn when setting nice and there is no rtkit fallback.
Always call set_rtlimit, it does not have an rtkit implementation, if just
has some properties with limits.
Only try to set nice again with rtkit when we required rtkit for some
functions.

Previously module-rt only checks if the user have the permission to use
realtime scheduling, and will unconditioally disable RtKit if they do,
even when they don't have the permission to set nice or rlimit.

A lot of code calls spa_hook_remove() from error paths where the hook
and therefore the list may not have been initialized.
This leads to null-derefences.

This function can be used to test if a list has been initialized.

Only update the resampler rate when we ask for more data, when we have
more input data, use the previously configured rate to calculate how
many samples we will consume.

Fixes resync errors with multiple sources. One source would do rate
matching, audioconvert would ask it to produce X samples, the source is
scheduled to produce the samples, the rate match is updated with the new
rate correction, audioconvert is scheduled again. It should now use the
X samples it asked to produce and apply the new rate correction for the
next iteration.

If no packets have been received and spa_bt_decode_buffer_process is
called, this->packet_size.max == INT32_MIN, which can give overflows.

Guard against this condition, although it should be harmless.

FreeBSD 13+ versions do have O_PATH flag, but FreeBSD 12, which is still supported
does not.

Some devices have really long names, especially when taking localisation
into account due to multibyte characters.

This fixes an assertion failure in PortAudio's JACK backend when the
monitor client name is longer than than JACK_CLIENT_NAME_SIZE.