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
3f6fe392 uses headers from /usr/include/lv2/atom/. but this leads compilation
failure for some distros (i.e. ubuntu 18.04) as they have some
different location for these headers. One can find these headers
at /usr/include/lv2/lv2plug.in/ns/ext/atom/ (for ubuntu 18.04)
instead /usr/include/lv2/atom/. So guard them with __has_include.
and mention other possibilities.
Fixes#2670
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.
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
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.
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.
