When reading the timerfd gives an error, we should return right away
because the timeout did not happen.
If we change the timerfd timeout before reading it, we can get -EAGAIN.
Don't log an error in that case but wait for the new timeout.
When we try to play data but the ringbuffer is full, we need to start
the device or else we will stay in this situation forever and stay
silent.
Fixes#2830
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.
uint32_t i;
for (i = 0; i < SPA_N_ELEMENTS(some_array); i++)
.. stuff with some_array[i].foo ...
becomes:
SPA_FOR_EACH_ELEMENT_VAR(some_array, p)
.. stuff with p->foo ..
DSD64 would be a rate of 44100 * 64 / 8. When packed in U32_BE, we would
negotiate 44100 * 8 / 4 (88200) with the device, this means all rates
from 88200 and up are allowed for DSD64 in U32_BE.
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
We need to check the last offset against the size of the buffer, not the
remaining size in the buffer.
When the writing is split, this could cause the buffer to be reused
wrongly.
See #2536
Use the NEAREST flag when setting a format. This only works for raw
formats and will update the format with the nearest accepted rate
or channels. We can then query the real configured format and use that
for the converter.
This makes things work when a driver tells us it can do 44100Hz but then
refuses and changes the rate to 48000.
See #2197, #2457, #2455, rhbz#2096193
Assume that capture and playback nodes from a device have different
clocks. This enables the adative resampler to match them. A lot of devices
actually have slightly different rates and would work out of the box
with this fix.
Make an exception when the card is configured in the pro audio profile.
Then we force the same clock on all device nodes and avoid resampling
and rate matching. This can still be changed with a session manager
override.
Use the max error to do a resync. Don't reset the dll, there is no
reason for that.
Don't use _rewind, but instead limit the amount of samples we read and
write
Should keep more stable sync in most cases.
Pass the current time around in various functions.
Make a higher precission htimestamp based get_delay() function. Seems to
work fine for playback but not for capturee.
When we get a commit error, try to resync our pointers. This fixes a
problem of endless commit errors after a quantum change because it never
manages to resync properly.
Parse the quantum_limit parameters and use this to scale the buffers so
that they can contain the maximum allowed samples instead of the
hardcoded 8192 value.
See #1931
