Add some guards against doing processing when there has been an error or
the node is not started. Set error status to IO. Continue driving on IO
errors.
In media-sink, there's no need to set RCVBUF.
In media-source, we don't need to set NONBLOCK, as reads are done with
DONTWAIT. Don't set SNDBUF as it's not needed there. Don't set RCVBUF,
but use the (big) kernel default value: decode-buffer will handle any
overruns. Small values of RCVBUF might cause problems if kernel is
sending packets in a burst faster than we wake up.
Delay output by one packet, so that we never need to wait for
node_process to supply more data when a packet is due out, and can write
audio packets at exactly equal intervals (up to timer/io accuracy).
In principle, this should not be necessary. However, enable it for now,
in case this improves the various stutter/etc. bug reports.
After flushing a packet, encode the next one immediately if we already
have the data. This makes the flush timing more accurate (std ~4x
smaller) as we don't need to wait for the encode.
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.
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.
The free buffer check must happen before writing to check for leftover data in buffers. In case data is left over from previous submission, bitpool mustn't be increased.
Also improved logging by adding bitpool to the log message.
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.
For backward compatibility with old Wireplumber releases, support the
old api.bluez5.a2dp.sink/source names, and use them in object events
instead of the media.sink/source names.
We can't determine which remote endpoint or device the
SelectConfiguration() call is associated with. For LE Audio BAP, as this
method is called only for the Initiator we set the whole instance as a
Central/Initiator.
This flag is unset on BAP media endpoint removal.
