... from the Audio thread to the PortAudio thread; the old
ScrubQueue::Consumer() function keeps only a vestigial purpose to prevent the
scrub queue from blocking.
... so there is only one update per track of the atomics in RingBuffer in each
pass of the loop in FillBuffers, which will be needed to synchronize RingBuffer
and TimeQueue correctly.
... this may be more than the batch size used in ongoing playback.
It is expected that this larger batch size is used only once when priming
the queue before starting play. But then FillBuffers() may attempt to
refill up to the minimum in case demand is outpacing supply.
Thus the new number defines a "yellow zone" for the queue.
1. The rate actually used by the sound card now sets the stopping sample number.
2. The tolerance in the 'stop position' is additionally trimmed to 20 samples.
... Rather than the confusing old terminology of "warped" time, meaning the
real time after accounting for time track, though that persists mostly in the
names of variables now mostly used only within PlaybackSchedule.
"Track" time refers to a position in a wave track, as indexed by the time
ruler.
"Real" time is indexed by the other scale of numbers drawn in the time track,
which is drawn on the screen with nonuniform spacing (so "warped" in that
sense), but it corresponds to uniform actual time.
... Adjust whenever recording, regardless whether there is any overdub
playback too.
May adjust by discarding samples when the Track Shift after Record preference
is negative (as by default); or, by inserting silence into the start of the
recording, when that preference is positive, though that may be less useful.
Perhaps the preference control should no longer be called "Track Shift after
Record."
1) When the program detects this, insert zeroes into the recording to keep the
other good parts synchronized.
2) When recording stops, a message box alerts the user, and a label track is
added showing the lost parts, labelled with consecutive numbers.
3) A menu item visible in alpha builds only is added to Tools, to simulate
recording errors at random times and test the reporting feature.
... This even makes it possible to remove the prohibition of undo and redo
during transport, not that we want to though. Playback and recording will
continue, using track objects that might not be in the current project.
... Write only one variable in audacityAudioCallback, to be read (maybe in
another thread) by AudioIO::MidiTime().
The non-Alsa case behaves essentially as before: it wasn't broken, so it
isn't fixed, though it is rearranged.
... I am not sure that this will solve all problems of jittery play, but I do
think this will solve the problem of a rush of slightly late notes at start of
play.
... and hoping it is positively helpful for Linux.
In AudioIO::MidiTime(), compute one of the terms by different means.
Use PaStreamInfo::outputLatency.
Do not use the difference of PaStreamCallbackTimeInfo::outputBufferDacTime
and PaStreamCallbackTimeInfo::currentTime.
Which debugging shows is very nearly the same value for Windows and Mac.
But we suspect the PaStreamCallbackTimeInfo fields are not correctly reported
on Linux.
... At least it fixes the gross problem; but there seems to be a small
accumulation of error still each time around the loop, that I don't understand
yet.
