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Since the frame timer implements a frame counter, and that frame count
is propagated through the get->encode pipeline for samples that get
through, any missed frames are noticed and dealt with making it not
lossy from the point of the timer down to the encoded stream in terms
of number of frames.
It's certainly lossy in terms of the contents of those frames, but
synchronization is all about the temporal domain and as long as the
frame counts all make it into the stream as frames, we can account
for them at the timer in terms of avd.
This in combination with the other commit moving the audio side of
avd maintenance immediately upon capture into the raw buffer, shrinks
the variable capacitance separating the audio timer and the frame
timer to virtually nil. As a consequence, the frame timer can now
be much more accurate in terms of how much longer/less to sleep or
if a frame should be dropped to get the timer caught up.
When avd was being maintained at points far removed from the actual
things they represented there was too much elastic capacitance in
there for sync_streams() to inform its adjustment accurately.
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This requires a bit of adjustment in the get_frame time_cond wait
loop so it still services the event loop when woken without advance
At least now get_frame has no explicit pause code, but it does
require the timer keep firing while paused so it signals time_cond.
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these concepts may return but not in this form
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This is focused on keeping --on-the-fly-encoding in sync even
over long videos. The existing code inevitably would fall into a
permanently negative pdata->avd value letting things get
increasingly out of sync and never correcting.
Before removing the vestigial negative avd "don't wait" logic
from get_frame when this permanently negative avd state was
entered, get_frame would just start sampling at an unregulated
fps.
The timer thread which drives get_frame now consults avd on every
tick, Depending on which which half is ahead, the timer will
either cause get_frame to drop frames by advancing the frameno by
more than one, or it will adjust its sleep delay in proportion to
the delta.
See comments in rmd_timer.c for more details.
Note that in testing especially with a loaded system I observed
some surprisingly large deltas where multi-second sleeps occurred
to let the sound catch back up. I expect to revisit this issue
more in the future, but would just like to get things more
correct for now.
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Name the timer and sound capture threads as well, and fixup the
rmd{Encode,Cache}Sounds names -> rmd{Encode,Cache}Sound
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usleep() is deprecated by posix in favor of nanosleep(), nanosleep
doesn't dick with signals so it's generally better anyways.
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This restores the recordmydesktop/ subdir as root from the mirror I
cloned by fork from.
I have no particular interest in the gtk/qt frontends and it doesn't
appear they were part of a single tree in the past. But I will
probably preserve backwards compatibility of the cli so they can
continue to work with this fork installed.
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