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Since the sound capture buffers all sound in newly allocated memory,
the "stream time" represented by those buffers can be accounted for
immediately upon reading into the buffer. Doing it later in the
different threads on the other side of the queue, especially after
encoding, is an unnecessary pile of variable capacitance that just
makes things less synchronized for zero gain.
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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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Now users can easily differentiate which rmd subtasks are
busy by using top-like tools in show-threads mode.
Also aids in troubleshooting...
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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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