The code actually handles this case correctly; the algorithm is linear
interpolation between the two closest samples, and the way it is written
should work correctly with any ratio.
The primary motivation here is to make sure we submit samples from the
CPU thread. This makes sure the timing of related interrupts accurate,
and generally keeps the different kinds of audio synchronized. This will also
allow improvements to audio dumping functionality.
The new code is also more concise because it gets rid of some duplicated
audio mixing code.
This reverts commit 4990b8910b.
The commit is causing substantial performance issues for the DSound
backend which I somehow didn't catch during testing.
Pretty straightforward; IDirectSoundNotify lets you register for
notifications after a certain amount of sound has played, so use that
instead of depending on Update() notifications from the CPU thread.
Also, while I'm here, reduce the buffer size by a factor of 4; this seems
to reduce the latency, although the difference is sort of subtle.
This is good for a couple of reasons: one, it gets rid of duplicated code,
and two, DSP emulation shouldn't need to interact with audio in the first
place.
degasus
Pierre Bourdon
Tillmann Karras
Lioncash
magumagu
shuffle2