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Switched from linear interpolation to antialiased point sampling

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when upconverting low-frequency sound streams. This should
de-muffle low-frequency streams, treating them has having
hard edges at sample boundaries.
This commit is contained in:
Aaron Giles 2009-11-25 14:41:09 +00:00
parent fe2987870a
commit 409223e919
1 changed files with 25 additions and 11 deletions
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36
src/emu/streams.c
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@ -22,7 +22,7 @@
rate.
Each stream also has a callback function. This function is called
with an array of input sample streams, and an array out output
with an array of input sample streams, and an array of output
sample streams. The input sample streams are automatically resampled
by the streaming engine to match the stream's current sample rate.
The output sample streams are expected to be generated at the
@ -30,7 +30,7 @@
Before the callback can be invoked, all the inputs that flow into it
must be updated as well. However, each stream can have an independent
sample rate, so this isn't as easy as it sounds.
sample rate, so this isn't as easy as it sounds.
To update a stream, the engine must iterate over all the inputs. For
each input, it requests that input to update to the current time.
@ -933,21 +933,35 @@ static stream_sample_t *generate_resampled_data(stream_input *input, UINT32 nums
}
}
/* input is undersampled: use linear interpolation */
/* input is undersampled: point sample except where our sample period covers a boundary */
else if (step < FRAC_ONE)
{
while (numsamples--)
while (numsamples != 0)
{
int interp_frac = basefrac >> (FRAC_BITS - 12);
/* compute the sample */
sample = (source[0] * (0x1000 - interp_frac) + source[1] * interp_frac) >> 12;
int nextfrac, startfrac, endfrac;
/* fill in with point samples until we hit a boundary */
while ((nextfrac = basefrac + step) < FRAC_ONE && numsamples--)
{
*dest++ = (source[0] * gain) >> 8;
basefrac = nextfrac;
}
/* if we're done, we're done */
if ((INT32)numsamples-- < 0)
break;
/* compute starting and ending fractional positions */
startfrac = basefrac >> (FRAC_BITS - 12);
endfrac = nextfrac >> (FRAC_BITS - 12);
/* blend between the two samples accordingly */
sample = (source[0] * (0x1000 - startfrac) + source[1] * (endfrac - 0x1000)) / (endfrac - startfrac);
*dest++ = (sample * gain) >> 8;
/* advance */
basefrac += step;
source += basefrac >> FRAC_BITS;
basefrac &= FRAC_MASK;
basefrac = nextfrac & FRAC_MASK;
source++;
}
}