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Fix for MAD sample scaling routine.

CQTexperiment
vspader 2008-05-03 19:37:10 +00:00
parent 10669a2db3
commit 549a0cf2da
1 changed files with 20 additions and 49 deletions
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69
Plugins/MAD/MADDecoder.m
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@ -321,62 +321,33 @@
} }
/** // Clipping and rounding code from madplay(audio.c):
* Scale PCM data /*
* madplay - MPEG audio decoder and player
* Copyright (C) 2000-2004 Robert Leslie
*/ */
static inline signed int scale (mad_fixed_t sample) static int32_t
audio_linear_round(unsigned int bits,
mad_fixed_t sample)
{ {
BOOL hard_limit = YES; enum {
MIN = -MAD_F_ONE,
double scale = 1.0; MAX = MAD_F_ONE - 1
};
/* hard-limit (clipping-prevention) */
if (hard_limit)
{
/* convert to double before computation, to avoid mad_fixed_t wrapping */
double x = mad_f_todouble(sample) * scale;
static const double k = 0.5; // -6dBFS
if (x > k)
{
x = tanh((x - k) / (1-k)) * (1-k) + k;
}
else if(x < -k)
{
x = tanh((x + k) / (1-k)) * (1-k) - k;
}
sample = x * (MAD_F_ONE);
}
else
sample *= scale;
int n_bits_to_loose = MAD_F_FRACBITS + 1 - 16;
/* round */ /* round */
/* add half of the bits_to_loose range to round */ sample += (1L << (MAD_F_FRACBITS - bits));
sample += (1L << (n_bits_to_loose - 1));
/* clip */ /* clip */
/* make sure we are between -1 and 1 */ if(MAX < sample)
if (sample >= MAD_F_ONE) sample = MAX;
{ else if(MIN > sample)
sample = MAD_F_ONE - 1; sample = MIN;
}
else if (sample < -MAD_F_ONE)
{
sample = -MAD_F_ONE;
}
/* quantize */ /* quantize and scale */
/* return sample >> (MAD_F_FRACBITS + 1 - bits);
* Turn our mad_fixed_t into an integer.
* Shift all but 16-bits of the fractional part
* off the right hand side and shift an extra place
* to get the sign bit.
*/
sample >>= n_bits_to_loose;
return sample;
} }
// End madplay code
- (void)writeOutput - (void)writeOutput
@ -433,7 +404,7 @@ static inline signed int scale (mad_fixed_t sample)
for (i = startingSample; i < sampleCount; i++) for (i = startingSample; i < sampleCount; i++)
{ {
signed short sample = scale(channel[i]); signed short sample = audio_linear_round(bitsPerSample, channel[i]);
outputPtr[0] = sample>>8; outputPtr[0] = sample>>8;
outputPtr[1] = sample & 0xff; outputPtr[1] = sample & 0xff;