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audio.c
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audio.c
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#define I_WANT_INTERNAL_AUDIO_STUFF
#include "audio.h"
/*
Differences from reference DFPWM implementation:
- Samples are 16-bit.
- Antijerk filter follows the C implementation.
- Compression when equal and strengthless MIGHT be around the other way.
*/
u16 audio_rb[AUDIO_RB_SIZE][2];
int audio_rb_start = 0;
int audio_rb_end = 0;
//
// UNOPTIMISED IMPLEMENTATION
//
inline void dfpwm_update_model(int *q, int *s, int ri, int rd, int lt, int t)
{
// calculate strength adjustments
int st, sr;
if(t == lt)
{
st = 0x3FFF;
sr = ri;
} else {
st = 0x00;
sr = rd;
}
// adjust charge
int nq = *q + ((*s * (t-*q) + 0x2000)>>14);
if(nq == *q && nq != t)
nq += (t ? 1 : -1);
*q = nq;
// adjust strength
int ns = *s + ((sr * (st-*s) + 0x2000)>>14);
if(ns == *s && ns != st)
ns += (st ? 1 : -1);
*s = ns;
}
inline int dfpwm_compress_bit(int *q, int *s, int ri, int rd, int *lt, int v)
{
int t = (v < *q || v == -0x8000)
? -0x8000
: 0x7FFF;
dfpwm_update_model(q,s,ri,rd,*lt,t);
*lt = t;
return t;
}
inline int dfpwm_decompress_bit(int *q, int *s, int ri, int rd, int *lt, float *fq, float *fq2, int bit)
{
int t = bit ? 0x7FFF : -0x8000;
int lq = *q;
dfpwm_update_model(q,s,ri,rd,*lt,t);
int ret = *q;
// antijerk
if(t != *lt)
*fq2 = (lq-ret);
else
*fq2 *= 0.999f;
ret += *fq2;
// low pass filter
*fq += (ret-*fq) * 0.5f;
ret = *fq;
*lt = t;
if(ret < -0x8000)
ret = 0x8000;
if(ret >= 0x7FFF)
ret = 0x7FFF;
return ret;
}
// NOTE: len is in compressed bytes!
void dfpwm_compress(int *q, int *s, int ri, int rd, int *lt, int len, s16 *rawbuf, u8 *cmpbuf)
{
int i,j;
u8 d;
for(i = 0; i < len; i++)
{
d = (0x01 & dfpwm_compress_bit(q,s,ri,rd,lt,*(rawbuf++)));
d |= (0x02 & dfpwm_compress_bit(q,s,ri,rd,lt,*(rawbuf++)));
d |= (0x04 & dfpwm_compress_bit(q,s,ri,rd,lt,*(rawbuf++)));
d |= (0x08 & dfpwm_compress_bit(q,s,ri,rd,lt,*(rawbuf++)));
d |= (0x10 & dfpwm_compress_bit(q,s,ri,rd,lt,*(rawbuf++)));
d |= (0x20 & dfpwm_compress_bit(q,s,ri,rd,lt,*(rawbuf++)));
d |= (0x40 & dfpwm_compress_bit(q,s,ri,rd,lt,*(rawbuf++)));
d |= (0x80 & dfpwm_compress_bit(q,s,ri,rd,lt,*(rawbuf++)));
*(cmpbuf++) = d;
}
}
void dfpwm_decompress(int *q, int *s, int ri, int rd, int *lt, float *fq, float *fq2, int len, s16 *rawbuf, u8 *cmpbuf)
{
int i,j;
u8 d;
for(i = 0; i < len; i++)
{
d = *(cmpbuf++);
*(rawbuf++) = dfpwm_decompress_bit(q,s,ri,rd,lt,fq,fq2,d&0x01);
*(rawbuf++) = dfpwm_decompress_bit(q,s,ri,rd,lt,fq,fq2,d&0x02);
*(rawbuf++) = dfpwm_decompress_bit(q,s,ri,rd,lt,fq,fq2,d&0x04);
*(rawbuf++) = dfpwm_decompress_bit(q,s,ri,rd,lt,fq,fq2,d&0x08);
*(rawbuf++) = dfpwm_decompress_bit(q,s,ri,rd,lt,fq,fq2,d&0x10);
*(rawbuf++) = dfpwm_decompress_bit(q,s,ri,rd,lt,fq,fq2,d&0x20);
*(rawbuf++) = dfpwm_decompress_bit(q,s,ri,rd,lt,fq,fq2,d&0x40);
*(rawbuf++) = dfpwm_decompress_bit(q,s,ri,rd,lt,fq,fq2,d&0x80);
}
}
/*int audio_loadchunk(int len, s16 *buf)
{
int ret = 0;
// deal to ring buffer
if(audio_rb_start+len > AUDIO_RB_SIZE)
{
int nlen = AUDIO_RB_SIZE - audio_rb_start;
int nret = audio_loadchunk(nlen, buf);
ret += nret;
buf += nlen*2;
len -= nlen;
}
// calculate remaining sample count
int smpcount = (audio_rb_start > audio_rb_end)
? audio_rb_end - audio_rb_start
: audio_rb_end + AUDIO_RB_SIZE - audio_rb_start;
if(((audio_rb_start+1)%AUDIO_RB_SIZE) == audio_rb_end)
{
return 0;
} else {
}
}
int audio_stealchunk(int len, s16 *buf)
{
int ret = 0;
// deal to ring buffer
while(audio_rb_start+len > AUDIO_RB_SIZE)
{
int nlen = AUDIO_RB_SIZE - audio_rb_start;
int nret = audio_stealchunk(nlen, buf);
ret += nret;
buf += nlen*2;
len -= nlen;
}
if(audio_rb_start == audio_rb_end)
{
}
}*/
// TODO: refactor it to something similar to that which is above
FILE *autest = NULL;
int dfp_q = 0x0000;
int dfp_s = 0;
float dfp_fq = 0.0f;
float dfp_fq2 = 0.0f;
int dfp_lt = 0;
int dfp_ri = 420;
int dfp_rd = 3500;
int audio_stealchunk(int len, s16 *buf)
{
if(autest == NULL)
{
autest = fopen("autest.raw","rb");
if(autest == NULL)
{
// XXX: tmpfile fails in Windows, I don't know why --GM
autest = tmpfile();
if(autest == NULL)
{
memset(buf, 0, len*4);
return len;
}
}
}
int i,j;
u8 tcmp;
s16 traw[8];
for(i = 0; i < len/8; i++)
{
// cmp/decmp
for(j = 0; j < 8; j++)
{
traw[j] = 0;
fread(&traw[j], 2, 1, autest);
}
dfpwm_compress(&dfp_q, &dfp_s, dfp_ri, dfp_rd, &dfp_lt, 1, traw, &tcmp);
dfpwm_decompress(&dfp_q, &dfp_s, dfp_ri, dfp_rd, &dfp_lt, &dfp_fq, &dfp_fq2, 1, traw, &tcmp);
for(j = 0; j < 8; j++)
{
*(buf++) = traw[j];
*(buf++) = traw[j];
}
}
return len;
}
int sfp_audio_init()
{
return sfp_audio_audio_init();
}