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/*
* FILE: convert_linear.c
* PROGRAM: RAT
* AUTHOR: O.Hodson <O.Hodson@cs.ucl.ac.uk>
*
* Copyright (c) 1998-2001 University College London
* All rights reserved.
*/
#ifndef HIDE_SOURCE_STRINGS
static const char cvsid[] =
"$Id: convert_linear.c,v 1.12 2001/01/18 00:24:23 ucacoxh Exp $";
#endif /* HIDE_SOURCE_STRINGS */
#include "config_unix.h"
#include "config_win32.h"
#include "audio_types.h"
#include "converter_types.h"
#include "convert_linear.h"
#include "convert_util.h"
#include "util.h"
#include "memory.h"
#include "debug.h"
/* Linear Interpolation Conversion *******************************************/
struct s_filter_state;
typedef void (*inter_cf)(int offset, int channels, sample *src, int src_len, sample *dst, int dst_len, struct s_filter_state *s);
typedef struct s_filter_state {
int scale;
sample *tmp_buf;
int tmp_len;
sample *last;
int last_len;
inter_cf convert_f;
} filter_state_t;
typedef struct s_linear_state {
int steps;
filter_state_t fs[2];
} linear_state_t;
static void
linear_upsample(int offset, int channels,
sample *src, int src_len,
sample *dst, int dst_len,
filter_state_t *l)
{
register int r, loop;
register short *sp, *dp;
short *last = l->last + offset;
sp = src + offset;
dp = dst + offset;
loop = min(src_len/channels, dst_len/(channels*l->scale));
/* On some platforms divisions by powers of 2 is way quicker */
/* than other divisions. To improve interpolation perf. */
/* approximate fractions which are not powers of two */
/* i.e. 1 / 3 -> 5 / 16 */
/* 1 / 5 -> 6 / 32 */
/* 1 / 6 -> 11 / 64 */
switch (l->scale) {
case 6:
while(loop--) {
register int il, ic;
il = *last; ic = *sp;
r = 55 * il + 11 * ic; r /= 64; *dp = (sample)r; dp += channels;
r = 44 * il + 22 * ic; r /= 64; *dp = (sample)r; dp += channels;
r = 33 * il + 33 * ic; r /= 64; *dp = (sample)r; dp += channels;
r = 22 * il + 44 * ic; r /= 64; *dp = (sample)r; dp += channels;
r = 11 * il + 55 * ic; r /= 64; *dp = (sample)r; dp += channels;
r = 66 * ic; r /= 64; *dp = (sample)r; dp += channels;
last = sp;
sp += channels;
}
break;
case 5:
while(loop--) {
register int il, ic;
il = *last; ic = *sp;
r = 24 * il + 6 * ic; r /= 32; *dp = (sample)r; dp += channels;
r = 18 * il + 12 * ic; r /= 32; *dp = (sample)r; dp += channels;
r = 12 * il + 18 * ic; r /= 32; *dp = (sample)r; dp += channels;
r = 6 * il + 24 * ic; r /= 32; *dp = (sample)r; dp += channels;
r = 30 * ic; r /= 32; *dp = (sample)r; dp += channels;
last = sp;
sp += channels;
}
break;
case 4:
while(loop--) {
register int il, ic;
il = *last; ic = *sp;
r = 3 * il + 1 * ic; r /= 4; *dp = (sample)r; dp += channels;
r = 2 * il + 2 * ic; r /= 4; *dp = (sample)r; dp += channels;
r = 1 * il + 3 * ic; r /= 4; *dp = (sample)r; dp += channels;
*dp = (sample)ic; dp += channels;
last = sp;
sp += channels;
}
break;
case 3:
while(loop--) {
register int il, ic;
il = *last; ic = *sp;
r = 10 * il + 5 * ic; r /= 16; *dp = (sample)r; dp += channels;
r = 5 * il + 10 * ic; r /= 16; *dp = (sample)r; dp += channels;
r = 15 * ic; r /= 16; *dp = (sample)r; dp += channels;
last = sp;
sp += channels;
}
break;
case 2:
while(loop--) {
register int il, ic;
il = *last; ic = *sp;
r = il + ic; r /= 2; *dp = (sample)r; dp += channels;
*dp = (sample)ic; dp += channels;
last = sp;
sp += channels;
}
break;
default:
assert(0); /* Should never get here */
}
l->last[offset] = src[src_len - channels + offset];
}
static void
linear_downsample(int offset, int channels,
sample *src, int src_len,
sample *dst, int dst_len,
filter_state_t *l)
{
register int loop, r, c, lim;
register short *sp, *dp;
loop = min(src_len / (channels * l->scale), dst_len / channels);
sp = src + offset;
dp = dst + offset;
lim = l->scale - 1;
while(loop--) {
r = (int)*sp; sp+=channels;
c = lim;
while(c--) {
r += (int)*sp; sp+=channels;
}
r /= l->scale;
*dp = (short)r;
dp+= channels;
}
}
static void
linear_init_state(filter_state_t *l, const converter_fmt_t *cfmt)
{
if (cfmt->src_freq > cfmt->dst_freq) {
l->scale = cfmt->src_freq / cfmt->dst_freq;
l->last_len = 0;
l->convert_f = linear_downsample;
} else if (cfmt->src_freq < cfmt->dst_freq) {
l->scale = cfmt->dst_freq / cfmt->src_freq;
l->last_len = cfmt->src_channels;
l->convert_f = linear_upsample;
l->last = (sample*)xmalloc(sizeof(sample) * l->last_len);
memset(l->last, 0, sizeof(sample) * cfmt->src_channels);
}
}
int
linear_create (const converter_fmt_t *cfmt, u_char **state, uint32_t *state_len)
{
converter_fmt_t sfmt, ufmt;
linear_state_t *l;
int g;
if (((cfmt->src_freq % 8000) == 0 && (cfmt->dst_freq % 8000)) ||
((cfmt->src_freq % 11025) == 0 && (cfmt->dst_freq % 11025))) {
/* 11025 - 8000 not supported */
return FALSE;
}
xmemchk();
g = gcd(cfmt->src_freq, cfmt->dst_freq);
l = (linear_state_t*)xmalloc(sizeof(linear_state_t));
memset(l, 0 , sizeof(linear_state_t));
l->steps = conversion_steps(cfmt->src_freq, cfmt->dst_freq);
xmemchk();
sfmt = *cfmt;
if (sfmt.src_channels != sfmt.dst_channels) {
/* In addition to rate conversion, this requires
* channel number conversion, this can happen
* either side of rate conversion - see sinc_convert() */
if (sfmt.src_channels == 2 && sfmt.dst_channels == 1) {
sfmt.src_channels = 1; /* Stereo->Mono, R1->R2 */
}
}
switch(l->steps) {
case 1:
linear_init_state(l->fs, &sfmt);
break;
case 2:
ufmt = sfmt;
ufmt.dst_freq = g;
linear_init_state(l->fs, &ufmt);
ufmt = sfmt;
ufmt.src_freq = g;
linear_init_state(l->fs + 1, &ufmt);
break;
}
xmemchk();
*state = (u_char*)l;
*state_len = sizeof(linear_state_t);
return TRUE;
}
void
linear_convert (const converter_fmt_t *cfmt,
u_char *state,
sample* src_buf, int src_len,
sample *dst_buf, int dst_len)
{
linear_state_t *l;
int channels, i;
channels = cfmt->src_channels;
l = (linear_state_t*)state;
if (cfmt->src_channels == 2 && cfmt->dst_channels == 1) {
/* stereo->mono then sample rate change */
if (l->steps) {
/* inplace conversion needed */
converter_change_channels(src_buf, src_len, 2, src_buf, src_len / 2, 1);
src_len /= 2;
} else {
/* this is only conversion */
converter_change_channels(src_buf, src_len, 2, dst_buf, dst_len, 1);
return;
}
channels = 1;
} else if (cfmt->src_channels == 1 && cfmt->dst_channels == 2) {
dst_len /= 2;
}
switch(l->steps) {
case 1:
assert(l->fs->convert_f);
for(i = 0; i < channels; i++) {
l->fs->convert_f(i, channels, src_buf, src_len, dst_buf, dst_len, l->fs);
}
break;
case 2:
/* first step is always downsampling for moment */
if (l->fs->tmp_buf == NULL) {
l->fs->tmp_len = src_len / l->fs->scale;
l->fs->tmp_buf = (sample*)xmalloc(sizeof(sample) * l->fs->tmp_len);
}
assert(l->fs[0].convert_f);
assert(l->fs[1].convert_f);
for(i = 0; i < channels; i++)
l->fs[0].convert_f(i, channels, src_buf, src_len, l->fs->tmp_buf, l->fs->tmp_len, l->fs);
for(i = 0; i < channels; i++)
l->fs[1].convert_f(i, channels, l->fs->tmp_buf, l->fs->tmp_len, dst_buf, dst_len, l->fs + 1);
break;
}
if (cfmt->src_channels == 1 && cfmt->dst_channels == 2) {
/* sample rate change before mono-> stereo */
if (l->steps) {
/* in place needed */
converter_change_channels(dst_buf, dst_len, 1, dst_buf, dst_len * 2, 2);
} else {
/* this is our only conversion here */
converter_change_channels(src_buf, src_len, 1, dst_buf, dst_len * 2, 2);
}
}
}
void
linear_destroy (u_char **state, uint32_t *state_len)
{
linear_state_t *l = (linear_state_t*)*state;
int i;
assert(*state_len == sizeof(linear_state_t));
for(i = 0; i < l->steps; i++) {
if (l->fs[i].last) xfree(l->fs[i].last);
if (l->fs[i].tmp_buf) xfree(l->fs[i].tmp_buf);
}
xfree(l);
*state = NULL;
*state_len = 0;
}
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