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sound, add FM filtering

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This commit is contained in:
kub 2022-04-14 17:32:40 +00:00
parent 21e0cd52e6
commit 68a950875c
13 changed files with 73 additions and 854 deletions
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40
pico/sound/resampler.c
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@ -154,7 +154,7 @@ resampler_t *resampler_new(unsigned taps, unsigned interpolation, unsigned decim
/* :cutoff is relative to the decimated frequency, but filtering is taking
* place at the interpolated frequency. It needs to be adapted if resampled
* rate is lower. Also needs more taps to keep the transistion band width */
* rate is lower. Also needs more taps to keep the transition band width */
if (decimation > interpolation) {
cutoff = cutoff * interpolation/decimation;
taps = taps * decimation/interpolation;
@ -164,7 +164,7 @@ resampler_t *resampler_new(unsigned taps, unsigned interpolation, unsigned decim
rs->decimation = decimation;
rs->taps = taps;
/* optimizers for resampler_update: */
rs->interp_inv = 0x100000000ULL / interpolation;
rs->interp_inv = (1ULL<<32) / interpolation;
rs->ratio_int = decimation / interpolation;
rs->filter = create_sinc(interpolation, taps, cutoff, beta);
@ -195,7 +195,7 @@ void resampler_update(resampler_t *rs, s32 *buffer, int length,
{
s16 *u;
s32 *p, *q = buffer;
int spf = (rs->stereo?2:1);
int spf = rs->stereo;
s32 inlen;
s32 l, r;
int n, i;
@ -206,29 +206,30 @@ void resampler_update(resampler_t *rs, s32 *buffer, int length,
* inlen = (length*decimation + interpolation-phase) / interpolation */
n = length*rs->decimation + rs->interpolation-rs->phase;
inlen = ((u64)n * rs->interp_inv) >> 32; /* input samples, n/interpolation */
if (inlen * rs->interpolation < n - rs->interpolation) inlen++; /* rounding */
if (n - inlen * rs->interpolation > rs->interpolation) inlen++; /* rounding */
/* reset buffer to start if the input doesn't fit into the buffer */
if (rs->buffer_idx + inlen+rs->taps >= rs->buffer_sz) {
memcpy(rs->buffer, rs->buffer + rs->buffer_idx*spf, rs->taps*spf*sizeof(*rs->buffer));
memcpy(rs->buffer, rs->buffer + (rs->buffer_idx<<spf), (rs->taps<<spf)*sizeof(*rs->buffer));
rs->buffer_idx = 0;
}
p = rs->buffer + rs->buffer_idx*spf;
p = rs->buffer + (rs->buffer_idx<<spf);
/* generate input samples */
if (inlen > 0)
get_samples(p + rs->taps*spf, inlen, rs->stereo);
get_samples(p + (rs->taps<<spf), inlen, rs->stereo);
if (rs->stereo) {
while (--length >= 0) {
/* compute filter output */
s32 *h = p;
u = rs->filter + (rs->phase * rs->taps);
for (i = 0, l = r = 0; i < rs->taps-1; i += 2)
{ n = *u++; l += n * p[2*i ]; r += n * p[2*i+1];
n = *u++; l += n * p[2*i+2]; r += n * p[2*i+3]; }
if (i < rs->taps)
{ n = *u++; l += n * p[2*i ]; r += n * p[2*i+1]; }
*q++ = l >> 16, *q++ = r >> 16;
for (i = rs->taps-1, l = r = 0; i > 0; i -= 2)
{ n = *u++; l += n * *h++; r += n * *h++;
n = *u++; l += n * *h++; r += n * *h++; }
if (i == 0)
{ n = *u++; l += n * *h++; r += n * *h++; }
*q++ = l >> 15, *q++ = r >> 15;
/* advance position to next sample */
rs->phase -= rs->decimation;
// if (rs->ratio_int) {
@ -241,13 +242,14 @@ void resampler_update(resampler_t *rs, s32 *buffer, int length,
} else {
while (--length >= 0) {
/* compute filter output */
s32 *h = p;
u = rs->filter + (rs->phase * rs->taps);
for (i = 0, l = r = 0; i < rs->taps-1; i += 2)
{ n = *u++; l += n * p[ i ];
n = *u++; l += n * p[ i+1]; }
if (i < rs->taps)
{ n = *u++; l += n * p[ i ]; }
*q++ = l >> 16;
for (i = rs->taps-1, l = r = 0; i > 0; i -= 2)
{ n = *u++; l += n * *h++;
n = *u++; l += n * *h++; }
if (i == 0)
{ n = *u++; l += n * *h++; }
*q++ = l >> 15;
/* advance position to next sample */
rs->phase -= rs->decimation;
// if (rs->ratio_int) {