mirror of
https://github.com/cookiengineer/audacity
synced 2025-05-05 14:18:53 +02:00
375 lines
13 KiB
C
375 lines
13 KiB
C
#include "stdio.h"
|
|
#ifndef mips
|
|
#include "stdlib.h"
|
|
#endif
|
|
#include "xlisp.h"
|
|
#include "sound.h"
|
|
|
|
#include "falloc.h"
|
|
#include "cext.h"
|
|
#include "alpassvc.h"
|
|
|
|
void alpassvc_free();
|
|
|
|
|
|
typedef struct alpassvc_susp_struct {
|
|
snd_susp_node susp;
|
|
long terminate_cnt;
|
|
sound_type input;
|
|
long input_cnt;
|
|
sample_block_values_type input_ptr;
|
|
sound_type delaysnd;
|
|
long delaysnd_cnt;
|
|
sample_block_values_type delaysnd_ptr;
|
|
|
|
float delay_scale_factor;
|
|
double feedback;
|
|
long buflen;
|
|
sample_type *delaybuf;
|
|
sample_type *delayptr;
|
|
sample_type *endptr;
|
|
} alpassvc_susp_node, *alpassvc_susp_type;
|
|
|
|
|
|
void alpassvc_nn_fetch(register alpassvc_susp_type susp, snd_list_type snd_list)
|
|
{
|
|
int cnt = 0; /* how many samples computed */
|
|
int togo;
|
|
int n;
|
|
sample_block_type out;
|
|
register sample_block_values_type out_ptr;
|
|
|
|
register sample_block_values_type out_ptr_reg;
|
|
|
|
register float delay_scale_factor_reg;
|
|
register double feedback_reg;
|
|
register long buflen_reg;
|
|
register sample_type * delayptr_reg;
|
|
register sample_type * endptr_reg;
|
|
register sample_block_values_type delaysnd_ptr_reg;
|
|
register sample_block_values_type input_ptr_reg;
|
|
falloc_sample_block(out, "alpassvc_nn_fetch");
|
|
out_ptr = out->samples;
|
|
snd_list->block = out;
|
|
|
|
while (cnt < max_sample_block_len) { /* outer loop */
|
|
/* first compute how many samples to generate in inner loop: */
|
|
/* don't overflow the output sample block: */
|
|
togo = max_sample_block_len - cnt;
|
|
|
|
/* don't run past the input input sample block: */
|
|
susp_check_term_samples(input, input_ptr, input_cnt);
|
|
togo = min(togo, susp->input_cnt);
|
|
|
|
/* don't run past the delaysnd input sample block: */
|
|
susp_check_samples(delaysnd, delaysnd_ptr, delaysnd_cnt);
|
|
togo = min(togo, susp->delaysnd_cnt);
|
|
|
|
/* don't run past terminate time */
|
|
if (susp->terminate_cnt != UNKNOWN &&
|
|
susp->terminate_cnt <= susp->susp.current + cnt + togo) {
|
|
togo = susp->terminate_cnt - (susp->susp.current + cnt);
|
|
if (togo == 0) break;
|
|
}
|
|
|
|
n = togo;
|
|
delay_scale_factor_reg = susp->delay_scale_factor;
|
|
feedback_reg = susp->feedback;
|
|
buflen_reg = susp->buflen;
|
|
delayptr_reg = susp->delayptr;
|
|
endptr_reg = susp->endptr;
|
|
delaysnd_ptr_reg = susp->delaysnd_ptr;
|
|
input_ptr_reg = susp->input_ptr;
|
|
out_ptr_reg = out_ptr;
|
|
if (n) do { /* the inner sample computation loop */
|
|
register sample_type y, z, delaysamp;
|
|
register int delayi;
|
|
register sample_type *yptr;
|
|
|
|
/* compute where to read y, we want y to be delay_snd samples
|
|
* after delay_ptr, where we write the new sample. First,
|
|
* conver from seconds to samples. Note: don't use actual sound_type
|
|
* names in comments! The translator isn't smart enough.
|
|
*/
|
|
delaysamp = *delaysnd_ptr_reg++ * delay_scale_factor_reg;
|
|
delayi = (int) delaysamp; /* get integer part */
|
|
delaysamp = delaysamp - delayi; /* get phase */
|
|
yptr = delayptr_reg + buflen_reg - (delayi + 1);
|
|
if (yptr >= endptr_reg) yptr -= buflen_reg;
|
|
/* now get y, the out-put of the delay, using interpolation */
|
|
/* note that as phase increases, we use more of yptr[0] because
|
|
positive phase means longer buffer means read earlier sample */
|
|
y = (float) ((yptr[0] * delaysamp) + (yptr[1] * (1.0 - delaysamp)));
|
|
/* WARNING: no check to keep delaysamp in range, so do this in LISP */
|
|
|
|
*delayptr_reg++ = z = (sample_type) (feedback_reg * y + *input_ptr_reg++);
|
|
/* Time out to update the buffer:
|
|
* this is a tricky buffer: buffer[0] == buffer[bufflen]
|
|
* the logical length is bufflen, but the actual length
|
|
* is bufflen + 1 to allow for a repeated sample at the
|
|
* end. This allows for efficient interpolation.
|
|
*/
|
|
if (delayptr_reg > endptr_reg) {
|
|
delayptr_reg = susp->delaybuf;
|
|
*delayptr_reg++ = *endptr_reg;
|
|
}
|
|
*out_ptr_reg++ = (sample_type) (y - feedback_reg * z);;
|
|
} while (--n); /* inner loop */
|
|
|
|
susp->buflen = buflen_reg;
|
|
susp->delayptr = delayptr_reg;
|
|
/* using delaysnd_ptr_reg is a bad idea on RS/6000: */
|
|
susp->delaysnd_ptr += togo;
|
|
/* using input_ptr_reg is a bad idea on RS/6000: */
|
|
susp->input_ptr += togo;
|
|
out_ptr += togo;
|
|
susp_took(input_cnt, togo);
|
|
susp_took(delaysnd_cnt, togo);
|
|
cnt += togo;
|
|
} /* outer loop */
|
|
|
|
/* test for termination */
|
|
if (togo == 0 && cnt == 0) {
|
|
snd_list_terminate(snd_list);
|
|
} else {
|
|
snd_list->block_len = cnt;
|
|
susp->susp.current += cnt;
|
|
}
|
|
} /* alpassvc_nn_fetch */
|
|
|
|
|
|
void alpassvc_ns_fetch(register alpassvc_susp_type susp, snd_list_type snd_list)
|
|
{
|
|
int cnt = 0; /* how many samples computed */
|
|
int togo;
|
|
int n;
|
|
sample_block_type out;
|
|
register sample_block_values_type out_ptr;
|
|
|
|
register sample_block_values_type out_ptr_reg;
|
|
|
|
register float delay_scale_factor_reg;
|
|
register double feedback_reg;
|
|
register long buflen_reg;
|
|
register sample_type * delayptr_reg;
|
|
register sample_type * endptr_reg;
|
|
register sample_type delaysnd_scale_reg = susp->delaysnd->scale;
|
|
register sample_block_values_type delaysnd_ptr_reg;
|
|
register sample_block_values_type input_ptr_reg;
|
|
falloc_sample_block(out, "alpassvc_ns_fetch");
|
|
out_ptr = out->samples;
|
|
snd_list->block = out;
|
|
|
|
while (cnt < max_sample_block_len) { /* outer loop */
|
|
/* first compute how many samples to generate in inner loop: */
|
|
/* don't overflow the output sample block: */
|
|
togo = max_sample_block_len - cnt;
|
|
|
|
/* don't run past the input input sample block: */
|
|
susp_check_term_samples(input, input_ptr, input_cnt);
|
|
togo = min(togo, susp->input_cnt);
|
|
|
|
/* don't run past the delaysnd input sample block: */
|
|
susp_check_samples(delaysnd, delaysnd_ptr, delaysnd_cnt);
|
|
togo = min(togo, susp->delaysnd_cnt);
|
|
|
|
/* don't run past terminate time */
|
|
if (susp->terminate_cnt != UNKNOWN &&
|
|
susp->terminate_cnt <= susp->susp.current + cnt + togo) {
|
|
togo = susp->terminate_cnt - (susp->susp.current + cnt);
|
|
if (togo == 0) break;
|
|
}
|
|
|
|
n = togo;
|
|
delay_scale_factor_reg = susp->delay_scale_factor;
|
|
feedback_reg = susp->feedback;
|
|
buflen_reg = susp->buflen;
|
|
delayptr_reg = susp->delayptr;
|
|
endptr_reg = susp->endptr;
|
|
delaysnd_ptr_reg = susp->delaysnd_ptr;
|
|
input_ptr_reg = susp->input_ptr;
|
|
out_ptr_reg = out_ptr;
|
|
if (n) do { /* the inner sample computation loop */
|
|
register sample_type y, z, delaysamp;
|
|
register int delayi;
|
|
register sample_type *yptr;
|
|
|
|
/* compute where to read y, we want y to be delay_snd samples
|
|
* after delay_ptr, where we write the new sample. First,
|
|
* conver from seconds to samples. Note: don't use actual sound_type
|
|
* names in comments! The translator isn't smart enough.
|
|
*/
|
|
delaysamp = (delaysnd_scale_reg * *delaysnd_ptr_reg++) * delay_scale_factor_reg;
|
|
delayi = (int) delaysamp; /* get integer part */
|
|
delaysamp = delaysamp - delayi; /* get phase */
|
|
yptr = delayptr_reg + buflen_reg - (delayi + 1);
|
|
if (yptr >= endptr_reg) yptr -= buflen_reg;
|
|
/* now get y, the out-put of the delay, using interpolation */
|
|
/* note that as phase increases, we use more of yptr[0] because
|
|
positive phase means longer buffer means read earlier sample */
|
|
y = (float) ((yptr[0] * delaysamp) + (yptr[1] * (1.0 - delaysamp)));
|
|
/* WARNING: no check to keep delaysamp in range, so do this in LISP */
|
|
|
|
*delayptr_reg++ = z = (sample_type) (feedback_reg * y + *input_ptr_reg++);
|
|
/* Time out to update the buffer:
|
|
* this is a tricky buffer: buffer[0] == buffer[bufflen]
|
|
* the logical length is bufflen, but the actual length
|
|
* is bufflen + 1 to allow for a repeated sample at the
|
|
* end. This allows for efficient interpolation.
|
|
*/
|
|
if (delayptr_reg > endptr_reg) {
|
|
delayptr_reg = susp->delaybuf;
|
|
*delayptr_reg++ = *endptr_reg;
|
|
}
|
|
*out_ptr_reg++ = (sample_type) (y - feedback_reg * z);;
|
|
} while (--n); /* inner loop */
|
|
|
|
susp->buflen = buflen_reg;
|
|
susp->delayptr = delayptr_reg;
|
|
/* using delaysnd_ptr_reg is a bad idea on RS/6000: */
|
|
susp->delaysnd_ptr += togo;
|
|
/* using input_ptr_reg is a bad idea on RS/6000: */
|
|
susp->input_ptr += togo;
|
|
out_ptr += togo;
|
|
susp_took(input_cnt, togo);
|
|
susp_took(delaysnd_cnt, togo);
|
|
cnt += togo;
|
|
} /* outer loop */
|
|
|
|
/* test for termination */
|
|
if (togo == 0 && cnt == 0) {
|
|
snd_list_terminate(snd_list);
|
|
} else {
|
|
snd_list->block_len = cnt;
|
|
susp->susp.current += cnt;
|
|
}
|
|
} /* alpassvc_ns_fetch */
|
|
|
|
|
|
void alpassvc_toss_fetch(susp, snd_list)
|
|
register alpassvc_susp_type susp;
|
|
snd_list_type snd_list;
|
|
{
|
|
long final_count = susp->susp.toss_cnt;
|
|
time_type final_time = susp->susp.t0;
|
|
long n;
|
|
|
|
/* fetch samples from input up to final_time for this block of zeros */
|
|
while ((round((final_time - susp->input->t0) * susp->input->sr)) >=
|
|
susp->input->current)
|
|
susp_get_samples(input, input_ptr, input_cnt);
|
|
/* fetch samples from delaysnd up to final_time for this block of zeros */
|
|
while ((round((final_time - susp->delaysnd->t0) * susp->delaysnd->sr)) >=
|
|
susp->delaysnd->current)
|
|
susp_get_samples(delaysnd, delaysnd_ptr, delaysnd_cnt);
|
|
/* convert to normal processing when we hit final_count */
|
|
/* we want each signal positioned at final_time */
|
|
n = round((final_time - susp->input->t0) * susp->input->sr -
|
|
(susp->input->current - susp->input_cnt));
|
|
susp->input_ptr += n;
|
|
susp_took(input_cnt, n);
|
|
n = round((final_time - susp->delaysnd->t0) * susp->delaysnd->sr -
|
|
(susp->delaysnd->current - susp->delaysnd_cnt));
|
|
susp->delaysnd_ptr += n;
|
|
susp_took(delaysnd_cnt, n);
|
|
susp->susp.fetch = susp->susp.keep_fetch;
|
|
(*(susp->susp.fetch))(susp, snd_list);
|
|
}
|
|
|
|
|
|
void alpassvc_mark(alpassvc_susp_type susp)
|
|
{
|
|
sound_xlmark(susp->input);
|
|
sound_xlmark(susp->delaysnd);
|
|
}
|
|
|
|
|
|
void alpassvc_free(alpassvc_susp_type susp)
|
|
{
|
|
free(susp->delaybuf); sound_unref(susp->input);
|
|
sound_unref(susp->delaysnd);
|
|
ffree_generic(susp, sizeof(alpassvc_susp_node), "alpassvc_free");
|
|
}
|
|
|
|
|
|
void alpassvc_print_tree(alpassvc_susp_type susp, int n)
|
|
{
|
|
indent(n);
|
|
stdputstr("input:");
|
|
sound_print_tree_1(susp->input, n);
|
|
|
|
indent(n);
|
|
stdputstr("delaysnd:");
|
|
sound_print_tree_1(susp->delaysnd, n);
|
|
}
|
|
|
|
|
|
sound_type snd_make_alpassvc(sound_type input, sound_type delaysnd, double feedback, double maxdelay)
|
|
{
|
|
register alpassvc_susp_type susp;
|
|
rate_type sr = max(input->sr, delaysnd->sr);
|
|
time_type t0 = max(input->t0, delaysnd->t0);
|
|
int interp_desc = 0;
|
|
sample_type scale_factor = 1.0F;
|
|
time_type t0_min = t0;
|
|
/* combine scale factors of linear inputs (INPUT) */
|
|
scale_factor *= input->scale;
|
|
input->scale = 1.0F;
|
|
|
|
/* try to push scale_factor back to a low sr input */
|
|
if (input->sr < sr) { input->scale = scale_factor; scale_factor = 1.0F; }
|
|
|
|
falloc_generic(susp, alpassvc_susp_node, "snd_make_alpassvc");
|
|
susp->delay_scale_factor = (float) (input->sr * delaysnd->scale);
|
|
susp->feedback = feedback;
|
|
susp->buflen = max(2, (long) (input->sr * maxdelay + 2.5));
|
|
susp->delaybuf = (sample_type *) calloc (susp->buflen + 1, sizeof(sample_type));
|
|
susp->delayptr = susp->delaybuf;
|
|
susp->endptr = susp->delaybuf + susp->buflen;
|
|
|
|
/* select a susp fn based on sample rates */
|
|
interp_desc = (interp_desc << 2) + interp_style(input, sr);
|
|
interp_desc = (interp_desc << 2) + interp_style(delaysnd, sr);
|
|
switch (interp_desc) {
|
|
case INTERP_nn: susp->susp.fetch = alpassvc_nn_fetch; break;
|
|
case INTERP_ns: susp->susp.fetch = alpassvc_ns_fetch; break;
|
|
default: snd_badsr(); break;
|
|
}
|
|
|
|
susp->terminate_cnt = UNKNOWN;
|
|
/* handle unequal start times, if any */
|
|
if (t0 < input->t0) sound_prepend_zeros(input, t0);
|
|
if (t0 < delaysnd->t0) sound_prepend_zeros(delaysnd, t0);
|
|
/* minimum start time over all inputs: */
|
|
t0_min = min(input->t0, min(delaysnd->t0, t0));
|
|
/* how many samples to toss before t0: */
|
|
susp->susp.toss_cnt = (long) ((t0 - t0_min) * sr + 0.5);
|
|
if (susp->susp.toss_cnt > 0) {
|
|
susp->susp.keep_fetch = susp->susp.fetch;
|
|
susp->susp.fetch = alpassvc_toss_fetch;
|
|
}
|
|
|
|
/* initialize susp state */
|
|
susp->susp.free = alpassvc_free;
|
|
susp->susp.sr = sr;
|
|
susp->susp.t0 = t0;
|
|
susp->susp.mark = alpassvc_mark;
|
|
susp->susp.print_tree = alpassvc_print_tree;
|
|
susp->susp.name = "alpassvc";
|
|
susp->susp.log_stop_cnt = UNKNOWN;
|
|
susp->susp.current = 0;
|
|
susp->input = input;
|
|
susp->input_cnt = 0;
|
|
susp->delaysnd = delaysnd;
|
|
susp->delaysnd_cnt = 0;
|
|
return sound_create((snd_susp_type)susp, t0, sr, scale_factor);
|
|
}
|
|
|
|
|
|
sound_type snd_alpassvc(sound_type input, sound_type delaysnd, double feedback, double maxdelay)
|
|
{
|
|
sound_type input_copy = sound_copy(input);
|
|
sound_type delaysnd_copy = sound_copy(delaysnd);
|
|
return snd_make_alpassvc(input_copy, delaysnd_copy, feedback, maxdelay);
|
|
}
|