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mirror of https://github.com/cookiengineer/audacity synced 2025-05-03 17:19:43 +02:00
2015-04-07 22:10:17 -05:00

654 lines
20 KiB
C

#include "stdio.h"
#ifndef mips
#include "stdlib.h"
#endif
#include "xlisp.h"
#include "sound.h"
#include "falloc.h"
#include "cext.h"
#include "siosc.h"
void siosc_free(snd_susp_type a_susp);
typedef struct siosc_susp_struct {
snd_susp_node susp;
boolean started;
long terminate_cnt;
boolean logically_stopped;
sound_type s_fm;
long s_fm_cnt;
sample_block_values_type s_fm_ptr;
/* support for interpolation of s_fm */
sample_type s_fm_x1_sample;
double s_fm_pHaSe;
double s_fm_pHaSe_iNcR;
/* support for ramp between samples of s_fm */
double output_per_s_fm;
long s_fm_n;
double table_len;
double ph_incr;
table_type table_a_ptr;
table_type table_b_ptr_ptr;
sample_type *table_a_samps;
sample_type *table_b_samps;
double table_sr;
double phase;
LVAL lis;
long next_breakpoint;
double ampramp_a;
double ampramp_b;
double ampslope;
} siosc_susp_node, *siosc_susp_type;
/* sisosc_table_init -- set up first two tables for interpolation */
/**/
void siosc_table_init(siosc_susp_type susp)
{
sound_type snd;
if (!susp->lis) xlfail("bad table list in SIOSC");
snd = getsound(car(susp->lis));
susp->table_b_ptr_ptr = sound_to_table(snd);
susp->table_b_samps = susp->table_b_ptr_ptr->samples;
susp->lis = cdr(susp->lis);
susp->table_sr = snd->sr;
susp->table_len = susp->table_b_ptr_ptr->length;
}
/* siosc_table_update -- outer loop processing, get next table */
/**/
long siosc_table_update(siosc_susp_type susp, long cur)
{
long n;
/* swap ampramps: */
susp->ampramp_a = 1.0;
susp->ampramp_b = 0.0;
/* swap tables: */
table_unref(susp->table_a_ptr);
susp->table_a_ptr = susp->table_b_ptr_ptr;
susp->table_a_samps = susp->table_b_samps;
susp->table_b_ptr_ptr = NULL; /* so we do not try to unref it */
if (susp->lis) {
sound_type snd;
/* compute slope */
susp->next_breakpoint = getfixnum(car(susp->lis));
susp->lis = cdr(susp->lis);
n = susp->next_breakpoint - cur;
susp->ampslope = 1.0 / n;
/* build new table: */
if (!susp->lis) xlfail("bad table list in SIOSC");
snd = getsound(car(susp->lis));
susp->table_b_ptr_ptr = sound_to_table(snd);
susp->table_b_samps = susp->table_b_ptr_ptr->samples;
if (susp->table_b_ptr_ptr->length != susp->table_len || susp->table_sr != snd->sr)
xlfail("mismatched tables passed to SIOSC") ;
susp->lis = cdr(susp->lis);
} else { /* use only table a */
susp->ampslope = 0.0;
susp->next_breakpoint = 0x7FFFFFFF;
n = 0x7FFFFFFF;
}
return n;
}
void siosc_s_fetch(snd_susp_type a_susp, snd_list_type snd_list)
{
siosc_susp_type susp = (siosc_susp_type) a_susp;
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 double table_len_reg;
register double ph_incr_reg;
register sample_type * table_a_samps_reg;
register sample_type * table_b_samps_reg;
register double phase_reg;
register double ampramp_a_reg;
register double ampramp_b_reg;
register double ampslope_reg;
register sample_type s_fm_scale_reg = susp->s_fm->scale;
register sample_block_values_type s_fm_ptr_reg;
falloc_sample_block(out, "siosc_s_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 s_fm input sample block: */
susp_check_term_log_samples(s_fm, s_fm_ptr, s_fm_cnt);
togo = min(togo, susp->s_fm_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) togo = 0; /* avoids rounding errros */
if (togo == 0) break;
}
/* don't run past logical stop time */
if (!susp->logically_stopped && susp->susp.log_stop_cnt != UNKNOWN) {
int to_stop = susp->susp.log_stop_cnt - (susp->susp.current + cnt);
/* break if to_stop == 0 (we're at the logical stop)
* AND cnt > 0 (we're not at the beginning of the
* output block).
*/
if (to_stop < 0) to_stop = 0; /* avoids rounding errors */
if (to_stop < togo) {
if (to_stop == 0) {
if (cnt) {
togo = 0;
break;
} else /* keep togo as is: since cnt == 0, we
* can set the logical stop flag on this
* output block
*/
susp->logically_stopped = true;
} else /* limit togo so we can start a new
* block at the LST
*/
togo = to_stop;
}
}
{ long cur = susp->susp.current + cnt;
n = susp->next_breakpoint - cur;
if (n == 0) n = siosc_table_update(susp, cur);
}
togo = min(n, togo);
n = togo;
table_len_reg = susp->table_len;
ph_incr_reg = susp->ph_incr;
table_a_samps_reg = susp->table_a_samps;
table_b_samps_reg = susp->table_b_samps;
phase_reg = susp->phase;
ampramp_a_reg = susp->ampramp_a;
ampramp_b_reg = susp->ampramp_b;
ampslope_reg = susp->ampslope;
s_fm_ptr_reg = susp->s_fm_ptr;
out_ptr_reg = out_ptr;
if (n) do { /* the inner sample computation loop */
long table_index;
double xa, xb;
table_index = (long) phase_reg;
xa = table_a_samps_reg[table_index];
xb = table_b_samps_reg[table_index];
*out_ptr_reg++ = (sample_type)
(ampramp_a_reg * (xa + (phase_reg - table_index) *
(table_a_samps_reg[table_index + 1] - xa)) +
ampramp_b_reg * (xb + (phase_reg - table_index) *
(table_b_samps_reg[table_index + 1] - xb)));
ampramp_a_reg -= ampslope_reg;
ampramp_b_reg += ampslope_reg;
phase_reg += ph_incr_reg + (s_fm_scale_reg * *s_fm_ptr_reg++);
while (phase_reg > table_len_reg) phase_reg -= table_len_reg;
/* watch out for negative frequencies! */
while (phase_reg < 0) phase_reg += table_len_reg;
} while (--n); /* inner loop */
susp->phase = phase_reg;
susp->ampramp_a = ampramp_a_reg;
susp->ampramp_b = ampramp_b_reg;
/* using s_fm_ptr_reg is a bad idea on RS/6000: */
susp->s_fm_ptr += togo;
out_ptr += togo;
susp_took(s_fm_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;
}
/* test for logical stop */
if (susp->logically_stopped) {
snd_list->logically_stopped = true;
} else if (susp->susp.log_stop_cnt == susp->susp.current) {
susp->logically_stopped = true;
}
} /* siosc_s_fetch */
void siosc_i_fetch(snd_susp_type a_susp, snd_list_type snd_list)
{
siosc_susp_type susp = (siosc_susp_type) a_susp;
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 double table_len_reg;
register double ph_incr_reg;
register sample_type * table_a_samps_reg;
register sample_type * table_b_samps_reg;
register double phase_reg;
register double ampramp_a_reg;
register double ampramp_b_reg;
register double ampslope_reg;
register double s_fm_pHaSe_iNcR_rEg = susp->s_fm_pHaSe_iNcR;
register double s_fm_pHaSe_ReG;
register sample_type s_fm_x1_sample_reg;
falloc_sample_block(out, "siosc_i_fetch");
out_ptr = out->samples;
snd_list->block = out;
/* make sure sounds are primed with first values */
if (!susp->started) {
susp->started = true;
susp_check_term_log_samples(s_fm, s_fm_ptr, s_fm_cnt);
susp->s_fm_x1_sample = susp_fetch_sample(s_fm, s_fm_ptr, s_fm_cnt);
}
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 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) togo = 0; /* avoids rounding errros */
if (togo == 0) break;
}
/* don't run past logical stop time */
if (!susp->logically_stopped && susp->susp.log_stop_cnt != UNKNOWN) {
int to_stop = susp->susp.log_stop_cnt - (susp->susp.current + cnt);
/* break if to_stop == 0 (we're at the logical stop)
* AND cnt > 0 (we're not at the beginning of the
* output block).
*/
if (to_stop < 0) to_stop = 0; /* avoids rounding errors */
if (to_stop < togo) {
if (to_stop == 0) {
if (cnt) {
togo = 0;
break;
} else /* keep togo as is: since cnt == 0, we
* can set the logical stop flag on this
* output block
*/
susp->logically_stopped = true;
} else /* limit togo so we can start a new
* block at the LST
*/
togo = to_stop;
}
}
{ long cur = susp->susp.current + cnt;
n = susp->next_breakpoint - cur;
if (n == 0) n = siosc_table_update(susp, cur);
}
togo = min(n, togo);
n = togo;
table_len_reg = susp->table_len;
ph_incr_reg = susp->ph_incr;
table_a_samps_reg = susp->table_a_samps;
table_b_samps_reg = susp->table_b_samps;
phase_reg = susp->phase;
ampramp_a_reg = susp->ampramp_a;
ampramp_b_reg = susp->ampramp_b;
ampslope_reg = susp->ampslope;
s_fm_pHaSe_ReG = susp->s_fm_pHaSe;
s_fm_x1_sample_reg = susp->s_fm_x1_sample;
out_ptr_reg = out_ptr;
if (n) do { /* the inner sample computation loop */
long table_index;
double xa, xb;
if (s_fm_pHaSe_ReG >= 1.0) {
/* fixup-depends s_fm */
/* pick up next sample as s_fm_x1_sample: */
susp->s_fm_ptr++;
susp_took(s_fm_cnt, 1);
s_fm_pHaSe_ReG -= 1.0;
susp_check_term_log_samples_break(s_fm, s_fm_ptr, s_fm_cnt, s_fm_x1_sample_reg);
s_fm_x1_sample_reg = susp_current_sample(s_fm, s_fm_ptr);
}
table_index = (long) phase_reg;
xa = table_a_samps_reg[table_index];
xb = table_b_samps_reg[table_index];
*out_ptr_reg++ = (sample_type)
(ampramp_a_reg * (xa + (phase_reg - table_index) *
(table_a_samps_reg[table_index + 1] - xa)) +
ampramp_b_reg * (xb + (phase_reg - table_index) *
(table_b_samps_reg[table_index + 1] - xb)));
ampramp_a_reg -= ampslope_reg;
ampramp_b_reg += ampslope_reg;
phase_reg += ph_incr_reg + s_fm_x1_sample_reg;
while (phase_reg > table_len_reg) phase_reg -= table_len_reg;
/* watch out for negative frequencies! */
while (phase_reg < 0) phase_reg += table_len_reg;
s_fm_pHaSe_ReG += s_fm_pHaSe_iNcR_rEg;
} while (--n); /* inner loop */
togo -= n;
susp->phase = phase_reg;
susp->ampramp_a = ampramp_a_reg;
susp->ampramp_b = ampramp_b_reg;
susp->s_fm_pHaSe = s_fm_pHaSe_ReG;
susp->s_fm_x1_sample = s_fm_x1_sample_reg;
out_ptr += 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;
}
/* test for logical stop */
if (susp->logically_stopped) {
snd_list->logically_stopped = true;
} else if (susp->susp.log_stop_cnt == susp->susp.current) {
susp->logically_stopped = true;
}
} /* siosc_i_fetch */
void siosc_r_fetch(snd_susp_type a_susp, snd_list_type snd_list)
{
siosc_susp_type susp = (siosc_susp_type) a_susp;
int cnt = 0; /* how many samples computed */
sample_type s_fm_val;
int togo;
int n;
sample_block_type out;
register sample_block_values_type out_ptr;
register sample_block_values_type out_ptr_reg;
register double table_len_reg;
register double ph_incr_reg;
register sample_type * table_a_samps_reg;
register sample_type * table_b_samps_reg;
register double phase_reg;
register double ampramp_a_reg;
register double ampramp_b_reg;
register double ampslope_reg;
falloc_sample_block(out, "siosc_r_fetch");
out_ptr = out->samples;
snd_list->block = out;
/* make sure sounds are primed with first values */
if (!susp->started) {
susp->started = true;
susp->s_fm_pHaSe = 1.0;
}
susp_check_term_log_samples(s_fm, s_fm_ptr, s_fm_cnt);
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;
/* grab next s_fm_x1_sample when phase goes past 1.0; */
/* use s_fm_n (computed below) to avoid roundoff errors: */
if (susp->s_fm_n <= 0) {
susp_check_term_log_samples(s_fm, s_fm_ptr, s_fm_cnt);
susp->s_fm_x1_sample = susp_fetch_sample(s_fm, s_fm_ptr, s_fm_cnt);
susp->s_fm_pHaSe -= 1.0;
/* s_fm_n gets number of samples before phase exceeds 1.0: */
susp->s_fm_n = (long) ((1.0 - susp->s_fm_pHaSe) *
susp->output_per_s_fm);
}
togo = min(togo, susp->s_fm_n);
s_fm_val = susp->s_fm_x1_sample;
/* 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) togo = 0; /* avoids rounding errros */
if (togo == 0) break;
}
/* don't run past logical stop time */
if (!susp->logically_stopped && susp->susp.log_stop_cnt != UNKNOWN) {
int to_stop = susp->susp.log_stop_cnt - (susp->susp.current + cnt);
/* break if to_stop == 0 (we're at the logical stop)
* AND cnt > 0 (we're not at the beginning of the
* output block).
*/
if (to_stop < 0) to_stop = 0; /* avoids rounding errors */
if (to_stop < togo) {
if (to_stop == 0) {
if (cnt) {
togo = 0;
break;
} else /* keep togo as is: since cnt == 0, we
* can set the logical stop flag on this
* output block
*/
susp->logically_stopped = true;
} else /* limit togo so we can start a new
* block at the LST
*/
togo = to_stop;
}
}
{ long cur = susp->susp.current + cnt;
n = susp->next_breakpoint - cur;
if (n == 0) n = siosc_table_update(susp, cur);
}
togo = min(n, togo);
n = togo;
table_len_reg = susp->table_len;
ph_incr_reg = susp->ph_incr;
table_a_samps_reg = susp->table_a_samps;
table_b_samps_reg = susp->table_b_samps;
phase_reg = susp->phase;
ampramp_a_reg = susp->ampramp_a;
ampramp_b_reg = susp->ampramp_b;
ampslope_reg = susp->ampslope;
out_ptr_reg = out_ptr;
if (n) do { /* the inner sample computation loop */
long table_index;
double xa, xb;
table_index = (long) phase_reg;
xa = table_a_samps_reg[table_index];
xb = table_b_samps_reg[table_index];
*out_ptr_reg++ = (sample_type)
(ampramp_a_reg * (xa + (phase_reg - table_index) *
(table_a_samps_reg[table_index + 1] - xa)) +
ampramp_b_reg * (xb + (phase_reg - table_index) *
(table_b_samps_reg[table_index + 1] - xb)));
ampramp_a_reg -= ampslope_reg;
ampramp_b_reg += ampslope_reg;
phase_reg += ph_incr_reg + s_fm_val;
while (phase_reg > table_len_reg) phase_reg -= table_len_reg;
/* watch out for negative frequencies! */
while (phase_reg < 0) phase_reg += table_len_reg;
} while (--n); /* inner loop */
susp->phase = phase_reg;
susp->ampramp_a = ampramp_a_reg;
susp->ampramp_b = ampramp_b_reg;
out_ptr += togo;
susp->s_fm_pHaSe += togo * susp->s_fm_pHaSe_iNcR;
susp->s_fm_n -= 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;
}
/* test for logical stop */
if (susp->logically_stopped) {
snd_list->logically_stopped = true;
} else if (susp->susp.log_stop_cnt == susp->susp.current) {
susp->logically_stopped = true;
}
} /* siosc_r_fetch */
void siosc_toss_fetch(snd_susp_type a_susp, snd_list_type snd_list)
{
siosc_susp_type susp = (siosc_susp_type) a_susp;
time_type final_time = susp->susp.t0;
long n;
/* fetch samples from s_fm up to final_time for this block of zeros */
while ((round((final_time - susp->s_fm->t0) * susp->s_fm->sr)) >=
susp->s_fm->current)
susp_get_samples(s_fm, s_fm_ptr, s_fm_cnt);
/* convert to normal processing when we hit final_count */
/* we want each signal positioned at final_time */
n = round((final_time - susp->s_fm->t0) * susp->s_fm->sr -
(susp->s_fm->current - susp->s_fm_cnt));
susp->s_fm_ptr += n;
susp_took(s_fm_cnt, n);
susp->susp.fetch = susp->susp.keep_fetch;
(*(susp->susp.fetch))(a_susp, snd_list);
}
void siosc_mark(snd_susp_type a_susp)
{
siosc_susp_type susp = (siosc_susp_type) a_susp;
if (susp->lis) mark(susp->lis);
sound_xlmark(susp->s_fm);
}
void siosc_free(snd_susp_type a_susp)
{
siosc_susp_type susp = (siosc_susp_type) a_susp;
table_unref(susp->table_a_ptr);
table_unref(susp->table_b_ptr_ptr);
sound_unref(susp->s_fm);
ffree_generic(susp, sizeof(siosc_susp_node), "siosc_free");
}
void siosc_print_tree(snd_susp_type a_susp, int n)
{
siosc_susp_type susp = (siosc_susp_type) a_susp;
indent(n);
stdputstr("s_fm:");
sound_print_tree_1(susp->s_fm, n);
}
sound_type snd_make_siosc(LVAL lis, rate_type sr, double hz, time_type t0, sound_type s_fm)
{
register siosc_susp_type susp;
/* sr specified as input parameter */
/* t0 specified as input parameter */
int interp_desc = 0;
sample_type scale_factor = 1.0F;
time_type t0_min = t0;
falloc_generic(susp, siosc_susp_node, "snd_make_siosc");
susp->table_len = 0.0;
susp->ph_incr = 0.0;
susp->table_a_ptr = NULL;
susp->table_b_ptr_ptr = NULL;
susp->table_a_samps = NULL;
susp->table_b_samps = NULL;
susp->table_sr = 0.0;
susp->phase = 0.0;
susp->lis = lis;
susp->next_breakpoint = 0;
susp->ampramp_a = 1.0;
susp->ampramp_b = 0.0;
susp->ampslope = 0.0;
siosc_table_init(susp);
susp->ph_incr = hz * susp->table_len / sr;
s_fm->scale = (sample_type) (s_fm->scale * (susp->table_len / sr));
/* make sure no sample rate is too high */
if (s_fm->sr > sr) {
sound_unref(s_fm);
snd_badsr();
}
/* select a susp fn based on sample rates */
interp_desc = (interp_desc << 2) + interp_style(s_fm, sr);
switch (interp_desc) {
case INTERP_n: /* handled below */
case INTERP_s: susp->susp.fetch = siosc_s_fetch; break;
case INTERP_i: susp->susp.fetch = siosc_i_fetch; break;
case INTERP_r: susp->susp.fetch = siosc_r_fetch; break;
default: snd_badsr(); break;
}
susp->terminate_cnt = UNKNOWN;
/* handle unequal start times, if any */
if (t0 < s_fm->t0) sound_prepend_zeros(s_fm, t0);
/* minimum start time over all inputs: */
t0_min = min(s_fm->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 = siosc_toss_fetch;
}
/* initialize susp state */
susp->susp.free = siosc_free;
susp->susp.sr = sr;
susp->susp.t0 = t0;
susp->susp.mark = siosc_mark;
susp->susp.print_tree = siosc_print_tree;
susp->susp.name = "siosc";
susp->logically_stopped = false;
susp->susp.log_stop_cnt = logical_stop_cnt_cvt(s_fm);
susp->started = false;
susp->susp.current = 0;
susp->s_fm = s_fm;
susp->s_fm_cnt = 0;
susp->s_fm_pHaSe = 0.0;
susp->s_fm_pHaSe_iNcR = s_fm->sr / sr;
susp->s_fm_n = 0;
susp->output_per_s_fm = sr / s_fm->sr;
return sound_create((snd_susp_type)susp, t0, sr, scale_factor);
}
sound_type snd_siosc(LVAL lis, rate_type sr, double hz, time_type t0, sound_type s_fm)
{
sound_type s_fm_copy = sound_copy(s_fm);
return snd_make_siosc(lis, sr, hz, t0, s_fm_copy);
}