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mirror of https://github.com/cookiengineer/audacity synced 2025-04-30 15:49:41 +02:00
Leland Lucius 15b9bb96cd Update nyquist to SVN r331 (r3.16+)
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   r331 | rbd | 2020-10-13 12:40:12 -0500 (Tue, 13 Oct 2020) | 2 lines

   Also forgot to install NyquistWords.txt

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   r330 | rbd | 2020-10-13 12:34:06 -0500 (Tue, 13 Oct 2020) | 2 lines

   Forgot to move nyquistman.pdf from docsrc/s2h to release

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   r329 | rbd | 2020-10-13 11:32:33 -0500 (Tue, 13 Oct 2020) | 2 lines

   Updated some version numbers for 3.16.

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   r328 | rbd | 2020-10-13 11:20:52 -0500 (Tue, 13 Oct 2020) | 2 lines

   Fixed NyquistIDE antialiasing for plot text, fix format of message.

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   r327 | rbd | 2020-10-12 21:01:53 -0500 (Mon, 12 Oct 2020) | 2 lines

   Fixed a couple of format problems in manual. This version of Nyquist has been tested wtih macOS, Linux, 32&64-bit Windows.

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   r326 | rbd | 2020-10-12 20:21:38 -0500 (Mon, 12 Oct 2020) | 1 line

   Modified WIN32 32-bit XLisp to use 64-bit FIXNUMs. This allows XLisp and Nyquist to handle big sounds even on 32-bit machines. Probably at some cost, but inner loops are mostly float and int32, and the Nyquist release is 64-bit anyway. Maybe we'll have to run some benchmarks on Audacity, which is still 32-bit on Windows.
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   r325 | rbd | 2020-10-12 13:16:57 -0500 (Mon, 12 Oct 2020) | 1 line

   Win64 passes bigfiletest.lsp now. This version should work on all 64-bit systems now. These changes untested on Linux and macOS.
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   r324 | rbd | 2020-10-11 21:31:53 -0500 (Sun, 11 Oct 2020) | 2 lines

   I couldn't free enough space on my linux box, so I adjusted the bigfiletest to write 8-bit ulaw. It's still >4GB and >4G samples. Works on Linux.

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   r323 | rbd | 2020-10-11 19:41:25 -0500 (Sun, 11 Oct 2020) | 2 lines

   Missing file from last commit.

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   r322 | rbd | 2020-10-11 19:36:08 -0500 (Sun, 11 Oct 2020) | 1 line

   Found another case where WIN64 needs int64_t instead of long for sample count.
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   r321 | rbd | 2020-10-11 19:33:25 -0500 (Sun, 11 Oct 2020) | 3 lines

   Fixed s-save to	handle optional	and keyword parameters (which should never have	been mixed in the first	place).	Documentation cleanup - should be final for this version.

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   r320 | rbd | 2020-10-11 14:44:37 -0500 (Sun, 11 Oct 2020) | 2 lines

   Fixes to handle IRCAM sound format and tests for big file io working on macOS.

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   r319 | rbd | 2020-10-10 21:31:58 -0500 (Sat, 10 Oct 2020) | 2 lines

   Changes for linux and to avoid compiler warnings on linux.

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   r318 | rbd | 2020-10-10 20:50:23 -0500 (Sat, 10 Oct 2020) | 1 line

   This is the test used for Win64 version.
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   r317 | rbd | 2020-10-10 20:34:34 -0500 (Sat, 10 Oct 2020) | 1 line

   This version works on Win64. Need to test changes on macOS and linux.
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   r316 | rbd | 2020-10-10 19:59:15 -0500 (Sat, 10 Oct 2020) | 2 lines

   PWL changes to avoid compiler warning.

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   r315 | rbd | 2020-10-10 19:34:04 -0500 (Sat, 10 Oct 2020) | 2 lines

   A few more changes for 64-bit sample counts on Win64

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   r314 | rbd | 2020-10-10 13:19:42 -0500 (Sat, 10 Oct 2020) | 2 lines

   Fixed int64_t declaration in gate.alg

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   r313 | rbd | 2020-10-10 12:07:40 -0500 (Sat, 10 Oct 2020) | 2 lines

   Fixes to gate for long sounds

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   r312 | rbd | 2020-10-10 11:47:29 -0500 (Sat, 10 Oct 2020) | 2 lines

   Fixed sound_save types for intgen

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   r311 | rbd | 2020-10-10 11:09:01 -0500 (Sat, 10 Oct 2020) | 2 lines

   Fixed a 64-bit sample count problem in siosc.alg

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   r310 | rbd | 2020-10-10 11:03:12 -0500 (Sat, 10 Oct 2020) | 2 lines

   Fixed sndmax to handle 64-bit sample counts.

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   r309 | rbd | 2020-10-10 10:57:04 -0500 (Sat, 10 Oct 2020) | 2 lines

   Forgot to re-translate all tran/*.alg files with fix for int64 cast to int32. This version compiles on macOS and ready for test on Win64.

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   r308 | rbd | 2020-10-10 10:16:05 -0500 (Sat, 10 Oct 2020) | 2 lines

   Everything seems to compile and run on macOS now. Moving changes to Windows for test.

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   r307 | rbd | 2020-10-10 09:23:45 -0500 (Sat, 10 Oct 2020) | 1 line

   Added casts to avoid compiler warnings and to review changes to support 64-bit sample counts on Windows. Still not complete, and waiting to regenerate and compile tran directory code after updates to translation code that will insert more casts.
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   r306 | rbd | 2020-10-09 21:55:15 -0500 (Fri, 09 Oct 2020) | 2 lines

   Rebuilt seqfnint.c from header files.

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   r305 | rbd | 2020-10-09 21:53:33 -0500 (Fri, 09 Oct 2020) | 1 line

   Changed some FIXNUMS to LONG to avoid compiler warnings in seqfnint.c
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   r304 | rbd | 2020-10-09 21:44:03 -0500 (Fri, 09 Oct 2020) | 2 lines

   I discovered forgotten regression-test.lsp and added test that requires 64-bit sample counts to pass. Fixed a few bugs revealed by running the type-checking regression tests.

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   r303 | rbd | 2020-10-09 12:28:58 -0500 (Fri, 09 Oct 2020) | 2 lines

   Changes for 64-bit sample counts broke mult-channel s-save. Fixed in the commit for macOS.

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   r302 | rbd | 2020-10-09 10:03:39 -0500 (Fri, 09 Oct 2020) | 2 lines

   Changed snd-play to return samples computed and used that to make a test for computing long sounds that would overflow 32-bit length counts.

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   r301 | rbd | 2020-10-09 09:11:26 -0500 (Fri, 09 Oct 2020) | 2 lines

   corrected mistake in delaycv.alg and re-translated

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   r300 | rbd | 2020-10-09 09:09:06 -0500 (Fri, 09 Oct 2020) | 2 lines

   Fix to delaycv.alg -- "s" changed to "input" to avoid matching "s" in "sample_type".

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   r299 | rbd | 2020-10-09 09:03:33 -0500 (Fri, 09 Oct 2020) | 4 lines

   To avoid compiler warnings, XLisp interfaces to C int and long are now
   specified as LONG rather than FIXNUM, and the stubs that call the C
   functions cast FIXNUMs from XLisp into longs before calling C functions.

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   r298 | rbd | 2020-10-08 22:20:26 -0500 (Thu, 08 Oct 2020) | 2 lines

   This commit has many more fixes to handle long (64-bit) sounds, including a lot of fixes for warnings by Visual Studio assigning int64_t to long (works on macOS, doesn't work on VS). This was compiled and tested on macOS, and even computed a 27.1-hour sound using OSC, LP, SUM and MULT (haven't tested I/O yet).

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   r297 | rbd | 2020-10-07 13:04:02 -0500 (Wed, 07 Oct 2020) | 2 lines

   This is a major cleanup. It started with the goal of changing long to int64_t for sample counts so that on 64-bit windows, where long is only 32-bits, the sample counts would nevertheless be 64-bit allowing long sounds, which was a limitation for long recordings in Audacity. Since I was using compiler warnings to track possible loss-of-precision conversions from 64-bit sample counts, and there were *many* warnings, I started cleaning up *all* the warnings and ended up with a very large set of changes, including "modernizing" C declarations that date back to XLisp and CMU MIDI Toolkit code and were never changed. This version runs all the examples.sal code on macOS, but will surely have problems on Windows and Linux given the number of changes.

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   r296 | rbd | 2020-10-06 13:34:20 -0500 (Tue, 06 Oct 2020) | 2 lines

   More changes from long to int64_t for sample counts.

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   r295 | rbd | 2020-10-06 11:53:49 -0500 (Tue, 06 Oct 2020) | 2 lines

   More work on using 64-bit sample counts. Changed MAX_STOP from 32-bit to 64-bit limit.

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   r294 | rbd | 2020-10-06 11:48:05 -0500 (Tue, 06 Oct 2020) | 2 lines

   Made some changes so that sample counts are int64_t (for windows) instead of long to support sample counts above 31 bits.

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   r293 | rbd | 2020-10-04 21:30:55 -0500 (Sun, 04 Oct 2020) | 2 lines

   Fixed a few minor things for Linux and tested on Linux.

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   r292 | rbd | 2020-10-04 21:00:28 -0500 (Sun, 04 Oct 2020) | 2 lines

   Update extensions: all are minor changes.

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   r291 | rbd | 2020-09-24 13:59:31 -0500 (Thu, 24 Sep 2020) | 2 lines

   New implementation of seq and seqrep, added get-real-time, documented get-real-time, fixed examples.sal and examples.lsp which are now in lib rather than extensions (so they are now back in the basic installation), other cleanup.

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   r290 | rbd | 2020-08-16 16:24:52 -0500 (Sun, 16 Aug 2020) | 2 lines

   Fixed bug in snd-gate, revised GATE and NOISE-GATE to handle multi-channel sound. RMS now handles multi-channel input. S-AVG added to take multichannel input (but not used, because RMS could not be written without making SND-SRATE convert multichannel sound to vector of floats. That seems to be going toward a fully vectorized model. Not going there for now.

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   r289 | rbd | 2020-07-09 16:27:45 -0500 (Thu, 09 Jul 2020) | 2 lines

   Added GET-REAL-TIME function to XLISP. May not work yet on Windows. Various fixes for compiler warnings. I noticed FLAC doesn't work (I guess it never did) and I cannot figure out how this even links because flac_min seems to be undefined. Something to look at later.
2021-01-27 23:45:25 -06:00

603 lines
21 KiB
C

#include "stdio.h"
#ifndef mips
#include "stdlib.h"
#endif
#include "xlisp.h"
#include "sound.h"
#include "falloc.h"
#include "cext.h"
#include "aresonvc.h"
void aresonvc_free(snd_susp_type a_susp);
typedef struct aresonvc_susp_struct {
snd_susp_node susp;
boolean started;
int64_t terminate_cnt;
boolean logically_stopped;
sound_type s1;
int s1_cnt;
sample_block_values_type s1_ptr;
sound_type hz;
int hz_cnt;
sample_block_values_type hz_ptr;
/* support for interpolation of hz */
sample_type hz_x1_sample;
double hz_pHaSe;
double hz_pHaSe_iNcR;
/* support for ramp between samples of hz */
double output_per_hz;
int64_t hz_n;
double c3co;
double c3p1;
double c3t4;
double omc3;
double c2;
double c1;
int normalization;
double y1;
double y2;
} aresonvc_susp_node, *aresonvc_susp_type;
void aresonvc_ns_fetch(snd_susp_type a_susp, snd_list_type snd_list)
{
aresonvc_susp_type susp = (aresonvc_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 c3co_reg;
register double c3p1_reg;
register double c3t4_reg;
register double omc3_reg;
register double c2_reg;
register double c1_reg;
register int normalization_reg;
register double y1_reg;
register double y2_reg;
register sample_type hz_scale_reg = susp->hz->scale;
register sample_block_values_type hz_ptr_reg;
register sample_block_values_type s1_ptr_reg;
falloc_sample_block(out, "aresonvc_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 s1 input sample block: */
susp_check_term_log_samples(s1, s1_ptr, s1_cnt);
togo = min(togo, susp->s1_cnt);
/* don't run past the hz input sample block: */
susp_check_term_samples(hz, hz_ptr, hz_cnt);
togo = min(togo, susp->hz_cnt);
/* don't run past terminate time */
if (susp->terminate_cnt != UNKNOWN &&
susp->terminate_cnt <= susp->susp.current + cnt + togo) {
togo = (int) (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) {
int64_t 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 = (int) to_stop;
}
}
n = togo;
c3co_reg = susp->c3co;
c3p1_reg = susp->c3p1;
c3t4_reg = susp->c3t4;
omc3_reg = susp->omc3;
c2_reg = susp->c2;
c1_reg = susp->c1;
normalization_reg = susp->normalization;
y1_reg = susp->y1;
y2_reg = susp->y2;
hz_ptr_reg = susp->hz_ptr;
s1_ptr_reg = susp->s1_ptr;
out_ptr_reg = out_ptr;
if (n) do { /* the inner sample computation loop */
register double y0, current; c2_reg = c3t4_reg * cos((hz_scale_reg * *hz_ptr_reg++)) / c3p1_reg;
c1_reg = (normalization_reg == 0 ? 0.0 :
(normalization_reg == 1 ? 1.0 - omc3_reg * sqrt(1.0 - c2_reg * c2_reg / c3t4_reg) :
1.0 - sqrt(c3p1_reg * c3p1_reg - c2_reg * c2_reg) * omc3_reg / c3p1_reg));
current = *s1_ptr_reg++;
y0 = c1_reg * current + c2_reg * y1_reg - c3co_reg * y2_reg;
*out_ptr_reg++ = (sample_type) y0;
y2_reg = y1_reg; y1_reg = y0 - current;
} while (--n); /* inner loop */
susp->y1 = y1_reg;
susp->y2 = y2_reg;
/* using hz_ptr_reg is a bad idea on RS/6000: */
susp->hz_ptr += togo;
/* using s1_ptr_reg is a bad idea on RS/6000: */
susp->s1_ptr += togo;
out_ptr += togo;
susp_took(s1_cnt, togo);
susp_took(hz_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;
}
} /* aresonvc_ns_fetch */
void aresonvc_ni_fetch(snd_susp_type a_susp, snd_list_type snd_list)
{
aresonvc_susp_type susp = (aresonvc_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 c3co_reg;
register double c3p1_reg;
register double c3t4_reg;
register double omc3_reg;
register double c2_reg;
register double c1_reg;
register int normalization_reg;
register double y1_reg;
register double y2_reg;
register double hz_pHaSe_iNcR_rEg = susp->hz_pHaSe_iNcR;
register double hz_pHaSe_ReG;
register sample_type hz_x1_sample_reg;
register sample_block_values_type s1_ptr_reg;
falloc_sample_block(out, "aresonvc_ni_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_samples(hz, hz_ptr, hz_cnt);
susp->hz_x1_sample = susp_fetch_sample(hz, hz_ptr, hz_cnt);
susp->c2 = susp->c3t4 * cos(susp->hz_x1_sample) / susp->c3p1;
susp->c1 = (susp->normalization == 0 ? 0.0 :
(susp->normalization == 1 ? 1.0 - susp->omc3 * sqrt(1.0 - susp->c2 * susp->c2 / susp->c3t4) :
1.0 - sqrt(susp->c3p1 * susp->c3p1 - susp->c2 * susp->c2) * susp->omc3 / susp->c3p1));
}
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 s1 input sample block: */
susp_check_term_log_samples(s1, s1_ptr, s1_cnt);
togo = min(togo, susp->s1_cnt);
/* don't run past terminate time */
if (susp->terminate_cnt != UNKNOWN &&
susp->terminate_cnt <= susp->susp.current + cnt + togo) {
togo = (int) (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) {
int64_t 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 = (int) to_stop;
}
}
n = togo;
c3co_reg = susp->c3co;
c3p1_reg = susp->c3p1;
c3t4_reg = susp->c3t4;
omc3_reg = susp->omc3;
c2_reg = susp->c2;
c1_reg = susp->c1;
normalization_reg = susp->normalization;
y1_reg = susp->y1;
y2_reg = susp->y2;
hz_pHaSe_ReG = susp->hz_pHaSe;
hz_x1_sample_reg = susp->hz_x1_sample;
s1_ptr_reg = susp->s1_ptr;
out_ptr_reg = out_ptr;
if (n) do { /* the inner sample computation loop */
register double y0, current; if (hz_pHaSe_ReG >= 1.0) {
/* fixup-depends hz */
/* pick up next sample as hz_x1_sample: */
susp->hz_ptr++;
susp_took(hz_cnt, 1);
hz_pHaSe_ReG -= 1.0;
susp_check_term_samples_break(hz, hz_ptr, hz_cnt, hz_x1_sample_reg);
hz_x1_sample_reg = susp_current_sample(hz, hz_ptr);
c2_reg = c3t4_reg * cos(hz_x1_sample_reg) / c3p1_reg;
c1_reg = (normalization_reg == 0 ? 0.0 :
(normalization_reg == 1 ? 1.0 - omc3_reg * sqrt(1.0 - c2_reg * c2_reg / c3t4_reg) :
1.0 - sqrt(c3p1_reg * c3p1_reg - c2_reg * c2_reg) * omc3_reg / c3p1_reg));
}
current = *s1_ptr_reg++;
y0 = c1_reg * current + c2_reg * y1_reg - c3co_reg * y2_reg;
*out_ptr_reg++ = (sample_type) y0;
y2_reg = y1_reg; y1_reg = y0 - current;
hz_pHaSe_ReG += hz_pHaSe_iNcR_rEg;
} while (--n); /* inner loop */
togo -= n;
susp->y1 = y1_reg;
susp->y2 = y2_reg;
susp->hz_pHaSe = hz_pHaSe_ReG;
susp->hz_x1_sample = hz_x1_sample_reg;
/* using s1_ptr_reg is a bad idea on RS/6000: */
susp->s1_ptr += togo;
out_ptr += togo;
susp_took(s1_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;
}
} /* aresonvc_ni_fetch */
void aresonvc_nr_fetch(snd_susp_type a_susp, snd_list_type snd_list)
{
aresonvc_susp_type susp = (aresonvc_susp_type) a_susp;
int cnt = 0; /* how many samples computed */
sample_type hz_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 c3co_reg;
register double c3p1_reg;
register double c3t4_reg;
register double omc3_reg;
register double c2_reg;
register double c1_reg;
register int normalization_reg;
register double y1_reg;
register double y2_reg;
register sample_block_values_type s1_ptr_reg;
falloc_sample_block(out, "aresonvc_nr_fetch");
out_ptr = out->samples;
snd_list->block = out;
/* make sure sounds are primed with first values */
if (!susp->started) {
susp->started = true;
susp->hz_pHaSe = 1.0;
}
susp_check_term_samples(hz, hz_ptr, hz_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 the s1 input sample block: */
susp_check_term_log_samples(s1, s1_ptr, s1_cnt);
togo = min(togo, susp->s1_cnt);
/* grab next hz_x1_sample when phase goes past 1.0; */
/* use hz_n (computed below) to avoid roundoff errors: */
if (susp->hz_n <= 0) {
susp_check_term_samples(hz, hz_ptr, hz_cnt);
susp->hz_x1_sample = susp_fetch_sample(hz, hz_ptr, hz_cnt);
susp->hz_pHaSe -= 1.0;
/* hz_n gets number of samples before phase exceeds 1.0: */
susp->hz_n = (int64_t) ((1.0 - susp->hz_pHaSe) *
susp->output_per_hz);
susp->c2 = susp->c3t4 * cos(susp->hz_x1_sample) / susp->c3p1;
susp->c1 = (susp->normalization == 0 ? 0.0 :
(susp->normalization == 1 ? 1.0 - susp->omc3 * sqrt(1.0 - susp->c2 * susp->c2 / susp->c3t4) :
1.0 - sqrt(susp->c3p1 * susp->c3p1 - susp->c2 * susp->c2) * susp->omc3 / susp->c3p1));
}
togo = (int) min(togo, susp->hz_n);
hz_val = susp->hz_x1_sample;
/* don't run past terminate time */
if (susp->terminate_cnt != UNKNOWN &&
susp->terminate_cnt <= susp->susp.current + cnt + togo) {
togo = (int) (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) {
int64_t 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 = (int) to_stop;
}
}
n = togo;
c3co_reg = susp->c3co;
c3p1_reg = susp->c3p1;
c3t4_reg = susp->c3t4;
omc3_reg = susp->omc3;
c2_reg = susp->c2;
c1_reg = susp->c1;
normalization_reg = susp->normalization;
y1_reg = susp->y1;
y2_reg = susp->y2;
s1_ptr_reg = susp->s1_ptr;
out_ptr_reg = out_ptr;
if (n) do { /* the inner sample computation loop */
register double y0, current; current = *s1_ptr_reg++;
y0 = c1_reg * current + c2_reg * y1_reg - c3co_reg * y2_reg;
*out_ptr_reg++ = (sample_type) y0;
y2_reg = y1_reg; y1_reg = y0 - current;
} while (--n); /* inner loop */
susp->y1 = y1_reg;
susp->y2 = y2_reg;
/* using s1_ptr_reg is a bad idea on RS/6000: */
susp->s1_ptr += togo;
out_ptr += togo;
susp_took(s1_cnt, togo);
susp->hz_pHaSe += togo * susp->hz_pHaSe_iNcR;
susp->hz_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;
}
} /* aresonvc_nr_fetch */
void aresonvc_toss_fetch(snd_susp_type a_susp, snd_list_type snd_list)
{
aresonvc_susp_type susp = (aresonvc_susp_type) a_susp;
time_type final_time = susp->susp.t0;
int n;
/* fetch samples from s1 up to final_time for this block of zeros */
while ((ROUNDBIG((final_time - susp->s1->t0) * susp->s1->sr)) >=
susp->s1->current)
susp_get_samples(s1, s1_ptr, s1_cnt);
/* fetch samples from hz up to final_time for this block of zeros */
while ((ROUNDBIG((final_time - susp->hz->t0) * susp->hz->sr)) >=
susp->hz->current)
susp_get_samples(hz, hz_ptr, hz_cnt);
/* convert to normal processing when we hit final_count */
/* we want each signal positioned at final_time */
n = (int) ROUNDBIG((final_time - susp->s1->t0) * susp->s1->sr -
(susp->s1->current - susp->s1_cnt));
susp->s1_ptr += n;
susp_took(s1_cnt, n);
n = (int) ROUNDBIG((final_time - susp->hz->t0) * susp->hz->sr -
(susp->hz->current - susp->hz_cnt));
susp->hz_ptr += n;
susp_took(hz_cnt, n);
susp->susp.fetch = susp->susp.keep_fetch;
(*(susp->susp.fetch))(a_susp, snd_list);
}
void aresonvc_mark(snd_susp_type a_susp)
{
aresonvc_susp_type susp = (aresonvc_susp_type) a_susp;
sound_xlmark(susp->s1);
sound_xlmark(susp->hz);
}
void aresonvc_free(snd_susp_type a_susp)
{
aresonvc_susp_type susp = (aresonvc_susp_type) a_susp;
sound_unref(susp->s1);
sound_unref(susp->hz);
ffree_generic(susp, sizeof(aresonvc_susp_node), "aresonvc_free");
}
void aresonvc_print_tree(snd_susp_type a_susp, int n)
{
aresonvc_susp_type susp = (aresonvc_susp_type) a_susp;
indent(n);
stdputstr("s1:");
sound_print_tree_1(susp->s1, n);
indent(n);
stdputstr("hz:");
sound_print_tree_1(susp->hz, n);
}
sound_type snd_make_aresonvc(sound_type s1, sound_type hz, double bw, int normalization)
{
register aresonvc_susp_type susp;
rate_type sr = s1->sr;
time_type t0 = max(s1->t0, hz->t0);
int interp_desc = 0;
sample_type scale_factor = 1.0F;
time_type t0_min = t0;
/* combine scale factors of linear inputs (S1) */
scale_factor *= s1->scale;
s1->scale = 1.0F;
/* try to push scale_factor back to a low sr input */
if (s1->sr < sr) { s1->scale = scale_factor; scale_factor = 1.0F; }
falloc_generic(susp, aresonvc_susp_node, "snd_make_aresonvc");
susp->c3co = exp(bw * -PI2 / s1->sr);
susp->c3p1 = susp->c3co + 1.0;
susp->c3t4 = susp->c3co * 4.0;
susp->omc3 = 1.0 - susp->c3co;
susp->c2 = 0.0;
susp->c1 = 0.0;
susp->normalization = normalization;
susp->y1 = 0.0;
susp->y2 = 0.0;
hz->scale = (sample_type) (hz->scale * (PI2 / s1->sr));
/* make sure no sample rate is too high */
if (hz->sr > sr) {
sound_unref(hz);
snd_badsr();
}
/* select a susp fn based on sample rates */
interp_desc = (interp_desc << 2) + interp_style(s1, sr);
interp_desc = (interp_desc << 2) + interp_style(hz, sr);
switch (interp_desc) {
case INTERP_nn: /* handled below */
case INTERP_ns: susp->susp.fetch = aresonvc_ns_fetch; break;
case INTERP_ni: susp->susp.fetch = aresonvc_ni_fetch; break;
case INTERP_nr: susp->susp.fetch = aresonvc_nr_fetch; break;
default: snd_badsr(); break;
}
susp->terminate_cnt = UNKNOWN;
/* handle unequal start times, if any */
if (t0 < s1->t0) sound_prepend_zeros(s1, t0);
if (t0 < hz->t0) sound_prepend_zeros(hz, t0);
/* minimum start time over all inputs: */
t0_min = min(s1->t0, min(hz->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 = aresonvc_toss_fetch;
}
/* initialize susp state */
susp->susp.free = aresonvc_free;
susp->susp.sr = sr;
susp->susp.t0 = t0;
susp->susp.mark = aresonvc_mark;
susp->susp.print_tree = aresonvc_print_tree;
susp->susp.name = "aresonvc";
susp->logically_stopped = false;
susp->susp.log_stop_cnt = logical_stop_cnt_cvt(s1);
susp->started = false;
susp->susp.current = 0;
susp->s1 = s1;
susp->s1_cnt = 0;
susp->hz = hz;
susp->hz_cnt = 0;
susp->hz_pHaSe = 0.0;
susp->hz_pHaSe_iNcR = hz->sr / sr;
susp->hz_n = 0;
susp->output_per_hz = sr / hz->sr;
return sound_create((snd_susp_type)susp, t0, sr, scale_factor);
}
sound_type snd_aresonvc(sound_type s1, sound_type hz, double bw, int normalization)
{
sound_type s1_copy = sound_copy(s1);
sound_type hz_copy = sound_copy(hz);
return snd_make_aresonvc(s1_copy, hz_copy, bw, normalization);
}