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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

328 lines
12 KiB
C

/* trigger.c -- return zero until input transitions from <=0 to >0, then
evaluate a closure to get a signal and convert to an add
of the new signal and a copy of this trigger object.
The sample rate of the output is the sample rate of the input, and
sounds returned by the closure must also have a matching sample rate.
The trigger will take place on the first input sample (zero delay) if the
first sample of the input is >0.
The input scale factor is assumed to be 1, so caller should force scaling
especially if the scale factor is negative (!)
The trigger terminates when the input signal terminates (but any adds
continue to run until all their inputs terminate).
Some implementation notes:
The closure gets evaluated at the time of the positive sample.
When the positive sample is encountered, first close off the
current output block.
Next, evaluate the closure, clone the trigger, and convert
the current trigger to an add. The next fetch will therefore
go to the add susp and it will add the closure result to the
zeros that continue to be generated by (a clone of) the trigger.
It would be simple if we could back the clone up one sample:
on the first call to the add, it will ask for samples from the
trigger clone and the closure, but the trigger clone has already
processed the positive sample, so it is one sample ahead of
everyone else. Since we've just read a sample, we CAN back up
just by carefully updating the input pointer to one less than
we actually read, forcing a reread later. (We'll still store
the previous value so re-reading will not re-trigger.)
*/
/* CHANGE LOG
* --------------------------------------------------------------------
* 13Dec06 rbd created from sndseq.c
*/
#include "stdio.h"
#ifndef mips
#include "stdlib.h"
#endif
#include "xlisp.h"
#include "sound.h"
#include "falloc.h"
#include "scale.h"
#include "add.h"
#include "extern.h"
#include "cext.h"
#include "assert.h"
#define TRIGGERDBG 1
#define D if (TRIGGERDBG)
/* Note: this structure is identical to an add_susp structure up
to the field output_per_s2 so that we can convert this into
an add after eval'ing the closure. Since this struct is bigger
than an add, make sure not to clobber the "free" routine
(trigger_free) or else we'll leak memory.
*/
typedef struct trigger_susp_struct {
snd_susp_node susp;
boolean started;
int terminate_bits;
int64_t terminate_cnt;
int logical_stop_bits;
boolean logically_stopped;
sound_type s1;
int s1_cnt;
sample_block_type s1_bptr; /* block pointer */
sample_block_values_type s1_ptr;
sound_type s2;
int s2_cnt;
sample_block_type s2_bptr; /* block pointer */
sample_block_values_type s2_ptr;
/* trigger-specific data starts here */
sample_type previous;
LVAL closure;
} trigger_susp_node, *trigger_susp_type;
void trigger_fetch(snd_susp_type, snd_list_type);
void trigger_free(snd_susp_type a_susp);
extern LVAL s_stdout;
void trigger_mark(snd_susp_type a_susp)
{
trigger_susp_type susp = (trigger_susp_type) a_susp;
sound_xlmark(susp->s1);
if (susp->closure) mark(susp->closure);
}
/* trigger_fetch returns zero blocks until s1 goes from <=0 to >0 */
/**/
void trigger_fetch(snd_susp_type a_susp, snd_list_type snd_list)
{
trigger_susp_type susp = (trigger_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 sample_block_values_type input_ptr_reg;
falloc_sample_block(out, "trigger_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 sample block: */
susp_check_term_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) break;
}
n = togo;
input_ptr_reg = susp->s1_ptr;
out_ptr_reg = out_ptr;
if (n) do { /* the inner sample computation loop */
sample_type s = *input_ptr_reg++;
if (susp->previous <= 0 && s > 0) {
trigger_susp_type new_trigger;
sound_type new_trigger_snd;
LVAL result;
int64_t delay; /* sample delay to s2 */
time_type now;
susp->previous = s; /* don't retrigger */
/**** close off block ****/
togo = togo - n;
susp->s1_ptr += togo;
susp_took(s1_cnt, togo);
cnt += togo;
snd_list->block_len = cnt;
susp->susp.current += cnt;
now = susp->susp.t0 + susp->susp.current / susp->susp.sr;
/**** eval closure and add result ****/
D nyquist_printf("trigger_fetch: about to eval closure at %g, "
"susp->susp.t0 %g, susp.current %d:\n",
now, susp->susp.t0, (int)susp->susp.current);
xlsave1(result);
result = xleval(cons(susp->closure, consa(cvflonum(now))));
if (exttypep(result, a_sound)) {
susp->s2 = sound_copy(getsound(result));
D nyquist_printf("trigger: copied result from closure is %p\n",
susp->s2);
} else xlerror("closure did not return a (monophonic) sound",
result);
D nyquist_printf("in trigger: after evaluation; "
"%p returned from evform\n",
susp->s2);
result = NIL;
/**** cloan this trigger to become s1 ****/
falloc_generic(new_trigger, trigger_susp_node,
"new_trigger");
memcpy(new_trigger, susp, sizeof(trigger_susp_node));
/* don't copy s2 -- it should only be referenced by add */
new_trigger->s2 = NULL;
new_trigger_snd = sound_create((snd_susp_type) new_trigger,
now, susp->susp.sr, 1.0F);
susp->s1 = new_trigger_snd;
/* add will have to ask new_trigger for samples, new_trigger
* will continue reading samples from s1 (the original input)
*/
susp->s1_cnt = 0;
susp->s1_ptr = NULL;
/**** convert to add ****/
susp->susp.mark = add_mark;
/* logical stop will be recomputed by add: */
susp->susp.log_stop_cnt = UNKNOWN;
susp->susp.print_tree = add_print_tree;
/* assume sample rates are the same */
if (susp->s1->sr != susp->s2->sr)
xlfail("in trigger: sample rates must match");
/* take care of scale factor, if any */
if (susp->s2->scale != 1.0) {
// stdputstr("normalizing next sound in a seq\n");
susp->s2 = snd_make_normalize(susp->s2);
}
/* figure out which add fetch routine to use */
delay = ROUNDBIG((susp->s2->t0 - now) * susp->s1->sr);
if (delay > 0) { /* fill hole between s1 and s2 */
D stdputstr("using add_s1_nn_fetch\n");
susp->susp.fetch = add_s1_nn_fetch;
susp->susp.name = "trigger:add_s1_nn_fetch";
} else {
susp->susp.fetch = add_s1_s2_nn_fetch;
susp->susp.name = "trigger:add_s1_s2_nn_fetch";
}
D stdputstr("in trigger: calling add's fetch\n");
/* fetch will get called later ..
(*(susp->susp.fetch))(a_susp, snd_list); */
D stdputstr("in trigger: returned from add's fetch\n");
xlpop();
susp->closure = NULL; /* allow garbage collection now */
/**** calculation tree modified, time to exit ****/
/* but if cnt == 0, then we haven't computed any samples */
/* call on new fetch routine to get some samples */
if (cnt == 0) {
// because adder will reallocate
ffree_sample_block(out, "trigger-pre-adder");
(*susp->susp.fetch)(a_susp, snd_list);
}
return;
} else {
susp->previous = s;
/* output zero until ready to add in closure */
*out_ptr_reg++ = 0;
}
} while (--n); /* inner loop */
/* using input_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 */
if (togo == 0 && cnt == 0) {
snd_list_terminate(snd_list);
} else {
snd_list->block_len = cnt;
susp->susp.current += cnt;
}
} /* trigger_fetch */
void trigger_free(snd_susp_type a_susp)
{
trigger_susp_type susp = (trigger_susp_type) a_susp;
sound_unref(susp->s1);
sound_unref(susp->s2);
ffree_generic(susp, sizeof(trigger_susp_node), "trigger_free");
}
void trigger_print_tree(snd_susp_type a_susp, int n)
{
trigger_susp_type susp = (trigger_susp_type) a_susp;
indent(n);
stdputstr("s1:");
sound_print_tree_1(susp->s1, n);
indent(n);
stdputstr("closure:");
stdprint(susp->closure);
indent(n);
stdputstr("s2:");
sound_print_tree_1(susp->s2, n);
}
sound_type snd_make_trigger(sound_type s1, LVAL closure)
{
register trigger_susp_type susp;
/* t0 specified as input parameter */
sample_type scale_factor = 1.0F;
sound_type result;
xlprot1(closure);
falloc_generic(susp, trigger_susp_node, "snd_make_trigger");
if (s1->scale != 1.0) {
/* stdputstr("normalizing first sound in a seq\n"); */
s1 = snd_make_normalize(s1);
}
susp->susp.fetch = trigger_fetch;
susp->terminate_cnt = UNKNOWN;
susp->terminate_bits = 0; /* bits for s1 and s2 termination */
susp->logical_stop_bits = 0; /* bits for s1 and s2 logical stop */
/* initialize susp state */
susp->susp.free = trigger_free;
susp->susp.sr = s1->sr;
susp->susp.t0 = s1->t0;
susp->susp.mark = trigger_mark;
susp->susp.print_tree = trigger_print_tree;
susp->susp.name = "trigger";
susp->logically_stopped = false;
susp->susp.log_stop_cnt = s1->logical_stop_cnt;
susp->started = false;
susp->susp.current = 0;
susp->s1 = s1;
susp->s1_cnt = 0;
susp->s2 = NULL;
susp->s2_cnt = 0;
susp->closure = closure;
susp->previous = 0;
result = sound_create((snd_susp_type)susp, susp->susp.t0, susp->susp.sr, scale_factor);
xlpopn(1);
return result;
}
sound_type snd_trigger(sound_type s1, LVAL closure)
{
sound_type s1_copy;
s1_copy = sound_copy(s1);
return snd_make_trigger(s1_copy, closure);
}