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mirror of https://github.com/cookiengineer/audacity synced 2025-06-25 16:48:44 +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

466 lines
17 KiB
C

/* convolve.c -- implements (non-"fast") convolution */
/*
* Note: this code is mostly generated by translate.lsp (see convole.tran
* in the tran directory), but it has been modified by hand to extend the
* stop time to include the "tail" of the convolution beyond the length
* of the first parameter.
*/
/* Original convolve.c modified to do fast convolution. Here are some
* notes:
* The first arg is arbitrary length. The second arg is the impulse
* response, which is converted into a table. The FFT size will be
* limited to 64K, which allows convolution with up to 32K samples.
* For longer impulse responses, we'll have to do convolutions one
* 32K block at a time. I considered just limiting the convolution
* size and handling longer impulse responses in Nyquist XLISP code,
* but that would require taking FFT's of each input block multiple
* times. Here, we save the FFT's and reuse them, which should gain
* a factor of 2 in speed (we still have to inverse FFT each block
* after multiplication, which should take 1/2 the time of doing
* FFT/inverse-FFT on each block).
*
* The fast convolution works like this:
* inputs are x_snd and h_snd.
* Compute the length of h_snd in samples.
* Set fft_size = MAX_FFT_SIZE
* If length <= MAX_FFT_SIZE / 4 then
* set fft_size = (round length to power of 2) * 2
* set N = fft_size/2
* Set h_len = (length rounded up to multiple of fft_size/2) * 2
* Let L = h_len/ fft_size
* Allocate H of h_len floats
* Iterate over i from 0 to L-1:
* Copy ht with zero fill into H[i] of size fft_size,
* where each H[i] of size fft_size is filled with
* fft_size/2 samples (except for the last H[i])
* Compute FFT of H[i] in place (FFT size is fft_size)
* Allocate X of h_len floats. This represents the history
* of x_snd, which is initially all zero, so the FFT, X is all zero
* Allocate output buffers Y and R, each of size fft_size
* Iterate over j (i.e. run this to generate MAX_CONVOLVE_LEN
* samples; then j = (j + 1) mod L.
* Copy 2nd half of R to first half and zero the 2nd half.
* Note: the first time does nothing because R is initially
* filled with zeros
* Copy fft_size/2 samples of x_snd into X[j],
* where X[j] is of size fft_size and filled with
* N samples (except when x_snd terminates)
* Zero fill X[j]
* Compute FFT of X[j] in place.
* Iterate k = 0 to L-1
* Multiply X[(j-k) mod L] by H[k] (result goes into Y).
* Compute IFFT of Y in place. Y is now time domain convolution
* of two blocks of samples.
* Add Y to R.
* Now N samples of R can be output.
* For simplicity, we'll keep processing x_snd input even after x_snd
* terminates. This will avoid special cases where we do not need all
* of X[j] at the end of the convolution.
*
* Length of output is length of x input + length of h
*/
// You can turn on debugging output with: #define D if (1)
#define D if (0)
#define MAX_IR_LEN 4000000 /* maximum impulse response length */
#define MAX_LOG_FFT_SIZE 16 /* maximum fft size for convolution */
//#define MAX_LOG_FFT_SIZE 4 /* maximum fft size for convolution */
#define _USE_MATH_DEFINES 1 /* for Visual C++ to get M_LN2 */
#include <math.h>
#include "stdio.h"
#ifndef mips
#include "stdlib.h"
#endif
#include "xlisp.h"
#include "sound.h"
#include "samples.h"
#include "falloc.h"
#include "cext.h"
#include "fftlib.h"
#include "fftext.h"
#include "convolve.h"
void convolve_free(snd_susp_type a_susp);
typedef struct convolve_susp_struct {
snd_susp_node susp;
int64_t terminate_cnt;
boolean know_end_of_x;
boolean logically_stopped;
sound_type x_snd;
int x_snd_cnt;
sample_block_values_type x_snd_ptr;
sample_type *X; // the FFTs of x_snd
int j; // which block are we processing? 0 <= j < L
sample_type *H; // the FFTs of h_snd
sample_type *Y; // product of X*H where we inverse FFT
int h_snd_len; // true length of h_snd in samples
int N; // length of convolution, FFTs are of size 2*N
int M; // log2 of 2*N, the FFT size
int L; // number of blocks: h_len / (2*N)
sample_type *R; // result buffer where output is summed
sample_type *R_current; // pointer to next sample to output
} convolve_susp_node, *convolve_susp_type;
/*
void h_reverse(sample_type *h, long len)
{
sample_type temp;
int i;
for (i = 0; i < len; i++) {
temp = h[i];
h[i] = h[len - 1];
h[len - 1] = temp;
len--;
}
}
*/
void convolve_s_fetch(snd_susp_type a_susp, snd_list_type snd_list)
{
convolve_susp_type susp = (convolve_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;
sample_type *R = susp->R;
sample_type *R_current;
int N = susp->N;
falloc_sample_block(out, "convolve_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;
/* if we need output samples, generate them here */
D printf("test R_current at offset %td\n", susp->R_current - R);
if (susp->R_current >= R + N) { // true when we output half of R
int i = 0;
int k;
sample_type *Xj = susp->X + susp->j * N * 2;
sample_type *H = susp->H;
sample_type *Y = susp->Y;
int to_copy;
/* Shift R, zero fill: */
memcpy(R, R + N, N * sizeof(*R));
memset(R + N, 0, N * sizeof(*R));
/* Copy N samples of x_snd into Xj and zero fill to size 2N */
D printf("Copying N samples of x_snd into Xj at offset %td\n", Xj - susp->X);
while (i < N) {
if (susp->x_snd_cnt == 0) {
susp_get_samples(x_snd, x_snd_ptr, x_snd_cnt);
if (susp->x_snd->logical_stop_cnt ==
susp->x_snd->current - susp->x_snd_cnt) {
min_cnt(&susp->susp.log_stop_cnt, susp->x_snd,
(snd_susp_type) susp, susp->x_snd_cnt);
}
}
/* This code is not standard. Since we extend the terminate
* count by susp->h_snd_len, the "standard" call to min_cnt()
* results in extending the terminate time forever. Instead,
* we make this code run once only by setting know_end_of_x.
*/
if (!susp->know_end_of_x &&
susp->x_snd_ptr == zero_block->samples) {
susp->terminate_cnt = susp->x_snd->current - susp->x_snd_cnt;
/* extend the output to include impulse response */
susp->terminate_cnt += susp->h_snd_len;
susp->know_end_of_x = TRUE;
}
/* copy no more than the remaining space and no more than
* the amount remaining in the block
*/
to_copy = min(N - i, susp->x_snd_cnt);
memcpy(Xj + i, susp->x_snd_ptr,
to_copy * sizeof(*susp->x_snd_ptr));
susp->x_snd_ptr += to_copy;
susp->x_snd_cnt -= to_copy;
i += to_copy;
}
/* zero fill to size 2N */
memset(Xj + N, 0, N * sizeof(Xj[0]));
D printf("Xj at offset %td: ", Xj - susp->X);
D for (i = 0; i < susp->N * 2; i++) {
printf("%g ", Xj[i]);
}
D printf("\n");
/* Compute FFT of Xj in place */
fftInit(susp->M);
rffts(Xj, susp->M, 1);
/* convolve pairs of blocks and sum into Y */
memset(Y, 0, N * sizeof(*Y)); /* initialize sum to zero */
for (k = 0; k < susp->L; k++) {
/* Multiply Xj by H (result goes into X) */
sample_type *X = susp->X + ((susp->L + susp->j - k) % susp->L) * N * 2;
rspectprod(X, H + k * N * 2, Y, N * 2);
/* Compute IFFT of Y in place */
riffts(Y, susp->M, 1);
/* R += Y */
D printf("Output block %d, X offset %td: ", k, X - susp->X);
for (i = 0; i < 2 * N; i++) {
R[i] += Y[i];
D printf("%g ", Y[i]);
}
D printf("\n");
}
/* now N samples of R can be output */
susp->R_current = R;
D printf("R: ");
D for (i = 0; i < susp->N; i++) {
printf("%g ", R[i]);
}
D printf("\n");
susp->j = (susp->j + 1) % susp->L;
}
/* compute togo, the number of samples to "compute" */
/* can't use more than what's left in R. R_current is
the next sample of R, so what's left is N - (R - R_current) */
R_current = susp->R_current;
togo = (int) min(togo, N - (R_current - R));
/* 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;
}
/* don't run past logical stop time */
if (!susp->logically_stopped &&
susp->susp.log_stop_cnt != UNKNOWN &&
susp->susp.log_stop_cnt <= susp->susp.current + cnt + togo) {
togo = (int) (susp->susp.log_stop_cnt - (susp->susp.current + cnt));
D printf("susp->susp.log_stop_cnt is set to %" PRId64 "\n",
susp->susp.log_stop_cnt);
if (togo == 0) break;
}
n = togo;
out_ptr_reg = out_ptr;
if (n) do { /* the inner sample computation loop */
*out_ptr_reg++ = (sample_type) *R_current++;
} while (--n); /* inner loop */
/* using R_current is a bad idea on RS/6000: */
susp->R_current += togo;
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;
}
} /* convolve_s_fetch */
void convolve_toss_fetch(snd_susp_type a_susp, snd_list_type snd_list)
{
convolve_susp_type susp = (convolve_susp_type) a_susp;
time_type final_time = susp->susp.t0;
long n;
/* fetch samples from x_snd up to final_time for this block of zeros */
while ((ROUNDBIG((final_time - susp->x_snd->t0) * susp->x_snd->sr)) >=
susp->x_snd->current)
susp_get_samples(x_snd, x_snd_ptr, x_snd_cnt);
/* convert to normal processing when we hit final_count */
/* we want each signal positioned at final_time */
n = (long) ROUNDBIG((final_time - susp->x_snd->t0) * susp->x_snd->sr -
(susp->x_snd->current - susp->x_snd_cnt));
susp->x_snd_ptr += n;
susp_took(x_snd_cnt, n);
susp->susp.fetch = susp->susp.keep_fetch;
(*(susp->susp.fetch))(a_susp, snd_list);
}
void convolve_mark(snd_susp_type a_susp)
{
convolve_susp_type susp = (convolve_susp_type) a_susp;
sound_xlmark(susp->x_snd);
}
void convolve_free(snd_susp_type a_susp)
{
convolve_susp_type susp = (convolve_susp_type) a_susp;
free(susp->R);
free(susp->X);
free(susp->Y);
free(susp->H);
sound_unref(susp->x_snd);
ffree_generic(susp, sizeof(convolve_susp_node), "convolve_free");
}
void convolve_print_tree(snd_susp_type a_susp, int n)
{
convolve_susp_type susp = (convolve_susp_type) a_susp;
indent(n);
stdputstr("x_snd:");
sound_print_tree_1(susp->x_snd, n);
}
void fill_with_samples(sample_type *x, sound_type s, long n)
{
/* this is based on snd_fetch in samples.c */
#define CNT extra[1]
#define INDEX extra[2]
#define FIELDS 3
#define SAMPLES list->block->samples
int i;
for (i = 0; i < n; i++) {
if (!s->extra) { /* this is the first call, so fix up s */
s->extra = (int64_t *) malloc(sizeof(s->extra[0]) * FIELDS);
s->extra[0] = sizeof(s->extra[0]) * FIELDS;
s->CNT = s->INDEX = 0;
}
int icnt = (int) s->CNT; /* need this to be int type */
if (icnt == s->INDEX) {
sound_get_next(s, &icnt);
s->CNT = icnt; /* save the count back into s->extra */
s->INDEX = 0;
}
x[i] = s->SAMPLES[s->INDEX++] * s->scale;
}
}
sound_type snd_make_convolve(sound_type x_snd, sound_type h_snd)
{
register convolve_susp_type susp;
rate_type sr = x_snd->sr;
time_type t0 = x_snd->t0;
sample_type scale_factor = 1.0F;
time_type t0_min = t0;
int64_t h_len;
int i;
// assume fft_size is maximal. We fix this later if it is wrong
long fft_size = 1 << MAX_LOG_FFT_SIZE;
if (sr != h_snd->sr) {
xlfail("convolve requires both inputs to have the same sample rates");
}
falloc_generic(susp, convolve_susp_node, "snd_make_convolve");
/* compute the length of h_snd in samples */
h_len = snd_length(h_snd, MAX_IR_LEN + 1);
if (h_len > MAX_IR_LEN) {
char emsg[100];
sprintf(emsg, "convolve maximum impulse length is %d", MAX_IR_LEN);
xlfail(emsg);
}
/* len is the impulse response length;
* the FFT size is at least double that */
if (h_len <= fft_size / 4) {
/* compute log-base-2(h_len): */;
double log_len = log((double) h_len) / M_LN2;
int log_len_int = (int) log_len;
if (log_len_int != log_len) log_len_int++; /* round up to power of 2 */
susp->M = log_len_int + 1;
} else {
susp->M = MAX_LOG_FFT_SIZE;
}
fft_size = (1 << susp->M);
D printf("fft_size %ld\n", fft_size);
susp->N = fft_size / 2;
// round h_len up to multiple of susp->N and multiply by 2
susp->h_snd_len = (int) h_len;
h_len = ((h_len + susp->N - 1) / susp->N) * susp->N * 2;
susp->L = (int) (h_len / fft_size);
// allocate memory
susp->H = (sample_type *) calloc((size_t) h_len, sizeof(susp->H[0]));
if (!susp->H) {
xlfail("memory allocation failure in convolve");
}
for (i = 0; i < susp->L; i++) {
/* copy fft_size/2 samples into each H[i] */
fill_with_samples(susp->H + i * susp->N * 2, h_snd, susp->N);
}
for (i = 0; i < susp->L; i++) {
int j;
float *H = susp->H + i * susp->N * 2;
D printf("H_%d at %td: ", i, H - susp->H);
D for (j = 0; j < susp->N * 2; j++) printf("%g ", H[j]);
D printf("\n");
}
sound_unref(h_snd);
h_snd = NULL;
/* remaining N samples are already zero-filled */
if (fftInit(susp->M)) {
free(susp->H);
xlfail("fft initialization error in convolve");
}
/* take the FFT of each block of the impulse response */
for (i = 0; i < susp->L; i++) {
rffts(susp->H + i * susp->N * 2, susp->M, 1);
}
susp->X = (sample_type *) calloc((size_t) h_len, sizeof(susp->X[0]));
susp->R = (sample_type *) calloc(fft_size, sizeof(susp->R[0]));
susp->Y = (sample_type *) calloc(fft_size, sizeof(susp->Y[0]));
if (!susp->X || !susp->R || !susp->Y) {
free(susp->H);
if (susp->X) free(susp->X);
if (susp->R) free(susp->R);
if (susp->Y) free(susp->Y);
xlfail("memory allocation failed in convolve");
}
susp->R_current = susp->R + susp->N;
susp->susp.fetch = &convolve_s_fetch;
susp->terminate_cnt = UNKNOWN;
susp->know_end_of_x = FALSE;
/* handle unequal start times, if any */
if (t0 < x_snd->t0) sound_prepend_zeros(x_snd, t0);
/* minimum start time over all inputs: */
t0_min = min(x_snd->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 = convolve_toss_fetch;
}
/* initialize susp state */
susp->susp.free = convolve_free;
susp->susp.sr = sr;
susp->susp.t0 = t0;
susp->susp.mark = convolve_mark;
susp->susp.print_tree = convolve_print_tree;
susp->susp.name = "convolve";
susp->logically_stopped = false;
susp->susp.log_stop_cnt = logical_stop_cnt_cvt(x_snd);
susp->susp.current = 0;
susp->x_snd = x_snd;
susp->x_snd_cnt = 0;
susp->j = 0;
return sound_create((snd_susp_type)susp, t0, sr, scale_factor);
}
sound_type snd_convolve(sound_type x_snd, sound_type h_snd)
{
sound_type x_snd_copy = sound_copy(x_snd);
sound_type h_snd_copy = sound_copy(h_snd);
return snd_make_convolve(x_snd_copy, h_snd_copy);
}