Update of /cvsroot/audacity/lib-src/libnyquist/nyquist/nyqsrc
In directory 23jxhf1.ch3.sourceforge.com:/tmp/cvs-serv8506/nyqsrc
Modified Files:
fft.c fftw.h nyx.c sndfnint.c sndfnint.lsp sndread.c
sndwritepa.c sound.c sound.h
Log Message:
Updating to Nyquist v3.03.
Index: nyx.c
===================================================================
RCS file: /cvsroot/audacity/lib-src/libnyquist/nyquist/nyqsrc/nyx.c,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -d -r1.1 -r1.2
--- nyx.c 29 Jan 2009 18:04:21 -0000 1.1
+++ nyx.c 5 Mar 2009 16:34:00 -0000 1.2
@@ -25,90 +25,480 @@
/* xlisp includes */
#include "switches.h"
#include "xlisp.h"
-#include "term.h"
#include "cext.h"
/* nyquist includes */
#include "sound.h"
+#include "samples.h"
#include "falloc.h"
[...1202 lines suppressed...]
+ }
+ return cvstring(tmp);
+}
+#endif
+
+#ifndef WIN32
+/* xget_user -- get a string identifying the user, for use in file names */
+LVAL xget_user()
+{
+ char *user = getenv("USER");
+ if (!user || !*user) {
+ user = getenv("USERNAME");
+ if (!user || !*user) {
+ errputstr("Warning: could not get user ID, using 'nyquist'\n");
+ user = "nyquist";
+ }
+ }
+ return cvstring(user);
+}
+#endif
Index: fft.c
===================================================================
RCS file: /cvsroot/audacity/lib-src/libnyquist/nyquist/nyqsrc/fft.c,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -d -r1.1 -r1.2
--- fft.c 29 Jan 2009 18:04:21 -0000 1.1
+++ fft.c 5 Mar 2009 16:34:00 -0000 1.2
@@ -1,5 +1,6 @@
/* fft.c -- implement snd_fft */
+#define _USE_MATH_DEFINES 1 /* for Visual C++ to get M_LN2 */
#include <math.h>
#include <stdio.h>
#ifndef mips
@@ -9,7 +10,7 @@
#include "sound.h"
#include "falloc.h"
#include "fft.h"
-#include "fftn.h"
+#include "fftext.h"
/* CHANGE LOG
* --------------------------------------------------------------------
@@ -31,8 +32,8 @@
* extra[3] -> FILLCNT (how many samples in buffer)
* extra[4] -> TERMCNT (how many samples until termination)
* extra[5 .. 5+len-1] -> samples (stored as floats)
- * extra[5+len .. 5+3*len-1] -> real and imag. arrays for fft
- * extra[5+3*len ... 5+4*len-1] -> window coefficients
+ * extra[5+len .. 5+2*len-1] -> array of samples to fft
+ * extra[5+2*len ... 5+3*len-1] -> window coefficients
*
* Termination details:
* Return NIL when the sound terminates.
@@ -87,7 +88,7 @@
LVAL snd_fft(sound_type s, long len, long step, LVAL winval)
{
- long i, maxlen, skip, fillptr;
+ long i, m, maxlen, skip, fillptr;
float *samples;
float *temp_fft;
float *window;
@@ -107,12 +108,12 @@
/* note: any storage required by fft must be allocated here in a
* contiguous block of memory who's size is given by the first long
* in the block. Here, there are 4 more longs after the size, and
- * then room for 4*len floats (assumes that floats and longs take
+ * then room for 3*len floats (assumes that floats and longs take
* equal space).
*
- * The reason for 4*len floats is to provide space for:
+ * The reason for 3*len floats is to provide space for:
* the samples to be transformed (len)
- * the complex FFT result (2*len)
+ * the complex FFT result (len)
* the window coefficients (len)
*
* The reason for this storage restriction is that when a sound is
@@ -121,12 +122,12 @@
* structure (this could be added in sound.c, however, if it's
* really necessary).
*/
- s->extra = (long *) malloc(sizeof(long) * (4 * len + OFFSET));
- s->extra[0] = sizeof(long) * (4 * len + OFFSET);
+ s->extra = (long *) malloc(sizeof(long) * (3 * len + OFFSET));
+ s->extra[0] = sizeof(long) * (3 * len + OFFSET);
s->CNT = s->INDEX = s->FILLCNT = 0;
s->TERMCNT = -1;
maxlen = len;
- window = (float *) &(s->extra[OFFSET + 3 * len]);
+ window = (float *) &(s->extra[OFFSET + 2 * len]);
/* fill the window from w */
if (!w) {
for (i = 0; i < len; i++) *window++ = 1.0F;
@@ -134,12 +135,12 @@
n_samples_from_sound(w, len, window);
}
} else {
- maxlen = ((s->extra[0] / sizeof(long)) - OFFSET) / 4;
+ maxlen = ((s->extra[0] / sizeof(long)) - OFFSET) / 3;
if (maxlen != len) xlfail("len changed from initial value");
}
samples = (float *) &(s->extra[OFFSET]);
temp_fft = samples + len;
- window = temp_fft + 2 * len;
+ window = temp_fft + len;
/* step 1: refill buffer with samples */
fillptr = s->FILLCNT;
while (fillptr < maxlen) {
@@ -174,17 +175,18 @@
*/
for (i = 0; i < len; i++) {
temp_fft[i] = samples[i] * *window++;
- temp_fft[i + len] = 0.0F;
}
/* perform the fft: */
- fftnf(1, (const int *) &len, temp_fft, temp_fft + len, 1, -1.0);
+ m = round(log(len) / M_LN2); /* compute log-base-2(len) */
+ if (!fftInit(m)) rffts(temp_fft, m, 1);
+ else xlfail("FFT initialization error");
+
+ /* move results to Lisp array */
setelement(result, 0, cvflonum(temp_fft[0]));
- for (i = 2; i < len; i += 2) {
- setelement(result, i - 1, cvflonum(temp_fft[i / 2] * 2));
- setelement(result, i, cvflonum(temp_fft[len + (i / 2)] * -2));
+ setelement(result, len - 1, cvflonum(temp_fft[1]));
+ for (i = 2; i < len; i++) {
+ setelement(result, i - 1, cvflonum(temp_fft[i]));
}
- if (len % 2 == 0)
- setelement(result, len - 1, cvflonum(temp_fft[len / 2]));
/* step 3: shift samples by step */
if (step < 0) xlfail("step < 0");
Index: sound.c
===================================================================
RCS file: /cvsroot/audacity/lib-src/libnyquist/nyquist/nyqsrc/sound.c,v
retrieving revision 1.3
retrieving revision 1.4
diff -u -d -r1.3 -r1.4
--- sound.c 1 Mar 2009 21:09:49 -0000 1.3
+++ sound.c 5 Mar 2009 16:34:00 -0000 1.4
@@ -85,7 +85,7 @@
* the nominal pitch (in half steps), the table length, and the sample
* rate, compute the sample number corresponding to the phase. This
* routine makes it easy to initialize the table pointer at the beginning
- * of various oscillator implementations in Fugue. Note that the table
+ * of various oscillator implementations in Nyquist. Note that the table
* may represent several periods, in which case phase 360 is not the same
* as 0. Also note that the phase increment is also computed and returned
* through incr_ptr.
@@ -332,7 +332,7 @@
/* compute the true t0 which corresponds to the time of first sample */
snd->true_t0 -= (n / snd->sr);
/* make caller happy by claiming the sound now starts at exactly t0;
- * this is always true within 0.5 samples as allowed by Fugue. */
+ * this is always true within 0.5 samples as allowed by Nyquist. */
snd->t0 = t0;
/* nyquist_printf("sound_prepend_zeros: snd %p true_t0 %g sr %g n %d\n",
snd, snd->true_t0, snd->sr, n);*/
@@ -506,39 +506,33 @@
void snd_list_unref(snd_list_type list)
{
void (*freefunc)();
- snd_list_type next;
- while (list != zero_snd_list) {
- if (list == NULL) {
+ if (list == NULL || list == zero_snd_list) {
+ if (list == NULL)
nyquist_printf("why did snd_list_unref get %p?\n", list);
- return;
+ return;
+ }
+ list->refcnt--;
+/* nyquist_printf("snd_list_unref "); print_snd_list_type(list);
stdputstr("\n"); */
+ if (list->refcnt == 0) {
+ if (list->block && list->block != zero_block) {
+ /* there is a next snd_list */
+/* stdputstr("["); */
+ sample_block_unref(list->block);
+/* stdputstr("]"); */
+ snd_list_unref(list->u.next);
}
- next = zero_snd_list;
-
- list->refcnt--;
-/* nyquist_printf("snd_list_unref "); print_snd_list_type(list);
stdputstr("\n"); */
- if (list->refcnt == 0) {
- if (list->block && list->block != zero_block) {
- /* there is a next snd_list */
-/* stdputstr("["); */
- sample_block_unref(list->block);
-/* stdputstr("]"); */
- next = list->u.next;
- }
- else if (list->block == NULL) { /* the next thing is the susp */
- /* free suspension structure */
- /* nyquist_printf("freeing s...@%p\n", list->u.susp); */
- freefunc = list->u.susp->free;
- (*freefunc)(list->u.susp);
- }
- /* nyquist_printf("freeing snd_l...@%p\n", list); */
- //DBY
- if (list == list_watch) printf("freeing watched snd_list %p\n",
list);
- //DBY
- ffree_snd_list(list, "snd_list_unref");
+ else if (list->block == NULL) { /* the next thing is the susp */
+ /* free suspension structure */
+ /* nyquist_printf("freeing s...@%p\n", list->u.susp); */
+ freefunc = list->u.susp->free;
+ (*freefunc)(list->u.susp);
}
-
- list = next;
+ /* nyquist_printf("freeing snd_l...@%p\n", list); */
+ //DBY
+ if (list == list_watch) printf("freeing watched snd_list %p\n", list);
+ //DBY
+ ffree_snd_list(list, "snd_list_unref");
}
}
Index: sound.h
===================================================================
RCS file: /cvsroot/audacity/lib-src/libnyquist/nyquist/nyqsrc/sound.h,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -d -r1.1 -r1.2
--- sound.h 29 Jan 2009 18:04:22 -0000 1.1
+++ sound.h 5 Mar 2009 16:34:00 -0000 1.2
@@ -131,7 +131,7 @@
/* used by sndwrite.c for output buffers. This should be
* eliminated:
*/
-#define MAX_SND_CHANNELS 8
+#define MAX_SND_CHANNELS 24
#define max_table_len 100000
/* Set to 4 for debugging block allocation stuff, 1012? for
Index: sndfnint.c
===================================================================
RCS file: /cvsroot/audacity/lib-src/libnyquist/nyquist/nyqsrc/sndfnint.c,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -d -r1.1 -r1.2
--- sndfnint.c 29 Jan 2009 18:04:21 -0000 1.1
+++ sndfnint.c 5 Mar 2009 16:34:00 -0000 1.2
@@ -1,39 +1,39 @@
-/* nyqsrc/sndfnint.c -- interface to nylsf/sndfile.h,
- * nyqsrc/sound.h, nyqsrc/add.h, nyqsrc/avg.h,
- * nyqsrc/compose.h, nyqsrc/convolve.h,
+/* nyqsrc/sndfnint.c -- interface to nyqsrc/sndfmt.h,
+ * nylsf/sndfile.h, nyqsrc/sound.h, nyqsrc/add.h,
+ * nyqsrc/avg.h, nyqsrc/compose.h, nyqsrc/convolve.h,
* nyqsrc/downsample.h, nyqsrc/fft.h, nyqsrc/inverse.h,
* nyqsrc/multiseq.h, nyqsrc/resamp.h, nyqsrc/resampv.h,
* nyqsrc/samples.h, nyqsrc/sndmax.h, nyqsrc/sndread.h,
* nyqsrc/sndseq.h, nyqsrc/sndsliders.h, nyqsrc/sndwrite.h,
- * nyqsrc/yin.h, nyqsrc/trigger.h, nyqsrc/lpanal.h,
- * tran/abs.h, tran/allpoles.h, tran/alpass.h,
- * tran/alpasscv.h, tran/alpassvv.h, tran/amosc.h,
- * tran/areson.h, tran/aresonvc.h, tran/aresoncv.h,
- * tran/aresonvv.h, tran/atone.h, tran/atonev.h,
- * tran/biquadfilt.h, tran/buzz.h, tran/chase.h,
- * tran/clip.h, tran/congen.h, tran/const.h,
- * tran/coterm.h, tran/delaycc.h, tran/delaycv.h,
- * tran/eqbandvvv.h, tran/exp.h, tran/follow.h,
- * tran/fmosc.h, tran/fromobject.h, tran/fromarraystream.h,
- * tran/gate.h, tran/ifft.h, tran/instrclar.h,
- * tran/instrclarall.h, tran/instrclarfreq.h,
- * tran/instrsax.h, tran/instrsaxall.h,
- * tran/instrsaxfreq.h, tran/integrate.h, tran/log.h,
- * tran/lpreson.h, tran/maxv.h, tran/offset.h,
- * tran/oneshot.h, tran/osc.h, tran/partial.h,
- * tran/pluck.h, tran/prod.h, tran/pwl.h,
- * tran/quantize.h, tran/recip.h, tran/reson.h,
- * tran/resonvc.h, tran/resoncv.h, tran/resonvv.h,
- * tran/sampler.h, tran/scale.h, tran/shape.h,
- * tran/sine.h, tran/siosc.h, tran/slope.h, tran/sqrt.h,
- * tran/tapf.h, tran/tapv.h, tran/tone.h, tran/tonev.h,
- * tran/upsample.h, tran/white.h, tran/stkrev.h,
- * tran/stkpitshift.h, tran/stkchorus.h, tran/instrbow.h,
- * tran/instrbowedfreq.h, tran/instrbanded.h,
- * tran/instrmandolin.h, tran/instrsitar.h,
- * tran/instrmodalbar.h, tran/instrflute.h,
- * tran/instrflutefreq.h, tran/instrfluteall.h,
- * tran/fmfb.h, tran/fmfbv.h */
+ * nyqsrc/yin.h, nyqsrc/nyq-osc-server.h, nyqsrc/trigger.h,
+ * nyqsrc/lpanal.h, tran/abs.h, tran/allpoles.h,
+ * tran/alpass.h, tran/alpasscv.h, tran/alpassvv.h,
+ * tran/amosc.h, tran/areson.h, tran/aresonvc.h,
+ * tran/aresoncv.h, tran/aresonvv.h, tran/atone.h,
+ * tran/atonev.h, tran/biquadfilt.h, tran/buzz.h,
+ * tran/chase.h, tran/clip.h, tran/congen.h,
+ * tran/const.h, tran/coterm.h, tran/delaycc.h,
+ * tran/delaycv.h, tran/eqbandvvv.h, tran/exp.h,
+ * tran/follow.h, tran/fmosc.h, tran/fromobject.h,
+ * tran/fromarraystream.h, tran/gate.h, tran/ifft.h,
+ * tran/instrclar.h, tran/instrclarall.h,
+ * tran/instrclarfreq.h, tran/instrsax.h,
+ * tran/instrsaxall.h, tran/instrsaxfreq.h,
+ * tran/integrate.h, tran/log.h, tran/lpreson.h,
+ * tran/maxv.h, tran/offset.h, tran/oneshot.h,
+ * tran/osc.h, tran/partial.h, tran/pluck.h, tran/prod.h,
+ * tran/pwl.h, tran/quantize.h, tran/recip.h,
+ * tran/reson.h, tran/resonvc.h, tran/resoncv.h,
+ * tran/resonvv.h, tran/sampler.h, tran/scale.h,
+ * tran/shape.h, tran/sine.h, tran/siosc.h, tran/slope.h,
+ * tran/sqrt.h, tran/tapf.h, tran/tapv.h, tran/tone.h,
+ * tran/tonev.h, tran/upsample.h, tran/white.h,
+ * tran/stkrev.h, tran/stkpitshift.h, tran/stkchorus.h,
+ * tran/instrbow.h, tran/instrbowedfreq.h,
+ * tran/instrbanded.h, tran/instrmandolin.h,
+ * tran/instrsitar.h, tran/instrmodalbar.h,
+ * tran/instrflute.h, tran/instrflutefreq.h,
+ * tran/instrfluteall.h, tran/fmfb.h, tran/fmfbv.h */
#ifndef mips
#include "stdlib.h"
@@ -51,6 +51,8 @@
extern LVAL RSLT_sym;
+#include "sndfmt.h"
+
#include "sndfile.h"
#include "sound.h"
@@ -772,6 +774,8 @@
}
+#include "nyq-osc-server.h"
+
#include "trigger.h"
/* xlc_snd_trigger -- interface to C routine snd_trigger */
Index: sndread.c
===================================================================
RCS file: /cvsroot/audacity/lib-src/libnyquist/nyquist/nyqsrc/sndread.c,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -d -r1.1 -r1.2
--- sndread.c 29 Jan 2009 18:04:21 -0000 1.1
+++ sndread.c 5 Mar 2009 16:34:00 -0000 1.2
@@ -273,7 +273,11 @@
nyquist_printf("Warning: more than 24 sound files are now open\n");
}
#endif
-
+ /* report info back to caller */
+ if ((susp->sf_info.format & SF_FORMAT_TYPEMASK) != SF_FORMAT_RAW) {
+ *flags = SND_HEAD_CHANNELS | SND_HEAD_MODE | SND_HEAD_BITS |
+ SND_HEAD_SRATE | SND_HEAD_LEN | SND_HEAD_TYPE;
+ }
if (susp->sf_info.channels == 1) {
susp->susp.fetch = read__fetch;
susp->susp.free = read_free;
Index: sndwritepa.c
===================================================================
RCS file: /cvsroot/audacity/lib-src/libnyquist/nyquist/nyqsrc/sndwritepa.c,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -d -r1.1 -r1.2
--- sndwritepa.c 29 Jan 2009 18:04:22 -0000 1.1
+++ sndwritepa.c 5 Mar 2009 16:34:00 -0000 1.2
@@ -37,6 +37,63 @@
#endif
#endif
+/* Previously, Nyquist would wrap samples that
+ * overflowed -- this produces horrible output,
+ * but makes it really easy to detect clipping,
+ * which I found helpful in my own work and good
+ * for students too since the effect is impossible
+ * to ignore. Now that Nyquist is doing IO to
+ * libraries that clip, we're going to artificially
+ * generate the wrapping here. This is floating point
+ * wrapping, so +1.0 does not wrap (it would if it
+ * were an integer since the maximum sample value for
+ * 16-bit data is a bit less than 1.) Since this is extra
+ * overhead, I'm trying to be a bit clever by using
+ * the compare to max_sample to eliminate compares
+ * for clipping in the common case.
+ *
+ * INPUTS: max_sample -- initially 0.0
+ * threshold -- initially 0.0
+ * s -- the value of the current sample
+ * x -- if s has to be wrapped, put the value here
+ */
+#define COMPUTE_MAXIMUM_AND_WRAP(x) \
+ if (s > threshold) { \
+ if (s > max_sample) { \
+ max_sample = s; \
+ threshold = min(1.0, s); \
+ } \
+ if (s > 1.0) { \
+ s = fmod(s + 1.0, 2.0) - 1.0; \
+ (x) = s; \
+ } \
+ } else if (s < -threshold) { \
+ if (s < -max_sample) { \
+ max_sample = -s; \
+ threshold = min(1.0, -s); \
+ } \
+ if (s < -1.0) { \
+ s = -(fmod(-s + 1.0, 2.0) - 1.0); \
+ (x) = s; \
+ } \
+ }
+// the s < -threshold case is tricky:
+// flip the signal, do the wrap, flip again
+// in order to pass positive values to fmod
+
+
+/* When not using PCM encodings, we do not wrap
+ * samples -- therefore float sample formats do
+ * not wrap or clip when written to sound files
+ */
+#define COMPUTE_MAXIMUM() \
+ if (s > max_sample) { \
+ max_sample = s; \
+ } else if (s < -max_sample) { \
+ max_sample = -s; \
+ }
+
+
/* jlh Changed these to the <> format, so it will be sought for in the
include path */
#include <portaudio.h>
@@ -300,6 +357,10 @@
*/
if (filename[0]) {
sndfile = sf_open((char *) filename, SFM_WRITE, &sf_info);
+ if (sndfile) {
+ /* use proper scale factor: 8000 vs 7FFF */
+ sf_command(sndfile, SFC_SET_CLIPPING, NULL, SF_TRUE);
+ }
}
if (play)
@@ -321,6 +382,10 @@
*sr = sf_info.samplerate;
if (filename[0]) {
sndfile = sf_open((char *) filename, SFM_WRITE, &sf_info);
+ if (sndfile) {
+ /* use proper scale factor: 8000 vs 7FFF */
+ sf_command(sndfile, SFC_SET_CLIPPING, NULL, SF_TRUE);
+ }
}
if (play)
play = prepare_audio(play, &sf_info, &audio_stream);
@@ -345,7 +410,7 @@
}
-/* open_for_write -- helper function for sound_save and sound_overwrite */
+/* open_for_write -- helper function for sound_overwrite */
/*
* if the format is RAW, then fill in sf_info according to
* sound sample rate and channels. Otherwise, open the file
@@ -373,10 +438,13 @@
sprintf(error, "snd_overwrite: cannot open file %s", filename);
xlabort(error);
}
+ /* use proper scale factor: 8000 vs 7FFF */
+ sf_command(sndfile, SFC_SET_CLIPPING, NULL, SF_TRUE);
+
frames = round(offset * sf_info->samplerate);
rslt = sf_seek(sndfile, frames, SEEK_SET);
if (rslt < 0) {
- sprintf(error, "snd_overwrite: cannot seek to frame %ld of %s",
+ sprintf(error, "snd_overwrite: cannot seek to frame %lld of %s",
frames, filename);
xlabort(error);
}
@@ -391,7 +459,7 @@
}
if (sf_info->samplerate != srate) {
- sprintf(error, "%s%g%s%g%s",
+ sprintf(error, "%s%g%s%ld%s",
"snd_overwrite: sample rate in sound (",
srate,
") does not match\n sample rate in file (",
@@ -480,6 +548,13 @@
return max_sample;
}
+int is_pcm(SF_INFO *sf_info)
+{
+ long subtype = sf_info->format & SF_FORMAT_SUBMASK;
+ return (subtype == SF_FORMAT_PCM_S8 || subtype == SF_FORMAT_PCM_16 ||
+ subtype == SF_FORMAT_PCM_24 || subtype == SF_FORMAT_PCM_32);
+}
+
sample_type sound_save_sound(LVAL s_as_lval, long n, SF_INFO *sf_info,
SNDFILE *sndfile, float *buf, long *ntotal, PaStream *audio_stream)
@@ -491,6 +566,7 @@
long debug_unit; /* print messages at intervals of this many samples */
long debug_count; /* next point at which to print a message */
sample_type max_sample = 0.0F;
+ sample_type threshold = 0.0F;
/* jlh cvtfn_type cvtfn; */
*ntotal = 0;
@@ -545,12 +621,17 @@
} else {
samps = sampblock->samples;
}
- for (i = 0; i < togo; i++) {
- sample_type s = samps[i];
- if (s > max_sample) max_sample = s;
- else if (s < -max_sample) max_sample = -s;
+ if (is_pcm(sf_info)) {
+ for (i = 0; i < togo; i++) {
+ sample_type s = samps[i];
+ COMPUTE_MAXIMUM_AND_WRAP(samps[i]);
+ }
+ } else {
+ for (i = 0; i < togo; i++) {
+ sample_type s = samps[i];
+ COMPUTE_MAXIMUM();
+ }
}
-
if (sndfile) {
sf_writef_float(sndfile, samps, togo);
}
@@ -584,6 +665,7 @@
long debug_unit; /* print messages at intervals of this many samples */
long debug_count; /* next point at which to print a message */
sample_type max_sample = 0.0F;
+ sample_type threshold = 0.0F;
/* cvtfn_type cvtfn; jlh */
*ntotal = 0;
@@ -684,12 +766,21 @@
if (terminated) break;
float_bufp = (float *) buf;
- for (j = 0; j < togo; j++) {
- for (i = 0; i < chans; i++) {
- float s = (float) (*(state[i].ptr++) * (float)
state[i].scale);
- *float_bufp++ = s;
- if (s > max_sample) max_sample = s;
- else if (s < -max_sample) max_sample = -s;
+ if (is_pcm(sf_info)) {
+ for (j = 0; j < togo; j++) {
+ for (i = 0; i < chans; i++) {
+ float s = (float) (*(state[i].ptr++) * (float)
state[i].scale);
+ COMPUTE_MAXIMUM_AND_WRAP(s);
+ *float_bufp++ = s;
+ }
+ }
+ } else {
+ for (j = 0; j < togo; j++) {
+ for (i = 0; i < chans; i++) {
+ float s = (float) (*(state[i].ptr++) * (float)
state[i].scale);
+ COMPUTE_MAXIMUM();
+ *float_bufp++ = s;
+ }
}
}
/* Here we have interleaved floats. Before converting to the sound
Index: sndfnint.lsp
===================================================================
RCS file: /cvsroot/audacity/lib-src/libnyquist/nyquist/nyqsrc/sndfnint.lsp,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -d -r1.1 -r1.2
--- sndfnint.lsp 29 Jan 2009 18:04:21 -0000 1.1
+++ sndfnint.lsp 5 Mar 2009 16:34:00 -0000 1.2
@@ -1,3 +1,83 @@
+ (setf snd-head-none 0)
+
+ (setf snd-head-AIFF 1)
+
+ (setf snd-head-IRCAM 2)
+
+ (setf snd-head-NeXT 3)
+
+ (setf snd-head-Wave 4)
+
+ (setf snd-head-PAF 5)
+
+ (setf snd-head-SVX 6)
+
+ (setf snd-head-NIST 7)
+
+ (setf snd-head-VOC 8)
+
+ (setf snd-head-W64 9)
+
+ (setf snd-head-MAT4 10)
+
+ (setf snd-head-MAT5 11)
+
+ (setf snd-head-PVF 12)
+
+ (setf snd-head-XI 13)
+
+ (setf snd-head-HTK 14)
+
+ (setf snd-head-SDS 15)
+
+ (setf snd-head-AVR 16)
+
+ (setf snd-head-SD2 17)
+
+ (setf snd-head-FLAC 18)
+
+ (setf snd-head-CAF 19)
+
+ (setf snd-head-raw 20)
+
+ (setf snd-head-channels 1)
+
+ (setf snd-head-mode 2)
+
+ (setf snd-head-bits 4)
+
+ (setf snd-head-srate 8)
+
+ (setf snd-head-dur 16)
+
+ (setf snd-head-latency 32)
+
+ (setf snd-head-type 64)
+
+ (setf snd-mode-adpcm 0)
+
+ (setf snd-mode-pcm 1)
+
+ (setf snd-mode-ulaw 2)
+
+ (setf snd-mode-alaw 3)
+
+ (setf snd-mode-float 4)
+
+ (setf snd-mode-upcm 5)
+
+ (setf snd-mode-unknown 6)
+
+ (setf snd-mode-double 7)
+
+ (setf snd-mode-GSM610 8)
+
+ (setf snd-mode-DWVW 9)
+
+ (setf snd-mode-DPCM 10)
+
+ (setf snd-mode-msadpcm 11)
+
(SETF MAX-STOP-TIME 10E20)
(SETF MIN-START-TIME -10E20)
------------------------------------------------------------------------------
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