ref: 13b765c1d1fc4a58c2bc494534fa4f5c8dab5605
parent: 1176b451913eea3407a08f925d8c23926aaf2915
author: cbagwell <cbagwell>
date: Thu Nov 22 22:42:57 EST 2001
Change LONG/ULONG to st_sample_t/st_size_t
--- a/Changelog
+++ b/Changelog
@@ -28,11 +28,12 @@
the headers did not contain the correct length or encoding type.
o Seperated st.h into 2 files. st.h for libst users and st_i.h for
internal use.
+ o Added new types used extensively by libst: st_sample_t & st_size_t.
+ This allows for more deterministic behavior on 64-bit machines and
+ also allows sox to possibly work with much larger file sizes.
o SoX was some times getting confused and thinking an EOF was an
error case when reading audio files. Removed unneeded aborts
when EOF was OK.
- o Change libst to use two new data types: st_sample_t and st_size_t.
- This allows for better support and maintaince on 64-bit machines.
o Silence effect was broken on stereo files. Also, made thresholds
relative to original bit percision of audio data. When 16-bit audio
is scaled up to 32-bits, a little bit of noise starts to look like a
--- a/TODO
+++ b/TODO
@@ -5,6 +5,8 @@
o Make a global version of MIN/MAX instead of sprinkled min/max/MIN/MAX
+ o Make a global version of clip() instead of sprinked in all files.
+
o Add support to all file handlers to handle 32-bit and float
data types since raw functions can handle them.
--- a/nul.c
+++ b/nul.c
@@ -69,7 +69,7 @@
{nul_t sk = (nul_t) ft->priv;
int done = 0;
- LONG l;
+ st_sample_t l;
for(; done < len; done++) {if (ft->file.eof)
break;
--- a/src/8svx.c
+++ b/src/8svx.c
@@ -359,7 +359,7 @@
#define SVXHEADERSIZE 100
static void svxwriteheader(ft_t ft, st_ssize_t nsamples)
{- LONG formsize = nsamples + SVXHEADERSIZE - 8;
+ int32_t formsize = nsamples + SVXHEADERSIZE - 8;
/* FORM size must be even */
if(formsize % 2 != 0) formsize++;
@@ -369,10 +369,10 @@
st_writes(ft, "8SVX"); /* File type */
st_writes(ft, "VHDR");
- st_writedw(ft, (LONG) 20); /* number of bytes to follow */
+ st_writedw(ft, 20); /* number of bytes to follow */
st_writedw(ft, nsamples); /* samples, 1-shot */
- st_writedw(ft, (LONG) 0); /* samples, repeat */
- st_writedw(ft, (LONG) 0); /* samples per repeat cycle */
+ st_writedw(ft, 0); /* samples, repeat */
+ st_writedw(ft, 0); /* samples per repeat cycle */
st_writew(ft, (int) ft->info.rate); /* samples per second */
st_writeb(ft,1); /* number of octabes */
st_writeb(ft,0); /* data compression (none) */
@@ -379,13 +379,13 @@
st_writew(ft,1); st_writew(ft,0); /* volume */
st_writes(ft, "ANNO");
- st_writedw(ft, (LONG) 32); /* length of block */
+ st_writedw(ft, 32); /* length of block */
st_writes(ft, "File created by Sound Exchange ");
st_writes(ft, "CHAN");
- st_writedw(ft, (LONG) 4);
- st_writedw(ft, (ft->info.channels == 2) ? (LONG) 6 :
- (ft->info.channels == 4) ? (LONG) 15 : (LONG) 2);
+ st_writedw(ft, 4);
+ st_writedw(ft, (ft->info.channels == 2) ? 6 :
+ (ft->info.channels == 4) ? 15 : 2);
st_writes(ft, "BODY");
st_writedw(ft, nsamples); /* samples in file */
--- a/src/adpcm.c
+++ b/src/adpcm.c
@@ -38,7 +38,7 @@
#include "adpcm.h"
typedef struct MsState {- LONG step; /* step size */
+ st_sample_t step; /* step size */
short iCoef[2];
} MsState_t;
@@ -52,7 +52,7 @@
* 1.0 is scaled to 0x100
*/
static const
-LONG stepAdjustTable[] = {+st_sample_t stepAdjustTable[] = {230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
@@ -71,25 +71,17 @@
{ 392,-232}};
-#if 0
-static LONG AdpcmDecode(LONG, MsState_t*, LONG, LONG)__attribute__((regparm(3)));
-#endif
-
-#ifdef __GNUC__
-inline
-#endif
-static LONG AdpcmDecode(c, state, sample1, sample2)
-LONG c, sample1, sample2;
-MsState_t *state;
+inline static st_sample_t AdpcmDecode(st_sample_t c, MsState_t *state,
+ st_sample_t sample1, st_sample_t sample2)
{- LONG vlin;
- LONG sample;
- LONG step;
+ st_sample_t vlin;
+ st_sample_t sample;
+ st_sample_t step;
/** Compute next step value **/
step = state->step;
{- LONG nstep;
+ st_sample_t nstep;
nstep = (stepAdjustTable[c] * step) >> 8;
state->step = (nstep < 16)? 16:nstep;
}
@@ -402,14 +394,14 @@
* samplesPerBlock which would go into a block of size blockAlign
* Yes, it is confusing usage.
*/
-ULONG AdpcmSamplesIn(
- ULONG dataLen,
+st_size_t AdpcmSamplesIn(
+ st_size_t dataLen,
unsigned short chans,
unsigned short blockAlign,
unsigned short samplesPerBlock
)
{- ULONG m, n;
+ st_size_t m, n;
if (samplesPerBlock) {n = (dataLen / blockAlign) * samplesPerBlock;
@@ -428,15 +420,15 @@
/* wSamplesPerBlock = 2*(wBlockAlign - 7*wChannels)/wChannels + 2; */
}
-ULONG AdpcmBytesPerBlock(
+st_size_t AdpcmBytesPerBlock(
unsigned short chans,
unsigned short samplesPerBlock
)
{- ULONG n;
+ st_size_t n;
n = 7*chans; /* header */
if (samplesPerBlock > 2)
- n += (((ULONG)samplesPerBlock-2)*chans + 1)/2;
+ n += (((st_size_t)samplesPerBlock-2)*chans + 1)/2;
return n;
}
--- a/src/adpcm.h
+++ b/src/adpcm.h
@@ -40,8 +40,8 @@
* samplesPerBlock which would go into a block of size blockAlign
* Yes, it is confusing usage.
*/
-extern ULONG AdpcmSamplesIn(
- ULONG dataLen,
+extern st_size_t AdpcmSamplesIn(
+ st_size_t dataLen,
unsigned short chans,
unsigned short blockAlign,
unsigned short samplesPerBlock
@@ -48,11 +48,11 @@
);
/*
- * ULONG AdpcmBytesPerBlock(chans, samplesPerBlock)
+ * st_size_t AdpcmBytesPerBlock(chans, samplesPerBlock)
* return minimum blocksize which would be required
* to encode number of chans with given samplesPerBlock
*/
-extern ULONG AdpcmBytesPerBlock(
+extern st_size_t AdpcmBytesPerBlock(
unsigned short chans,
unsigned short samplesPerBlock
);
--- a/src/aiff.c
+++ b/src/aiff.c
@@ -56,14 +56,14 @@
/* Private data used by writer */
typedef struct aiffpriv {- ULONG nsamples; /* number of 1-channel samples read or written */
- /*Decrements for read increments for write */
- LONG dataStart; /* need to for seeking */
+ st_size_t nsamples; /* number of 1-channel samples read or written */
+ /* Decrements for read increments for write */
+ st_size_t dataStart; /* need to for seeking */
} *aiff_t;
/* forward declarations */
static double read_ieee_extended(ft_t);
-static int aiffwriteheader(ft_t, LONG);
+static int aiffwriteheader(ft_t, st_size_t);
static void write_ieee_extended(ft_t, double);
static double ConvertFromIeeeExtended(unsigned char*);
static void ConvertToIeeeExtended(double, char *);
@@ -71,7 +71,7 @@
static int commentChunk(char **text, char *chunkDescription, ft_t ft);
static void reportInstrument(ft_t ft);
-int st_aiffseek(ft_t ft,st_size_t offset)
+int st_aiffseek(ft_t ft, st_size_t offset)
{aiff_t aiff = (aiff_t ) ft->priv;
@@ -106,7 +106,7 @@
unsigned short nmarks = 0;
unsigned short sustainLoopBegin = 0, sustainLoopEnd = 0,
releaseLoopBegin = 0, releaseLoopEnd = 0;
- LONG seekto = 0L, ssndsize = 0L;
+ st_size_t seekto = 0L, ssndsize = 0L;
char *author;
char *copyright;
char *nametext;
@@ -352,7 +352,7 @@
st_fail_errno(ft,ST_EHDR,"AIFF header specifies nonzero blocksize?!?!");
return(ST_EOF);
}
- while ((LONG) (--offset) >= 0) {+ while (offset-- > 0) {if (st_readb(ft, (unsigned char *)&trash8) == ST_EOF)
{st_fail_errno(ft,errno,"unexpected EOF while skipping AIFF offset");
@@ -574,7 +574,7 @@
st_ssize_t st_aiffread(ft_t ft, st_sample_t *buf, st_ssize_t len)
{aiff_t aiff = (aiff_t ) ft->priv;
- LONG done;
+ st_ssize_t done;
/* just read what's left of SSND chunk */
if (len > aiff->nsamples)
@@ -590,7 +590,7 @@
{char buf[5];
uint32_t chunksize;
- ULONG trash;
+ uint32_t trash;
if (!ft->seekable)
{@@ -607,7 +607,7 @@
buf, chunksize);
if (! strcmp(buf, "MARK") || ! strcmp(buf, "INST"))
st_warn(" You're stripping MIDI/loop info!\n");- while ((LONG) (--chunksize) >= 0)
+ while (chunksize-- > 0)
{if (st_readb(ft, (unsigned char *)&trash) == ST_EOF)
break;
@@ -697,7 +697,7 @@
return(aiffwriteheader(ft, aiff->nsamples / ft->info.channels));
}
-static int aiffwriteheader(ft_t ft, LONG nframes)
+static int aiffwriteheader(ft_t ft, st_size_t nframes)
{int hsize =
8 /*COMM hdr*/ + 18 /*COMM chunk*/ +
@@ -706,7 +706,7 @@
int i;
int padded_comment_size = 0;
int comment_size = 0;
- LONG comment_chunk_size = 0L;
+ st_size_t comment_chunk_size = 0L;
/* MARK and INST chunks */
if (ft->instr.nloops) {@@ -737,7 +737,7 @@
padded_comment_size = ((comment_size % 2) == 0) ?
comment_size : comment_size + 1;
/* one comment, timestamp, marker ID and text count */
- comment_chunk_size = (LONG) (2 + 4 + 2 + 2 + padded_comment_size);
+ comment_chunk_size = (2L + 4 + 2 + 2 + padded_comment_size);
hsize += 8 /* COMT hdr */ + comment_chunk_size;
}
@@ -757,7 +757,7 @@
/* time stamp of comment, Unix knows of time from 1/1/1970,
Apple knows time from 1/1/1904 */
- st_writedw(ft, (LONG) ((LONG) time(NULL)) + 2082844800L);
+ st_writedw(ft, ((int32_t) time(NULL)) + 2082844800L);
/* A marker ID of 0 indicates the comment is not associated
with a marker */
@@ -772,7 +772,7 @@
/* COMM chunk -- describes encoding (and #frames) */
st_writes(ft, "COMM");
- st_writedw(ft, (LONG) 18); /* COMM chunk size */
+ st_writedw(ft, 18); /* COMM chunk size */
st_writew(ft, ft->info.channels); /* nchannels */
st_writedw(ft, nframes); /* number of frames */
st_writew(ft, bits); /* sample width, in bits */
@@ -828,8 +828,8 @@
st_writes(ft, "SSND");
/* chunk size */
st_writedw(ft, 8 + nframes * ft->info.channels * ft->info.size);
- st_writedw(ft, (LONG) 0); /* offset */
- st_writedw(ft, (LONG) 0); /* block size */
+ st_writedw(ft, 0); /* offset */
+ st_writedw(ft, 0); /* block size */
return(ST_SUCCESS);
}
@@ -897,7 +897,7 @@
# define HUGE_VAL HUGE
#endif /*HUGE_VAL*/
-# define FloatToUnsigned(f) ((ULONG)(((LONG)(f - 2147483648.0)) + 2147483647L) + 1)
+# define FloatToUnsigned(f) ((uint32_t)(((int32_t)(f - 2147483648.0)) + 2147483647L) + 1)
static void ConvertToIeeeExtended(double num, char *bytes)
{@@ -904,7 +904,7 @@
int sign;
int expon;
double fMant, fsMant;
- ULONG hiMant, loMant;
+ uint32_t hiMant, loMant;
if (num < 0) {sign = 0x8000;
@@ -990,7 +990,7 @@
# define HUGE_VAL HUGE
#endif /*HUGE_VAL*/
-# define UnsignedToFloat(u) (((double)((LONG)(u - 2147483647L - 1))) + 2147483648.0)
+# define UnsignedToFloat(u) (((double)((int32_t)(u - 2147483647L - 1))) + 2147483648.0)
/****************************************************************
* Extended precision IEEE floating-point conversion routine.
@@ -1000,17 +1000,17 @@
{double f;
int expon;
- ULONG hiMant, loMant;
+ uint32_t hiMant, loMant;
expon = ((bytes[0] & 0x7F) << 8) | (bytes[1] & 0xFF);
- hiMant = ((ULONG)(bytes[2] & 0xFF) << 24)
- | ((ULONG)(bytes[3] & 0xFF) << 16)
- | ((ULONG)(bytes[4] & 0xFF) << 8)
- | ((ULONG)(bytes[5] & 0xFF));
- loMant = ((ULONG)(bytes[6] & 0xFF) << 24)
- | ((ULONG)(bytes[7] & 0xFF) << 16)
- | ((ULONG)(bytes[8] & 0xFF) << 8)
- | ((ULONG)(bytes[9] & 0xFF));
+ hiMant = ((uint32_t)(bytes[2] & 0xFF) << 24)
+ | ((uint32_t)(bytes[3] & 0xFF) << 16)
+ | ((uint32_t)(bytes[4] & 0xFF) << 8)
+ | ((uint32_t)(bytes[5] & 0xFF));
+ loMant = ((uint32_t)(bytes[6] & 0xFF) << 24)
+ | ((uint32_t)(bytes[7] & 0xFF) << 16)
+ | ((uint32_t)(bytes[8] & 0xFF) << 8)
+ | ((uint32_t)(bytes[9] & 0xFF));
if (expon == 0 && hiMant == 0 && loMant == 0) {f = 0;
--- a/src/au.c
+++ b/src/au.c
@@ -50,9 +50,9 @@
/* Private data */
typedef struct aupriv {/* For writer: size in bytes */
- ULONG data_size;
+ st_size_t data_size;
/* For seeking */
- LONG dataStart;
+ st_size_t dataStart;
/* For G72x decoding: */
struct g72x_state state;
int (*dec_routine)();
@@ -61,7 +61,7 @@
int in_bits;
} *au_t;
-static void auwriteheader(ft_t ft, ULONG data_size);
+static void auwriteheader(ft_t ft, st_size_t data_size);
static int st_auencodingandsize(int sun_encoding, char *encoding, char *size)
{@@ -381,11 +381,11 @@
static void auwriteheader(ft_t ft, st_size_t data_size)
{- ULONG magic;
- ULONG hdr_size;
- ULONG encoding;
- ULONG sample_rate;
- ULONG channels;
+ uint32_t magic;
+ uint32_t hdr_size;
+ uint32_t encoding;
+ uint32_t sample_rate;
+ uint32_t channels;
int x;
int comment_size;
--- a/src/avr.c
+++ b/src/avr.c
@@ -262,9 +262,7 @@
return(ST_SUCCESS);
}
-LONG st_avrwrite(ft, buf, nsamp)
-ft_t ft;
-LONG *buf, nsamp;
+st_ssize_t st_avrwrite(ft_t ft, st_sample_t *buf, st_ssize_t nsamp)
{avr_t avr = (avr_t)ft->priv;
@@ -273,10 +271,7 @@
return (st_rawwrite (ft, buf, nsamp));
}
-
-
-int st_avrstopwrite(ft)
-ft_t ft;
+int st_avrstopwrite(ft_t ft)
{avr_t avr = (avr_t)ft->priv;
int rc;
--- a/src/band.c
+++ b/src/band.c
@@ -118,7 +118,7 @@
band_t band = (band_t) effp->priv;
int len, done;
double d;
- LONG l;
+ st_sample_t l;
len = ((*isamp > *osamp) ? *osamp : *isamp);
--- a/src/cdr.c
+++ b/src/cdr.c
@@ -28,7 +28,7 @@
/* Private data for SKEL file */
typedef struct cdrstuff {- LONG samples; /* number of samples written */
+ st_size_t samples; /* number of samples written */
} *cdr_t;
/*
--- a/src/chorus.c
+++ b/src/chorus.c
@@ -76,12 +76,12 @@
int num_chorus;
int modulation[MAX_CHORUS];
int counter;
- LONG phase[MAX_CHORUS];
+ long phase[MAX_CHORUS];
float *chorusbuf;
float in_gain, out_gain;
float delay[MAX_CHORUS], decay[MAX_CHORUS];
float speed[MAX_CHORUS], depth[MAX_CHORUS];
- LONG length[MAX_CHORUS];
+ long length[MAX_CHORUS];
int *lookup_tab[MAX_CHORUS];
int depth_samples[MAX_CHORUS], samples[MAX_CHORUS];
int maxsamples, fade_out;
@@ -260,7 +260,7 @@
int i;
float d_in, d_out;
- LONG out;
+ st_sample_t out;
len = ((*isamp > *osamp) ? *osamp : *isamp);
for(done = 0; done < len; done++) {@@ -274,7 +274,7 @@
chorus->maxsamples] * chorus->decay[i];
/* Adjust the output volume and size to 24 bit */
d_out = d_out * chorus->out_gain;
- out = st_clip24((LONG) d_out);
+ out = st_clip24((st_sample_t) d_out);
*obuf++ = out * 256;
/* Mix decay of delay and input */
chorus->chorusbuf[chorus->counter] = d_in;
@@ -298,7 +298,7 @@
int i;
float d_in, d_out;
- LONG out;
+ st_sample_t out;
done = 0;
while ( ( done < *osamp ) && ( done < chorus->fade_out ) ) {@@ -311,7 +311,7 @@
chorus->maxsamples] * chorus->decay[i];
/* Adjust the output volume and size to 24 bit */
d_out = d_out * chorus->out_gain;
- out = st_clip24((LONG) d_out);
+ out = st_clip24((st_sample_t) d_out);
*obuf++ = out * 256;
/* Mix decay of delay and input */
chorus->chorusbuf[chorus->counter] = d_in;
--- a/src/compand.c
+++ b/src/compand.c
@@ -56,10 +56,10 @@
double *volume; /* Current "volume" of each channel */
double outgain; /* Post processor gain */
double delay; /* Delay to apply before companding */
- LONG *delay_buf; /* Old samples, used for delay processing */
- LONG delay_buf_size; /* Size of delay_buf in samples */
- LONG delay_buf_ptr; /* Index into delay_buf */
- LONG delay_buf_cnt; /* No. of active entries in delay_buf */
+ st_sample_t *delay_buf; /* Old samples, used for delay processing */
+ st_ssize_t delay_buf_size;/* Size of delay_buf in samples */
+ st_ssize_t delay_buf_ptr; /* Index into delay_buf */
+ st_ssize_t delay_buf_cnt; /* No. of active entries in delay_buf */
} *compand_t;
/*
@@ -258,7 +258,7 @@
static void doVolume(double *v, double samp, compand_t l, int chan)
{- double s = samp/(~((LONG)1<<31));
+ double s = samp/(~((st_sample_t)1<<31));
double delta = s - *v;
if (delta > 0.0) /* increase volume according to attack rate */
--- a/src/copy.c
+++ b/src/copy.c
@@ -49,7 +49,7 @@
int done;
done = ((*isamp < *osamp) ? *isamp : *osamp);
- memcpy(obuf, ibuf, done * sizeof(LONG));
+ memcpy(obuf, ibuf, done * sizeof(st_sample_t));
*isamp = *osamp = done;
return (ST_SUCCESS);
}
--- a/src/cvsd.c
+++ b/src/cvsd.c
@@ -34,7 +34,6 @@
/* ---------------------------------------------------------------------- */
-#include <limits.h>
#include <math.h>
#include <string.h>
#include <time.h>
@@ -315,7 +314,7 @@
p->com.v_max = oval;
if (oval < p->com.v_min)
p->com.v_min = oval;
- *buf++ = (oval * ((float)LONG_MAX));
+ *buf++ = (oval * ((float)ST_SAMPLE_MAX));
done++;
}
p->com.phase &= 3;
@@ -359,7 +358,7 @@
memmove(p->c.enc.input_filter+1, p->c.enc.input_filter,
sizeof(p->c.enc.input_filter)-sizeof(float));
p->c.enc.input_filter[0] = (*buf++) /
- ((float)LONG_MAX);
+ ((float)ST_SAMPLE_MAX);
done++;
}
p->com.phase &= 3;
@@ -418,7 +417,7 @@
time_t Unixtime;
unsigned Usender;
unsigned Ureceiver;
- ULONG Length;
+ st_size_t Length;
unsigned Srate;
unsigned Days;
unsigned Custom1;
@@ -432,15 +431,15 @@
/* ---------------------------------------------------------------------- */
/* FIXME: Move these to misc.c */
-static ULONG get32(unsigned char **p)
+static uint32_t get32(unsigned char **p)
{- ULONG val = (((*p)[3]) << 24) | (((*p)[2]) << 16) |
+ uint32_t val = (((*p)[3]) << 24) | (((*p)[2]) << 16) |
(((*p)[1]) << 8) | (**p);
(*p) += 4;
return val;
}
-static unsigned get16(unsigned char **p)
+static uint16_t get16(unsigned char **p)
{unsigned val = (((*p)[1]) << 8) | (**p);
(*p) += 2;
@@ -447,7 +446,7 @@
return val;
}
-static void put32(unsigned char **p, ULONG val)
+static void put32(unsigned char **p, uint32_t val)
{*(*p)++ = val & 0xff;
*(*p)++ = (val >> 8) & 0xff;
@@ -455,7 +454,7 @@
*(*p)++ = (val >> 24) & 0xff;
}
-static void put16(unsigned char **p, unsigned val)
+static void put16(unsigned char **p, int16_t val)
{*(*p)++ = val & 0xff;
*(*p)++ = (val >> 8) & 0xff;
--- a/src/dat.c
+++ b/src/dat.c
@@ -28,7 +28,7 @@
} *dat_t;
/* FIXME: Move this to misc.c */
-static LONG roundoff(double x)
+static st_sample_t roundoff(double x)
{if (x < 0.0) return(x - 0.5);
else return(x + 0.5);
--- a/src/dcshift.c
+++ b/src/dcshift.c
@@ -14,7 +14,6 @@
#include "st_i.h"
#include <math.h> /* exp(), sqrt() */
-#include <limits.h> /* LONG_MAX */
/* type used for computations.
*/
@@ -76,10 +75,13 @@
}
dcs->uselimiter = 1; /* ok, we'll use it */
- /* The following equation is derived so that there is no discontinuity in output amplitudes */
- /* and a LONG_MAX input always maps to a LONG_MAX output when the limiter is activated. */
- /* (NOTE: There **WILL** be a discontinuity in the slope of the output amplitudes when using the limiter.) */
- dcs->limiterthreshhold = LONG_MAX * (ONE - (fabs(dcs->dcshift) - dcs->limitergain));
+ /* The following equation is derived so that there is no
+ * discontinuity in output amplitudes */
+ /* and a ST_SAMPLE_MAX input always maps to a ST_SAMPLE_MAX output
+ * when the limiter is activated. */
+ /* (NOTE: There **WILL** be a discontinuity in the slope of the
+ * output amplitudes when using the limiter.) */
+ dcs->limiterthreshhold = ST_SAMPLE_MAX * (ONE - (fabs(dcs->dcshift) - dcs->limitergain));
}
return ST_SUCCESS;
@@ -116,20 +118,20 @@
* this could be a function on its own, with clip count and report
* handled by eff_t and caller.
*/
-static LONG clip(dcs_t dcs, const DCSHIFT_FLOAT v)
+static st_sample_t clip(dcs_t dcs, const DCSHIFT_FLOAT v)
{
- if (v > LONG_MAX)
+ if (v > ST_SAMPLE_MAX)
{
dcs->clipped++;
- return LONG_MAX;
+ return ST_SAMPLE_MAX;
}
- else if (v < -LONG_MAX)
+ else if (v < -ST_SAMPLE_MAX)
{
dcs->clipped++;
- return -LONG_MAX;
+ return -ST_SAMPLE_MAX;
}
/* else */
- return (LONG) v;
+ return (st_sample_t) v;
}
#ifndef MIN
@@ -147,7 +149,7 @@
register DCSHIFT_FLOAT limitergain = dcs->limitergain;
register DCSHIFT_FLOAT limiterthreshhold = dcs->limiterthreshhold;
register DCSHIFT_FLOAT sample;
- register LONG len;
+ register st_size_t len;
len = MIN(*osamp, *isamp);
@@ -164,17 +166,17 @@
if (sample > limiterthreshhold && dcshift > 0)
{
- sample = (sample - limiterthreshhold) * limitergain / (LONG_MAX - limiterthreshhold) + limiterthreshhold + dcshift;
+ sample = (sample - limiterthreshhold) * limitergain / (ST_SAMPLE_MAX - limiterthreshhold) + limiterthreshhold + dcshift;
dcs->limited++;
}
else if (sample < -limiterthreshhold && dcshift < 0)
{
- sample = (sample + limiterthreshhold) * limitergain / (LONG_MAX - limiterthreshhold) - limiterthreshhold + dcshift;
+ sample = (sample + limiterthreshhold) * limitergain / (ST_SAMPLE_MAX - limiterthreshhold) - limiterthreshhold + dcshift;
dcs->limited++;
}
else
{
- sample = dcshift * LONG_MAX + sample;
+ sample = dcshift * ST_SAMPLE_MAX + sample;
}
*obuf++ = clip(dcs, sample);
@@ -184,7 +186,7 @@
{
/* quite basic, with clipping */
for (;len>0; len--)
- *obuf++ = clip(dcs, dcshift * LONG_MAX + *ibuf++);
+ *obuf++ = clip(dcs, dcshift * ST_SAMPLE_MAX + *ibuf++);
}
return ST_SUCCESS;
}
--- a/src/deemphas.c
+++ b/src/deemphas.c
@@ -103,8 +103,8 @@
/* Private data for deemph file */
typedef struct deemphstuff {- LONG lastin;
- double lastout;
+ st_sample_t lastin;
+ double lastout;
} *deemph_t;
/*
--- a/src/earwax.c
+++ b/src/earwax.c
@@ -32,44 +32,45 @@
/* A stereo fir filter. One side filters as if the signal was from
30 degrees from the ear, the other as if 330 degrees. */
/* 30 330 */
-static const LONG filt[] ={4, -6,- 4, -11,
- -1, -5,
- 3, 3,
- -2, 5,
- -5, 0,
- 9, 1,
- 6, 3,
- -4, -1,
- -5, -3,
- -2, -5,
- -7, 1,
- 6, -7,
- 30, -29,
- 12, -3,
- -11, 4,
- -3, 7,
- -20, 23,
- 2, 0,
- 1, -6,
- -14, -5,
- 15, -18,
- 6, 7,
- 15, -10,
- -14, 22,
- -7, -2,
- -4, 9,
- 6, -12,
- 6, -6,
- 0, -11,
- 0, -5,
- 4, 0};
+static const st_sample_t filt[] =
+{ 4, -6,+ 4, -11,
+ -1, -5,
+ 3, 3,
+ -2, 5,
+ -5, 0,
+ 9, 1,
+ 6, 3,
+ -4, -1,
+ -5, -3,
+ -2, -5,
+ -7, 1,
+ 6, -7,
+ 30, -29,
+ 12, -3,
+ -11, 4,
+ -3, 7,
+ -20, 23,
+ 2, 0,
+ 1, -6,
+ -14, -5,
+ 15, -18,
+ 6, 7,
+ 15, -10,
+ -14, 22,
+ -7, -2,
+ -4, 9,
+ 6, -12,
+ 6, -6,
+ 0, -11,
+ 0, -5,
+ 4, 0};
/* 32 tap stereo FIR filter needs 64 taps */
#define EARWAX_NUMTAPS 64
typedef struct earwaxstuff {- LONG *tap; /* taps are z^-1 delays for the FIR filter */
+ st_sample_t *tap; /* taps are z^-1 delays for the FIR filter */
} *earwax_t;
/*
@@ -102,10 +103,10 @@
}
/* allocate tap memory */
- earwax->tap = (LONG*)malloc( sizeof(LONG) * EARWAX_NUMTAPS );
+ earwax->tap = (st_sample_t*)malloc( sizeof(st_sample_t) * EARWAX_NUMTAPS );
if( !earwax->tap ){ st_fail("earwax: Cannot malloc %d bytes!\n", - sizeof(LONG) * EARWAX_NUMTAPS );
+ sizeof(st_sample_t) * EARWAX_NUMTAPS );
return (ST_EOF);
}
@@ -128,7 +129,7 @@
earwax_t earwax = (earwax_t) effp->priv;
int len, done;
int i;
- LONG output;
+ st_sample_t output;
len = ((*isamp > *osamp) ? *osamp : *isamp);
@@ -158,7 +159,7 @@
{earwax_t earwax = (earwax_t) effp->priv;
int i,j;
- LONG output;
+ st_sample_t output;
for(i = EARWAX_NUMTAPS-1; i >= 0; i--){output = 0;
--- a/src/echo.c
+++ b/src/echo.c
@@ -70,7 +70,7 @@
double *delay_buf;
float in_gain, out_gain;
float delay[MAX_ECHOS], decay[MAX_ECHOS];
- LONG samples[MAX_ECHOS], maxsamples, fade_out;
+ st_ssize_t samples[MAX_ECHOS], maxsamples, fade_out;
} *echo_t;
/* Private data for SKEL file */
@@ -190,7 +190,7 @@
int j;
double d_in, d_out;
- LONG out;
+ st_sample_t out;
len = ((*isamp > *osamp) ? *osamp : *isamp);
for(done = 0; done < len; done++) {@@ -205,7 +205,7 @@
}
/* Adjust the output volume and size to 24 bit */
d_out = d_out * echo->out_gain;
- out = st_clip24((LONG) d_out);
+ out = st_clip24((st_sample_t) d_out);
*obuf++ = out * 256;
/* Store input in delay buffer */
echo->delay_buf[echo->counter] = d_in;
@@ -223,7 +223,7 @@
{echo_t echo = (echo_t) effp->priv;
double d_in, d_out;
- LONG out;
+ st_sample_t out;
int j;
long done;
@@ -239,7 +239,7 @@
}
/* Adjust the output volume and size to 24 bit */
d_out = d_out * echo->out_gain;
- out = st_clip24((LONG) d_out);
+ out = st_clip24((st_sample_t) d_out);
*obuf++ = out * 256;
/* Store input in delay buffer */
echo->delay_buf[echo->counter] = d_in;
--- a/src/echos.c
+++ b/src/echos.c
@@ -61,7 +61,7 @@
double *delay_buf;
float in_gain, out_gain;
float delay[MAX_ECHOS], decay[MAX_ECHOS];
- LONG samples[MAX_ECHOS], pointer[MAX_ECHOS], sumsamples;
+ st_ssize_t samples[MAX_ECHOS], pointer[MAX_ECHOS], sumsamples;
} *echos_t;
/* Private data for SKEL file */
@@ -182,7 +182,7 @@
int j;
double d_in, d_out;
- LONG out;
+ st_sample_t out;
len = ((*isamp > *osamp) ? *osamp : *isamp);
for(done = 0; done < len; done++) {@@ -195,7 +195,7 @@
}
/* Adjust the output volume and size to 24 bit */
d_out = d_out * echos->out_gain;
- out = st_clip24((LONG) d_out);
+ out = st_clip24((st_sample_t) d_out);
*obuf++ = out * 256;
/* Mix decay of delays and input */
for ( j = 0; j < echos->num_delays; j++ ) {@@ -221,7 +221,7 @@
{echos_t echos = (echos_t) effp->priv;
double d_in, d_out;
- LONG out;
+ st_sample_t out;
int j;
long done;
@@ -235,7 +235,7 @@
}
/* Adjust the output volume and size to 24 bit */
d_out = d_out * echos->out_gain;
- out = st_clip24((LONG) d_out);
+ out = st_clip24((st_sample_t) d_out);
*obuf++ = out * 256;
/* Mix decay of delays and input */
for ( j = 0; j < echos->num_delays; j++ ) {--- a/src/fade.c
+++ b/src/fade.c
@@ -27,7 +27,7 @@
/* Private data for fade file */
typedef struct fadestuff
{ /* These are measured as samples */- ULONG in_start, in_stop, out_start, out_stop, samplesdone;
+ st_size_t in_start, in_stop, out_start, out_stop, samplesdone;
char *in_stop_str, *out_start_str, *out_stop_str;
char in_fadetype, out_fadetype;
int endpadwarned;
@@ -36,7 +36,7 @@
#define FADE_USAGE "Usage: fade [ type ] fade-in-length [ stop-time [ fade-out-length ] ]\nTimes in seconds.\nFade type one of q, h, t, l or p.\n"
/* prototypes */
-static double fade_gain(ULONG index, ULONG range, char fadetype);
+static double fade_gain(st_size_t index, st_size_t range, char fadetype);
/*
* Process options
@@ -216,7 +216,7 @@
fade_t fade = (fade_t) effp->priv;
/* len is total samples, chcnt counts channels */
int len = 0, chcnt = 0, t_output = 0;
- LONG t_ibuf;
+ st_sample_t t_ibuf;
len = ((*isamp > *osamp) ? *osamp : *isamp);
@@ -337,7 +337,7 @@
/* Function returns gain value 0.0 - 1.0 according index / range ratio
* and -1.0 if type is invalid
* todo: to optimize performance calculate gain every now and then and interpolate */
-static double fade_gain(ULONG index, ULONG range, char type)
+static double fade_gain(st_size_t index, st_size_t range, char type)
{double retval = 0.0, findex = 0.0;
--- a/src/filter.c
+++ b/src/filter.c
@@ -39,22 +39,22 @@
/* Private data for Lerp via LCM file */
typedef struct filterstuff {- LONG rate;
- LONG freq0; /* low corner freq */
- LONG freq1; /* high corner freq */
- double beta; /* >2 is kaiser window beta, <=2 selects nuttall window */
- LONG Nwin;
- Float *Fp; /* [Xh+1] Filter coefficients */
- LONG Xh; /* number of past/future samples needed by filter */
- LONG Xt; /* target to enter new data into X */
- Float *X, *Y; /* I/O buffers */
+ st_rate_t rate;
+ st_sample_t freq0;/* low corner freq */
+ st_sample_t freq1;/* high corner freq */
+ double beta;/* >2 is kaiser window beta, <=2 selects nuttall window */
+ long Nwin;
+ Float *Fp;/* [Xh+1] Filter coefficients */
+ long Xh;/* number of past/future samples needed by filter */
+ long Xt;/* target to enter new data into X */
+ Float *X, *Y;/* I/O buffers */
} *filter_t;
/* makeFilter() declared in resample.c */
extern int
-makeFilter(Float Fp[], LONG Nwing, double Froll, double Beta, LONG Num, int Normalize);
+makeFilter(Float Fp[], long Nwing, double Froll, double Beta, long Num, int Normalize);
-static void FiltWin(filter_t f, LONG Nx);
+static void FiltWin(filter_t f, long Nx);
/*
* Process options
@@ -86,7 +86,7 @@
return (ST_EOF);
}
- if ((n >= 2) && !sscanf(argv[1], "%d", &f->Nwin))
+ if ((n >= 2) && !sscanf(argv[1], "%ld", &f->Nwin))
{ st_fail("Usage: filter low-high [ windowlength ]");return (ST_EOF);
@@ -113,7 +113,7 @@
{filter_t f = (filter_t) effp->priv;
Float *Fp0, *Fp1;
- LONG Xh0, Xh1, Xh;
+ long Xh0, Xh1, Xh;
int i;
f->rate = effp->ininfo.rate;
@@ -191,7 +191,7 @@
st_size_t *isamp, st_size_t *osamp)
{filter_t f = (filter_t) effp->priv;
- LONG i, Nx, Nproc;
+ long i, Nx, Nproc;
/* constrain amount we actually process */
/* fprintf(stderr,"Xh %d, Xt %d, isamp %d, ",f->Xh, f->Xt, *isamp);fflush(stderr); */
@@ -242,7 +242,8 @@
int st_filter_drain(eff_t effp, st_sample_t *obuf, st_size_t *osamp)
{filter_t f = (filter_t) effp->priv;
- LONG isamp_res, *Obuf, osamp_res;
+ long isamp_res, osamp_res;
+ st_sample_t *Obuf;
/* fprintf(stderr,"Xh %d, Xt %d <--- DRAIN\n",f->Xh, f->Xt); */
@@ -251,7 +252,7 @@
osamp_res = *osamp;
Obuf = obuf;
while (isamp_res>0 && osamp_res>0) {- LONG Isamp, Osamp;
+ st_sample_t Isamp, Osamp;
Isamp = isamp_res;
Osamp = osamp_res;
st_filter_flow(effp, NULL, Obuf, &Isamp, &Osamp);
@@ -281,7 +282,7 @@
return (ST_SUCCESS);
}
-static double jprod(const Float *Fp, const Float *Xp, LONG ct)
+static double jprod(const Float *Fp, const Float *Xp, long ct)
{const Float *fp, *xp, *xq;
double v = 0;
@@ -297,7 +298,7 @@
return v;
}
-static void FiltWin(filter_t f, LONG Nx)
+static void FiltWin(filter_t f, long Nx)
{Float *Y;
Float *X, *Xend;
--- a/src/flanger.c
+++ b/src/flanger.c
@@ -69,9 +69,9 @@
float in_gain, out_gain;
float delay, decay;
float speed;
- LONG length;
+ st_size_t length;
int *lookup_tab;
- LONG maxsamples, fade_out;
+ st_size_t maxsamples, fade_out;
} *flanger_t;
/* Private data for SKEL file */
@@ -212,7 +212,7 @@
int len, done;
double d_in, d_out;
- LONG out;
+ st_sample_t out;
len = ((*isamp > *osamp) ? *osamp : *isamp);
for(done = 0; done < len; done++) {@@ -225,7 +225,7 @@
flanger->maxsamples] * flanger->decay;
/* Adjust the output volume and size to 24 bit */
d_out = d_out * flanger->out_gain;
- out = st_clip24((LONG) d_out);
+ out = st_clip24((st_sample_t) d_out);
*obuf++ = out * 256;
/* Mix decay of delay and input */
flanger->flangerbuf[flanger->counter] = d_in;
@@ -246,7 +246,7 @@
int done;
double d_in, d_out;
- LONG out;
+ st_sample_t out;
done = 0;
while ( ( done < *osamp ) && ( done < flanger->fade_out ) ) {@@ -258,7 +258,7 @@
flanger->maxsamples] * flanger->decay;
/* Adjust the output volume and size to 24 bit */
d_out = d_out * flanger->out_gain;
- out = st_clip24((LONG) d_out);
+ out = st_clip24((st_sample_t) d_out);
*obuf++ = out * 256;
/* Mix decay of delay and input */
flanger->flangerbuf[flanger->counter] = d_in;
--- a/src/g72x.h
+++ b/src/g72x.h
@@ -47,7 +47,7 @@
* included in this Recommendation.
*/
struct g72x_state {- LONG yl; /* Locked or steady state step size multiplier. */
+ long yl; /* Locked or steady state step size multiplier. */
short yu; /* Unlocked or non-steady state step size multiplier. */
short dms; /* Short term energy estimate. */
short dml; /* Long term energy estimate. */
--- a/src/hcom.c
+++ b/src/hcom.c
@@ -30,7 +30,7 @@
/* Dictionary entry for Huffman (de)compression */
typedef struct {- LONG frequ;
+ long frequ;
short dict_leftson;
short dict_rightson;
} dictent;
@@ -39,11 +39,11 @@
struct readpriv {/* Static data from the header */
dictent *dictionary;
- LONG checksum;
+ int32_t checksum;
int deltacompression;
/* Engine state */
- LONG huffcount;
- LONG cksum;
+ long huffcount;
+ long cksum;
int dictentry;
int nrbits;
uint32_t current;
@@ -310,8 +310,8 @@
st_ssize_t st_hcomwrite(ft_t ft, st_sample_t *buf, st_ssize_t len)
{register struct writepriv *p = (struct writepriv *) ft->priv;
- LONG datum;
- LONG save_len = len;
+ st_sample_t datum;
+ st_ssize_t save_len = len;
if (len == 0)
return (0);
@@ -348,9 +348,9 @@
static dictent dictionary[511];
static dictent *de;
-static LONG codes[256];
-static LONG codesize[256];
-static LONG checksum;
+static long codes[256];
+static long codesize[256];
+static int32_t checksum;
static void makecodes(int e, int c, int s, int b)
{@@ -363,11 +363,11 @@
}
}
-static LONG curword;
static int nbits;
+static int32_t curword;
/* FIXME: Place in misc.c */
-static void putlong(unsigned char *c, LONG v)
+static void putlong(unsigned char *c, int32_t v)
{*c++ = (v >> 24) & 0xff;
*c++ = (v >> 16) & 0xff;
@@ -384,8 +384,9 @@
static void putcode(unsigned char c, unsigned char **df)
{-LONG code, size;
+long code, size;
int i;
+
code = codes[c];
size = codesize[c];
for(i = 0; i < size; i++) {@@ -403,9 +404,9 @@
}
}
-static int compress(unsigned char **df, LONG *dl, float fr)
+static int compress(unsigned char **df, int32_t *dl, float fr)
{- LONG samplerate;
+ int32_t samplerate;
unsigned char *datafork = *df;
unsigned char *ddf;
short dictsize;
@@ -503,7 +504,7 @@
putlong(datafork + 4, *dl);
putlong(datafork + 8, checksum);
putlong(datafork + 12, 1L);
- samplerate = 22050 / (LONG)fr;
+ samplerate = 22050 / (int32_t)fr;
putlong(datafork + 16, samplerate);
putshort(datafork + 20, dictsize);
*df = datafork; /* reassign passed pointer to new datafork */
@@ -526,7 +527,7 @@
{register struct writepriv *p = (struct writepriv *) ft->priv;
unsigned char *compressed_data = p->data;
- LONG compressed_len = p->pos;
+ uint32_t compressed_len = p->pos;
int rc;
/* Compress it all at once */
@@ -543,8 +544,8 @@
padbytes(ft, 65-3);
st_writes(ft, "FSSD");
padbytes(ft, 83-69);
- st_writedw(ft, (ULONG) compressed_len); /* compressed_data size */
- st_writedw(ft, (ULONG) 0); /* rsrc size */
+ st_writedw(ft, (uint32_t) compressed_len); /* compressed_data size */
+ st_writedw(ft, (uint32_t) 0); /* rsrc size */
padbytes(ft, 128 - 91);
if (ferror(ft->fp))
{--- a/src/highp.c
+++ b/src/highp.c
@@ -84,7 +84,7 @@
highp_t highp = (highp_t) effp->priv;
int len, done;
double d;
- LONG l;
+ st_sample_t l;
len = ((*isamp > *osamp) ? *osamp : *isamp);
--- a/src/ima_rw.c
+++ b/src/ima_rw.c
@@ -335,14 +335,14 @@
* samplesPerBlock which would go into a block of size blockAlign
* Yes, it is confusing.
*/
-ULONG ImaSamplesIn(
- ULONG dataLen,
+st_size_t ImaSamplesIn(
+ st_size_t dataLen,
unsigned short chans,
unsigned short blockAlign,
unsigned short samplesPerBlock
)
{- ULONG m, n;
+ st_size_t m, n;
if (samplesPerBlock) {n = (dataLen / blockAlign) * samplesPerBlock;
@@ -363,22 +363,22 @@
}
/*
- * ULONG ImaBytesPerBlock(chans, samplesPerBlock)
+ * st_size_t ImaBytesPerBlock(chans, samplesPerBlock)
* return minimum blocksize which would be required
* to encode number of chans with given samplesPerBlock
*/
-ULONG ImaBytesPerBlock(
+st_size_t ImaBytesPerBlock(
unsigned short chans,
unsigned short samplesPerBlock
)
{- ULONG n;
+ st_size_t n;
/* per channel, ima has blocks of len 4, the 1st has 1st sample, the others
* up to 8 samples per block,
* so number of later blocks is (nsamp-1 + 7)/8, total blocks/chan is
* (nsamp-1+7)/8 + 1 = (nsamp+14)/8
*/
- n = ((ULONG)samplesPerBlock + 14)/8 * 4 * chans;
+ n = ((st_size_t)samplesPerBlock + 14)/8 * 4 * chans;
return n;
}
--- a/src/ima_rw.h
+++ b/src/ima_rw.h
@@ -77,8 +77,8 @@
* samplesPerBlock which would go into a block of size blockAlign
* Yes, it is confusing usage.
*/
-extern ULONG ImaSamplesIn(
- ULONG dataLen,
+extern st_size_t ImaSamplesIn(
+ st_size_t dataLen,
unsigned short chans,
unsigned short blockAlign,
unsigned short samplesPerBlock
@@ -85,11 +85,11 @@
);
/*
- * ULONG ImaBytesPerBlock(chans, samplesPerBlock)
+ * st_size_t ImaBytesPerBlock(chans, samplesPerBlock)
* return minimum blocksize which would be required
* to encode number of chans with given samplesPerBlock
*/
-extern ULONG ImaBytesPerBlock(
+extern st_size_t ImaBytesPerBlock(
unsigned short chans,
unsigned short samplesPerBlock
);
--- a/src/lowp.c
+++ b/src/lowp.c
@@ -80,7 +80,7 @@
lowp_t lowp = (lowp_t) effp->priv;
int len, done;
double d;
- LONG l;
+ st_sample_t l;
len = ((*isamp > *osamp) ? *osamp : *isamp);
--- a/src/mask.c
+++ b/src/mask.c
@@ -49,8 +49,8 @@
{int len, done;
- LONG l;
- LONG tri16; /* 16 signed bits of triangular noise */
+ st_sample_t l;
+ st_sample_t tri16; /* 16 signed bits of triangular noise */
len = ((*isamp > *osamp) ? *osamp : *isamp);
switch (effp->outinfo.encoding) {--- a/src/maud.c
+++ b/src/maud.c
@@ -323,8 +323,8 @@
st_writes(ft, "MAUD"); /* File type */
st_writes(ft, "MHDR");
- st_writedw(ft, (LONG) 8*4); /* number of bytes to follow */
- st_writedw(ft, (LONG) (p->nsamples )); /* number of samples stored in MDAT */
+ st_writedw(ft, 8*4); /* number of bytes to follow */
+ st_writedw(ft, p->nsamples); /* number of samples stored in MDAT */
switch (ft->info.encoding) {@@ -346,7 +346,7 @@
}
- st_writedw(ft, (LONG) ft->info.rate); /* clock source frequency */
+ st_writedw(ft, ft->info.rate); /* clock source frequency */
st_writew(ft, (int) 1); /* clock devide */
if (ft->info.channels == 1) {@@ -375,12 +375,12 @@
}
- st_writedw(ft, (LONG) 0); /* reserved */
- st_writedw(ft, (LONG) 0); /* reserved */
- st_writedw(ft, (LONG) 0); /* reserved */
+ st_writedw(ft, 0); /* reserved */
+ st_writedw(ft, 0); /* reserved */
+ st_writedw(ft, 0); /* reserved */
st_writes(ft, "ANNO");
- st_writedw(ft, (LONG) 30); /* length of block */
+ st_writedw(ft, 30); /* length of block */
st_writes(ft, "file create by Sound eXchange ");
st_writes(ft, "MDAT");
--- a/src/misc.c
+++ b/src/misc.c
@@ -312,7 +312,7 @@
st_sample_t st_lcm(st_sample_t a, st_sample_t b)
{- /* parenthesize this way to avoid LONG overflow in the product term */
+ /* parenthesize this way to avoid st_sample_t overflow in product term */
return a * (b / st_gcd(a, b));
}
@@ -466,5 +466,5 @@
fstat(fileno(ft->fp), &st);
- return (LONG)st.st_size;
+ return (st_size_t)st.st_size;
}
--- a/src/pan.c
+++ b/src/pan.c
@@ -18,9 +18,6 @@
#include "st_i.h"
-/* should be taken care in st.h? */
-#include <limits.h> /* LONG_MAX */
-
/* type for computations.
float mantissa is okay for 8 or 16 bits signals.
maybe a little bit short for 24 bits.
@@ -100,20 +97,20 @@
* Okay, it might be quite slow to have such a function for so small
* a task. Hopefully the function can be inlined by the compiler?
*/
-static LONG clip(pan_t pan, PAN_FLOAT value)
+static st_sample_t clip(pan_t pan, PAN_FLOAT value)
{- if (value < -LONG_MAX)
+ if (value < -ST_SAMPLE_MAX)
{pan->clipped++;
- return -LONG_MAX;
+ return -ST_SAMPLE_MAX;
}
- else if (value > LONG_MAX)
+ else if (value > ST_SAMPLE_MAX)
{pan->clipped++;
- return LONG_MAX;
+ return ST_SAMPLE_MAX;
} /* else */
- return (LONG) value;
+ return (st_sample_t) value;
}
#ifndef MIN
@@ -131,7 +128,7 @@
st_size_t *isamp, st_size_t *osamp)
{pan_t pan = (pan_t) effp->priv;
- register LONG len;
+ register st_size_t len;
register int done, ich, och;
register PAN_FLOAT left, right, dir, hdir;
--- a/src/phaser.c
+++ b/src/phaser.c
@@ -71,9 +71,9 @@
float in_gain, out_gain;
float delay, decay;
float speed;
- LONG length;
+ st_size_t length;
int *lookup_tab;
- LONG maxsamples, fade_out;
+ st_size_t maxsamples, fade_out;
} *phaser_t;
/*
@@ -199,7 +199,7 @@
int len, done;
double d_in, d_out;
- LONG out;
+ st_sample_t out;
len = ((*isamp > *osamp) ? *osamp : *isamp);
for(done = 0; done < len; done++) {@@ -212,7 +212,7 @@
phaser->maxsamples] * phaser->decay * -1.0;
/* Adjust the output volume and size to 24 bit */
d_out = d_in * phaser->out_gain;
- out = st_clip24((LONG) d_out);
+ out = st_clip24((st_sample_t) d_out);
*obuf++ = out * 256;
/* Mix decay of delay and input */
phaser->phaserbuf[phaser->counter] = d_in;
@@ -233,7 +233,7 @@
int done;
double d_in, d_out;
- LONG out;
+ st_sample_t out;
done = 0;
while ( ( done < *osamp ) && ( done < phaser->fade_out ) ) {@@ -245,7 +245,7 @@
phaser->maxsamples] * phaser->decay * -1.0;
/* Adjust the output volume and size to 24 bit */
d_out = d_in * phaser->out_gain;
- out = st_clip24((LONG) d_out);
+ out = st_clip24((st_sample_t) d_out);
*obuf++ = out * 256;
/* Mix decay of delay and input */
phaser->phaserbuf[phaser->counter] = d_in;
--- a/src/pitch.c
+++ b/src/pitch.c
@@ -41,7 +41,6 @@
#include <string.h> /* memcpy() */
#include <math.h> /* cos(), pow() */
-#include <limits.h> /* LONG_MAX */
#ifndef MIN
#define MIN(a,b) (((a)<(b))?(a):(b))
@@ -132,9 +131,9 @@
int iacc; /* part of acc already output */
- LONG size; /* size of buffer for processing chunks. */
+ st_size_t size; /* size of buffer for processing chunks. */
int index; /* index of next empty input item. */
- LONG * buf; /* bufferize input */
+ st_sample_t *buf; /* bufferize input */
pitch_state_t state; /* buffer management status. */
@@ -202,7 +201,7 @@
*/
static void interpolation(
pitch_t pitch,
- LONG * ibuf, int ilen,
+ st_sample_t *ibuf, int ilen,
PITCH_FLOAT * out, int olen,
PITCH_FLOAT rate) /* signed */
{@@ -276,20 +275,20 @@
pitch->acc[i] += pitch->fade[pitch->step-i-1]*pitch->tmp[i];
}
-static LONG clip(pitch_t pitch, PITCH_FLOAT v)
+static st_sample_t clip(pitch_t pitch, PITCH_FLOAT v)
{- if (v < -LONG_MAX)
+ if (v < -ST_SAMPLE_MAX)
{pitch->clipped++;
- return -LONG_MAX;
+ return -ST_SAMPLE_MAX;
}
- else if (v > LONG_MAX)
+ else if (v > ST_SAMPLE_MAX)
{pitch->clipped++;
- return LONG_MAX;
+ return ST_SAMPLE_MAX;
}
else
- return (LONG) v;
+ return (st_sample_t) v;
}
/*
@@ -427,7 +426,7 @@
pitch->fade = (PITCH_FLOAT *) malloc(pitch->step*sizeof(PITCH_FLOAT));
pitch->tmp = (PITCH_FLOAT *) malloc(pitch->step*sizeof(PITCH_FLOAT));
pitch->acc = (PITCH_FLOAT *) malloc(pitch->step*sizeof(PITCH_FLOAT));
- pitch->buf = (LONG *) malloc(pitch->size*sizeof(LONG));
+ pitch->buf = (st_sample_t *) malloc(pitch->size*sizeof(st_sample_t));
if (!pitch->fade || !pitch->tmp || !pitch->acc || !pitch->buf)
{@@ -519,7 +518,7 @@
{register int tocopy = MIN(pitch->size-pitch->index, len);
- memcpy(pitch->buf+pitch->index, ibuf+iindex, tocopy*sizeof(LONG));
+ memcpy(pitch->buf+pitch->index, ibuf+iindex, tocopy*sizeof(st_sample_t));
len -= tocopy;
pitch->index += tocopy;
--- a/src/polyphas.c
+++ b/src/polyphas.c
@@ -31,7 +31,6 @@
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
-#include <limits.h>
#include <string.h>
#include "st_i.h"
@@ -51,11 +50,11 @@
typedef struct polyphase {- ULONG lcmrate; /* least common multiple of rates */
- ULONG inskip, outskip; /* LCM increments for I & O rates */
+ st_rate_t lcmrate; /* least common multiple of rates */
+ st_rate_t inskip, outskip; /* LCM increments for I & O rates */
double Factor; /* out_rate/in_rate */
- ULONG total; /* number of filter stages */
- ULONG oskip; /* output samples to skip at start*/
+ ulong total; /* number of filter stages */
+ st_size_t oskip; /* output samples to skip at start*/
double inpipe; /* output samples 'in the pipe' */
polystage *stage[MF]; /* array of pointers to polystage structs */
@@ -201,8 +200,8 @@
for (k=ct; k>1; ) {int tmp;
- ULONG j;
- j = (ULONG)(rand()%32768L) + ((ULONG)(rand()%32768L)<<13); /* reasonably big */
+ unsigned long j;
+ j = (rand()%32768L) + ((rand()%32768L)<<13); /* reasonably big */
j = j % k; /* non-negative! */
k--;
if (j != k) {@@ -387,7 +386,7 @@
poly_t rate = (poly_t) effp->priv;
static int l1[MF], l2[MF];
double skip = 0;
- int total, size, uprate;
+ int total, size, uprate;
int k;
if (effp->ininfo.rate == effp->outinfo.rate)
@@ -398,7 +397,8 @@
st_initrand();
- rate->lcmrate = st_lcm((LONG)effp->ininfo.rate, (LONG)effp->outinfo.rate);
+ rate->lcmrate = st_lcm((st_sample_t)effp->ininfo.rate,
+ (st_sample_t)effp->outinfo.rate);
/* Cursory check for LCM overflow.
* If both rate are below 65k, there should be no problem.
@@ -535,14 +535,12 @@
}
-static Float FLONG_MAX = LONG_MAX;
-
-static LONG clipfloat(Float sample)
+static st_sample_t clipfloat(Float sample)
{- if (sample > FLONG_MAX)
- return LONG_MAX;
- if (sample < -FLONG_MAX)
- return -LONG_MAX;
+ if (sample > ST_SAMPLE_MAX)
+ return ST_SAMPLE_MAX;
+ if (sample < -ST_SAMPLE_MAX)
+ return -ST_SAMPLE_MAX;
return sample;
}
@@ -606,9 +604,9 @@
}
{- LONG *q;
- LONG out_size;
- LONG oskip;
+ st_sample_t *q;
+ st_size_t out_size;
+ st_size_t oskip;
Float *out_buf;
int k;
@@ -646,7 +644,7 @@
*/
int st_poly_drain(eff_t effp, st_sample_t *obuf, st_size_t *osamp)
{- LONG in_size;
+ st_size_t in_size;
/* Call "flow" with NULL input. */
st_poly_flow(effp, NULL, obuf, &in_size, osamp);
return (ST_SUCCESS);
--- a/src/rate.c
+++ b/src/rate.c
@@ -39,15 +39,15 @@
/* Private data */
typedef struct ratestuff {- ULONG opos_frac; /* fractional position of the output stream in input stream unit */
- ULONG opos;
+ unsigned long opos_frac; /* fractional position of the output stream in input stream unit */
+ unsigned long opos;
- ULONG opos_inc_frac; /* fractional position increment in the output stream */
- ULONG opos_inc;
+ unsigned long opos_inc_frac; /* fractional position increment in the output stream */
+ unsigned long opos_inc;
- ULONG ipos; /* position in the input stream (integer) */
+ unsigned long ipos; /* position in the input stream (integer) */
- LONG ilast; /* last sample in the input stream */
+ st_sample_t ilast; /* last sample in the input stream */
} *rate_t;
/*
@@ -69,7 +69,7 @@
int st_rate_start(eff_t effp)
{rate_t rate = (rate_t) effp->priv;
- ULONG incr;
+ unsigned long incr;
if (effp->ininfo.rate == effp->outinfo.rate)
{@@ -92,7 +92,7 @@
rate->opos=0;
/* increment */
- incr=(ULONG)((double)effp->ininfo.rate / (double)effp->outinfo.rate *
+ incr=(unsigned long)((double)effp->ininfo.rate / (double)effp->outinfo.rate *
(double) ((unsigned long) 1 << FRAC_BITS));
rate->opos_inc_frac = incr & (((unsigned long) 1 << FRAC_BITS)-1);
@@ -112,10 +112,10 @@
st_size_t *isamp, st_size_t *osamp)
{rate_t rate = (rate_t) effp->priv;
- LONG *istart,*iend;
- LONG *ostart,*oend;
- LONG ilast,icur,out;
- ULONG tmp;
+ st_sample_t *istart,*iend;
+ st_sample_t *ostart,*oend;
+ st_sample_t ilast,icur,out;
+ unsigned long tmp;
double t;
ilast=rate->ilast;
@@ -148,7 +148,7 @@
/* output sample & increment position */
- *obuf++=(LONG) out;
+ *obuf++=(st_sample_t) out;
tmp = rate->opos_frac + rate->opos_inc_frac;
rate->opos = rate->opos + rate->opos_inc + (tmp >> FRAC_BITS);
@@ -226,11 +226,11 @@
/* Private data for Lerp via LCM file */
typedef struct ratestuff {- ULONG lcmrate; /* least common multiple of rates */
- ULONG inskip, outskip; /* LCM increments for I & O rates */
- ULONG total;
- ULONG intot, outtot; /* total samples in LCM basis */
- LONG lastsamp; /* history */
+ st_rate_t lcmrate; /* least common multiple of rates */
+ unsigned long inskip, outskip; /* LCM increments for I & O rates */
+ unsigned long total;
+ unsigned long intot, outtot; /* total samples in LCM basis */
+ st_sample_t lastsamp; /* history */
} *rate_t;
/*
@@ -253,7 +253,7 @@
{rate_t rate = (rate_t) effp->priv;
- rate->lcmrate = lcm((LONG)effp->ininfo.rate, (LONG)effp->outinfo.rate);
+ rate->lcmrate = st_lcm((st_sample_t)effp->ininfo.rate, (st_sample_t)effp->outinfo.rate);
/* Cursory check for LCM overflow.
* If both rate are below 65k, there should be no problem.
* 16 bits x 16 bits = 32 bits, which we can handle.
@@ -274,8 +274,8 @@
{rate_t rate = (rate_t) effp->priv;
int len, done;
- LONG *istart = ibuf;
- LONG last;
+ st_sample_t *istart = ibuf;
+ st_sample_t last;
done = 0;
if (rate->total == 0) {--- a/src/resampl.h
+++ b/src/resampl.h
@@ -54,7 +54,7 @@
* the input array (X), and to the right, it is interpreted as a number
* between 0 and 1 sample of the input X. The default value of 23 is
* about right. There is a constraint that the filter window must be
- * "addressable" in a LONG int, more precisely, if Nmult is the number
+ * "addressable" in a int32_t, more precisely, if Nmult is the number
* of sinc zero-crossings in the right wing of the filter window, then
* (Nwing<<Lp) must be expressible in 31 bits.
*
--- a/src/resample.c
+++ b/src/resample.c
@@ -73,41 +73,41 @@
double rolloff; /* roll-off frequency */
double beta; /* passband/stopband tuning magic */
int quadr; /* non-zero to use qprodUD quadratic interpolation */
- LONG Nmult;
- LONG Nwing;
- LONG Nq;
+ long Nmult;
+ long Nwing;
+ long Nq;
Float *Imp; /* impulse [Nwing+1] Filter coefficients */
double Time; /* Current time/pos in input sample */
- LONG dhb;
+ long dhb;
- LONG a,b; /* gcd-reduced input,output rates */
- LONG t; /* Current time/pos for exact-coeff's method */
+ long a,b; /* gcd-reduced input,output rates */
+ long t; /* Current time/pos for exact-coeff's method */
- LONG Xh; /* number of past/future samples needed by filter */
- LONG Xoff; /* Xh plus some room for creep */
- LONG Xread; /* X[Xread] is start-position to enter new samples */
- LONG Xp; /* X[Xp] is position to start filter application */
- LONG Xsize,Ysize; /* size (Floats) of X[],Y[] */
+ long Xh; /* number of past/future samples needed by filter */
+ long Xoff; /* Xh plus some room for creep */
+ long Xread; /* X[Xread] is start-position to enter new samples */
+ long Xp; /* X[Xp] is position to start filter application */
+ long Xsize,Ysize; /* size (Floats) of X[],Y[] */
Float *X, *Y; /* I/O buffers */
} *resample_t;
static void LpFilter(double c[],
- LONG N,
+ long N,
double frq,
double Beta,
- LONG Num);
+ long Num);
/* makeFilter is used by filter.c */
int makeFilter(Float Imp[],
- LONG Nwing,
+ long Nwing,
double Froll,
double Beta,
- LONG Num,
+ long Num,
int Normalize);
-static LONG SrcUD(resample_t r, LONG Nx);
-static LONG SrcEX(resample_t r, LONG Nx);
+static long SrcUD(resample_t r, long Nx);
+static long SrcEX(resample_t r, long Nx);
/*
@@ -174,9 +174,8 @@
int st_resample_start(eff_t effp)
{resample_t r = (resample_t) effp->priv;
- LONG Xoff, gcdrate;
+ long Xoff, gcdrate;
int i;
- extern LONG st_gcd(LONG a,LONG b);
if (effp->ininfo.rate == effp->outinfo.rate)
{@@ -186,7 +185,7 @@
r->Factor = (double)effp->outinfo.rate / (double)effp->ininfo.rate;
- gcdrate = st_gcd((LONG)effp->ininfo.rate, (LONG)effp->outinfo.rate);
+ gcdrate = st_gcd((long)effp->ininfo.rate, (long)effp->outinfo.rate);
r->a = effp->ininfo.rate / gcdrate;
r->b = effp->outinfo.rate / gcdrate;
@@ -270,7 +269,7 @@
st_size_t *isamp, st_size_t *osamp)
{resample_t r = (resample_t) effp->priv;
- LONG i, last, Nout, Nx, Nproc;
+ long i, last, Nout, Nx, Nproc;
/* constrain amount we actually process */
/*fprintf(stderr,"Xp %d, Xread %d, isamp %d, ",r->Xp, r->Xread,*isamp);*/
@@ -309,7 +308,7 @@
return (ST_SUCCESS);
}
if (r->quadr < 0) { /* exact coeff's method */- LONG creep;
+ long creep;
Nout = SrcEX(r, Nproc);
/*fprintf(stderr,"Nproc %d --> %d\n",Nproc,Nout);*/
/* Move converter Nproc samples back in time */
@@ -325,7 +324,7 @@
/*fprintf(stderr,"Nproc %ld, creep %ld\n",Nproc,creep);*/
}
} else { /* approx coeff's method */- LONG creep;
+ long creep;
Nout = SrcUD(r, Nproc);
/*fprintf(stderr,"Nproc %d --> %d\n",Nproc,Nout);*/
/* Move converter Nproc samples back in time */
@@ -343,7 +342,7 @@
}
{- LONG i,k;
+ long i,k;
/* Copy back portion of input signal that must be re-used */
k = r->Xp - r->Xoff;
/*fprintf(stderr,"k %d, last %d\n",k,last);*/
@@ -370,7 +369,8 @@
int st_resample_drain(eff_t effp, st_sample_t *obuf, st_size_t *osamp)
{resample_t r = (resample_t) effp->priv;
- LONG isamp_res, *Obuf, osamp_res;
+ long isamp_res, osamp_res;
+ st_sample_t *Obuf;
int rc;
/* fprintf(stderr,"Xoff %d, Xt %d <--- DRAIN\n",r->Xoff, r->Xt); */
@@ -380,7 +380,7 @@
osamp_res = *osamp;
Obuf = obuf;
while (isamp_res>0 && osamp_res>0) {- LONG Isamp, Osamp;
+ st_sample_t Isamp, Osamp;
Isamp = isamp_res;
Osamp = osamp_res;
rc = st_resample_flow(effp, NULL, Obuf, &Isamp, &Osamp);
@@ -415,12 +415,12 @@
/* over 90% of CPU time spent in this iprodUD() function */
/* quadratic interpolation */
-static double qprodUD(const Float Imp[], const Float *Xp, LONG Inc, double T0,
- LONG dhb, LONG ct)
+static double qprodUD(const Float Imp[], const Float *Xp, long Inc, double T0,
+ long dhb, long ct)
{const double f = 1.0/(1<<La);
double v;
- LONG Ho;
+ long Ho;
Ho = T0 * dhb;
Ho += (ct-1)*dhb; /* so Float sum starts with smallest coef's */
@@ -428,7 +428,7 @@
v = 0;
do {Float coef;
- LONG Hoh;
+ long Hoh;
Hoh = Ho>>La;
coef = Imp[Hoh];
{@@ -447,12 +447,12 @@
}
/* linear interpolation */
-static double iprodUD(const Float Imp[], const Float *Xp, LONG Inc,
- double T0, LONG dhb, LONG ct)
+static double iprodUD(const Float Imp[], const Float *Xp, long Inc,
+ double T0, long dhb, long ct)
{const double f = 1.0/(1<<La);
double v;
- LONG Ho;
+ long Ho;
Ho = T0 * dhb;
Ho += (ct-1)*dhb; /* so Float sum starts with smallest coef's */
@@ -460,7 +460,7 @@
v = 0;
do {Float coef;
- LONG Hoh;
+ long Hoh;
Hoh = Ho>>La;
/* if (Hoh >= End) break; */
coef = Imp[Hoh] + (Imp[Hoh+1]-Imp[Hoh]) * (Ho & Amask) * f;
@@ -475,7 +475,7 @@
/* From resample:filters.c */
/* Sampling rate conversion subroutine */
-static LONG SrcUD(resample_t r, LONG Nx)
+static long SrcUD(resample_t r, long Nx)
{Float *Ystart, *Y;
double Factor;
@@ -503,7 +503,7 @@
double v;
double T;
T = time-floor(time); /* fractional part of Time */
- Xp = r->X + (LONG)time; /* Ptr to current input sample */
+ Xp = r->X + (long)time; /* Ptr to current input sample */
/* Past inner product: */
v = (*prodUD)(r->Imp, Xp, -1, T, r->dhb, r->Xh); /* needs Np*Nmult in 31 bits */
@@ -521,7 +521,7 @@
/* exact coeff's */
static double prodEX(const Float Imp[], const Float *Xp,
- LONG Inc, LONG T0, LONG dhb, LONG ct)
+ long Inc, long T0, long dhb, long ct)
{double v;
const Float *Cp;
@@ -537,12 +537,12 @@
return v;
}
-static LONG SrcEX(resample_t r, LONG Nx)
+static long SrcEX(resample_t r, long Nx)
{Float *Ystart, *Y;
double Factor;
- LONG a,b;
- LONG time;
+ long a,b;
+ long time;
int n;
Factor = r->Factor;
@@ -555,7 +555,7 @@
{Float *Xp;
double v;
- LONG T;
+ long T;
T = time % b; /* fractional part of Time */
Xp = r->X + (time/b); /* Ptr to current input sample */
@@ -572,11 +572,11 @@
return (Y - Ystart); /* Return the number of output samples */
}
-int makeFilter(Float Imp[], LONG Nwing, double Froll, double Beta,
- LONG Num, int Normalize)
+int makeFilter(Float Imp[], long Nwing, double Froll, double Beta,
+ long Num, int Normalize)
{double *ImpR;
- LONG Mwing, i;
+ long Mwing, i;
if (Nwing > MAXNWING) /* Check for valid parameters */
return(-1);
@@ -595,7 +595,7 @@
LpFilter(ImpR, Mwing, Froll, Beta, Num);
if (Normalize) { /* 'correct' the DC gain of the lowpass filter */- LONG Dh;
+ long Dh;
double DCgain;
DCgain = 0;
Dh = Num; /* Filter sampling period for factors>=1 */
@@ -663,7 +663,7 @@
static double Izero(double x)
{double sum, u, halfx, temp;
- LONG n;
+ long n;
sum = u = n = 1;
halfx = x/2.0;
@@ -677,9 +677,9 @@
return(sum);
}
-static void LpFilter(double *c, LONG N, double frq, double Beta, LONG Num)
+static void LpFilter(double *c, long N, double frq, double Beta, long Num)
{- LONG i;
+ long i;
/* Calculate filter coeffs: */
c[0] = frq;
--- a/src/reverb.c
+++ b/src/reverb.c
@@ -107,7 +107,7 @@
float in_gain, out_gain, time;
float delay[MAXREVERBS], decay[MAXREVERBS];
long samples[MAXREVERBS], maxsamples;
- LONG pl, ppl, pppl;
+ st_sample_t pl, ppl, pppl;
} *reverb_t;
/*
@@ -214,7 +214,7 @@
int i, j;
float d_in, d_out;
- LONG out;
+ st_sample_t out;
i = reverb->counter;
len = ((*isamp > *osamp) ? *osamp : *isamp);
@@ -227,7 +227,7 @@
d_in +=
reverb->reverbbuf[(i + reverb->maxsamples - reverb->samples[j]) % reverb->maxsamples] * reverb->decay[j];
d_out = d_in * reverb->out_gain;
- out = st_clip24((LONG) d_out);
+ out = st_clip24((st_sample_t) d_out);
*obuf++ = out * 256;
reverb->reverbbuf[i] = d_in;
i++; /* XXX need a % maxsamples here ? */
@@ -245,7 +245,7 @@
{reverb_t reverb = (reverb_t) effp->priv;
float d_in, d_out;
- LONG out, l;
+ st_sample_t out, l;
int i, j, done;
i = reverb->counter;
@@ -258,10 +258,10 @@
d_in +=
reverb->reverbbuf[(i + reverb->maxsamples - reverb->samples[j]) % reverb->maxsamples] * reverb->decay[j];
d_out = d_in * reverb->out_gain;
- out = st_clip24((LONG) d_out);
+ out = st_clip24((st_sample_t) d_out);
obuf[done++] = out * 256;
reverb->reverbbuf[i] = d_in;
- l = st_clip24((LONG) d_in);
+ l = st_clip24((st_sample_t) d_in);
reverb->pppl = reverb->ppl;
reverb->ppl = reverb->pl;
reverb->pl = l;
--- a/src/reverse.c
+++ b/src/reverse.c
@@ -23,7 +23,7 @@
/* Private data */
typedef struct reversestuff {FILE *fp;
- LONG pos;
+ st_size_t pos;
int phase;
} *reverse_t;
@@ -75,7 +75,7 @@
st_fail("Internal error: reverse_flow called in wrong phase");return(ST_EOF);
}
- if (fwrite((char *)ibuf, sizeof(LONG), *isamp, reverse->fp)
+ if (fwrite((char *)ibuf, sizeof(st_sample_t), *isamp, reverse->fp)
!= *isamp)
{ st_fail("Reverse effect write error on temporary file\n");@@ -94,13 +94,13 @@
reverse_t reverse = (reverse_t) effp->priv;
int len, nbytes;
register int i, j;
- LONG temp;
+ st_sample_t temp;
if (reverse->phase == WRITING) {fflush(reverse->fp);
fseek(reverse->fp, 0L, SEEK_END);
reverse->pos = ftell(reverse->fp);
- if (reverse->pos % sizeof(LONG) != 0)
+ if (reverse->pos % sizeof(st_sample_t) != 0)
{ st_fail("Reverse effect finds odd temporary file\n");return(ST_EOF);
@@ -108,14 +108,14 @@
reverse->phase = READING;
}
len = *osamp;
- nbytes = len * sizeof(LONG);
+ nbytes = len * sizeof(st_sample_t);
if (reverse->pos < nbytes) {nbytes = reverse->pos;
- len = nbytes / sizeof(LONG);
+ len = nbytes / sizeof(st_sample_t);
}
reverse->pos -= nbytes;
fseek(reverse->fp, reverse->pos, SEEK_SET);
- if (fread((char *)obuf, sizeof(LONG), len, reverse->fp) != len)
+ if (fread((char *)obuf, sizeof(st_sample_t), len, reverse->fp) != len)
{ st_fail("Reverse effect read error from temporary file\n");return(ST_EOF);
--- a/src/sf.c
+++ b/src/sf.c
@@ -27,7 +27,7 @@
typedef struct sfstuff {struct sfinfo info;
/* needed for seek */
- LONG dataStart;
+ st_size_t dataStart;
} *sf_t;
/*
--- a/src/sfircam.h
+++ b/src/sfircam.h
@@ -24,19 +24,19 @@
#define SF_NEXT 4
#ifdef vax
#define SF_MACHINE SF_VAX
-#define SF_MAGIC ((LONG)(SF_MAGIC1 | SF_MAGIC2 << 8 | SF_MACHINE << 16))
+#define SF_MAGIC ((uint32_t)(SF_MAGIC1 | SF_MAGIC2 << 8 | SF_MACHINE << 16))
#endif
#ifdef sun
#define SF_MACHINE SF_SUN
-#define SF_MAGIC ((LONG)(SF_MAGIC1 << 24 | SF_MAGIC2 << 16 | SF_MACHINE << 8))
+#define SF_MAGIC ((uint32_t)(SF_MAGIC1 << 24 | SF_MAGIC2 << 16 | SF_MACHINE << 8))
#endif
#ifdef mips
#define SF_MACHINE SF_MIPS
-#define SF_MAGIC ((LONG)(SF_MAGIC1 | SF_MAGIC2 << 8 | SF_MACHINE << 16))
+#define SF_MAGIC ((uint32_t)(SF_MAGIC1 | SF_MAGIC2 << 8 | SF_MACHINE << 16))
#endif
#ifdef NeXT
#define SF_MACHINE SF_NEXT
-#define SF_MAGIC ((LONG)(SF_MAGIC1 << 24 | SF_MAGIC2 << 16 | SF_MACHINE << 8))
+#define SF_MAGIC ((uint32_t)(SF_MAGIC1 << 24 | SF_MAGIC2 << 16 | SF_MACHINE << 8))
#endif
@@ -54,12 +54,12 @@
*
* Possible values of sf_packmode:
*/
-#define SF_CHAR ((LONG) sizeof(char))
-#define SF_ALAW ((LONG) sizeof(char) | 0x10000)
-#define SF_ULAW ((LONG) sizeof(char) | 0x20000)
-#define SF_SHORT ((LONG) sizeof(short))
-#define SF_LONG ((LONG) sizeof(LONG) | 0x40000)
-#define SF_FLOAT ((LONG) sizeof(float))
+#define SF_CHAR ((uint32_t) sizeof(char))
+#define SF_ALAW ((uint32_t) sizeof(char) | 0x10000)
+#define SF_ULAW ((uint32_t) sizeof(char) | 0x20000)
+#define SF_SHORT ((uint32_t) sizeof(short))
+#define SF_LONG ((uint32_t) sizeof(long) | 0x40000)
+#define SF_FLOAT ((uint32_t) sizeof(float))
/* For marking data after fixed section of soundfile header -- see man (3carl)
* sfcodes and defintions of SFCODE and related structures, below.
@@ -89,11 +89,11 @@
unsigned char sf_machine; /* 3 */
unsigned char sf_param; /* 4 */
} _magic_bytes;
- LONG sf_magic; /* magic as a 4-byte long */
+ uint32_t sf_magic; /* magic as a 4-byte long */
} magic_union;
float sf_srate;
- LONG sf_chans;
- LONG sf_packmode;
+ uint32_t sf_chans;
+ uint32_t sf_packmode;
char sf_codes;
} sfinfo;
char filler[SIZEOF_HEADER];
@@ -117,8 +117,8 @@
typedef struct Sfmaxamp {float value[SF_MAXCHAN];
- LONG samploc[SF_MAXCHAN];
- LONG timetag;
+ uint32_t samploc[SF_MAXCHAN];
+ uint32_t timetag;
} SFMAXAMP;
typedef struct sfcomment {--- a/src/silence.c
+++ b/src/silence.c
@@ -292,7 +292,7 @@
else
silence->mode = SILENCE_COPY;
- silence->start_holdoff = malloc(sizeof(LONG)*silence->start_duration);
+ silence->start_holdoff = malloc(sizeof(st_sample_t)*silence->start_duration);
if (!silence->start_holdoff)
{ st_fail("Could not allocate memory");@@ -302,7 +302,7 @@
silence->start_holdoff_end = 0;
silence->start_found_periods = 0;
- silence->stop_holdoff = malloc(sizeof(LONG)*silence->stop_duration);
+ silence->stop_holdoff = malloc(sizeof(st_sample_t)*silence->stop_duration);
if (!silence->stop_holdoff)
{ st_fail("Could not allocate memory");@@ -380,7 +380,7 @@
{silence_t silence = (silence_t) effp->priv;
int threshold, i, j;
- ULONG nrOfTicks, nrOfInSamplesRead, nrOfOutSamplesWritten;
+ st_size_t nrOfTicks, nrOfInSamplesRead, nrOfOutSamplesWritten;
nrOfInSamplesRead = 0;
nrOfOutSamplesWritten = 0;
@@ -611,7 +611,7 @@
{silence_t silence = (silence_t) effp->priv;
int i;
- LONG nrOfTicks, nrOfOutSamplesWritten = 0;
+ st_size_t nrOfTicks, nrOfOutSamplesWritten = 0;
/* Only if in flush mode will there be possible samples to write
* out during drain() call.
--- a/src/skel.c
+++ b/src/skel.c
@@ -59,18 +59,8 @@
st_ssize_t st_skelread(ft_t ft, st_sample_t *buf, st_ssize_t len)
{skel_t sk = (skel_t) ft->priv;
- int abs;
- float amp;
int done = 0;
-
- char c;
- unsigned char uc;
- short s;
- unsigned short us;
- LONG l;
- ULONG ul;
- float f;
- double d;
+ st_sample_t l;
for(; done < len; done++) {if no more samples
--- a/src/smp.c
+++ b/src/smp.c
@@ -63,7 +63,7 @@
/* Private data for SMP file */
typedef struct smpstuff {uint32_t NoOfSamps; /* Sample data count in words */
- LONG dataStart;
+ st_size_t dataStart;
/* comment memory resides in private data because it's small */
char comment[COMMENTLEN + NAMELEN + 3];
} *smp_t;
@@ -77,7 +77,7 @@
static int readtrailer(ft_t ft, struct smptrailer *trailer)
{int i;
- ULONG trash;
+ int32_t trash;
st_readw(ft, (unsigned short *)&trash); /* read reserved word */
for(i = 0; i < 8; i++) { /* read the 8 loops */@@ -109,7 +109,7 @@
/*
* set the trailer data - loops and markers, to reasonably benign values
*/
-static void settrailer(ft_t ft, struct smptrailer *trailer, ULONG rate)
+static void settrailer(ft_t ft, struct smptrailer *trailer, st_rate_t rate)
{int i;
@@ -192,7 +192,7 @@
smp_t smp = (smp_t) ft->priv;
int i;
int namelen, commentlen;
- LONG samplestart;
+ long samplestart;
struct smpheader header;
struct smptrailer trailer;
@@ -386,7 +386,7 @@
{smp_t smp = (smp_t) ft->priv;
register int datum;
- LONG done = 0;
+ st_ssize_t done = 0;
while(done < len) {datum = (int) RIGHT(*buf++, 16);
--- a/src/sndrtool.c
+++ b/src/sndrtool.c
@@ -21,8 +21,8 @@
/* Private data used by writer */
typedef struct sndpriv {- ULONG nsamples;
- LONG dataStart;
+ st_size_t nsamples;
+ st_size_t dataStart;
} *snd_t;
#ifndef SEEK_CUR
@@ -29,7 +29,7 @@
#define SEEK_CUR 1
#endif
-static void sndtwriteheader(ft_t ft,LONG nsamples);
+static void sndtwriteheader(ft_t ft, st_size_t nsamples);
int st_sndseek(ft_t ft, st_size_t offset)
{@@ -224,7 +224,7 @@
/*======================================================================*/
/* SNDTWRITEHEADER */
/*======================================================================*/
-static void sndtwriteheader(ft_t ft, st_ssize_t nsamples)
+static void sndtwriteheader(ft_t ft, st_size_t nsamples)
{char name_buf[97];
@@ -231,9 +231,9 @@
/* sndtool header */
st_writes(ft, "SOUND"); /* magic */
st_writeb(ft, 0x1a);
-st_writew (ft,(LONG)0); /* hGSound */
+st_writew (ft,0); /* hGSound */
st_writedw (ft,nsamples);
-st_writedw (ft,(LONG)0);
+st_writedw (ft,0);
st_writedw (ft,nsamples);
st_writew (ft,(int) ft->info.rate);
st_writew (ft,0);
--- a/src/sox.c
+++ b/src/sox.c
@@ -64,10 +64,10 @@
static int writing = 0; /* are we writing to a file? */
static int soxpreview = 0; /* preview mode */
-static LONG ibufl[BUFSIZ/2]; /* Left/right interleave buffers */
-static LONG ibufr[BUFSIZ/2];
-static LONG obufl[BUFSIZ/2];
-static LONG obufr[BUFSIZ/2];
+static st_sample_t ibufl[BUFSIZ/2]; /* Left/right interleave buffers */
+static st_sample_t ibufr[BUFSIZ/2];
+static st_sample_t obufl[BUFSIZ/2];
+static st_sample_t obufr[BUFSIZ/2];
/* local forward declarations */
static void doopts(ft_t, int, char **);
@@ -79,7 +79,7 @@
static int filetype(int);
static void process(void);
static void statistics(void);
-static LONG volumechange(LONG *buf, LONG ct, double vol);
+static st_sample_t volumechange(st_sample_t *buf, st_ssize_t ct, double vol);
static void checkeffect(void);
static int flow_effect_out(void);
static int flow_effect(int);
@@ -544,8 +544,8 @@
int e, f, flowstatus;
#ifdef SOXMIX
int s;
- ULONG ilen[MAX_INPUT_FILES];
- LONG *ibuf[MAX_INPUT_FILES];
+ st_ssize_t ilen[MAX_INPUT_FILES];
+ st_sample_t *ibuf[MAX_INPUT_FILES];
#endif
for (f = 0; f < input_count; f++)
@@ -744,7 +744,7 @@
if (writing&&efftab[neffects-1].olen > 0)
(* outformat->h->write)(outformat, efftab[neffects-1].obuf,
- (LONG) efftab[neffects-1].olen);
+ (st_ssize_t) efftab[neffects-1].olen);
if (efftab[f].olen != BUFSIZ)
break;
@@ -836,7 +836,7 @@
static int flow_effect(e)
int e;
{- st_size_t i, done, idone, odone, idonel, odonel, idoner, odoner;
+ st_ssize_t i, done, idone, odone, idonel, odonel, idoner, odoner;
st_sample_t *ibuf, *obuf;
int effstatus;
@@ -905,8 +905,8 @@
static int drain_effect(e)
int e;
{- LONG i, olen, olenl, olenr;
- LONG *obuf;
+ st_ssize_t i, olen, olenl, olenr;
+ st_sample_t *obuf;
if (! efftabR[e].name) {efftab[e].olen = BUFSIZ;
@@ -1113,14 +1113,12 @@
st_report("Volume change clipped %d samples", clipped);}
-static LONG volumechange(buf, ct, vol)
-LONG *buf;
-LONG ct;
-double vol;
+static st_sample_t volumechange(st_sample_t *buf, st_ssize_t ct,
+ double vol)
{double y;
- LONG *p,*top;
- LONG clips=0;
+ st_sample_t *p,*top;
+ st_ssize_t clips=0;
p = buf;
top = buf+ct;
--- a/src/speed.c
+++ b/src/speed.c
@@ -13,7 +13,6 @@
#include "st_i.h"
-#include <limits.h> /* LONG_MAX */
#include <math.h> /* pow */
#include <string.h>
@@ -54,7 +53,7 @@
int compression; /* integer compression of the signal. */
int index; /* how much of the input buffer is filled */
- LONG * ibuf; /* small internal input buffer for compression */
+ st_sample_t *ibuf; /* small internal input buffer for compression */
SPEED_FLOAT cbuf[4]; /* computation buffer for interpolation */
SPEED_FLOAT frac; /* current index position in cbuf */
@@ -94,20 +93,20 @@
}
/* clip if necessary, and report. */
-static LONG clip(speed_t speed, SPEED_FLOAT v)
+static st_sample_t clip(speed_t speed, SPEED_FLOAT v)
{- if (v < -LONG_MAX)
+ if (v < -ST_SAMPLE_MAX)
{speed->clipped++;
- return -LONG_MAX;
+ return -ST_SAMPLE_MAX;
}
- else if (v > LONG_MAX)
+ else if (v > ST_SAMPLE_MAX)
{speed->clipped++;
- return LONG_MAX;
+ return ST_SAMPLE_MAX;
}
else
- return (LONG) v;
+ return (st_sample_t) v;
}
/* get options. */
@@ -157,7 +156,8 @@
speed->rate = speed->factor;
}
- speed->ibuf = (LONG*) malloc(speed->compression*sizeof(LONG));
+ speed->ibuf = (st_sample_t *) malloc(speed->compression*
+ sizeof(st_sample_t));
speed->index = 0;
speed->state = sp_input;
--- a/src/sphere.c
+++ b/src/sphere.c
@@ -18,8 +18,8 @@
/* Private data for SKEL file */
typedef struct spherestuff {- char shorten_check[4];
- ULONG numSamples;
+ char shorten_check[4];
+ st_size_t numSamples;
} *sphere_t;
/*
--- a/src/st.h
+++ b/src/st.h
@@ -73,14 +73,6 @@
#define ST_SSIZE_MAX 0x7fffffffL
#define ST_SSIZE_MIN (-ST_SSIZE_MAX - 1L)
-/* FIXME: Remove from usage by libst */
-#define LONG int32_t
-#define ULONG uint32_t
-
-/* FIXME: Get rid of this and usage of LONG_MAX */
-#define MAXLONG 0x7fffffffL
-#define MAXULONG 0xffffffffL
-
/* Signal parameters */
typedef struct st_signalinfo
--- a/src/st_i.h
+++ b/src/st_i.h
@@ -37,12 +37,12 @@
#endif
/* declared in misc.c */
-LONG st_clip24(LONG) REGPARM(1);
-void st_sine(int *, LONG, int, int);
-void st_triangle(int *, LONG, int, int);
+st_sample_t st_clip24(st_sample_t) REGPARM(1);
+void st_sine(int *buf, st_ssize_t len, int max, int depth);
+void st_triangle(int *buf, st_ssize_t len, int max, int depth);
-LONG st_gcd(LONG,LONG) REGPARM(2);
-LONG st_lcm(LONG,LONG) REGPARM(2);
+st_sample_t st_gcd(st_sample_t a, st_sample_t b) REGPARM(2);
+st_sample_t st_lcm(st_sample_t a, st_sample_t b) REGPARM(2);
#ifndef HAVE_RAND
int rand(void);
@@ -79,7 +79,7 @@
#ifdef HAVE_BYTESWAP_H
#define st_swapw(x) bswap_16(x)
#define st_swapdw(x) bswap_32(x)
-#define st_swapf(x) (float)bswap_32((ULONG)(x))
+#define st_swapf(x) (float)bswap_32((uint32_t)(x))
#else
uint16_t st_swapw(uint16_t uw);
uint32_t st_swapdw(uint32_t udw);
--- a/src/stat.c
+++ b/src/stat.c
@@ -29,16 +29,16 @@
double dsum1, dsum2; /* deltas */
double scale; /* scale-factor */
double last; /* previous sample */
- ULONG read; /* samples processed */
+ st_size_t read; /* samples processed */
int volume;
int srms;
int fft;
- ULONG bin[4];
+ ulong bin[4];
double *re;
double *im;
- ULONG fft_bits;
- ULONG fft_size;
- ULONG fft_offset;
+ ulong fft_bits;
+ ulong fft_size;
+ ulong fft_offset;
} *stat_t;
@@ -49,7 +49,7 @@
{stat_t stat = (stat_t) effp->priv;
- stat->scale = MAXLONG;
+ stat->scale = ST_SAMPLE_MAX;
stat->volume = 0;
stat->srms = 0;
stat->fft = 0;
@@ -107,7 +107,7 @@
{stat_t stat = (stat_t) effp->priv;
int i;
- LONG bitmask;
+ ulong bitmask;
stat->min = stat->max = stat->mid = 0;
stat->asum = 0;
@@ -344,7 +344,7 @@
/* Just print the volume adjustment */
if (stat->volume == 1 && amp > 0) {- fprintf(stderr, "%.3f\n", MAXLONG/(amp*scale));
+ fprintf(stderr, "%.3f\n", ST_SAMPLE_MAX/(amp*scale));
return (ST_SUCCESS);
}
if (stat->volume == 2) {@@ -371,7 +371,7 @@
freq = sqrt(stat->dsum2/stat->sum2)*effp->ininfo.rate/(M_PI*2);
fprintf(stderr, "Rough frequency: %12d\n", (int)freq);
- if (amp>0) fprintf(stderr, "Volume adjustment: %12.3f\n", MAXLONG/(amp*scale));
+ if (amp>0) fprintf(stderr, "Volume adjustment: %12.3f\n", ST_SAMPLE_MAX/(amp*scale));
if (stat->bin[2] == 0 && stat->bin[3] == 0)
fprintf(stderr, "\nProbably text, not sound\n");
--- a/src/stretch.c
+++ b/src/stretch.c
@@ -22,7 +22,6 @@
#include "st_i.h"
#include <stdlib.h> /* malloc and free */
-#include <limits.h> /* LONG_MAX */
#include <string.h> /* memcpy() */
#ifndef MIN
@@ -80,7 +79,7 @@
int size; /* buffer size */
int index; /* next available element */
- LONG * ibuf; /* input buffer */
+ st_sample_t *ibuf; /* input buffer */
int ishift; /* input shift */
int oindex; /* next evailable element */
@@ -106,21 +105,21 @@
/* clip amplitudes and count number of clipped values.
*/
-static LONG clip(stretch_t stretch, STRETCH_FLOAT v)
+static st_sample_t clip(stretch_t stretch, STRETCH_FLOAT v)
{- if (v < -LONG_MAX)
+ if (v < -ST_SAMPLE_MAX)
{stretch->clipped++;
- return -LONG_MAX;
+ return -ST_SAMPLE_MAX;
}
- else if (v > LONG_MAX)
+ else if (v > ST_SAMPLE_MAX)
{stretch->clipped++;
- return LONG_MAX;
+ return ST_SAMPLE_MAX;
}
else
{- return (LONG) v;
+ return (st_sample_t) v;
}
}
@@ -231,7 +230,8 @@
stretch->size = (int)(effp->outinfo.rate * ONETHOUSANDS * stretch->window);
/* start in the middle of an input to avoid initial fading... */
stretch->index = stretch->size/2;
- stretch->ibuf = (LONG *) malloc(stretch->size * sizeof(LONG));
+ stretch->ibuf = (st_sample_t *) malloc(stretch->size *
+ sizeof(st_sample_t));
/* the shift ratio deal with the longest of ishift/oshift
hence ishift<=size and oshift<=size. should be asserted.
@@ -328,7 +328,7 @@
stretch->size-stretch->index);
memcpy(stretch->ibuf+stretch->index,
- ibuf+iindex, tocopy*sizeof(LONG));
+ ibuf+iindex, tocopy*sizeof(st_sample_t));
iindex += tocopy;
stretch->index += tocopy;
--- a/src/synth.c
+++ b/src/synth.c
@@ -12,7 +12,6 @@
#include <signal.h>
#include <string.h>
-#include <limits.h>
#include <math.h>
#include <ctype.h>
#include "st_i.h"
@@ -161,10 +160,10 @@
double par[MAXCHAN][5];
/* internal stuff */
- LONG max;
- ULONG samples_done;
+ st_sample_t max;
+ st_size_t samples_done;
int rate;
- ULONG length; /* length in number of samples */
+ st_size_t length; /* length in number of samples */
double h[MAXCHAN]; /* store values necessary for creation */
PinkNoise pinkn[MAXCHAN];
} *synth_t;
@@ -243,7 +242,7 @@
/* set default parameters */
synth->length = 0; /* use length of input file */
synth->length_str = 0;
- synth->max = LONG_MAX;
+ synth->max = ST_SAMPLE_MAX;
for(c=0;c<MAXCHAN;c++){synth->freq[c] = 440.0;
synth->freq2[c] = 440.0;
@@ -515,8 +514,8 @@
-static LONG do_synth(LONG iv, synth_t synth, int c){- LONG ov=iv;
+static st_sample_t do_synth(st_sample_t iv, synth_t synth, int c){+ st_sample_t ov=iv;
double r=0.0; /* -1 .. +1 */
double f;
double om;
@@ -674,7 +673,7 @@
ov = iv/2 + r*synth->max/2;
break;
case SYNTH_AMOD:
- ov = (LONG)(0.5*(r+1.0)*(double)iv);
+ ov = (st_sample_t)(0.5*(r+1.0)*(double)iv);
break;
case SYNTH_FMOD:
ov = iv * r ;
--- a/src/trim.c
+++ b/src/trim.c
@@ -26,12 +26,12 @@
char *length_str;
/* options converted to values */
- ULONG start;
- ULONG length;
+ st_size_t start;
+ st_size_t length;
/* internal stuff */
- ULONG index;
- ULONG trimmed;
+ st_size_t index;
+ st_size_t trimmed;
int done;
} * trim_t;
@@ -191,7 +191,7 @@
trim->trimmed += done;
}
- memcpy(obuf, ibuf+offset, done * sizeof(LONG));
+ memcpy(obuf, ibuf+offset, done * sizeof(st_sample_t));
*osamp = done;
return (ST_SUCCESS);
}
--- a/src/tx16w.c
+++ b/src/tx16w.c
@@ -45,7 +45,7 @@
/* Private data for TX16 file */
typedef struct txwstuff {- LONG rest; /* bytes remaining in sample file */
+ st_size_t rest; /* bytes remaining in sample file */
} *txw_t;
struct WaveHeader_ {@@ -64,8 +64,8 @@
static const unsigned char magic2[4] = {0, 0x52, 0x00, 0x52};/* SJB: dangerous static variables */
-static LONG tx16w_len=0;
-static LONG writedone=0;
+static st_size_t tx16w_len=0;
+static st_size_t writedone=0;
/*
* Do anything required before you start reading samples.
@@ -80,10 +80,10 @@
char filetype[7];
char format;
char sample_rate;
- LONG num_samp_bytes = 0;
+ st_size_t num_samp_bytes = 0;
char gunk[8];
int blewIt;
- ULONG trash;
+ int32_t trash;
txw_t sk = (txw_t) ft->priv;
/* If you need to seek around the input file. */
@@ -181,7 +181,7 @@
/*
* Read up to len samples from file.
- * Convert to signed LONGs.
+ * Convert to st_sample_ts.
* Place in buf[].
* Return number of samples read.
*/
@@ -211,9 +211,9 @@
* two unsigned short samples.
* TCC 3.0 appeared to do unwanted things, so we really specify
* exactly what we want to happen.
- * Convert unsigned short to LONG then shift up the result
+ * Convert unsigned short to st_sample_t then shift up the result
* so that the 12-bit sample lives in the most significant
- * 12-bits of the LONG.
+ * 12-bits of the st_sample_t.
* This gets our two samples into the internal format which we
* deposit into the given buffer and adjust our counts respectivly.
*/
@@ -225,10 +225,10 @@
sk->rest -= 3; /* adjust remaining for bytes we just read */
s1 = (unsigned short) (uc1 << 4) | (((uc2 >> 4) & 017));
s2 = (unsigned short) (uc3 << 4) | (( uc2 & 017 ));
- *buf = (LONG) s1;
+ *buf = (st_sample_t) s1;
*buf = (*buf << 20);
buf++; /* sample one is done */
- *buf = (LONG) s2;
+ *buf = (st_sample_t) s2;
*buf = (*buf << 20);
buf++; /* sample two is done */
done += 2; /* adjust converted & stored sample count */
--- a/src/util.c
+++ b/src/util.c
@@ -468,7 +468,7 @@
* # of samples.
* Returns ST_EOF on error.
*/
-int st_parsesamples(ULONG rate, char *str, ULONG *samples, char def)
+int st_parsesamples(st_rate_t rate, char *str, st_size_t *samples, char def)
{int found_samples = 0, found_time = 0;
int time;
--- a/src/vibro.c
+++ b/src/vibro.c
@@ -109,7 +109,7 @@
register int counter, tablen;
int len, done;
short *sinetab;
- LONG l;
+ st_sample_t l;
len = ((*isamp > *osamp) ? *osamp : *isamp);
--- a/src/voc.c
+++ b/src/voc.c
@@ -143,12 +143,12 @@
/* Private data for VOC file */
typedef struct vocstuff {- LONG rest; /* bytes remaining in current block */
- LONG rate; /* rate code (byte) of this chunk */
+ long rest; /* bytes remaining in current block */
+ long rate; /* rate code (byte) of this chunk */
int silent; /* sound or silence? */
- LONG srate; /* rate code (byte) of silence */
- LONG blockseek; /* start of current output block */
- LONG samples; /* number of samples output */
+ long srate; /* rate code (byte) of silence */
+ long blockseek; /* start of current output block */
+ long samples; /* number of samples output */
int size; /* word length of data */
unsigned char channels; /* number of sound channels */
int extended; /* Has an extended block been read? */
@@ -337,7 +337,7 @@
vs_t v = (vs_t) ft->priv;
unsigned char uc;
int sw;
- LONG done = 0;
+ st_ssize_t done = 0;
if (v->samples == 0) {/* No silence packing yet. */
@@ -372,11 +372,11 @@
{vs_t v = (vs_t) ft->priv;
unsigned char uc, block;
- ULONG sblen;
+ uint32_t sblen;
uint16_t new_rate_16;
uint32_t new_rate_32;
int i;
- ULONG trash;
+ uint32_t trash;
v->silent = 0;
while (v->rest == 0) {@@ -395,9 +395,9 @@
st_readb(ft, &uc);
sblen = uc;
st_readb(ft, &uc);
- sblen |= ((LONG) uc) << 8;
+ sblen |= ((uint32_t) uc) << 8;
st_readb(ft, &uc);
- sblen |= ((LONG) uc) << 16;
+ sblen |= ((uint32_t) uc) << 16;
switch(block) {case VOC_DATA:
st_readb(ft, &uc);
@@ -612,7 +612,7 @@
static void blockstop(ft_t ft)
{vs_t v = (vs_t) ft->priv;
- LONG datum;
+ st_sample_t datum;
st_writeb(ft, 0); /* End of file block code */
fseek(ft->fp, v->blockseek, 0); /* seek back to block length */
--- a/src/vol.c
+++ b/src/vol.c
@@ -11,7 +11,6 @@
#include "st_i.h"
#include <math.h> /* exp(), sqrt() */
-#include <limits.h> /* LONG_MAX */
/* type used for computations.
*/
@@ -94,10 +93,13 @@
}
vol->uselimiter = 1; /* ok, we'll use it */
- /* The following equation is derived so that there is no discontinuity in output amplitudes */
- /* and a LONG_MAX input always maps to a LONG_MAX output when the limiter is activated. */
- /* (NOTE: There **WILL** be a discontinuity in the slope of the output amplitudes when using the limiter.) */
- vol->limiterthreshhold = LONG_MAX * (ONE - vol->limitergain) / (fabs(vol->gain) - vol->limitergain);
+ /* The following equation is derived so that there is no
+ * discontinuity in output amplitudes */
+ /* and a ST_SAMPLE_MAX input always maps to a ST_SAMPLE_MAX output
+ * when the limiter is activated. */
+ /* (NOTE: There **WILL** be a discontinuity in the slope
+ * of the output amplitudes when using the limiter.) */
+ vol->limiterthreshhold = ST_SAMPLE_MAX * (ONE - vol->limitergain) / (fabs(vol->gain) - vol->limitergain);
}
@@ -135,20 +137,20 @@
* this could be a function on its own, with clip count and report
* handled by eff_t and caller.
*/
-static LONG clip(vol_t vol, const VOL_FLOAT v)
+static st_sample_t clip(vol_t vol, const VOL_FLOAT v)
{- if (v > LONG_MAX)
+ if (v > ST_SAMPLE_MAX)
{vol->clipped++;
- return LONG_MAX;
+ return ST_SAMPLE_MAX;
}
- else if (v < -LONG_MAX)
+ else if (v < -ST_SAMPLE_MAX)
{vol->clipped++;
- return -LONG_MAX;
+ return -ST_SAMPLE_MAX;
}
/* else */
- return (LONG) v;
+ return (st_sample_t) v;
}
#ifndef MIN
@@ -165,7 +167,7 @@
register VOL_FLOAT gain = vol->gain;
register VOL_FLOAT limiterthreshhold = vol->limiterthreshhold;
register VOL_FLOAT sample;
- register LONG len;
+ register st_size_t len;
len = MIN(*osamp, *isamp);
@@ -182,12 +184,12 @@
if (sample > limiterthreshhold)
{- sample = (LONG_MAX - vol->limitergain * (LONG_MAX - sample));
+ sample = (ST_SAMPLE_MAX - vol->limitergain * (ST_SAMPLE_MAX - sample));
vol->limited++;
}
else if (sample < -limiterthreshhold)
{- sample = -(LONG_MAX - vol->limitergain * (LONG_MAX + sample));
+ sample = -(ST_SAMPLE_MAX - vol->limitergain * (ST_SAMPLE_MAX + sample));
vol->limited++;
}
else
--- a/src/vorbis.c
+++ b/src/vorbis.c
@@ -221,7 +221,7 @@
vorbis_t vb = (vorbis_t) ft->priv;
int i;
int ret;
- LONG l;
+ st_sample_t l;
for(i = 0; i < len; i++) {
@@ -359,9 +359,9 @@
{
vorbis_t vb = (vorbis_t) ft->priv;
vorbis_enc_t *ve = vb->vorbis_enc_data;
- LONG samples = len / ft->info.channels;
+ st_ssize_t samples = len / ft->info.channels;
float **buffer = vorbis_analysis_buffer(&ve->vd, samples);
- LONG i, j;
+ st_ssize_t i, j;
int ret;
int eos = 0;
--- a/src/wav.c
+++ b/src/wav.c
@@ -80,13 +80,13 @@
/* Private data for .wav file */
typedef struct wavstuff {- LONG numSamples; /* samples/channel reading: starts at total count and decremented */
+ st_size_t numSamples; /* samples/channel reading: starts at total count and decremented */
/* writing: starts at 0 and counts samples written */
- ULONG dataLength; /* needed for ADPCM writing */
+ st_size_t dataLength; /* needed for ADPCM writing */
unsigned short formatTag; /* What type of encoding file is using */
unsigned short samplesPerBlock;
unsigned short blockAlign;
- LONG dataStart; /* need to for seeking */
+ st_size_t dataStart; /* need to for seeking */
/* following used by *ADPCM wav files */
unsigned short nCoefs; /* ADPCM: number of coef sets */
@@ -435,13 +435,13 @@
unsigned short wChannels; /* number of channels */
uint32_t dwSamplesPerSecond; /* samples per second per channel */
uint32_t dwAvgBytesPerSec;/* estimate of bytes per second needed */
- unsigned short wBitsPerSample; /* bits per sample */
- unsigned short wFmtSize;
- unsigned short wExtSize = 0; /* extended field for non-PCM */
+ uint16_t wBitsPerSample; /* bits per sample */
+ uint32_t wFmtSize;
+ uint16_t wExtSize = 0; /* extended field for non-PCM */
uint32_t dwDataLength; /* length of sound data in bytes */
- ULONG bytesPerBlock = 0;
- ULONG bytespersample; /* bytes per sample (per channel */
+ st_size_t bytesPerBlock = 0;
+ int bytespersample; /* bytes per sample (per channel */
char text[256];
uint32_t dwLoopPos;
@@ -629,7 +629,6 @@
switch (wav->formatTag)
{- /* ULONG max_spb; */
case WAVE_FORMAT_ADPCM:
if (wExtSize < 4)
{@@ -966,7 +965,7 @@
st_ssize_t st_wavread(ft_t ft, st_sample_t *buf, st_ssize_t len)
{wav_t wav = (wav_t) ft->priv;
- LONG done;
+ st_ssize_t done;
ft->st_errno = ST_SUCCESS;
@@ -1001,7 +1000,7 @@
wav->samplePtr = wav->samples;
}
- /* Copy interleaved data into buf, converting short to LONG */
+ /* Copy interleaved data into buf, converting to st_sample_t */
{short *p, *top;
int ct;
@@ -1175,8 +1174,8 @@
ULAW, ALAW formats:
36 - 37 wExtSize = 0 the length of the format extension
38 - 41 'fact'
-42 - 45 wFactSize = 4 length of the fact chunk minus 8 byte header
-46 - 49 wSamplesWritten actual number of samples written out
+42 - 45 dwFactSize = 4 length of the fact chunk minus 8 byte header
+46 - 49 dwSamplesWritten actual number of samples written out
50 - 53 'data'
54 - 57 dwDataLength length of data chunk minus 8 byte header
58 - (dwDataLength + 57) the data
@@ -1186,8 +1185,8 @@
36 - 37 wExtSize = 2 the length in bytes of the format-dependent extension
38 - 39 320 number of samples per block
40 - 43 'fact'
-44 - 47 wFactSize = 4 length of the fact chunk minus 8 byte header
-48 - 51 wSamplesWritten actual number of samples written out
+44 - 47 dwFactSize = 4 length of the fact chunk minus 8 byte header
+48 - 51 dwSamplesWritten actual number of samples written out
52 - 55 'data'
56 - 59 dwDataLength length of data chunk minus 8 byte header
60 - (dwDataLength + 59) the data
@@ -1200,7 +1199,7 @@
and padded compressed formats:
wRiffLength - (riff header) the length of the file, minus 8
-wSamplesWritten - (fact header) the number of samples written (after padding
+dwSamplesWritten - (fact header) the number of samples written (after padding
to a complete block eg for GSM)
dwDataLength - (data chunk header) the number of (valid) data bytes written
@@ -1208,273 +1207,273 @@
static int wavwritehdr(ft_t ft, int second_header)
{- wav_t wav = (wav_t) ft->priv;
+ wav_t wav = (wav_t) ft->priv;
- /* variables written to wav file header */
- /* RIFF header */
- ULONG wRiffLength ; /* length of file after 8 byte riff header */
- /* fmt chunk */
- ULONG wFmtSize = 16; /* size field of the fmt chunk */
- unsigned short wFormatTag = 0; /* data format */
- unsigned short wChannels; /* number of channels */
- ULONG dwSamplesPerSecond; /* samples per second per channel*/
- uint32_t dwAvgBytesPerSec=0; /* estimate of bytes per second needed */
- unsigned short wBlockAlign=0; /* byte alignment of a basic sample block */
- unsigned short wBitsPerSample=0; /* bits per sample */
- /* fmt chunk extension (not PCM) */
- unsigned short wExtSize=0; /* extra bytes in the format extension */
- unsigned short wSamplesPerBlock; /* samples per channel per block */
- /* wSamplesPerBlock and other things may go into format extension */
+ /* variables written to wav file header */
+ /* RIFF header */
+ uint32_t wRiffLength ; /* length of file after 8 byte riff header */
+ /* fmt chunk */
+ uint16_t wFmtSize = 16; /* size field of the fmt chunk */
+ uint16_t wFormatTag = 0; /* data format */
+ uint16_t wChannels; /* number of channels */
+ uint32_t dwSamplesPerSecond; /* samples per second per channel*/
+ uint32_t dwAvgBytesPerSec=0; /* estimate of bytes per second needed */
+ uint16_t wBlockAlign=0; /* byte alignment of a basic sample block */
+ uint16_t wBitsPerSample=0; /* bits per sample */
+ /* fmt chunk extension (not PCM) */
+ uint16_t wExtSize=0; /* extra bytes in the format extension */
+ uint16_t wSamplesPerBlock; /* samples per channel per block */
+ /* wSamplesPerBlock and other things may go into format extension */
- /* fact chunk (not PCM) */
- ULONG wFactSize=4; /* length of the fact chunk */
- ULONG wSamplesWritten=0; /* windows doesnt seem to use this*/
+ /* fact chunk (not PCM) */
+ uint32_t dwFactSize=4; /* length of the fact chunk */
+ uint32_t dwSamplesWritten=0; /* windows doesnt seem to use this*/
- /* data chunk */
- uint32_t dwDataLength=0x7ffff000L; /* length of sound data in bytes */
- /* end of variables written to header */
+ /* data chunk */
+ uint32_t dwDataLength=0x7ffff000L; /* length of sound data in bytes */
+ /* end of variables written to header */
- /* internal variables, intermediate values etc */
- ULONG bytespersample; /* (uncompressed) bytes per sample (per channel) */
- ULONG blocksWritten = 0;
+ /* internal variables, intermediate values etc */
+ int bytespersample; /* (uncompressed) bytes per sample (per channel) */
+ long blocksWritten = 0;
- dwSamplesPerSecond = ft->info.rate;
- wChannels = ft->info.channels;
+ dwSamplesPerSecond = ft->info.rate;
+ wChannels = ft->info.channels;
- /* Check to see if encoding is ADPCM or not. If ADPCM
- * possibly override the size to be bytes. It isn't needed
- * by this routine will look nicer (and more correct)
- * on verbose output.
- */
- if ((ft->info.encoding == ST_ENCODING_ADPCM ||
- ft->info.encoding == ST_ENCODING_IMA_ADPCM ||
- ft->info.encoding == ST_ENCODING_GSM) &&
- ft->info.size != ST_SIZE_BYTE)
- {- st_warn("Overriding output size to bytes for compressed data.");- ft->info.size = ST_SIZE_BYTE;
- }
+ /* Check to see if encoding is ADPCM or not. If ADPCM
+ * possibly override the size to be bytes. It isn't needed
+ * by this routine will look nicer (and more correct)
+ * on verbose output.
+ */
+ if ((ft->info.encoding == ST_ENCODING_ADPCM ||
+ ft->info.encoding == ST_ENCODING_IMA_ADPCM ||
+ ft->info.encoding == ST_ENCODING_GSM) &&
+ ft->info.size != ST_SIZE_BYTE)
+ {+ st_warn("Overriding output size to bytes for compressed data.");+ ft->info.size = ST_SIZE_BYTE;
+ }
- switch (ft->info.size)
- {- case ST_SIZE_BYTE:
- wBitsPerSample = 8;
- if (ft->info.encoding != ST_ENCODING_UNSIGNED &&
- ft->info.encoding != ST_ENCODING_ULAW &&
- ft->info.encoding != ST_ENCODING_ALAW &&
- ft->info.encoding != ST_ENCODING_GSM &&
- ft->info.encoding != ST_ENCODING_ADPCM &&
- ft->info.encoding != ST_ENCODING_IMA_ADPCM)
- {- st_warn("Do not support %s with 8-bit data. Forcing to unsigned",st_encodings_str[(unsigned char)ft->info.encoding]);- ft->info.encoding = ST_ENCODING_UNSIGNED;
- }
- break;
- case ST_SIZE_WORD:
- wBitsPerSample = 16;
- if (ft->info.encoding != ST_ENCODING_SIGN2)
- {- st_warn("Do not support %s with 16-bit data. Forcing to Signed.",st_encodings_str[(unsigned char)ft->info.encoding]);- ft->info.encoding = ST_ENCODING_SIGN2;
- }
- break;
- case ST_SIZE_DWORD:
- wBitsPerSample = 32;
- if (ft->info.encoding != ST_ENCODING_SIGN2)
- {- st_warn("Do not support %s with 16-bit data. Forcing to Signed.",st_encodings_str[(unsigned char)ft->info.encoding]);- ft->info.encoding = ST_ENCODING_SIGN2;
- }
+ switch (ft->info.size)
+ {+ case ST_SIZE_BYTE:
+ wBitsPerSample = 8;
+ if (ft->info.encoding != ST_ENCODING_UNSIGNED &&
+ ft->info.encoding != ST_ENCODING_ULAW &&
+ ft->info.encoding != ST_ENCODING_ALAW &&
+ ft->info.encoding != ST_ENCODING_GSM &&
+ ft->info.encoding != ST_ENCODING_ADPCM &&
+ ft->info.encoding != ST_ENCODING_IMA_ADPCM)
+ {+ st_warn("Do not support %s with 8-bit data. Forcing to unsigned",st_encodings_str[(unsigned char)ft->info.encoding]);+ ft->info.encoding = ST_ENCODING_UNSIGNED;
+ }
+ break;
+ case ST_SIZE_WORD:
+ wBitsPerSample = 16;
+ if (ft->info.encoding != ST_ENCODING_SIGN2)
+ {+ st_warn("Do not support %s with 16-bit data. Forcing to Signed.",st_encodings_str[(unsigned char)ft->info.encoding]);+ ft->info.encoding = ST_ENCODING_SIGN2;
+ }
+ break;
+ case ST_SIZE_DWORD:
+ wBitsPerSample = 32;
+ if (ft->info.encoding != ST_ENCODING_SIGN2)
+ {+ st_warn("Do not support %s with 16-bit data. Forcing to Signed.",st_encodings_str[(unsigned char)ft->info.encoding]);+ ft->info.encoding = ST_ENCODING_SIGN2;
+ }
- break;
- default:
- st_warn("Do not support %s in WAV files. Forcing to Signed Words.",st_sizes_str[(unsigned char)ft->info.size]);- ft->info.encoding = ST_ENCODING_SIGN2;
- ft->info.size = ST_SIZE_WORD;
- wBitsPerSample = 16;
- break;
- }
+ break;
+ default:
+ st_warn("Do not support %s in WAV files. Forcing to Signed Words.",st_sizes_str[(unsigned char)ft->info.size]);+ ft->info.encoding = ST_ENCODING_SIGN2;
+ ft->info.size = ST_SIZE_WORD;
+ wBitsPerSample = 16;
+ break;
+ }
- wSamplesPerBlock = 1; /* common default for PCM data */
+ wSamplesPerBlock = 1; /* common default for PCM data */
- switch (ft->info.encoding)
- {- case ST_ENCODING_UNSIGNED:
- case ST_ENCODING_SIGN2:
- wFormatTag = WAVE_FORMAT_PCM;
- bytespersample = (wBitsPerSample + 7)/8;
- wBlockAlign = wChannels * bytespersample;
- break;
- case ST_ENCODING_ALAW:
- wFormatTag = WAVE_FORMAT_ALAW;
- wBlockAlign = wChannels;
- break;
- case ST_ENCODING_ULAW:
- wFormatTag = WAVE_FORMAT_MULAW;
- wBlockAlign = wChannels;
- break;
- case ST_ENCODING_IMA_ADPCM:
- if (wChannels>16)
- {- st_fail_errno(ft,ST_EOF,"Channels(%d) must be <= 16\n",wChannels);
- return ST_EOF;
- }
- wFormatTag = WAVE_FORMAT_IMA_ADPCM;
- wBlockAlign = wChannels * 256; /* reasonable default */
- wBitsPerSample = 4;
- wExtSize = 2;
- wSamplesPerBlock = ImaSamplesIn(0, wChannels, wBlockAlign, 0);
- break;
- case ST_ENCODING_ADPCM:
- if (wChannels>16)
- {- st_fail_errno(ft,ST_EOF,"Channels(%d) must be <= 16\n",wChannels);
- return ST_EOF;
- }
- wFormatTag = WAVE_FORMAT_ADPCM;
- wBlockAlign = wChannels * 128; /* reasonable default */
- wBitsPerSample = 4;
- wExtSize = 4+4*7; /* Ext fmt data length */
- wSamplesPerBlock = AdpcmSamplesIn(0, wChannels, wBlockAlign, 0);
- break;
- case ST_ENCODING_GSM:
+ switch (ft->info.encoding)
+ {+ case ST_ENCODING_UNSIGNED:
+ case ST_ENCODING_SIGN2:
+ wFormatTag = WAVE_FORMAT_PCM;
+ bytespersample = (wBitsPerSample + 7)/8;
+ wBlockAlign = wChannels * bytespersample;
+ break;
+ case ST_ENCODING_ALAW:
+ wFormatTag = WAVE_FORMAT_ALAW;
+ wBlockAlign = wChannels;
+ break;
+ case ST_ENCODING_ULAW:
+ wFormatTag = WAVE_FORMAT_MULAW;
+ wBlockAlign = wChannels;
+ break;
+ case ST_ENCODING_IMA_ADPCM:
+ if (wChannels>16)
+ {+ st_fail_errno(ft,ST_EOF,"Channels(%d) must be <= 16\n",wChannels);
+ return ST_EOF;
+ }
+ wFormatTag = WAVE_FORMAT_IMA_ADPCM;
+ wBlockAlign = wChannels * 256; /* reasonable default */
+ wBitsPerSample = 4;
+ wExtSize = 2;
+ wSamplesPerBlock = ImaSamplesIn(0, wChannels, wBlockAlign, 0);
+ break;
+ case ST_ENCODING_ADPCM:
+ if (wChannels>16)
+ {+ st_fail_errno(ft,ST_EOF,"Channels(%d) must be <= 16\n",wChannels);
+ return ST_EOF;
+ }
+ wFormatTag = WAVE_FORMAT_ADPCM;
+ wBlockAlign = wChannels * 128; /* reasonable default */
+ wBitsPerSample = 4;
+ wExtSize = 4+4*7; /* Ext fmt data length */
+ wSamplesPerBlock = AdpcmSamplesIn(0, wChannels, wBlockAlign, 0);
+ break;
+ case ST_ENCODING_GSM:
#ifdef ENABLE_GSM
- if (wChannels!=1)
- {- st_warn("Overriding GSM audio from %d channel to 1\n",wChannels);- wChannels = ft->info.channels = 1;
- }
- wFormatTag = WAVE_FORMAT_GSM610;
- /* dwAvgBytesPerSec = 1625*(dwSamplesPerSecond/8000.)+0.5; */
- wBlockAlign=65;
- wBitsPerSample=0; /* not representable as int */
- wExtSize=2; /* length of format extension */
- wSamplesPerBlock = 320;
+ if (wChannels!=1)
+ {+ st_warn("Overriding GSM audio from %d channel to 1\n",wChannels);+ wChannels = ft->info.channels = 1;
+ }
+ wFormatTag = WAVE_FORMAT_GSM610;
+ /* dwAvgBytesPerSec = 1625*(dwSamplesPerSecond/8000.)+0.5; */
+ wBlockAlign=65;
+ wBitsPerSample=0; /* not representable as int */
+ wExtSize=2; /* length of format extension */
+ wSamplesPerBlock = 320;
#else
- st_fail_errno(ft,ST_EOF,"sorry, no GSM6.10 support, recompile sox with gsm library");
- return ST_EOF;
+ st_fail_errno(ft,ST_EOF,"sorry, no GSM6.10 support, recompile sox with gsm library");
+ return ST_EOF;
#endif
- break;
- }
- wav->formatTag = wFormatTag;
- wav->blockAlign = wBlockAlign;
- wav->samplesPerBlock = wSamplesPerBlock;
+ break;
+ }
+ wav->formatTag = wFormatTag;
+ wav->blockAlign = wBlockAlign;
+ wav->samplesPerBlock = wSamplesPerBlock;
- if (!second_header) { /* adjust for blockAlign */- blocksWritten = dwDataLength/wBlockAlign;
- dwDataLength = blocksWritten * wBlockAlign;
- wSamplesWritten = blocksWritten * wSamplesPerBlock;
- } else { /* fixup with real length */- wSamplesWritten = wav->numSamples;
- switch(wFormatTag)
- {- case WAVE_FORMAT_ADPCM:
- case WAVE_FORMAT_IMA_ADPCM:
- dwDataLength = wav->dataLength;
- break;
+ if (!second_header) { /* adjust for blockAlign */+ blocksWritten = dwDataLength/wBlockAlign;
+ dwDataLength = blocksWritten * wBlockAlign;
+ dwSamplesWritten = blocksWritten * wSamplesPerBlock;
+ } else { /* fixup with real length */+ dwSamplesWritten = wav->numSamples;
+ switch(wFormatTag)
+ {+ case WAVE_FORMAT_ADPCM:
+ case WAVE_FORMAT_IMA_ADPCM:
+ dwDataLength = wav->dataLength;
+ break;
#ifdef ENABLE_GSM
- case WAVE_FORMAT_GSM610:
- /* intentional case fallthrough! */
+ case WAVE_FORMAT_GSM610:
+ /* intentional case fallthrough! */
#endif
- default:
- wSamplesWritten /= wChannels; /* because how rawwrite()'s work */
- blocksWritten = (wSamplesWritten+wSamplesPerBlock-1)/wSamplesPerBlock;
- dwDataLength = blocksWritten * wBlockAlign;
- }
+ default:
+ dwSamplesWritten /= wChannels; /* because how rawwrite()'s work */
+ blocksWritten = (dwSamplesWritten+wSamplesPerBlock-1)/wSamplesPerBlock;
+ dwDataLength = blocksWritten * wBlockAlign;
}
+ }
#ifdef ENABLE_GSM
- if (wFormatTag == WAVE_FORMAT_GSM610)
- dwDataLength = (dwDataLength+1) & ~1; /*round up to even */
+ if (wFormatTag == WAVE_FORMAT_GSM610)
+ dwDataLength = (dwDataLength+1) & ~1; /*round up to even */
#endif
- if (wFormatTag != WAVE_FORMAT_PCM)
- wFmtSize += 2+wExtSize; /* plus ExtData */
+ if (wFormatTag != WAVE_FORMAT_PCM)
+ wFmtSize += 2+wExtSize; /* plus ExtData */
- wRiffLength = 4 + (8+wFmtSize) + (8+dwDataLength);
- if (wFormatTag != WAVE_FORMAT_PCM) /* PCM omits the "fact" chunk */
- wRiffLength += (8+wFactSize);
-
- /* dwAvgBytesPerSec <-- this is BEFORE compression, isn't it? guess not. */
- dwAvgBytesPerSec = (double)wBlockAlign*ft->info.rate / (double)wSamplesPerBlock + 0.5;
+ wRiffLength = 4 + (8+wFmtSize) + (8+dwDataLength);
+ if (wFormatTag != WAVE_FORMAT_PCM) /* PCM omits the "fact" chunk */
+ wRiffLength += (8+dwFactSize);
- /* figured out header info, so write it */
- st_writes(ft, "RIFF");
- st_writedw(ft, wRiffLength);
- st_writes(ft, "WAVE");
- st_writes(ft, "fmt ");
- st_writedw(ft, wFmtSize);
- st_writew(ft, wFormatTag);
- st_writew(ft, wChannels);
- st_writedw(ft, dwSamplesPerSecond);
- st_writedw(ft, dwAvgBytesPerSec);
- st_writew(ft, wBlockAlign);
- st_writew(ft, wBitsPerSample); /* end info common to all fmts */
+ /* dwAvgBytesPerSec <-- this is BEFORE compression, isn't it? guess not. */
+ dwAvgBytesPerSec = (double)wBlockAlign*ft->info.rate / (double)wSamplesPerBlock + 0.5;
- /* if not PCM, we need to write out wExtSize even if wExtSize=0 */
- if (wFormatTag != WAVE_FORMAT_PCM)
- st_writew(ft,wExtSize);
+ /* figured out header info, so write it */
+ st_writes(ft, "RIFF");
+ st_writedw(ft, wRiffLength);
+ st_writes(ft, "WAVE");
+ st_writes(ft, "fmt ");
+ st_writedw(ft, wFmtSize);
+ st_writew(ft, wFormatTag);
+ st_writew(ft, wChannels);
+ st_writedw(ft, dwSamplesPerSecond);
+ st_writedw(ft, dwAvgBytesPerSec);
+ st_writew(ft, wBlockAlign);
+ st_writew(ft, wBitsPerSample); /* end info common to all fmts */
- switch (wFormatTag)
- {+ /* if not PCM, we need to write out wExtSize even if wExtSize=0 */
+ if (wFormatTag != WAVE_FORMAT_PCM)
+ st_writew(ft,wExtSize);
+
+ switch (wFormatTag)
+ {int i;
case WAVE_FORMAT_IMA_ADPCM:
- st_writew(ft, wSamplesPerBlock);
- break;
+ st_writew(ft, wSamplesPerBlock);
+ break;
case WAVE_FORMAT_ADPCM:
- st_writew(ft, wSamplesPerBlock);
- st_writew(ft, 7); /* nCoefs */
- for (i=0; i<7; i++) {- st_writew(ft, iCoef[i][0]);
- st_writew(ft, iCoef[i][1]);
- }
- break;
+ st_writew(ft, wSamplesPerBlock);
+ st_writew(ft, 7); /* nCoefs */
+ for (i=0; i<7; i++) {+ st_writew(ft, iCoef[i][0]);
+ st_writew(ft, iCoef[i][1]);
+ }
+ break;
#ifdef ENABLE_GSM
case WAVE_FORMAT_GSM610:
- st_writew(ft, wSamplesPerBlock);
- break;
+ st_writew(ft, wSamplesPerBlock);
+ break;
#endif
default:
- break;
- }
+ break;
+ }
- /* if not PCM, write the 'fact' chunk */
- if (wFormatTag != WAVE_FORMAT_PCM){- st_writes(ft, "fact");
- st_writedw(ft,wFactSize);
- st_writedw(ft,wSamplesWritten);
- }
+ /* if not PCM, write the 'fact' chunk */
+ if (wFormatTag != WAVE_FORMAT_PCM){+ st_writes(ft, "fact");
+ st_writedw(ft,dwFactSize);
+ st_writedw(ft,dwSamplesWritten);
+ }
- st_writes(ft, "data");
- st_writedw(ft, dwDataLength); /* data chunk size */
+ st_writes(ft, "data");
+ st_writedw(ft, dwDataLength); /* data chunk size */
- if (!second_header) {- st_report("Writing Wave file: %s format, %d channel%s, %d samp/sec",- wav_format_str(wFormatTag), wChannels,
- wChannels == 1 ? "" : "s", dwSamplesPerSecond);
- st_report(" %d byte/sec, %d block align, %d bits/samp",- dwAvgBytesPerSec, wBlockAlign, wBitsPerSample);
- } else {- st_report("Finished writing Wave file, %u data bytes %u samples\n",- dwDataLength,wav->numSamples);
+ if (!second_header) {+ st_report("Writing Wave file: %s format, %d channel%s, %d samp/sec",+ wav_format_str(wFormatTag), wChannels,
+ wChannels == 1 ? "" : "s", dwSamplesPerSecond);
+ st_report(" %d byte/sec, %d block align, %d bits/samp",+ dwAvgBytesPerSec, wBlockAlign, wBitsPerSample);
+ } else {+ st_report("Finished writing Wave file, %u data bytes %u samples\n",+ dwDataLength,wav->numSamples);
#ifdef ENABLE_GSM
- if (wFormatTag == WAVE_FORMAT_GSM610){- st_report("GSM6.10 format: %u blocks %u padded samples %u padded data bytes\n",- blocksWritten, wSamplesWritten, dwDataLength);
- if (wav->gsmbytecount != dwDataLength)
- st_warn("help ! internal inconsistency - data_written %u gsmbytecount %u",- dwDataLength, wav->gsmbytecount);
+ if (wFormatTag == WAVE_FORMAT_GSM610){+ st_report("GSM6.10 format: %u blocks %u padded samples %u padded data bytes\n",+ blocksWritten, dwSamplesWritten, dwDataLength);
+ if (wav->gsmbytecount != dwDataLength)
+ st_warn("help ! internal inconsistency - data_written %u gsmbytecount %u",+ dwDataLength, wav->gsmbytecount);
- }
-#endif
}
- return ST_SUCCESS;
+#endif
+ }
+ return ST_SUCCESS;
}
st_ssize_t st_wavwrite(ft_t ft, st_sample_t *buf, st_ssize_t len)
{wav_t wav = (wav_t) ft->priv;
- LONG total_len = len;
+ st_ssize_t total_len = len;
ft->st_errno = ST_SUCCESS;
--- a/src/wve.c
+++ b/src/wve.c
@@ -22,7 +22,7 @@
short padding;
short repeats;
/* For seeking */
- LONG dataStart;
+ st_size_t dataStart;
} *wve_t;
static void wvewriteheader(ft_t ft);
@@ -41,7 +41,7 @@
short version;
int rc;
- ULONG trash;
+ uint32_t trash;
/* Needed for rawread() */
rc = st_rawstartread(ft);
--
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