ref: 6ab46f5ce1be3069d817e83304b716c728a30b13
parent: 8c3e9929c87493382bf82983aa1cb0b20aa99e3b
author: robs <robs>
date: Sat Nov 25 13:30:06 EST 2006
Split -V messages in two via st_report and newly created st_debug
--- a/Changelog
+++ b/Changelog
@@ -47,6 +47,8 @@
o Make sox and soxmix a single binary. (Reuben Thomas)
o Consistent (and hopefully complete) clipping detection and
reporting. (robs)
+ o -V now gives only user-relevant messages, use -V -V to get
+ developer-relevant messages. (robs)
sox-12.18.2
-----------
--- a/sox.1
+++ b/sox.1
@@ -154,13 +154,48 @@
.TP 10
\fB--version\fR
Print version number and exit.
-.TP 10
-\fB-V\fR
-Print a description of processing phases.
+.IP "\fB\-V[level]\fP"
+Set verbosity.
+.I Sox
+prints messages to the console (stderr) according to the following
+verbosity levels:
+.IP
+.RS
+.IP 0
+No messages are printed at all; use the exit status to determine
+if an error has ocurred.
+.IP 1
+Only error messages are printed. These are generated if
+.I SoX
+cannot complete the requested commands.
+.IP 2
+Warning messages are also printed. These are generated if
+.I SoX
+can complete the requested commands,
+but not exactly according to the requested command parameters,
+or if clipping occurs.
+.IP 3
+Descriptions of
+.I SoX's
+processing phases are also printed.
Useful for figuring out exactly how
.I SoX
-.PP
is mangling your sound samples.
+.IP "4 and above"
+Messages to help with debugging
+.I SoX
+are also printed.
+.RE
+.IP
+By default, the verbosity level is set to 2. Each occurrence of the \fI-V\fR
+option increases the verbosity level by 1. Alternatively, the verbosity
+level can be set to an absolute number by specifying it immediately after
+the
+.I -V
+e.g.
+.I -V0
+sets it to 0.
+.IP
.PP
\fBFormat options:\fR
.PP
--- a/src/8svx.c
+++ b/src/8svx.c
@@ -111,7 +111,7 @@
return(ST_EOF);
}
chunk_buf[chunksize] = '\0';
- st_report("%s",chunk_buf);+ st_debug("%s",chunk_buf);free(chunk_buf);
continue;
@@ -134,7 +134,7 @@
return(ST_EOF);
}
chunk_buf[chunksize] = '\0';
- st_report("%s",chunk_buf);+ st_debug("%s",chunk_buf);free(chunk_buf);
continue;
--- a/src/FFT.c
+++ b/src/FFT.c
@@ -51,6 +51,8 @@
#include <math.h>
#include <assert.h>
+#include "st_i.h"
+
#include "FFT.h"
int **gFFTBitTable = NULL;
@@ -75,7 +77,7 @@
int i;
if (PowerOfTwo < 2) {- fprintf(stderr, "Error: FFT called with size %d\n", PowerOfTwo);
+ st_debug("Error: FFT called with size %d", PowerOfTwo);exit(1);
}
@@ -140,7 +142,7 @@
float tr, ti; /* temp real, temp imaginary */
if (!IsPowerOfTwo(NumSamples)) {- fprintf(stderr, "%d is not a power of two\n", NumSamples);
+ st_debug("%d is not a power of two", NumSamples);exit(1);
}
--- a/src/aiff.c
+++ b/src/aiff.c
@@ -366,7 +366,7 @@
if (st_eof(ft))
break;
buf[4] = 0;
- st_report("AIFFstartread: ignoring '%s' chunk\n", buf);+ st_debug("AIFFstartread: ignoring '%s' chunk", buf);st_readdw(ft, &chunksize);
if (st_eof(ft))
break;
@@ -441,7 +441,7 @@
|| (ft->info.rate == 0)
|| (ft->info.encoding == -1)
|| (ft->info.size == -1)) {- st_report("You must specify # channels, sample rate, signed/unsigned,\n");+ st_report("You must specify # channels, sample rate, signed/unsigned,"); st_report("and 8/16 on the command line.");st_fail_errno(ft,ST_EFMT,"Bogus AIFF file: no COMM section.");
return(ST_EOF);
@@ -460,12 +460,12 @@
if (foundmark && !foundinstr)
{- st_report("Ignoring MARK chunk since no INSTR found.");+ st_debug("Ignoring MARK chunk since no INSTR found.");foundmark = 0;
}
if (!foundmark && foundinstr)
{- st_report("Ignoring INSTR chunk since no MARK found.");+ st_debug("Ignoring INSTR chunk since no MARK found.");foundinstr = 0;
}
if (foundmark && foundinstr) {@@ -524,7 +524,7 @@
int loopNum;
if(ft->instr.nloops > 0)
- st_report("AIFF Loop markers:\n");+ st_report("AIFF Loop markers:"); for(loopNum = 0; loopNum < ft->instr.nloops; loopNum++) { if (ft->loops[loopNum].count) { st_report("Loop %d: start: %6d", loopNum, ft->loops[loopNum].start);@@ -533,15 +533,15 @@
st_report(" count: %6d", ft->loops[loopNum].count); st_report(" type: "); switch(ft->loops[loopNum].type & ~ST_LOOP_SUSTAIN_DECAY) {- case 0: st_report("off\n"); break;- case 1: st_report("forward\n"); break;- case 2: st_report("forward/backward\n"); break;+ case 0: st_report("off"); break;+ case 1: st_report("forward"); break;+ case 2: st_report("forward/backward"); break;}
}
}
- st_report("Unity MIDI Note: %d\n", ft->instr.MIDInote);- st_report("Low MIDI Note: %d\n", ft->instr.MIDIlow);- st_report("High MIDI Note: %d\n", ft->instr.MIDIhi);+ st_report("Unity MIDI Note: %d", ft->instr.MIDInote);+ st_report("Low MIDI Note: %d", ft->instr.MIDIlow);+ st_report("High MIDI Note: %d", ft->instr.MIDIhi);}
/* Process a text chunk, allocate memory, display it if verbose and return */
@@ -572,7 +572,7 @@
return(ST_EOF);
}
}
- st_report("%-10s \"%s\"\n", chunkDescription, *text);+ st_debug("%-10s \"%s\"", chunkDescription, *text);return(ST_SUCCESS);
}
@@ -626,7 +626,7 @@
}
}
}
- st_report("%-10s \"%s\"\n", chunkDescription, *text);+ st_debug("%-10s \"%s\"", chunkDescription, *text);/* make sure we read the whole chunk */
if (totalReadLength < chunksize) {int i;
@@ -671,10 +671,10 @@
if (st_eof(ft))
break;
buf[4] = '\0';
- st_warn("Ignoring AIFF tail chunk: '%s', %d bytes long\n", + st_warn("Ignoring AIFF tail chunk: '%s', %d bytes long", buf, chunksize);
if (! strcmp(buf, "MARK") || ! strcmp(buf, "INST"))
- st_warn(" You're stripping MIDI/loop info!\n");+ st_warn(" You're stripping MIDI/loop info!");while (chunksize-- > 0)
{if (st_readb(ft, (unsigned char *)&trash) == ST_EOF)
@@ -1042,7 +1042,7 @@
char buf[10];
ConvertToIeeeExtended(x, buf);
/*
- st_report("converted %g to %o %o %o %o %o %o %o %o %o %o",+ st_debug("converted %g to %o %o %o %o %o %o %o %o %o %o",x,
buf[0], buf[1], buf[2], buf[3], buf[4],
buf[5], buf[6], buf[7], buf[8], buf[9]);
--- a/src/alsa.c
+++ b/src/alsa.c
@@ -452,7 +452,7 @@
{if (xrun_recovery(alsa->pcm_handle, err) < 0)
{- st_fail_errno(ft, ST_EPERM, "ALSA write error\n");
+ st_fail_errno(ft, ST_EPERM, "ALSA write error");
return ST_EOF;
}
}
--- a/src/au.c
+++ b/src/au.c
@@ -99,7 +99,7 @@
*size = ST_SIZE_32BIT;
break;
default:
- st_report("encoding: 0x%lx", encoding);+ st_debug("encoding: 0x%lx", encoding);return(ST_EOF);
}
return(ST_SUCCESS);
@@ -171,17 +171,17 @@
* endianess. Its not hurting though.
*/
ft->swap = ft->swap ? 0 : 1;
- st_report("Found inverted DEC magic word. Swapping bytes.");+ st_debug("Found inverted DEC magic word. Swapping bytes.");}
else if (magic == SUN_INV_MAGIC) {ft->swap = ft->swap ? 0 : 1;
- st_report("Found inverted Sun/NeXT magic word. Swapping bytes.");+ st_debug("Found inverted Sun/NeXT magic word. Swapping bytes.");}
else if (magic == SUN_MAGIC) {- st_report("Found Sun/NeXT magic word");+ st_debug("Found Sun/NeXT magic word");}
else if (magic == DEC_MAGIC) {- st_report("Found DEC magic word");+ st_debug("Found DEC magic word");}
else
{@@ -424,7 +424,7 @@
if ((encoding = st_ausunencoding(ft->info.size, ft->info.encoding)) == -1) { st_report("Unsupported output encoding/size for Sun/NeXT header or .AU format not specified."); st_report("Only U-law, A-law signed bytes, and signed words are supported.");- st_report("Defaulting to 8khz u-law\n");+ st_report("Defaulting to 8khz u-law");encoding = SUN_ULAW;
ft->info.encoding = ST_ENCODING_ULAW;
ft->info.size = ST_SIZE_BYTE;
--- a/src/auto.c
+++ b/src/auto.c
@@ -185,7 +185,7 @@
if(!type)
{- st_fail_errno(ft,ST_EFMT, "Could not determine file type.\n");
+ st_fail_errno(ft,ST_EFMT, "Could not determine file type.");
return (ST_EOF);
}
free(ft->filetype);
@@ -193,11 +193,11 @@
rc = st_gettype(ft); /* Change ft->h to the new format */
if(rc != ST_SUCCESS)
{- st_fail_errno(ft,ST_EFMT,"Do not understand format type: %s\n",type);
+ st_fail_errno(ft,ST_EFMT,"Do not understand format type: %s",type);
return (rc);
}
- st_report("Detected file format type: %s\n", type);+ st_report("Detected file format type: %s", type);return ((* ft->h->startread)(ft));
}
--- a/src/band.c
+++ b/src/band.c
@@ -93,7 +93,7 @@
band_t band = (band_t) effp->priv;
if (band->center > effp->ininfo.rate/2)
{- st_fail("Band: center must be < minimum data rate/2\n");+ st_fail("Band: center must be < minimum data rate/2");return (ST_EOF);
}
--- a/src/chorus.c
+++ b/src/chorus.c
@@ -146,17 +146,17 @@
if ( chorus->in_gain < 0.0 )
{- st_fail("chorus: gain-in must be positive!\n");+ st_fail("chorus: gain-in must be positive!");return (ST_EOF);
}
if ( chorus->in_gain > 1.0 )
{- st_fail("chorus: gain-in must be less than 1.0!\n");+ st_fail("chorus: gain-in must be less than 1.0!");return (ST_EOF);
}
if ( chorus->out_gain < 0.0 )
{- st_fail("chorus: gain-out must be positive!\n");+ st_fail("chorus: gain-out must be positive!");return (ST_EOF);
}
for ( i = 0; i < chorus->num_chorus; i++ ) {@@ -167,42 +167,42 @@
if ( chorus->delay[i] < 20.0 )
{- st_fail("chorus: delay must be more than 20.0 msec!\n");+ st_fail("chorus: delay must be more than 20.0 msec!");return (ST_EOF);
}
if ( chorus->delay[i] > 100.0 )
{- st_fail("chorus: delay must be less than 100.0 msec!\n");+ st_fail("chorus: delay must be less than 100.0 msec!");return (ST_EOF);
}
if ( chorus->speed[i] < 0.1 )
{- st_fail("chorus: speed must be more than 0.1 Hz!\n");+ st_fail("chorus: speed must be more than 0.1 Hz!");return (ST_EOF);
}
if ( chorus->speed[i] > 5.0 )
{- st_fail("chorus: speed must be less than 5.0 Hz!\n");+ st_fail("chorus: speed must be less than 5.0 Hz!");return (ST_EOF);
}
if ( chorus->depth[i] < 0.0 )
{- st_fail("chorus: delay must be more positive!\n");+ st_fail("chorus: delay must be more positive!");return (ST_EOF);
}
if ( chorus->depth[i] > 10.0 )
{- st_fail("chorus: delay must be less than 10.0 msec!\n");+ st_fail("chorus: delay must be less than 10.0 msec!");return (ST_EOF);
}
if ( chorus->decay[i] < 0.0 )
{- st_fail("chorus: decay must be positive!\n" );+ st_fail("chorus: decay must be positive!" );return (ST_EOF);
}
if ( chorus->decay[i] > 1.0 )
{- st_fail("chorus: decay must be less that 1.0!\n" );+ st_fail("chorus: decay must be less that 1.0!" );return (ST_EOF);
}
chorus->length[i] = effp->ininfo.rate / chorus->speed[i];
@@ -209,7 +209,7 @@
if (! (chorus->lookup_tab[i] =
(int *) malloc(sizeof (int) * chorus->length[i])))
{- st_fail("chorus: Cannot malloc %d bytes!\n", + st_fail("chorus: Cannot malloc %d bytes!", sizeof(int) * chorus->length[i]);
return (ST_EOF);
}
@@ -238,7 +238,7 @@
if (! (chorus->chorusbuf =
(float *) malloc(sizeof (float) * chorus->maxsamples)))
{- st_fail("chorus: Cannot malloc %d bytes!\n", + st_fail("chorus: Cannot malloc %d bytes!", sizeof(float) * chorus->maxsamples);
return (ST_EOF);
}
--- a/src/compand.c
+++ b/src/compand.c
@@ -124,7 +124,7 @@
if (commas % 2 == 0) /* There must be an even number of
transfer parameters */
{- st_fail("compander: Odd number of transfer function parameters\n"+ st_fail("compander: Odd number of transfer function parameters""Each input value in dB must have a corresponding output value");
return (ST_EOF);
}
@@ -196,26 +196,22 @@
compand_t l = (compand_t) effp->priv;
int i;
-# ifdef DEBUG
- {- fprintf(stderr, "Starting compand effect\n");
- fprintf(stderr, "\nRate %ld, size %d, encoding %d, output gain %g.\n",
- effp->outinfo.rate, effp->outinfo.size, effp->outinfo.encoding,
- l->outgain);
- fprintf(stderr, "%d input channel(s) expected: actually %d\n",
- l->expectedChannels, effp->outinfo.channels);
- fprintf(stderr, "\nAttack and decay rates\n"
- "======================\n");
- for (i = 0; i < l->expectedChannels; ++i)
- fprintf(stderr, "Channel %d: attack = %-12g decay = %-12g\n",
- i, l->attackRate[i], l->decayRate[i]);
- fprintf(stderr, "\nTransfer function (linear values)\n"
- "================= =============\n");
- for (i = 0; i < l->transferPoints; ++i)
- fprintf(stderr, "%12g -> %-12g\n",
- l->transferIns[i], l->transferOuts[i]);
- }
-# endif
+ st_debug("Starting compand effect");+ st_debug("Rate %ld, size %d, encoding %d, output gain %g.",+ effp->outinfo.rate, effp->outinfo.size, effp->outinfo.encoding,
+ l->outgain);
+ st_debug("%d input channel(s) expected: actually %d",+ l->expectedChannels, effp->outinfo.channels);
+ st_debug("Attack and decay rates"+ "======================");
+ for (i = 0; i < l->expectedChannels; ++i)
+ st_debug("Channel %d: attack = %-12g decay = %-12g",+ i, l->attackRate[i], l->decayRate[i]);
+ st_debug("Transfer function (linear values)"+ "================= =============");
+ for (i = 0; i < l->transferPoints; ++i)
+ st_debug("%12g -> %-12g",+ l->transferIns[i], l->transferOuts[i]);
/* Convert attack and decay rates using number of samples */
--- a/src/cvsd.c
+++ b/src/cvsd.c
@@ -153,7 +153,7 @@
p->bytes_written = 0;
p->com.v_min = 1;
p->com.v_max = -1;
- st_report("cvsd: bit rate %dbit/s, bits from %s\n", p->cvsd_rate,+ st_report("cvsd: bit rate %dbit/s, bits from %s", p->cvsd_rate,p->swapbits ? "msb to lsb" : "lsb to msb");
}
@@ -215,7 +215,7 @@
st_writeb(ft, p->bit.shreg);
p->bytes_written++;
}
- st_report("cvsd: min slope %f, max slope %f\n", + st_debug("cvsd: min slope %f, max slope %f", p->com.v_min, p->com.v_max);
return (ST_SUCCESS);
@@ -227,7 +227,7 @@
{struct cvsdpriv *p = (struct cvsdpriv *) ft->priv;
- st_report("cvsd: min value %f, max value %f\n", + st_debug("cvsd: min value %f, max value %f", p->com.v_min, p->com.v_max);
return(ST_SUCCESS);
@@ -496,7 +496,7 @@
hdr->Crc = get16(&pch);
if (sum != hdr->Crc)
{- st_report("DVMS header checksum error, read %u, calculated %u\n",+ st_report("DVMS header checksum error, read %u, calculated %u",hdr->Crc, sum);
return (ST_EOF);
}
@@ -543,7 +543,7 @@
}
if (st_writebuf(ft, hdrbuf, sizeof(hdrbuf), 1) != 1)
{- st_report("%s\n",strerror(errno));+ st_report("%s",strerror(errno));return (ST_EOF);
}
return (ST_SUCCESS);
@@ -585,25 +585,25 @@
rc = dvms_read_header(ft, &hdr);
if (rc){- st_fail_errno(ft,ST_EHDR,"unable to read DVMS header\n");
+ st_fail_errno(ft,ST_EHDR,"unable to read DVMS header");
return rc;
}
- st_report("DVMS header of source file \"%s\":");- st_report(" filename \"%.14s\"",ft->filename);- st_report(" id 0x%x", hdr.Filename);- st_report(" state 0x%x", hdr.Id, hdr.State);- st_report(" time %s",ctime(&hdr.Unixtime)); /* ctime generates lf */- st_report(" usender %u", hdr.Usender);- st_report(" ureceiver %u", hdr.Ureceiver);- st_report(" length %u", hdr.Length);- st_report(" srate %u", hdr.Srate);- st_report(" days %u", hdr.Days);- st_report(" custom1 %u", hdr.Custom1);- st_report(" custom2 %u", hdr.Custom2);- st_report(" info \"%.16s\"\n", hdr.Info);+ st_debug("DVMS header of source file \"%s\":");+ st_debug(" filename \"%.14s\"",ft->filename);+ st_debug(" id 0x%x", hdr.Filename);+ st_debug(" state 0x%x", hdr.Id, hdr.State);+ st_debug(" time %s",ctime(&hdr.Unixtime)); /* ctime generates lf */+ st_debug(" usender %u", hdr.Usender);+ st_debug(" ureceiver %u", hdr.Ureceiver);+ st_debug(" length %u", hdr.Length);+ st_debug(" srate %u", hdr.Srate);+ st_debug(" days %u", hdr.Days);+ st_debug(" custom1 %u", hdr.Custom1);+ st_debug(" custom2 %u", hdr.Custom2);+ st_debug(" info \"%.16s\"", hdr.Info);ft->info.rate = (hdr.Srate < 240) ? 16000 : 32000;
- st_report("DVMS rate %dbit/s using %dbit/s deviation %d%%\n", + st_debug("DVMS rate %dbit/s using %dbit/s deviation %d%%", hdr.Srate*100, ft->info.rate,
((ft->info.rate - hdr.Srate*100) * 100) / ft->info.rate);
rc = st_cvsdstartread(ft);
@@ -629,7 +629,7 @@
make_dvms_hdr(ft, &hdr);
rc = dvms_write_header(ft, &hdr);
if (rc){- st_fail_errno(ft,rc,"cannot write DVMS header\n");
+ st_fail_errno(ft,rc,"cannot write DVMS header");
return rc;
}
@@ -661,7 +661,7 @@
make_dvms_hdr(ft, &hdr);
rc = dvms_write_header(ft, &hdr);
if(rc){- st_fail_errno(ft,rc,"cannot write DVMS header\n");
+ st_fail_errno(ft,rc,"cannot write DVMS header");
return rc;
}
return rc;
--- a/src/deemphas.c
+++ b/src/deemphas.c
@@ -109,6 +109,9 @@
double lastout;
} *deemph_t;
+assert_static(sizeof(struct deemphstuff) <= ST_MAX_EFFECT_PRIVSIZE,
+ /* else */ deemph_PRIVSIZE_too_big);
+
/*
* Process options
*
@@ -120,11 +123,6 @@
if (n)
{st_fail(st_deemph_effect.usage);
- return (ST_EOF);
- }
- if (sizeof(double)*ST_MAX_EFFECT_PRIVSIZE < sizeof(struct deemphstuff))
- {- st_fail("Internal error: PRIVSIZE too small.\n");return (ST_EOF);
}
return (ST_SUCCESS);
--- a/src/earwax.c
+++ b/src/earwax.c
@@ -100,7 +100,7 @@
if (effp->ininfo.encoding != ST_ENCODING_SIGN2
|| effp->ininfo.rate != 44100
|| effp->ininfo.channels != 2){- st_fail("the earwax effect works only with audio cd (44.1 kHz, twos-complement signed linear, stereo) samples.\n");+ st_fail("the earwax effect works only with audio cd (44.1 kHz, twos-complement signed linear, stereo) samples.");return (ST_EOF);
}
@@ -107,7 +107,7 @@
/* allocate tap memory */
earwax->tap = (st_sample_t*)malloc( sizeof(st_sample_t) * EARWAX_NUMTAPS );
if( !earwax->tap ){- st_fail("earwax: Cannot malloc %d bytes!\n", + st_fail("earwax: Cannot malloc %d bytes!", sizeof(st_sample_t) * EARWAX_NUMTAPS );
return (ST_EOF);
}
--- a/src/echo.c
+++ b/src/echo.c
@@ -123,17 +123,17 @@
echo->maxsamples = 0L;
if ( echo->in_gain < 0.0 )
{- st_fail("echo: gain-in must be positive!\n");+ st_fail("echo: gain-in must be positive!");return (ST_EOF);
}
if ( echo->in_gain > 1.0 )
{- st_fail("echo: gain-in must be less than 1.0!\n");+ st_fail("echo: gain-in must be less than 1.0!");return (ST_EOF);
}
if ( echo->out_gain < 0.0 )
{- st_fail("echo: gain-in must be positive!\n");+ st_fail("echo: gain-in must be positive!");return (ST_EOF);
}
for ( i = 0; i < echo->num_delays; i++ ) {@@ -140,23 +140,23 @@
echo->samples[i] = echo->delay[i] * effp->ininfo.rate / 1000.0;
if ( echo->samples[i] < 1 )
{- st_fail("echo: delay must be positive!\n");+ st_fail("echo: delay must be positive!");return (ST_EOF);
}
if ( echo->samples[i] > DELAY_BUFSIZ )
{- st_fail("echo: delay must be less than %g seconds!\n",+ st_fail("echo: delay must be less than %g seconds!",DELAY_BUFSIZ / (float) effp->ininfo.rate );
return (ST_EOF);
}
if ( echo->decay[i] < 0.0 )
{- st_fail("echo: decay must be positive!\n" );+ st_fail("echo: decay must be positive!" );return (ST_EOF);
}
if ( echo->decay[i] > 1.0 )
{- st_fail("echo: decay must be less than 1.0!\n" );+ st_fail("echo: decay must be less than 1.0!" );return (ST_EOF);
}
if ( echo->samples[i] > echo->maxsamples )
@@ -164,7 +164,7 @@
}
if (! (echo->delay_buf = (double *) malloc(sizeof (double) * echo->maxsamples)))
{- st_fail("echo: Cannot malloc %d bytes!\n", + st_fail("echo: Cannot malloc %d bytes!", sizeof(long) * echo->maxsamples);
return (ST_EOF);
}
--- a/src/echos.c
+++ b/src/echos.c
@@ -116,17 +116,17 @@
if ( echos->in_gain < 0.0 )
{- st_fail("echos: gain-in must be positive!\n");+ st_fail("echos: gain-in must be positive!");return (ST_EOF);
}
if ( echos->in_gain > 1.0 )
{- st_fail("echos: gain-in must be less than 1.0!\n");+ st_fail("echos: gain-in must be less than 1.0!");return (ST_EOF);
}
if ( echos->out_gain < 0.0 )
{- st_fail("echos: gain-in must be positive!\n");+ st_fail("echos: gain-in must be positive!");return (ST_EOF);
}
for ( i = 0; i < echos->num_delays; i++ ) {@@ -133,23 +133,23 @@
echos->samples[i] = echos->delay[i] * effp->ininfo.rate / 1000.0;
if ( echos->samples[i] < 1 )
{- st_fail("echos: delay must be positive!\n");+ st_fail("echos: delay must be positive!");return (ST_EOF);
}
if ( echos->samples[i] > DELAY_BUFSIZ )
{- st_fail("echos: delay must be less than %g seconds!\n",+ st_fail("echos: delay must be less than %g seconds!",DELAY_BUFSIZ / (float) effp->ininfo.rate );
return (ST_EOF);
}
if ( echos->decay[i] < 0.0 )
{- st_fail("echos: decay must be positive!\n" );+ st_fail("echos: decay must be positive!" );return (ST_EOF);
}
if ( echos->decay[i] > 1.0 )
{- st_fail("echos: decay must be less than 1.0!\n" );+ st_fail("echos: decay must be less than 1.0!" );return (ST_EOF);
}
echos->counter[i] = 0;
@@ -158,7 +158,7 @@
}
if (! (echos->delay_buf = (double *) malloc(sizeof (double) * echos->sumsamples)))
{- st_fail("echos: Cannot malloc %d bytes!\n", + st_fail("echos: Cannot malloc %d bytes!", sizeof(double) * echos->sumsamples);
return(ST_EOF);
}
--- a/src/equalizer.c
+++ b/src/equalizer.c
@@ -101,13 +101,13 @@
amp = pow( 10, eq->gain/40 );
alpha = sin(w0)/( 2*eq->Q );
- st_report("Debug: cfreq: %fHz", eq->cfreq);- st_report("Debug: Q: %f", eq->Q);- st_report("Debug: gain: %fdB", eq->gain);- st_report("Debug: rate: %f", eq->rate);- st_report("Debug: w0: %f", w0);- st_report("Debug: amp: %f", amp);- st_report("Debug: alpha: %f", alpha);+ st_debug("cfreq: %fHz", eq->cfreq);+ st_debug("Q: %f", eq->Q);+ st_debug("gain: %fdB", eq->gain);+ st_debug("rate: %f", eq->rate);+ st_debug("w0: %f", w0);+ st_debug("amp: %f", amp);+ st_debug("alpha: %f", alpha);// Initialisation
eq->b[0] = 1 + alpha*amp;
--- a/src/fade.c
+++ b/src/fade.c
@@ -206,7 +206,7 @@
fade->endpadwarned = 0;
- /* fprintf(stderr, "fade: in_start = %d in_stop = %d out_start = %d out_stop = %d\n", fade->in_start, fade->in_stop, fade->out_start, fade->out_stop); */
+ /* st_debug("fade: in_start = %d in_stop = %d out_start = %d out_stop = %d", fade->in_start, fade->in_stop, fade->out_start, fade->out_stop); */return(ST_SUCCESS);
}
--- a/src/filter.c
+++ b/src/filter.c
@@ -81,7 +81,7 @@
}
if (*p) f->freq1 = f->freq0 = 0;
}
- /* fprintf(stderr,"freq: %d-%d\n", f->freq0, f->freq1);fflush(stderr); */
+ st_debug("freq: %d-%d", f->freq0, f->freq1);if (f->freq0 == 0 && f->freq1 == 0)
{st_fail(st_filter_effect.usage);
@@ -104,7 +104,7 @@
return (ST_EOF);
}
- st_report("filter opts: %d-%d, window-len %d, beta %f\n", f->freq0, f->freq1, f->Nwin, f->beta);+ st_debug("filter opts: %d-%d, window-len %d, beta %f", f->freq0, f->freq1, f->Nwin, f->beta);return (ST_SUCCESS);
}
@@ -137,7 +137,7 @@
Xh0 = makeFilter(Fp0, Xh, 2.0*(double)f->freq0/f->rate, f->beta, 1, 0);
if (Xh0 <= 1)
{- st_fail("filter: Unable to make low filter\n");+ st_fail("filter: Unable to make low filter");return (ST_EOF);
}
} else {@@ -149,7 +149,7 @@
Xh1 = makeFilter(Fp1, Xh, 2.0*(double)f->freq1/f->rate, f->beta, 1, 0);
if (Xh1 <= 1)
{- st_fail("filter: Unable to make high filter\n");+ st_fail("filter: Unable to make high filter");return (ST_EOF);
}
} else {@@ -196,7 +196,7 @@
long i, Nx, Nproc;
/* constrain amount we actually process */
- /* fprintf(stderr,"Xh %d, Xt %d, isamp %d, ",f->Xh, f->Xt, *isamp);fflush(stderr); */
+ /* st_debug("Xh %d, Xt %d, isamp %d, ",f->Xh, f->Xt, *isamp); */Nx = BUFFSIZE + 2*f->Xh - f->Xt;
if (Nx > *isamp) Nx = *isamp;
if (Nx > *osamp) Nx = *osamp;
@@ -222,7 +222,7 @@
*osamp = 0;
return (ST_SUCCESS);
}
- /* fprintf(stderr,"flow Nproc %d\n",Nproc); */
+ st_debug("flow Nproc %d",Nproc);FiltWin(f, Nproc);
/* Copy back portion of input signal that must be re-used */
@@ -247,7 +247,7 @@
long isamp_res, osamp_res;
st_sample_t *Obuf;
- /* fprintf(stderr,"Xh %d, Xt %d <--- DRAIN\n",f->Xh, f->Xt); */
+ st_debug("Xh %d, Xt %d <--- DRAIN",f->Xh, f->Xt);/* stuff end with Xh zeros */
isamp_res = f->Xh;
@@ -258,7 +258,7 @@
Isamp = isamp_res;
Osamp = osamp_res;
st_filter_flow(effp, NULL, Obuf, (st_size_t *)&Isamp, (st_size_t *)&Osamp);
- /* fprintf(stderr,"DRAIN isamp,osamp (%d,%d) -> (%d,%d)\n",
+ /* st_debug("DRAIN isamp,osamp (%d,%d) -> (%d,%d)",* isamp_res,osamp_res,Isamp,Osamp); */
Obuf += Osamp;
osamp_res -= Osamp;
@@ -265,9 +265,9 @@
isamp_res -= Isamp;
};
*osamp -= osamp_res;
- /* fprintf(stderr,"DRAIN osamp %d\n", *osamp); */
+ /* st_debug("DRAIN osamp %d", *osamp); */if (isamp_res)
- st_warn("drain overran obuf by %d\n", isamp_res); fflush(stderr);+ st_warn("drain overran obuf by %d", isamp_res);/* FIXME: This is very picky. osamp better be big enough to grab
* all remaining samples or they will be discarded.
*/
--- a/src/flanger.c
+++ b/src/flanger.c
@@ -123,47 +123,47 @@
if ( flanger->in_gain < 0.0 )
{- st_fail("flanger: gain-in must be positive!\n");+ st_fail("flanger: gain-in must be positive!");return (ST_EOF);
}
if ( flanger->in_gain > 1.0 )
{- st_fail("flanger: gain-in must be less than 1.0!\n");+ st_fail("flanger: gain-in must be less than 1.0!");return (ST_EOF);
}
if ( flanger->out_gain < 0.0 )
{- st_fail("flanger: gain-out must be positive!\n");+ st_fail("flanger: gain-out must be positive!");return (ST_EOF);
}
if ( flanger->delay < 0.0 )
{- st_fail("flanger: delay must be positive!\n");+ st_fail("flanger: delay must be positive!");return (ST_EOF);
}
if ( flanger->delay > 5.0 )
{- st_fail("flanger: delay must be less than 5.0 msec!\n");+ st_fail("flanger: delay must be less than 5.0 msec!");return (ST_EOF);
}
if ( flanger->speed < 0.1 )
{- st_fail("flanger: speed must be more than 0.1 Hz!\n");+ st_fail("flanger: speed must be more than 0.1 Hz!");return (ST_EOF);
}
if ( flanger->speed > 2.0 )
{- st_fail("flanger: speed must be less than 2.0 Hz!\n");+ st_fail("flanger: speed must be less than 2.0 Hz!");return (ST_EOF);
}
if ( flanger->decay < 0.0 )
{- st_fail("flanger: decay must be positive!\n" );+ st_fail("flanger: decay must be positive!" );return (ST_EOF);
}
if ( flanger->decay > 1.0 )
{- st_fail("flanger: decay must be less that 1.0!\n" );+ st_fail("flanger: decay must be less that 1.0!" );return (ST_EOF);
}
/* Be nice and check the hint with warning, if... */
@@ -175,7 +175,7 @@
if (! (flanger->flangerbuf =
(double *) malloc(sizeof (double) * flanger->maxsamples)))
{- st_fail("flanger: Cannot malloc %d bytes!\n", + st_fail("flanger: Cannot malloc %d bytes!", sizeof(double) * flanger->maxsamples);
return (ST_EOF);
}
@@ -184,7 +184,7 @@
if (! (flanger->lookup_tab =
(int *) malloc(sizeof (int) * flanger->length)))
{- st_fail("flanger: Cannot malloc %d bytes!\n", + st_fail("flanger: Cannot malloc %d bytes!", sizeof(int) * flanger->length);
return(ST_EOF);
}
--- a/src/hcom.c
+++ b/src/hcom.c
@@ -140,7 +140,7 @@
st_readw(ft, (unsigned short *)&(p->dictionary[i].dict_leftson));
st_readw(ft, (unsigned short *)&(p->dictionary[i].dict_rightson));
/*
- st_report("%d %d",+ st_debug("%d %d",p->dictionary[i].dict_leftson,
p->dictionary[i].dict_rightson);
*/
@@ -153,7 +153,7 @@
p->checksum = checksum;
p->deltacompression = compresstype;
if (!p->deltacompression)
- st_report("HCOM data using value compression");+ st_debug("HCOM data using value compression");p->huffcount = huffcount;
p->cksum = 0;
p->dictentry = 0;
@@ -419,10 +419,8 @@
d = (datafork[i] - (sample & 0xff)) & 0xff; /* creates absolute entries LMS */
sample = datafork[i];
datafork[i] = d;
-#if 0 /* checking our table is accessed correctly */
- if(d < 0 || d > 255)
- printf("d is outside array bounds %d\n", d);-#endif
+ if(d < 0 || d > 255) /* checking our table is accessed correctly */
+ st_debug("d is outside array bounds %d", d);frequtable[d]++;
}
de = dictionary;
@@ -476,8 +474,8 @@
l += frequtable[i] * codesize[i];
}
l = (((l + 31) >> 5) << 2) + 24 + dictsize * 4;
- st_report(" Original size: %6d bytes", *dl);- st_report("Compressed size: %6d bytes", l);+ st_debug(" Original size: %6d bytes", *dl);+ st_debug("Compressed size: %6d bytes", l);if((datafork = (unsigned char *)malloc((unsigned)l)) == NULL)
{return (ST_ENOMEM);
--- a/src/highp.c
+++ b/src/highp.c
@@ -64,7 +64,7 @@
highp_t highp = (highp_t) effp->priv;
if (highp->cutoff > effp->ininfo.rate/2)
{- st_fail("Highpass: cutoff must be < sample rate / 2 (Nyquest rate)\n");+ st_fail("Highpass: cutoff must be < sample rate / 2 (Nyquest rate)");return (ST_EOF);
}
--- a/src/ima_rw.c
+++ b/src/ima_rw.c
@@ -83,8 +83,7 @@
val = (short)(ip[0] + (ip[1]<<8)); /* need cast for sign-extend */
state = ip[2];
if (state > ISSTMAX) {- st_warn("IMA_ADPCM block ch%d initial-state (%d) out of range\n", ch, state);- fflush(stderr);
+ st_warn("IMA_ADPCM block ch%d initial-state (%d) out of range", ch, state);state = 0;
}
/* specs say to ignore ip[3] , but write it as 0 */
@@ -247,9 +246,6 @@
}
d2 /= n; /* be sure it's non-negative */
- if (sho) {- fflush(stderr);
- }
*st = state;
return (int) sqrt(d2);
}
--- a/src/lowp.c
+++ b/src/lowp.c
@@ -62,7 +62,7 @@
lowp_t lowp = (lowp_t) effp->priv;
if (lowp->cutoff > effp->ininfo.rate / 2)
{- st_fail("Lowpass: cutoff must be < sample rate / 2 (Nyquest rate)\n");+ st_fail("Lowpass: cutoff must be < sample rate / 2 (Nyquest rate)");return (ST_EOF);
}
--- a/src/maud.c
+++ b/src/maud.c
@@ -93,7 +93,7 @@
/*
buf[4] = 0;
- st_report("chunk %s",buf);+ st_debug("chunk %s",buf);*/
if (strncmp(buf,"MHDR",4) == 0) {@@ -196,7 +196,7 @@
return(ST_EOF);
}
chunk_buf[chunksize] = '\0';
- st_report("%s",chunk_buf);+ st_debug("%s",chunk_buf);free(chunk_buf);
continue;
--- a/src/mcompand.c
+++ b/src/mcompand.c
@@ -311,7 +311,7 @@
l->transferIns[i] = pow(10.0, atof(s)/20.0);
if (l->transferIns[i] > 1.0)
{- st_fail("dB values are relative to maximum input, and, ipso facto, "+ st_fail("dB values are relative to maximum input, and, ipso facto, \n""cannot exceed 0");
return (ST_EOF);
}
@@ -319,7 +319,7 @@
--(l->transferPoints);
if (i > 0 && l->transferIns[i] <= l->transferIns[i-1])
{- st_fail("Transfer function points don't have strictly ascending "+ st_fail("Transfer function points don't have strictly ascending \n""input amplitude");
return (ST_EOF);
}
--- a/src/mp3.c
+++ b/src/mp3.c
@@ -327,7 +327,7 @@
continue;
else
{- st_report("unrecoverable frame level error (%s).\n",+ st_report("unrecoverable frame level error (%s).",mad_stream_errorstr(p->Stream));
return done;
}
--- a/src/noisered.c
+++ b/src/noisered.c
@@ -114,7 +114,7 @@
fchannels ++;
}
if (fchannels != channels) {- st_fail("noisered: channel mismatch: %d in input, %d in profile.\n",+ st_fail("noisered: channel mismatch: %d in input, %d in profile.",channels, fchannels);
return ST_EOF;
}
--- a/src/oss.c
+++ b/src/oss.c
@@ -147,7 +147,7 @@
if (tmp != dsp_stereo)
{- st_warn("Sound card appears to only support %d channels. Overriding format\n", tmp+1);+ st_warn("Sound card appears to only support %d channels. Overriding format", tmp+1);ft->info.channels = tmp + 1;
}
--- a/src/phaser.c
+++ b/src/phaser.c
@@ -123,32 +123,32 @@
if ( phaser->delay < 0.0 )
{- st_fail("phaser: delay must be positive!\n");+ st_fail("phaser: delay must be positive!");return (ST_EOF);
}
if ( phaser->delay > 5.0 )
{- st_fail("phaser: delay must be less than 5.0 msec!\n");+ st_fail("phaser: delay must be less than 5.0 msec!");return (ST_EOF);
}
if ( phaser->speed < 0.1 )
{- st_fail("phaser: speed must be more than 0.1 Hz!\n");+ st_fail("phaser: speed must be more than 0.1 Hz!");return (ST_EOF);
}
if ( phaser->speed > 2.0 )
{- st_fail("phaser: speed must be less than 2.0 Hz!\n");+ st_fail("phaser: speed must be less than 2.0 Hz!");return (ST_EOF);
}
if ( phaser->decay < 0.0 )
{- st_fail("phaser: decay must be positive!\n" );+ st_fail("phaser: decay must be positive!" );return (ST_EOF);
}
if ( phaser->decay >= 1.0 )
{- st_fail("phaser: decay must be less that 1.0!\n" );+ st_fail("phaser: decay must be less that 1.0!" );return (ST_EOF);
}
/* Be nice and check the hint with warning, if... */
@@ -162,7 +162,7 @@
if (! (phaser->phaserbuf =
(double *) malloc(sizeof (double) * phaser->maxsamples)))
{- st_fail("phaser: Cannot malloc %d bytes!\n", + st_fail("phaser: Cannot malloc %d bytes!", sizeof(double) * phaser->maxsamples);
return (ST_EOF);
}
@@ -171,7 +171,7 @@
if (! (phaser->lookup_tab =
(int *) malloc(sizeof (int) * phaser->length)))
{- st_fail("phaser: Cannot malloc %d bytes!\n", + st_fail("phaser: Cannot malloc %d bytes!", sizeof(int) * phaser->length);
return (ST_EOF);
}
--- a/src/pitch.c
+++ b/src/pitch.c
@@ -148,8 +148,7 @@
static void debug(pitch_t pitch, char * where)
{- fprintf(stderr,
- "%s: ind=%d sz=%ld step=%d o=%d rate=%f ia=%d st=%d fo=%s\n",
+ st_debug("%s: ind=%d sz=%ld step=%d o=%d rate=%f ia=%d st=%d fo=%s", where, pitch->index, pitch->size, pitch->step, pitch->overlap,
pitch->rate, pitch->iacc, pitch->state, fadeoptname(pitch->fadeopt));
}
--- a/src/polyphas.c
+++ b/src/polyphas.c
@@ -171,11 +171,11 @@
p = primes;
q = q0;
- st_report("factors(%d) =",n);+ st_debug("factors(%d) =",n); while (n > 1) {while ((pr = *p) && (n % pr)) p++;
if (!pr) {- st_fail("Number %d too large of a prime.\n",n);+ st_fail("Number %d too large of a prime.",n);pr = n;
}
*q++ = pr;
@@ -182,8 +182,8 @@
n /= pr;
}
*q = 0;
- for (pr=0; pr<q-q0; pr++) st_report(" %d",q0[pr]);- st_report("\n");+ for (pr=0; pr<q-q0; pr++) st_debug(" %d",q0[pr]);+ st_debug("");return (q-q0);
}
@@ -221,8 +221,8 @@
}
if (n) *p++=n;
*p = 0;
- /*for (k=0; k<p-m; k++) st_report(" %d",m[k]);*/- /*st_report("\n");*/+ /*for (k=0; k<p-m; k++) st_debug(" %d",m[k]);*/+ /*st_debug("");*/return (p-m);
}
@@ -253,9 +253,9 @@
cost = 0;
f = denom;
u = min(ct1,ct2) + 1;
- /*st_report("pfacts(%d): ", numer);*/+ /*st_debug("pfacts(%d): ", numer);*/u1 = permute(m1,l1,ct1,u,amalg);
- /*st_report("pfacts(%d): ", denom);*/+ /*st_debug("pfacts(%d): ", denom);*/u2 = permute(m2,l2,ct2,u,amalg);
u = max(u1,u2);
for (j=0; j<u; j++) {@@ -269,10 +269,10 @@
c_min = cost;
u_min = u;
# if 0
- st_report("c_min %d, [%d-%d]:",c_min,numer,denom);+ st_debug("c_min %d, [%d-%d]:",c_min,numer,denom);for (j=0; j<u; j++)
- st_report(" (%d,%d)",m1[j],m2[j]);- st_report("\n");+ st_debug(" (%d,%d)",m1[j],m2[j]);+ st_debug("");# endif
memcpy(b1,m1,u*sizeof(int));
memcpy(b2,m2,u*sizeof(int));
@@ -306,7 +306,7 @@
int N1;
if(buffer == NULL || length <= 0)
- st_fail("Illegal buffer %p or length %d to nuttall.\n", buffer, length);+ st_fail("Illegal buffer %p or length %d to nuttall.", buffer, length);/* Initial variable setups. */
N = length;
@@ -329,7 +329,7 @@
int N1;
if(buffer == NULL || length <= 0)
- st_fail("Illegal buffer %p or length %d to hamming.\n",buffer,length);+ st_fail("Illegal buffer %p or length %d to hamming.",buffer,length);N1 = length/2;
for(j=0;j<length;j++)
@@ -355,7 +355,7 @@
double sum;
if(buffer == NULL || length < 0 || cutoff < 0 || cutoff > PI)
- st_fail("Illegal buffer %p, length %d, or cutoff %f.\n",buffer,length,cutoff);+ st_fail("Illegal buffer %p, length %d, or cutoff %f.",buffer,length,cutoff);/* Use the user-option of window type */
if(win_type == 0)
@@ -363,12 +363,12 @@
else
hamming(buffer,length); /* Design Hamming window: 43 dB cutoff */
- /* st_report("# fir_design length=%d, cutoff=%8.4f\n",length,cutoff); */+ /* st_debug("# fir_design length=%d, cutoff=%8.4f",length,cutoff); *//* Design filter: windowed sinc function */
sum = 0.0;
for(j=0;j<length;j++) {buffer[j] *= sinc(PI*cutoff*(j-length/2)); /* center at length/2 */
- /* st_report("%.1f %.6f\n",(float)j,buffer[j]); */+ /* st_debug("%.1f %.6f",(float)j,buffer[j]); */sum += buffer[j];
}
sum = (double)1.0/sum;
@@ -376,7 +376,7 @@
for(j=0;j<length;j++) {buffer[j] *= sum;
}
- /* st_report("# end\n\n"); */+ /* st_debug("# end"); */}
#define RIBLEN 2048
@@ -420,9 +420,9 @@
rate->total = total;
/* l1 and l2 are now lists of the up/down factors for conversion */
- st_report("Poly: input rate %d, output rate %d. %d stages.",+ st_debug("Poly: input rate %d, output rate %d. %d stages.",effp->ininfo.rate, effp->outinfo.rate,total);
- st_report("Poly: window: %s size: %d cutoff: %f.",+ st_debug("Poly: window: %s size: %d cutoff: %f.", (win_type == 0) ? ("nut") : ("ham"), win_width, cutoff);/* Create an array of filters and past history */
@@ -442,7 +442,7 @@
s->size = size;
s->hsize = f_len/s->up; /* this much of window is past-history */
s->held = 0;
- st_report("Poly: stage %d: Up by %d, down by %d, i_samps %d, hsize %d",+ st_debug("Poly: stage %d: Up by %d, down by %d, i_samps %d, hsize %d",k+1,s->up,s->down,size, s->hsize);
s->filt_len = f_len;
s->filt_array = (Float *) malloc(sizeof(Float) * f_len);
@@ -452,7 +452,7 @@
s->window[j] = 0.0;
uprate *= s->up;
- st_report("Poly: : filt_len %d, cutoff freq %.1f",+ st_debug("Poly: : filt_len %d, cutoff freq %.1f",f_len, uprate*cutoff/f_cutoff);
uprate /= s->down;
fir_design(s->filt_array, f_len, cutoff/f_cutoff);
@@ -476,7 +476,7 @@
s->filt_array = NULL;
s->window = (Float *) malloc(sizeof(Float) * size);
}
- st_report("Poly: output samples %d, oskip %d",size, rate->oskip);+ st_debug("Poly: output samples %d, oskip %d",size, rate->oskip);return (ST_SUCCESS);
}
@@ -510,10 +510,10 @@
Float *o_top;
in = s->window + s->hsize;
- /*for (mm=0; mm<s->filt_len; mm++) st_report("cf_%d %f\n",mm,s->filt_array[mm]);*/+ /*for (mm=0; mm<s->filt_len; mm++) st_debug("cf_%d %f",mm,s->filt_array[mm]);*//* assumes s->size divisible by down (now true) */
o_top = output + (s->size * up) / down;
- /*st_report(" isize %d, osize %d, up %d, down %d, N %d", s->size, o_top-output, up, down, f_len);*/+ /*st_debug(" isize %d, osize %d, up %d, down %d, N %d", s->size, o_top-output, up, down, f_len);*/ for (mm=0, o=output; o < o_top; mm+=down, o++) {double sum;
const Float *p, *q;
@@ -542,7 +542,7 @@
polystage *s0,*s1;
/* Sanity check: how much can we tolerate? */
- /* st_report("*isamp=%d *osamp=%d\n",*isamp,*osamp); fflush(stderr); */+ /* st_debug("*isamp=%d *osamp=%d",*isamp,*osamp); */s0 = rate->stage[0]; /* the first stage */
s1 = rate->stage[rate->total]; /* the 'last' stage is output buffer */
{@@ -581,7 +581,7 @@
out = rate->stage[k+1]->window + rate->stage[k+1]->hsize;
- /* st_report("k=%d insize=%d\n",k,in_size); fflush(stderr); */+ /* st_debug("k=%d insize=%d",k,in_size); */polyphase(out, s);
/* copy input history into lower portion of rate->window[k] */
--- a/src/prc.c
+++ b/src/prc.c
@@ -86,7 +86,7 @@
/* Check the header */
st_readbuf(ft, head,1, sizeof(header));
if (memcmp(head, header, sizeof(header))==0) {- st_report("Found Psion record.app header");+ st_debug("Found Psion record.app header");}
else
{@@ -96,7 +96,7 @@
st_readw(ft, &(len));
p->length=len;
- st_report("Found length=%d",len);+ st_debug("Found length=%d",len);/* dummy read rest */
st_readbuf(ft, head,1,14+2+2);
@@ -172,7 +172,7 @@
{prc_t p = (prc_t ) ft->priv;
p->length += samp * ft->info.size;
- st_report("length now = %d", p->length);+ st_debug("length now = %d", p->length);return st_rawwrite(ft, buf, samp);
}
@@ -201,7 +201,7 @@
char nullbuf[15];
prc_t p = (prc_t ) ft->priv;
- st_report("Final length=%d",p->length);+ st_debug("Final length=%d",p->length);memset(nullbuf,0,14);
st_writebuf(ft, header, 1, sizeof(header));
st_writew(ft, p->length);
--- a/src/repeat.c
+++ b/src/repeat.c
@@ -77,7 +77,7 @@
if (fwrite((char *)ibuf, sizeof(st_sample_t), *isamp, repeat->fp) !=
*isamp) {- st_fail("repeat: write error on temporary file\n");+ st_fail("repeat: write error on temporary file");return (ST_EOF);
}
@@ -102,7 +102,7 @@
repeat->total = ftell(repeat->fp);
if ((repeat->total % sizeof(st_sample_t)) != 0) {- st_fail("repeat: corrupted temporary file\n");+ st_fail("repeat: corrupted temporary file");return (ST_EOF);
}
@@ -131,7 +131,7 @@
repeat->fp);
if (read != samp) {perror(strerror(errno));
- st_fail("repeat1: read error on temporary file\n");+ st_fail("repeat1: read error on temporary file");return(ST_EOF);
}
@@ -176,7 +176,7 @@
repeat->fp);
if (read != *osamp) {perror(strerror(errno));
- st_fail("repeat3: read error on temporary file\n");+ st_fail("repeat3: read error on temporary file");return(ST_EOF);
}
repeat->remaining -= *osamp;
--- a/src/resample.c
+++ b/src/resample.c
@@ -163,9 +163,9 @@
}
else if (r->beta <= 2.0) {r->beta = 0;
- st_report("resample opts: Nuttall window, cutoff %f\n", r->rolloff);+ st_debug("resample opts: Nuttall window, cutoff %f", r->rolloff); } else {- st_report("resample opts: Kaiser window, cutoff %f, beta %f\n", r->rolloff, r->beta);+ st_debug("resample opts: Kaiser window, cutoff %f, beta %f", r->rolloff, r->beta);}
return (ST_SUCCESS);
}
@@ -214,15 +214,15 @@
i = makeFilter(r->Imp, r->Nwing, r->rolloff, r->beta, r->Nq, 1);
if (i <= 0)
{- st_fail("resample: Unable to make filter\n");+ st_fail("resample: Unable to make filter");return (ST_EOF);
}
- /*st_report("Nmult: %ld, Nwing: %ld, Nq: %ld\n",r->Nmult,r->Nwing,r->Nq);*/+ /*st_debug("Nmult: %ld, Nwing: %ld, Nq: %ld",r->Nmult,r->Nwing,r->Nq);*/ if (r->quadr < 0) { /* exact coeff's method */r->Xh = r->Nwing/r->b;
- st_report("resample: rate ratio %ld:%ld, coeff interpolation not needed\n", r->a, r->b);+ st_debug("resample: rate ratio %ld:%ld, coeff interpolation not needed", r->a, r->b); } else {r->dhb = Np; /* Fixed-point Filter sampling-time-increment */
if (r->Factor<1.0) r->dhb = r->Factor*Np + 0.5;
@@ -252,7 +252,7 @@
r->Xsize = 2*Xoff + i/(1.0+r->Factor);
r->Ysize = BUFFSIZE - r->Xsize;
- /* st_report("Xsize %d, Ysize %d, Xoff %d",r->Xsize,r->Ysize,r->Xoff); */+ /* st_debug("Xsize %d, Ysize %d, Xoff %d",r->Xsize,r->Ysize,r->Xoff); */r->X = (Float *) malloc(sizeof(Float) * (BUFFSIZE));
r->Y = r->X + r->Xsize;
@@ -274,7 +274,7 @@
long i, last, Nout, Nx, Nproc;
/* constrain amount we actually process */
- /*fprintf(stderr,"Xp %d, Xread %d, isamp %d, ",r->Xp, r->Xread,*isamp);*/
+ /*st_debug("Xp %d, Xread %d, isamp %d, ",r->Xp, r->Xread,*isamp);*/Nproc = r->Xsize - r->Xp;
@@ -290,7 +290,7 @@
}
if ((unsigned long)Nx > *isamp)
Nx = *isamp;
- /*fprintf(stderr,"Nx %d\n",Nx);*/
+ st_debug("Nx %d",Nx); if (ibuf == NULL) {for(i = r->Xread; i < Nx + r->Xread ; i++)
@@ -312,7 +312,7 @@
if (r->quadr < 0) { /* exact coeff's method */long creep;
Nout = SrcEX(r, Nproc);
- /*fprintf(stderr,"Nproc %d --> %d\n",Nproc,Nout);*/
+ st_debug("Nproc %d --> %d",Nproc,Nout);/* Move converter Nproc samples back in time */
r->t -= Nproc * r->b;
/* Advance by number of samples processed */
@@ -323,12 +323,12 @@
{r->t -= creep * r->b; /* Remove time accumulation */
r->Xp += creep; /* and add it to read pointer */
- /*fprintf(stderr,"Nproc %ld, creep %ld\n",Nproc,creep);*/
+ /*st_debug("Nproc %ld, creep %ld",Nproc,creep);*/}
} else { /* approx coeff's method */long creep;
Nout = SrcUD(r, Nproc);
- /*fprintf(stderr,"Nproc %d --> %d\n",Nproc,Nout);*/
+ /*st_debug("Nproc %d --> %d",Nproc,Nout);*//* Move converter Nproc samples back in time */
r->Time -= Nproc;
/* Advance by number of samples processed */
@@ -339,7 +339,7 @@
{r->Time -= creep; /* Remove time accumulation */
r->Xp += creep; /* and add it to read pointer */
- /* fprintf(stderr,"Nproc %ld, creep %ld\n",Nproc,creep); */
+ /* st_debug("Nproc %ld, creep %ld",Nproc,creep); */}
}
@@ -347,7 +347,7 @@
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);*/
+ /*st_debug("k %d, last %d",k,last);*/for (i=0; i<last - k; i++)
r->X[i] = r->X[i+k];
@@ -380,7 +380,7 @@
st_sample_t *Obuf;
int rc;
- /* fprintf(stderr,"Xoff %d <--- DRAIN\n",r->Xoff); */
+ /* st_debug("Xoff %d <--- DRAIN",r->Xoff); *//* stuff end with Xoff zeros */
isamp_res = r->Xoff;
@@ -393,7 +393,7 @@
rc = st_resample_flow(effp, NULL, Obuf, (st_size_t *)&Isamp, (st_size_t *)&Osamp);
if (rc)
return rc;
- /* fprintf(stderr,"DRAIN isamp,osamp (%d,%d) -> (%d,%d)\n",
+ /* st_debug("DRAIN isamp,osamp (%d,%d) -> (%d,%d)",isamp_res,osamp_res,Isamp,Osamp); */
Obuf += Osamp;
osamp_res -= Osamp;
@@ -400,9 +400,9 @@
isamp_res -= Isamp;
}
*osamp -= osamp_res;
- /* fprintf(stderr,"DRAIN osamp %d\n", *osamp); */
+ /* st_debug("DRAIN osamp %d", *osamp); */if (isamp_res)
- st_warn("drain overran obuf by %d\n", isamp_res); fflush(stderr);+ st_warn("drain overran obuf by %d", isamp_res);/* FIXME: This is very picky. IF obuf is not big enough to
* drain remaining samples, they will be lost.
*/
@@ -498,12 +498,12 @@
Factor = r->Factor;
time = r->Time;
dt = 1.0/Factor; /* Output sampling period */
- /*fprintf(stderr,"Factor %f, dt %f, ",Factor,dt); */
- /*fprintf(stderr,"Time %f, ",r->Time);*/
+ /*st_debug("Factor %f, dt %f, ",Factor,dt); */+ /*st_debug("Time %f, ",r->Time);*//* (Xh * dhb)>>La is max index into Imp[] */
- /*fprintf(stderr,"ct=%d\n",ct);*/
- /*fprintf(stderr,"ct=%.2f %d\n",(double)r->Nwing*Na/r->dhb, r->Xh);*/
- /*fprintf(stderr,"ct=%ld, T=%.6f, dhb=%6f, dt=%.6f\n",
+ /*st_debug("ct=%d",ct);*/+ /*st_debug("ct=%.2f %d",(double)r->Nwing*Na/r->dhb, r->Xh);*/+ /*st_debug("ct=%ld, T=%.6f, dhb=%6f, dt=%.6f",r->Xh, time-floor(time),(double)r->dhb/Na,dt);*/
Ystart = Y = r->Y;
n = (int)ceil((double)Nx/dt);
@@ -525,7 +525,7 @@
time += dt; /* Move to next sample by time increment */
}
r->Time = time;
- /*fprintf(stderr,"Time %f\n",r->Time);*/
+ /*st_debug("Time %f",r->Time);*/return (Y - Ystart); /* Return the number of output samples */
}
@@ -612,7 +612,7 @@
for (i=Dh; i<Mwing; i+=Dh)
DCgain += ImpR[i];
DCgain = 2*DCgain + ImpR[0]; /* DC gain of real coefficients */
- /*st_report("DCgain err=%.12f",DCgain-1.0);*/+ /*st_debug("DCgain err=%.12f",DCgain-1.0);*/DCgain = 1.0/DCgain;
for (i=0; i<Mwing; i++)
--- a/src/reverb.c
+++ b/src/reverb.c
@@ -173,12 +173,12 @@
reverb->samples[i] = reverb->delay[i] * effp->ininfo.rate / 1000.0;
if ( reverb->samples[i] < 1 )
{- st_fail("reverb: delay must be positive!\n");+ st_fail("reverb: delay must be positive!");return (ST_EOF);
}
if ( reverb->samples[i] > DELAY_BUFSIZ )
{- st_fail("reverb: delay must be less than %g seconds!\n",+ st_fail("reverb: delay must be less than %g seconds!",DELAY_BUFSIZ / (float) effp->ininfo.rate );
return(ST_EOF);
}
@@ -189,7 +189,7 @@
}
if (! (reverb->reverbbuf = (float *) malloc(sizeof (float) * reverb->maxsamples)))
{- st_fail("reverb: Cannot malloc %d bytes!\n", + st_fail("reverb: Cannot malloc %d bytes!", sizeof(float) * reverb->maxsamples);
return(ST_EOF);
}
--- a/src/reverse.c
+++ b/src/reverse.c
@@ -56,7 +56,7 @@
reverse->fp = tmpfile();
if (reverse->fp == NULL)
{- st_fail("Reverse effect can't create temporary file\n");+ st_fail("Reverse effect can't create temporary file");return (ST_EOF);
}
reverse->phase = WRITING;
@@ -80,7 +80,7 @@
if (fwrite((char *)ibuf, sizeof(st_sample_t), *isamp, reverse->fp)
!= *isamp)
{- st_fail("Reverse effect write error on temporary file\n");+ st_fail("Reverse effect write error on temporary file");return(ST_EOF);
}
*osamp = 0;
@@ -104,7 +104,7 @@
reverse->pos = ftell(reverse->fp);
if (reverse->pos % sizeof(st_sample_t) != 0)
{- st_fail("Reverse effect finds odd temporary file\n");+ st_fail("Reverse effect finds odd temporary file");return(ST_EOF);
}
reverse->phase = READING;
@@ -119,7 +119,7 @@
fseek(reverse->fp, reverse->pos, SEEK_SET);
if (fread((char *)obuf, sizeof(st_sample_t), len, reverse->fp) != len)
{- st_fail("Reverse effect read error from temporary file\n");+ st_fail("Reverse effect read error from temporary file");return(ST_EOF);
}
for (i = 0, j = len-1; i < j; i++, j--) {--- a/src/sf.c
+++ b/src/sf.c
@@ -137,7 +137,7 @@
samplesize = sizeof(float);
break;
default:
- st_fail("Soundfile input: unknown format 0x%x\n",+ st_fail("Soundfile input: unknown format 0x%x",sf->info.sf_packmode);
return(ST_EOF);
}
--- a/src/smp.c
+++ b/src/smp.c
@@ -233,12 +233,12 @@
}
if (strncmp(header.Id, SVmagic, 17) != 0)
{- st_fail_errno(ft,ST_EHDR,"SMP header does not begin with magic word %s\n", SVmagic);
+ st_fail_errno(ft,ST_EHDR,"SMP header does not begin with magic word %s", SVmagic);
return(ST_EOF);
}
if (strncmp(header.version, SVvers, 4) != 0)
{- st_fail_errno(ft,ST_EHDR,"SMP header is not version %s\n", SVvers);
+ st_fail_errno(ft,ST_EHDR,"SMP header is not version %s", SVvers);
return(ST_EOF);
}
@@ -288,7 +288,7 @@
smp->dataStart = samplestart;
ft->length = smp->NoOfSamps;
- st_report("SampleVision trailer:\n");+ st_report("SampleVision trailer:"); for(i = 0; i < 8; i++) if (1 || trailer.loops[i].count) {#ifdef __alpha__
st_report("Loop %d: start: %6d", i, trailer.loops[i].start);@@ -299,12 +299,12 @@
#endif
st_report(" count: %6d", trailer.loops[i].count); switch(trailer.loops[i].type) {- case 0: st_report("type: off\n"); break;- case 1: st_report("type: forward\n"); break;- case 2: st_report("type: forward/backward\n"); break;+ case 0: st_report("type: off"); break;+ case 1: st_report("type: forward"); break;+ case 2: st_report("type: forward/backward"); break;}
}
- st_report("MIDI Note number: %d\n\n", trailer.MIDInote);+ st_report("MIDI Note number: %d", trailer.MIDInote);ft->instr.nloops = 0;
for(i = 0; i < 8; i++)
--- a/src/sndrtool.c
+++ b/src/sndrtool.c
@@ -117,7 +117,7 @@
st_fail_errno(ft,ST_EHDR,"SND: unexpected EOF in SND header");
return(ST_EOF);
}
- st_report("%s",buf);+ st_debug("%s",buf);}
ft->info.channels = 1;
--- a/src/sox.c
+++ b/src/sox.c
@@ -165,6 +165,23 @@
static struct st_effect user_efftab[MAX_USER_EFF];
static int nuser_effects;
+static int verbosity_level = 2; /* be noisy on stderr */
+static char * myname = 0;
+
+
+
+static void sox_output_message(int level, st_output_message_t m)
+{+ if (verbosity_level >= level)
+ {+ fprintf(stderr, "%s ", myname);
+ st_output_message(stderr, m);
+ fprintf(stderr, "\n");
+ }
+}
+
+
+
int main(int argc, char **argv)
{file_options_t *fo;
@@ -176,6 +193,8 @@
if (i >= sizeof("soxmix") - 1) soxmix = strcmp(myname + i - (sizeof("soxmix") - 1), "soxmix") == 0;+ st_output_message_handler = sox_output_message;
+
/* Loop over arguments and filenames, stop when an effect name is
* found.
*/
@@ -183,7 +202,7 @@
{if (file_count >= MAX_FILES)
{- st_fail("to many filenames. max of %d input files and 1 output file\n", MAX_INPUT_FILES);+ st_fail("too many filenames. max of %d input files and 1 output file", MAX_INPUT_FILES);exit(1);
}
@@ -295,7 +314,7 @@
return(0);
}
-static char *getoptstr = "+r:v:t:c:C:phsuUAaig1b2w34lfdxVSqo";
+static char *getoptstr = "+r:v:t:c:C:phsuUAaig1b2w34lfdxV::Sqo";
static struct option long_options[] =
{@@ -368,7 +387,7 @@
}
fo->uservolume = 1;
if (fo->volume < 0.0)
- st_report("Volume adjustment is negative. This will result in a phase change\n");+ st_report("Volume adjustment is negative. This will result in a phase change");break;
case 'c':
@@ -452,7 +471,17 @@
break;
case 'V':
- verbose = 1;
+ str = optarg;
+ if (optarg == NULL)
+ {+ ++verbosity_level;
+ }
+ else if (sscanf(str, "%i", &verbosity_level) == 0)
+ {+ st_fail("argument for -V must be an integer");+ cleanup();
+ exit(1);
+ }
break;
case 'S':
@@ -537,7 +566,7 @@
(file_desc[f]->info.channels > 1) ? "channels" : "channel");
if (file_desc[f]->comment)
- st_report("Input file %s: comment \"%s\"\n",+ st_report("Input file %s: comment \"%s\"",file_desc[f]->filename, file_desc[f]->comment);
}
@@ -621,7 +650,7 @@
(file_desc[file_count-1]->info.channels > 1) ? "channels" : "channel");
if (file_desc[file_count-1]->comment)
- st_report("Output file: comment \"%s\"\n", + st_report("Output file: comment \"%s\"", file_desc[file_count-1]->comment);
}
@@ -692,7 +721,7 @@
(st_ssize_t)ST_BUFSIZ);
if (ilen[0] > ST_BUFSIZ)
{- st_warn("WARNING: Corrupt value of %d! Assuming 0 bytes read.\n", ilen);+ st_warn("WARNING: Corrupt value of %d! Assuming 0 bytes read.", ilen);ilen[0] = 0;
}
@@ -886,7 +915,7 @@
{if (nuser_effects >= MAX_USER_EFF)
{- st_fail("Too many effects specified.\n");+ st_fail("Too many effects specified.");cleanup();
exit(2);
}
--- a/src/speed.c
+++ b/src/speed.c
@@ -64,7 +64,7 @@
/*
static void debug(char * where, speed_t s)
{- fprintf(stderr, "%s: f=%f r=%f comp=%d i=%d ic=%d frac=%f state=%d v=%f\n",
+ st_debug("%s: f=%f r=%f comp=%d i=%d ic=%d frac=%f state=%d v=%f",where, s->factor, s->rate, s->compression, s->index,
s->icbuf, s->frac, s->state, s->cbuf[0]);
}
@@ -107,7 +107,7 @@
if (n && (!sscanf(argv[0], SPEED_FLOAT_SCAN, &speed->factor) ||
(cent==0 && speed->factor<=ZERO)))
{- printf("n = %d cent = %d speed = %f\n",n,cent,speed->factor);+ st_debug("n = %d cent = %d speed = %f",n,cent,speed->factor);st_fail(st_speed_effect.usage);
return ST_EOF;
}
@@ -114,7 +114,7 @@
else if (cent != 0) /* CONST==2**(1/1200) */
{speed->factor = pow((double)1.00057778950655, speed->factor);
- /* fprintf(stderr, "Speed factor: %f\n", speed->factor);*/
+ /* st_debug("Speed factor: %f", speed->factor);*/}
return ST_SUCCESS;
--- a/src/sphere.c
+++ b/src/sphere.c
@@ -177,7 +177,7 @@
if (!strcmp(sphere->shorten_check,"ajkg"))
{- st_fail_errno(ft,ST_EFMT,"File uses shorten compression, can not handle this.\n");
+ st_fail_errno(ft,ST_EFMT,"File uses shorten compression, can not handle this.");
free(buf);
return(ST_EOF);
}
@@ -268,7 +268,7 @@
if (st_seeki(ft, 0L, 0) != 0)
{- st_fail_errno(ft,errno,"Could not rewird output file to rewrite sphere header.\n");
+ st_fail_errno(ft,errno,"Could not rewird output file to rewrite sphere header.");
return (ST_EOF);
}
--- a/src/st.h
+++ b/src/st.h
@@ -356,9 +356,7 @@
ft_t st_initformat(void);
int st_parsesamples(st_rate_t rate, char *str, st_size_t *samples, char def);
-/* FIXME: these declared in util.c, global is inappropriate for lib */
-extern int verbose; /* be noisy on stderr */
-extern char *myname;
+extern char const * filename;
#define ST_EOF (-1)
#define ST_SUCCESS (0)
--- a/src/st_i.h
+++ b/src/st_i.h
@@ -102,9 +102,26 @@
double st_swapd(double d);
/* util.c */
-void st_report(const char *, ...);
-void st_warn(const char *, ...);
+struct st_output_message_s;
+typedef struct st_output_message_s * st_output_message_t;
+typedef void (* st_output_message_handler_t)(int level, st_output_message_t);
+extern st_output_message_handler_t st_output_message_handler;
+void st_output_message(FILE * file, st_output_message_t);
+
void st_fail(const char *, ...);
+void st_warn(const char *, ...);
+void st_report(const char *, ...);
+void st_debug(const char *, ...);
+void st_debug_more(char const * fmt, ...);
+void st_debug_most(char const * fmt, ...);
+
+#define st_fail filename=__FILE__,st_fail
+#define st_warn filename=__FILE__,st_warn
+#define st_report filename=__FILE__,st_report
+#define st_debug filename=__FILE__,st_debug
+#define st_debug_more filename=__FILE__,st_debug_more
+#define st_debug_most filename=__FILE__,st_debug_most
+
void st_fail_errno(ft_t, int, const char *, ...);
int st_is_bigendian(void);
--- a/src/stat.c
+++ b/src/stat.c
@@ -139,7 +139,7 @@
if (bitmask && (stat->fft_size & ~bitmask))
{- st_fail("FFT can only use sample buffers of 2^n. Buffer size used is %ld\n",stat->fft_size);+ st_fail("FFT can only use sample buffers of 2^n. Buffer size used is %ld",stat->fft_size);return(ST_EOF);
}
@@ -148,7 +148,7 @@
if (!stat->re || !stat->im)
{- st_fail("Unable to allocate memory for FFT buffers.\n");+ st_fail("Unable to allocate memory for FFT buffers.");return (ST_EOF);
}
}
--- a/src/stretch.c
+++ b/src/stretch.c
@@ -86,7 +86,7 @@
/*
static void debug(stretch_t s, char * where)
{- fprintf(stderr,
+ st_debug(
"%s: (f=%.2f w=%.2f r=%.2f f=%.2f)"
" st=%d s=%d ii=%d is=%d oi=%d os=%d fs=%d\n",
where, s->factor, s->window, s->shift, s->fading,
--- a/src/sunaudio.c
+++ b/src/sunaudio.c
@@ -76,7 +76,7 @@
st_fail_errno(ft,errno,"Unable to get device information.");
return(ST_EOF);
}
- st_report("Hardware detected: %s\n",audio_dev.name);+ st_report("Hardware detected: %s",audio_dev.name); if (strcmp("SUNW,am79c30",audio_dev.name) == 0) {simple_hw = 1;
@@ -210,7 +210,7 @@
st_fail_errno(ft,errno,"Unable to get device information.");
return(ST_EOF);
}
- st_report("Hardware detected: %s\n",audio_dev.name);+ st_report("Hardware detected: %s",audio_dev.name); if (strcmp("SUNW,am79c30",audio_dev.name) == 0) {simple_hw = 1;
--- a/src/tx16w.c
+++ b/src/tx16w.c
@@ -123,7 +123,7 @@
*/
/* Check to make sure we got a good filetype ID from file */
- st_report("Found header filetype %s",filetype);+ st_debug("Found header filetype %s",filetype);if(strcmp(filetype,"LM8953"))
{st_fail_errno(ft,ST_EHDR,"Invalid filetype ID in input file header, != LM8953");
@@ -166,11 +166,11 @@
break;
}
if ( blewIt ) {- st_report("Invalid sample rate identifier found %d", (int)sample_rate);+ st_debug("Invalid sample rate identifier found %d", (int)sample_rate);ft->info.rate = 33333;
}
}
- st_report("Sample rate = %ld",ft->info.rate);+ st_debug("Sample rate = %ld",ft->info.rate);ft->info.channels = 1 ; /* not sure about stereo sample data yet ??? */
ft->info.size = ST_SIZE_WORD; /* this is close enough */
@@ -249,7 +249,7 @@
{struct WaveHeader_ WH;
- st_report("tx16w selected output");+ st_debug("tx16w selected output");memset(&WH, 0, sizeof(struct WaveHeader_));
@@ -308,7 +308,7 @@
/* If file header needs fixing up, for example it needs the */
/* the number of samples in a field, seek back and write them here. */
- st_report("tx16w:output finished");+ st_debug("tx16w:output finished");memset(&WH, 0, sizeof(struct WaveHeader_));
strncpy(WH.filetype,"LM8953",6);
@@ -326,7 +326,7 @@
else WH.sample_rate = 2;
if (tx16w_len >= TXMAXLEN) {- st_warn("Sound too large for TX16W. Truncating, Loop Off\n");+ st_warn("Sound too large for TX16W. Truncating, Loop Off");AttackLength = TXMAXLEN/2;
LoopLength = TXMAXLEN/2;
}
--- a/src/util.c
+++ b/src/util.c
@@ -25,56 +25,109 @@
* and utility routines for other main programs to use.
*/
-/* export flags */
-/* FIXME: To be moved inside of fileop structure per handler. */
-int verbose = 0; /* be noisy on stderr */
-/* FIXME: These functions are user level concepts. Move them outside
- * the ST library.
- */
-char *myname = 0;
-void st_report(const char *fmt, ...)
+
+struct st_output_message_s
{- va_list args;
+ char const * filename;
+ char const * fmt;
+ va_list ap;
+};
- if (! verbose)
- return;
- fprintf(stderr, "%s: ", myname);
- va_start(args, fmt);
- vfprintf(stderr, fmt, args);
- va_end(args);
- fprintf(stderr, "\n");
-}
-void st_warn(const char *fmt, ...)
+st_output_message_handler_t st_output_message_handler = NULL;
+
+
+
+/* This is a bit of a hack. It's useful to have the ST library
+ * report which driver (i.e. format or effect handler) is outputing
+ * the message. Using the filename for this purpose is only an
+ * approximation, but it saves a lot of work. ;)
+ */
+char const * filename = 0;
+
+
+
+void st_output_message(FILE * file, st_output_message_t message)
{- va_list args;
+ char buffer[10];
+ char const * drivername;
+ char const * dot_pos;
+
+ drivername = strrchr(message->filename, '/');
+ if (drivername != NULL)
+ {+ ++drivername;
+ }
+ else
+ {+ drivername = strrchr(message->filename, '\\');
+ if (drivername != NULL)
+ {+ ++drivername;
+ }
+ else
+ {+ drivername = message->filename;
+ }
+ }
- fprintf(stderr, "%s: ", myname);
- va_start(args, fmt);
+ dot_pos = strrchr(drivername, '.');
+ if (dot_pos != NULL && dot_pos - drivername <= sizeof(buffer) - 1)
+ {+ strncpy(buffer, drivername, dot_pos - drivername);
+ buffer[dot_pos - drivername] = '\0';
+ drivername = buffer;
+ }
- vfprintf(stderr, fmt, args);
- va_end(args);
- fprintf(stderr, "\n");
+ fprintf(file, "%s: ", drivername);
+ vfprintf(file, message->fmt, message->ap);
}
-/* Warning: This function is depricated. st_fail_errno() is
- * the preferred way so that applications can control printing
- * to their choice; not just stderr.
- */
-void st_fail(const char *fmt, ...)
+
+
+static void st_emit_message(int level, char const * fmt, va_list ap)
{- va_list args;
+ if (st_output_message_handler != NULL)
+ {+ struct st_output_message_s m;
+ m.filename = filename;
+ m.fmt = fmt;
+ m.ap = ap;
+ (*st_output_message_handler)(level, &m);
+ }
+}
- fprintf(stderr, "%s: ", myname);
- va_start(args, fmt);
- vfprintf(stderr, fmt, args);
- va_end(args);
- fprintf(stderr, "\n");
+
+#undef st_fail
+#undef st_warn
+#undef st_report
+#undef st_debug
+#undef st_debug_more
+#undef st_debug_most
+
+#define ST_MESSAGE_FUNCTION(name,level) \
+void name(char const * fmt, ...) \
+{ \+ va_list args; \
+\
+ va_start(args, fmt); \
+ st_emit_message(level, fmt, args); \
+ va_end(args); \
}
+
+ST_MESSAGE_FUNCTION(st_fail , 1)
+ST_MESSAGE_FUNCTION(st_warn , 2)
+ST_MESSAGE_FUNCTION(st_report, 3)
+ST_MESSAGE_FUNCTION(st_debug , 4)
+ST_MESSAGE_FUNCTION(st_debug_more , 5)
+ST_MESSAGE_FUNCTION(st_debug_most , 6)
+
+#undef ST_MESSAGE_FUNCTION
+
/* Warning: no error checking is done with errstr. Be sure not to
--- a/src/voc.c
+++ b/src/voc.c
@@ -275,15 +275,15 @@
ft->info.encoding = ST_ENCODING_UNSIGNED;
break;
case VOC_FMT_CRLADPCM4: /* 1 Creative 8-bit to 4-bit ADPCM */
- st_warn ("Unsupported VOC format CRLADPCM4 %d", v->format);+ st_fail ("Unsupported VOC format CRLADPCM4 %d", v->format);rtn=ST_EOF;
break;
case VOC_FMT_CRLADPCM3: /* 2 Creative 8-bit to 3-bit ADPCM */
- st_warn ("Unsupported VOC format CRLADPCM3 %d", v->format);+ st_fail ("Unsupported VOC format CRLADPCM3 %d", v->format);rtn=ST_EOF;
break;
case VOC_FMT_CRLADPCM2: /* 3 Creative 8-bit to 2-bit ADPCM */
- st_warn ("Unsupported VOC format CRLADPCM2 %d", v->format);+ st_fail ("Unsupported VOC format CRLADPCM2 %d", v->format);rtn=ST_EOF;
break;
case VOC_FMT_LIN16: /* 4 16-bit signed PCM */
@@ -296,11 +296,11 @@
ft->info.encoding = ST_ENCODING_ULAW;
break;
case VOC_FMT_CRLADPCM4A: /*0x200 Creative 16-bit to 4-bit ADPCM */
- printf ("Unsupported VOC format CRLADPCM4A %d", v->format);+ st_fail ("Unsupported VOC format CRLADPCM4A %d", v->format);rtn=ST_EOF;
break;
default:
- printf ("Unknown VOC format %d", v->format);+ st_fail ("Unknown VOC format %d", v->format);rtn=ST_EOF;
break;
}
@@ -655,27 +655,37 @@
st_readb(ft, &uc);
/* Falling! Falling! */
case VOC_TEXT:
- {- uint32_t i;
- /* Could add to comment in SF? */
- for(i = 0; i < sblen; i++) {- st_readb(ft, (unsigned char *)&trash);
- /* uncomment lines below to display text */
- /* Note, if this is uncommented, studio */
- /* will not be able to read the VOC file */
- /* ANN: added verbose dump of text info */
- /* */
- if (verbose) {- if ((trash != '\0') && (trash != '\r'))
- putc (trash, stderr);
- }
- /* */
+ { + /* TODO: Could add to comment in SF? */
+
+ /* Note, if this is sent to stderr, studio */
+ /* will not be able to read the VOC file */
+
+ uint32_t i = sblen;
+ char c/*, line_buf[80];
+ int len = 0*/;
+
+ while (i--)
+ {+ st_readb(ft, (unsigned char *)&c);
+ /* FIXME: this needs to be tested but I couldn't
+ * find a voc file with a VOC_TEXT chunk :(
+ if (c != '\0' && c != '\r')
+ line_buf[len++] = c;
+ if (len && (c == '\0' || c == '\r' ||
+ i == 0 || len == sizeof(line_buf) - 1))
+ {+ st_report("%s", line_buf);+ line_buf[len] = '\0';
+ len = 0;
}
+ */
+ }
}
continue; /* get next block */
case VOC_LOOP:
case VOC_LOOPEND:
- st_report("File %s: skipping repeat loop");+ st_debug("File %s: skipping repeat loop");for(i = 0; i < sblen; i++)
st_readb(ft, (unsigned char *)&trash);
break;
@@ -720,7 +730,7 @@
/* can be grabed. */
continue;
default:
- st_report("File %s: skipping unknown block code %d",+ st_debug("File %s: skipping unknown block code %d",ft->filename, block);
for(i = 0; i < sblen; i++)
st_readb(ft, (unsigned char *)&trash);
--- a/src/vorbis.c
+++ b/src/vorbis.c
@@ -211,7 +211,7 @@
if (num_read == 0)
return (BUF_EOF);
else if (num_read == OV_HOLE)
- st_warn("Warning: hole in stream; probably harmless\n");
+ st_warn("Warning: hole in stream; probably harmless");
else if (num_read < 0)
return (BUF_ERROR);
else
--- a/src/wav.c
+++ b/src/wav.c
@@ -409,7 +409,7 @@
Label);
return ST_EOF;
}
- st_report("WAV Chunk %s", magic);+ st_debug("WAV Chunk %s", magic);if (st_readdw(ft, len) == ST_EOF)
{st_fail_errno(ft, ST_EHDR, "WAVE file %s chunk is to short",
@@ -474,7 +474,7 @@
/* RIFX is a Big-endian RIFF */
if (strncmp("RIFX", magic, 4) == 0) {- st_report("Found RIFX header, swapping bytes");+ st_debug("Found RIFX header, swapping bytes");ft->swap = ft->swap ? 0 : 1;
}
@@ -728,7 +728,7 @@
st_readw(ft, &(wav->nCoefs));
if (wav->nCoefs < 7 || wav->nCoefs > 0x100) {- st_fail_errno(ft,ST_EOF,"ADPCM file nCoefs (%.4hx) makes no sense\n", wav->nCoefs);
+ st_fail_errno(ft,ST_EOF,"ADPCM file nCoefs (%.4hx) makes no sense", wav->nCoefs);
return ST_EOF;
}
wav->packet = (unsigned char *)malloc(wav->blockAlign);
@@ -766,7 +766,7 @@
st_readw(ft, (unsigned short *)&(wav->iCoefs[i]));
len -= 2;
if (i<14) errct += (wav->iCoefs[i] != iCoef[i/2][i%2]);
- /* fprintf(stderr,"iCoefs[%2d] %4d\n",i,wav->iCoefs[i]); */
+ /* st_debug("iCoefs[%2d] %4d",i,wav->iCoefs[i]); */}
if (errct) st_warn("base iCoefs differ in %d/14 positions",errct);}
@@ -925,7 +925,7 @@
wav->numSamples =
AdpcmSamplesIn(dwDataLength, ft->info.channels,
wav->blockAlign, wav->samplesPerBlock);
- /*st_report("datalen %d, numSamples %d",dwDataLength, wav->numSamples);*/+ /*st_debug("datalen %d, numSamples %d",dwDataLength, wav->numSamples);*/wav->blockSamplesRemaining = 0; /* Samples left in buffer */
ft->length = wav->numSamples*ft->info.channels;
break;
@@ -936,7 +936,7 @@
wav->numSamples =
ImaSamplesIn(dwDataLength, ft->info.channels,
wav->blockAlign, wav->samplesPerBlock);
- /*st_report("datalen %d, numSamples %d",dwDataLength, wav->numSamples);*/+ /*st_debug("datalen %d, numSamples %d",dwDataLength, wav->numSamples);*/wav->blockSamplesRemaining = 0; /* Samples left in buffer */
initImaTable();
ft->length = wav->numSamples*ft->info.channels;
@@ -955,10 +955,10 @@
ft->length = wav->numSamples*ft->info.channels;
}
- st_report("Reading Wave file: %s format, %d channel%s, %d samp/sec",+ st_debug("Reading Wave file: %s format, %d channel%s, %d samp/sec",wav_format_str(wav->formatTag), ft->info.channels,
wChannels == 1 ? "" : "s", dwSamplesPerSecond);
- st_report(" %d byte/sec, %d block align, %d bits/samp, %u data bytes",+ st_debug(" %d byte/sec, %d block align, %d bits/samp, %u data bytes",dwAvgBytesPerSec, wav->blockAlign, wBitsPerSample, dwDataLength);
/* Can also report extended fmt information */
@@ -965,13 +965,13 @@
switch (wav->formatTag)
{case WAVE_FORMAT_ADPCM:
- st_report(" %d Extsize, %d Samps/block, %d bytes/block %d Num Coefs, %d Samps/chan",+ st_debug(" %d Extsize, %d Samps/block, %d bytes/block %d Num Coefs, %d Samps/chan",wExtSize,wav->samplesPerBlock,bytesPerBlock,wav->nCoefs,
wav->numSamples);
break;
case WAVE_FORMAT_IMA_ADPCM:
- st_report(" %d Extsize, %d Samps/block, %d bytes/block %d Samps/chan",+ st_debug(" %d Extsize, %d Samps/block, %d bytes/block %d Samps/chan",wExtSize, wav->samplesPerBlock, bytesPerBlock,
wav->numSamples);
break;
@@ -978,13 +978,13 @@
#ifdef ENABLE_GSM
case WAVE_FORMAT_GSM610:
- st_report("GSM .wav: %d Extsize, %d Samps/block, %d Samples/chan",+ st_debug("GSM .wav: %d Extsize, %d Samps/block, %d Samples/chan",wExtSize, wav->samplesPerBlock, wav->numSamples);
break;
#endif
default:
- st_report(" %d Samps/chans", wav->numSamples);+ st_debug(" %d Samps/chans", wav->numSamples);}
/* Horrible way to find Cool Edit marker points. Taken from Quake source*/
@@ -1015,12 +1015,12 @@
if (strncmp(magic, "INFO", 4) == 0)
{/*Skip*/
- st_report("Type INFO");+ st_debug("Type INFO");}
else if (strncmp(magic, "adtl", 4) == 0)
{/* Skip */
- st_report("Type adtl");+ st_debug("Type adtl");}
else
{@@ -1028,10 +1028,10 @@
break;
if (strncmp(magic,"ICRD",4) == 0)
{- st_report("Chunk ICRD");+ st_debug("Chunk ICRD");if (len > 254)
{- st_warn("Possible buffer overflow hack attack (ICRD)!\n");+ st_warn("Possible buffer overflow hack attack (ICRD)!");break;
}
st_reads(ft,text,len);
@@ -1047,10 +1047,10 @@
}
else if (strncmp(magic,"ISFT",4) == 0)
{- st_report("Chunk ISFT");+ st_debug("Chunk ISFT");if (len > 254)
{- st_warn("Possible buffer overflow hack attack (ISFT)!\n");+ st_warn("Possible buffer overflow hack attack (ISFT)!");break;
}
st_reads(ft,text,len);
@@ -1066,7 +1066,7 @@
}
else if (strncmp(magic,"cue ",4) == 0)
{- st_report("Chunk cue ");+ st_debug("Chunk cue ");st_seeki(ft,len-4,SEEK_CUR);
st_readdw(ft,&dwLoopPos);
ft->loops[0].start = dwLoopPos;
@@ -1073,7 +1073,7 @@
}
else if (strncmp(magic,"ltxt",4) == 0)
{- st_report("Chunk ltxt");+ st_debug("Chunk ltxt");st_readdw(ft,&dwLoopPos);
ft->loops[0].length = dwLoopPos - ft->loops[0].start;
if (len > 4)
@@ -1081,7 +1081,7 @@
}
else
{- st_report("Attempting to seek beyond unsupported chunk '%c%c%c%c' of length %d bytes\n", magic[0], magic[1], magic[2], magic[3], len);+ st_debug("Attempting to seek beyond unsupported chunk '%c%c%c%c' of length %d bytes", magic[0], magic[1], magic[2], magic[3], len);len = (len + 1) & ~1;
st_seeki(ft, len, SEEK_CUR);
}
@@ -1493,7 +1493,7 @@
case ST_ENCODING_IMA_ADPCM:
if (wChannels>16)
{- st_fail_errno(ft,ST_EOF,"Channels(%d) must be <= 16\n",wChannels);
+ st_fail_errno(ft,ST_EOF,"Channels(%d) must be <= 16",wChannels);
return ST_EOF;
}
wFormatTag = WAVE_FORMAT_IMA_ADPCM;
@@ -1505,7 +1505,7 @@
case ST_ENCODING_ADPCM:
if (wChannels>16)
{- st_fail_errno(ft,ST_EOF,"Channels(%d) must be <= 16\n",wChannels);
+ st_fail_errno(ft,ST_EOF,"Channels(%d) must be <= 16",wChannels);
return ST_EOF;
}
wFormatTag = WAVE_FORMAT_ADPCM;
@@ -1518,7 +1518,7 @@
#ifdef ENABLE_GSM
if (wChannels!=1)
{- st_report("Overriding GSM audio from %d channel to 1\n",wChannels);+ st_report("Overriding GSM audio from %d channel to 1",wChannels);wChannels = ft->info.channels = 1;
}
wFormatTag = WAVE_FORMAT_GSM610;
@@ -1657,17 +1657,17 @@
st_writedw(ft, dwDataLength); /* data chunk size */
if (!second_header) {- st_report("Writing Wave file: %s format, %d channel%s, %d samp/sec",+ st_debug("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",+ st_debug(" %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",+ st_debug("Finished writing Wave file, %u data bytes %u samples",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",+ st_debug("GSM6.10 format: %u blocks %u padded samples %u padded data bytes",blocksWritten, dwSamplesWritten, dwDataLength);
if (wav->gsmbytecount != dwDataLength)
st_warn("help ! internal inconsistency - data_written %u gsmbytecount %u",--- a/src/wve.c
+++ b/src/wve.c
@@ -72,7 +72,7 @@
/* Check the magic word (null-terminated) */
st_reads(ft, magic, 16);
if (strncmp(magic, PSION_MAGIC, 15)==0) {- st_report("Found Psion magic word");+ st_debug("Found Psion magic word");}
else
{@@ -90,11 +90,11 @@
* it doesn't hurt.
*/
ft->swap = ft->swap ? 0 : 1;
- st_report("Found inverted PSION magic word. Swapping bytes.");+ st_debug("Found inverted PSION magic word. Swapping bytes.");}
else if (version == PSION_VERSION)
{- st_report("Found PSION magic word");+ st_debug("Found PSION magic word");}
else
{--- a/src/xa.c
+++ b/src/xa.c
@@ -117,20 +117,20 @@
if (st_readw(ft, &xa->header.bits) != ST_SUCCESS) return ST_EOF;
/* Output the data from the header */
- st_report("XA Header:");- st_report(" szID: %02x %02x %02x %02x |%c%c%c%c|",+ st_debug("XA Header:");+ st_debug(" szID: %02x %02x %02x %02x |%c%c%c%c|",magic[0], magic[1], magic[2], magic[3],
(magic[0] >= 0x20 && magic[0] <= 0x7e) ? magic[0] : '.',
(magic[1] >= 0x20 && magic[1] <= 0x7e) ? magic[1] : '.',
(magic[2] >= 0x20 && magic[2] <= 0x7e) ? magic[2] : '.',
(magic[3] >= 0x20 && magic[3] <= 0x7e) ? magic[3] : '.');
- st_report(" dwOutSize: %u", xa->header.outSize);- st_report(" wTag: 0x%04x", xa->header.tag);- st_report(" wChannels: %u", xa->header.channels);- st_report(" dwSampleRate: %u", xa->header.sampleRate);- st_report(" dwAvgByteRate: %u", xa->header.avgByteRate);- st_report(" wAlign: %u", xa->header.align);- st_report(" wBits: %u", xa->header.bits);+ st_debug(" dwOutSize: %u", xa->header.outSize);+ st_debug(" wTag: 0x%04x", xa->header.tag);+ st_debug(" wChannels: %u", xa->header.channels);+ st_debug(" dwSampleRate: %u", xa->header.sampleRate);+ st_debug(" dwAvgByteRate: %u", xa->header.avgByteRate);+ st_debug(" wAlign: %u", xa->header.align);+ st_debug(" wBits: %u", xa->header.bits);/* Populate the st_soundstream structure */
ft->info.encoding = ST_ENCODING_SIGN2;