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// ----------------------------------------------------------------------------
//
// Copyright (C) 2006-2011 Fons Adriaensen <fons@linuxaudio.org>
//
// This program is free software; you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation; either version 3 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program. If not, see <http://www.gnu.org/licenses/>.
//
// ----------------------------------------------------------------------------
#include <stdlib.h>
#include <string.h>
#include <getopt.h>
#include <stdio.h>
#include <math.h>
#include <zita-resampler/vresampler.h>
#include "audiofile.h"
enum { HELP, CAF, WAV, AMB, AIFF, FLAC, BIT16, BIT24, FLOAT, CENT, REC, TRI, LIPS, PAD };
enum { BUFFSIZE = 0x4000, FILTSIZE = 96 };
static unsigned int type = Audiofile::TYPE_WAV;
static unsigned int form = Audiofile::FORM_24BIT;
static double cent = 0;
static unsigned int dith = Audiofile::DITHER_NONE;
static bool zpad = false;
static void help (void)
{
fprintf (stderr, "\nzretune %s\n", VERSION);
fprintf (stderr, "(C) 2007-2012 Fons Adriaensen <fons@linuxaudio.org>\n");
fprintf (stderr, "Usage: zretune <options> <input file> <output file>.\n");
fprintf (stderr, "Options:\n");
fprintf (stderr, " Display this text: --help\n");
fprintf (stderr, " Output file type: --caf, --wav, --amb, --aiff, --flac\n");
fprintf (stderr, " Resampling ratio: --cent <pitch change>\n");
fprintf (stderr, " Output sample format: --16bit, --24bit, --float\n");
fprintf (stderr, " Dither type (16 bit): --rec, --tri, --lips\n");
fprintf (stderr, " Add zero padding : --pad\n");
fprintf (stderr, "The default output file format is wav, 24-bit, no dithering.\n");
fprintf (stderr, "Integer output formats are clipped, float output is not.\n\n");
exit (1);
}
static struct option options [] =
{
{ "help", 0, 0, HELP },
{ "caf", 0, 0, CAF },
{ "wav", 0, 0, WAV },
{ "amb", 0, 0, AMB },
{ "aiff", 0, 0, AIFF },
{ "flac", 0, 0, FLAC },
{ "16bit", 0, 0, BIT16 },
{ "24bit", 0, 0, BIT24 },
{ "float", 0, 0, FLOAT },
{ "cent", 1, 0, CENT },
{ "rec", 0, 0, REC },
{ "tri", 0, 0, TRI },
{ "lips", 0, 0, LIPS },
{ "pad", 0, 0, PAD },
{ 0, 0, 0, 0 }
};
static void procoptions (int ac, char *av [])
{
int k;
while ((k = getopt_long (ac, av, "", options, 0)) != -1)
{
switch (k)
{
case '?':
case HELP:
help ();
break;
case CAF:
type = Audiofile::TYPE_CAF;
break;
case WAV:
type = Audiofile::TYPE_WAV;
break;
case AMB:
type = Audiofile::TYPE_AMB;
break;
case AIFF:
type = Audiofile::TYPE_AIFF;
break;
case FLAC:
type = Audiofile::TYPE_FLAC;
break;
case BIT16:
form = Audiofile::FORM_16BIT;
break;
case BIT24:
form = Audiofile::FORM_24BIT;
break;
case FLOAT:
form = Audiofile::FORM_FLOAT;
break;
case CENT:
if (sscanf (optarg, "%lf", ¢) != 1)
{
fprintf (stderr, "Illegal value for --rate option: '%s'.\n", optarg);
exit (1);
}
break;
case REC:
dith = Audiofile::DITHER_RECT;
break;
case TRI:
dith = Audiofile::DITHER_TRIA;
break;
case LIPS:
dith = Audiofile::DITHER_LIPS;
break;
case PAD:
zpad = true;
break;
}
}
}
int main (int ac, char *av [])
{
Audiofile Ainp;
Audiofile Aout;
VResampler R;
unsigned int k, chan, z1, z2;
float *inpb, *outb;
bool done;
double ratio;
procoptions (ac, av);
if (ac - optind < 2)
{
fprintf (stderr, "Missing arguments, try --help.\n");
return 1;
}
if (ac - optind > 2 )
{
fprintf (stderr, "Too many arguments, try --help.\n");
return 1;
}
if (Ainp.open_read (av [optind]))
{
fprintf (stderr, "Can't open input file '%s'.\n", av [optind]);
return 1;
}
if ((cent < -1200) || (cent > 1200))
{
fprintf (stderr, "Pitch change %3.1lf is out of range.\n", cent);
Ainp.close ();
return 1;
}
ratio = pow (2.0, -cent / 1200.0);
R.setup (ratio, Ainp.chan (), FILTSIZE);
optind++;
if (Aout.open_write (av [optind], type, form, Ainp.rate(), Ainp.chan ()))
{
fprintf (stderr, "Can't open output file '%s'.\n", av [optind]);
Ainp.close ();
return 1;
}
if (dith != Audiofile::DITHER_NONE)
{
Aout.set_dither (dith);
}
if (zpad)
{
z1 = R.inpsize () - 1;
z2 = R.inpsize () - 1;
}
else
{
z1 = R.inpsize () / 2 - 1;
z2 = R.inpsize () / 2;
}
chan = Ainp.chan ();
inpb = new float [chan * BUFFSIZE];
if (cent != 0.0)
{
outb = new float [chan * BUFFSIZE];
// Insert zero samples at start.
R.inp_count = z1;
R.inp_data = 0;
R.out_count = BUFFSIZE;
R.out_data = outb;
done = false;
while (true)
{
R.process ();
if (R.inp_count == 0)
{
// Input buffer empty, read more samples, insert
// zeros at the end, or terminate.
if (done)
{
// We already inserted final zero samples.
// Write out any remaining output samples and terminate.
Aout.write (outb, BUFFSIZE - R.out_count);
break;
}
k = Ainp.read (inpb, BUFFSIZE);
if (k)
{
// Process next 'k' input samples.
R.inp_count = k;
R.inp_data = inpb;
}
else
{
// At end of input, insert zero samples.
R.inp_count = z2;
R.inp_data = 0;
done = true;
}
}
if (R.out_count == 0)
{
// Output buffer full, write to file.
Aout.write (outb, BUFFSIZE);
R.out_count = BUFFSIZE;
R.out_data = outb;
}
}
delete[] outb;
}
else
{
// No resampling, just copy.
while (1)
{
k = Ainp.read (inpb, BUFFSIZE);
if (k) Aout.write (inpb, k);
else break;
}
}
Ainp.close ();
Aout.close ();
delete[] inpb;
return 0;
}
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