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/* ScummVM - Scumm Interpreter
* Copyright (C) 2001-2003 The ScummVM project
*
* 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 2
* 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, write to the Free Software
* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*
* $Header$
*
*/
/*
* The code in this file is based on code with Copyright 1998 Fabrice Bellard
* Fabrice original code is part of SoX (http://sox.sourceforge.net).
* Max Horn adapted that code to the needs of ScummVM and rewrote it partial,
* in the process removing any use of floating point arithmetic. Various other
* improvments over the original code were made.
*/
#include "stdafx.h"
#include "sound/rate.h"
#define FRAC_BITS 16
/**
* Audio rate converter based on simple linear Interpolation.
*
* The use of fractional increment allows us to use no buffer. It
* avoid the problems at the end of the buffer we had with the old
* method which stored a possibly big buffer of size
* lcm(in_rate,out_rate).
*
* Limited to sampling frequency <= 65535 Hz.
*/
template<bool stereo, bool reverseStereo>
class LinearRateConverter : public RateConverter {
protected:
bool _reverseStereo;
/** fractional position of the output stream in input stream unit */
unsigned long opos, opos_frac;
/** fractional position increment in the output stream */
unsigned long opos_inc, opos_inc_frac;
/** position in the input stream (integer) */
unsigned long ipos;
/** last sample(s) in the input stream (left/right channel) */
st_sample_t ilast[2];
/** current sample(s) in the input stream (left/right channel) */
st_sample_t icur[2];
public:
LinearRateConverter(st_rate_t inrate, st_rate_t outrate);
int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol);
int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
return (ST_SUCCESS);
}
};
/*
* Prepare processing.
*/
template<bool stereo, bool reverseStereo>
LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate, st_rate_t outrate) {
unsigned long incr;
if (inrate == outrate) {
error("Input and Output rates must be different to use rate effect");
}
if (inrate >= 65536 || outrate >= 65536) {
error("rate effect can only handle rates < 65536");
}
opos_frac = 0;
opos = 0;
/* increment */
incr = (inrate << FRAC_BITS) / outrate;
opos_inc_frac = incr & ((1UL << FRAC_BITS) - 1);
opos_inc = incr >> FRAC_BITS;
ipos = 0;
ilast[0] = ilast[1] = 0;
icur[0] = icur[1] = 0;
}
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
template<bool stereo, bool reverseStereo>
int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol)
{
st_sample_t *ostart, *oend;
st_sample_t out[2], tmpOut;
ostart = obuf;
oend = obuf + osamp * 2;
while (obuf < oend) {
// read enough input samples so that ipos > opos
while (ipos <= opos + 1) {
// Abort if we reached the end of the input buffer
if (input.eof())
goto the_end;
ilast[0] = icur[0];
icur[0] = input.read();
if (stereo) {
ilast[1] = icur[1];
icur[1] = input.read();
}
ipos++;
}
// Loop as long as the outpos trails behind, and as long as there is
// still space in the output buffer.
while (ipos > opos + 1) {
// interpolate
tmpOut = (st_sample_t)(ilast[0] + (((icur[0] - ilast[0]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));
// adjust volume
out[0] = out[1] = (st_sample_t)((tmpOut * vol) >> 8);
if (stereo) {
// interpolate
tmpOut = (st_sample_t)(ilast[1] + (((icur[1] - ilast[1]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));
// adjust volume
out[reverseStereo ? 0 : 1] = (st_sample_t)((tmpOut * vol) >> 8);
}
// output left channel
clampedAdd(*obuf++, out[0]);
// output right channel
clampedAdd(*obuf++, out[1]);
// Increment output position
unsigned long tmp = opos_frac + opos_inc_frac;
opos += opos_inc + (tmp >> FRAC_BITS);
opos_frac = tmp & ((1UL << FRAC_BITS) - 1);
// Abort if we reached the end of the output buffer
if (obuf >= oend)
goto the_end;
}
}
the_end:
return (ST_SUCCESS);
}
#pragma mark -
/**
* Simple audio rate converter for the case that the inrate equals the outrate.
*/
template<bool stereo, bool reverseStereo>
class CopyRateConverter : public RateConverter {
public:
virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
int16 tmp[2];
st_size_t len = osamp;
assert(input.isStereo() == stereo);
while (!input.eof() && len--) {
tmp[0] = tmp[1] = (input.read() * vol) >> 8;
if (stereo)
tmp[reverseStereo ? 0 : 1] = (input.read() * vol) >> 8;
clampedAdd(*obuf++, tmp[0]);
clampedAdd(*obuf++, tmp[1]);
}
return (ST_SUCCESS);
}
virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
return (ST_SUCCESS);
}
};
#pragma mark -
/**
* Create and return a RateConverter object for the specified input and output rates.
*/
RateConverter *makeRateConverter(st_rate_t inrate, st_rate_t outrate, bool stereo, bool reverseStereo) {
if (inrate != outrate) {
if (stereo) {
if (reverseStereo)
return new LinearRateConverter<true, true>(inrate, outrate);
else
return new LinearRateConverter<true, false>(inrate, outrate);
} else
return new LinearRateConverter<false, false>(inrate, outrate);
//return new ResampleRateConverter(inrate, outrate, 1);
} else {
if (stereo) {
if (reverseStereo)
return new CopyRateConverter<true, true>();
else
return new CopyRateConverter<true, false>();
} else
return new CopyRateConverter<false, false>();
}
}
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