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/*
* August 21, 1998
* Copyright 1998 Fabrice Bellard.
*
* [Rewrote completly the code of Lance Norskog And Sundry
* Contributors with a more efficient algorithm.]
*
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Lance Norskog And Sundry Contributors are not responsible for
* the consequences of using this software.
*/
/*
* Sound Tools rate change effect file.
*/
#include "rate.h"
#include <math.h>
/*
* 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 16 bit samples and sampling frequency <= 65535 Hz. If
* the input & output frequencies are equal, a delay of one sample is
* introduced. Limited to processing 32-bit count worth of samples.
*
* 1 << FRAC_BITS evaluating to zero in several places. Changed with
* an (unsigned long) cast to make it safe. MarkMLl 2/1/99
*
* Replaced all uses of floating point arithmetic by fixed point
* calculations (Max Horn 2003-07-18).
*/
#define FRAC_BITS 16
/* Private data */
typedef struct ratestuff
{
unsigned long opos_frac; /* fractional position of the output stream in input stream unit */
unsigned long opos;
unsigned long opos_inc_frac; /* fractional position increment in the output stream */
unsigned long opos_inc;
unsigned long ipos; /* position in the input stream (integer) */
st_sample_t ilast[2]; /* last sample(s) in the input stream (left/right channel) */
} *rate_t;
/*
* Prepare processing.
*/
int st_rate_start(eff_t effp, st_rate_t inrate, st_rate_t outrate)
{
rate_t rate = (rate_t) effp->priv;
unsigned long incr;
if (inrate == outrate) {
st_fail("Input and Output rates must be different to use rate effect");
return (ST_EOF);
}
if (inrate >= 65536 || outrate >= 65536) {
st_fail("rate effect can only handle rates < 65536");
return (ST_EOF);
}
rate->opos_frac = 0;
rate->opos = 0;
/* increment */
incr = (inrate << FRAC_BITS) / outrate;
rate->opos_inc_frac = incr & ((1UL << FRAC_BITS) - 1);
rate->opos_inc = incr >> FRAC_BITS;
rate->ipos = 0;
rate->ilast[0] = 0;
rate->ilast[1] = 0;
return (ST_SUCCESS);
}
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
template<bool stereo>
int st_rate_flow(eff_t effp, AudioInputStream &input, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol)
{
rate_t rate = (rate_t) effp->priv;
st_sample_t *ostart, *oend;
st_sample_t ilast[2], icur[2], out;
unsigned long tmp;
ilast[0] = rate->ilast[0];
if (stereo)
ilast[1] = rate->ilast[1];
ostart = obuf;
oend = obuf + *osamp * 2;
if (stereo)
assert(input.size() % 2 == 0); // Stereo code assumes even number of input samples
// If the input position exceeds the output position, then we aborted the
// previous conversion run because the output buffer was full. Resume!
if (rate->ipos > rate->opos)
goto resume;
while (obuf < oend && !input.eof()) {
/* read enough input samples so that ipos > opos */
while (rate->ipos <= rate->opos) {
ilast[0] = input.read();
if (stereo)
ilast[1] = input.read();
rate->ipos++;
/* See if we finished the input buffer yet */
if (input.eof())
goto the_end;
}
// read the input sample(s)
icur[0] = input.read();
if (stereo)
icur[1] = input.read();
resume:
// Loop as long as the outpos trails behind, and as long as there is
// still space in the output buffer.
while (rate->ipos > rate->opos && obuf < oend) {
// interpolate
out = ilast[0] + (((icur[0] - ilast[0]) * rate->opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS);
// adjust volume
out = out * vol / 256;
// output left channel sample
clampedAdd(*obuf++, out);
if (stereo) {
// interpolate
out = ilast[1] + (((icur[1] - ilast[1]) * rate->opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS);
// adjust volume
out = out * vol / 256;
}
// output right channel sample
clampedAdd(*obuf++, out);
// Increment output position
tmp = rate->opos_frac + rate->opos_inc_frac;
rate->opos = rate->opos + rate->opos_inc + (tmp >> FRAC_BITS);
rate->opos_frac = tmp & ((1UL << FRAC_BITS) - 1);
}
// Increment input position again (for the sample we read now)
rate->ipos++;
ilast[0] = icur[0];
if (stereo)
ilast[1] = icur[1];
}
the_end:
*osamp = (obuf - ostart) / 2;
rate->ilast[0] = ilast[0];
if (stereo)
rate->ilast[1] = ilast[1];
return (ST_SUCCESS);
}
#pragma mark -
LinearRateConverter::LinearRateConverter(st_rate_t inrate, st_rate_t outrate) {
st_rate_start(&effp, inrate, outrate);
}
int LinearRateConverter::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) {
if (input.isStereo())
return st_rate_flow<true>(&effp, input, obuf, osamp, vol);
else
return st_rate_flow<false>(&effp, input, obuf, osamp, vol);
}
int LinearRateConverter::drain(st_sample_t *obuf, st_size_t *osamp, st_volume_t vol) {
return (ST_SUCCESS);
}
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