qemu-e2k/audio/mixeng.c

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/*
* QEMU Mixing engine
*
* Copyright (c) 2004 Vassili Karpov (malc)
* Copyright (c) 1998 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include "vl.h"
//#define DEBUG_FP
#include "audio/mixeng.h"
#define IN_T int8_t
#define IN_MIN CHAR_MIN
#define IN_MAX CHAR_MAX
#define SIGNED
#include "mixeng_template.h"
#undef SIGNED
#undef IN_MAX
#undef IN_MIN
#undef IN_T
#define IN_T uint8_t
#define IN_MIN 0
#define IN_MAX UCHAR_MAX
#include "mixeng_template.h"
#undef IN_MAX
#undef IN_MIN
#undef IN_T
#define IN_T int16_t
#define IN_MIN SHRT_MIN
#define IN_MAX SHRT_MAX
#define SIGNED
#include "mixeng_template.h"
#undef SIGNED
#undef IN_MAX
#undef IN_MIN
#undef IN_T
#define IN_T uint16_t
#define IN_MIN 0
#define IN_MAX USHRT_MAX
#include "mixeng_template.h"
#undef IN_MAX
#undef IN_MIN
#undef IN_T
t_sample *mixeng_conv[2][2][2] = {
{
{
conv_uint8_t_to_mono,
conv_uint16_t_to_mono
},
{
conv_int8_t_to_mono,
conv_int16_t_to_mono
}
},
{
{
conv_uint8_t_to_stereo,
conv_uint16_t_to_stereo
},
{
conv_int8_t_to_stereo,
conv_int16_t_to_stereo
}
}
};
f_sample *mixeng_clip[2][2][2] = {
{
{
clip_uint8_t_from_mono,
clip_uint16_t_from_mono
},
{
clip_int8_t_from_mono,
clip_int16_t_from_mono
}
},
{
{
clip_uint8_t_from_stereo,
clip_uint16_t_from_stereo
},
{
clip_int8_t_from_stereo,
clip_int16_t_from_stereo
}
}
};
/*
* 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.
*/
/*
* 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
*/
/* Private data */
typedef struct ratestuff {
uint64_t opos;
uint64_t opos_inc;
uint32_t ipos; /* position in the input stream (integer) */
st_sample_t ilast; /* last sample in the input stream */
} *rate_t;
/*
* Prepare processing.
*/
void *st_rate_start (int inrate, int outrate)
{
rate_t rate = (rate_t) qemu_mallocz (sizeof (struct ratestuff));
if (!rate) {
exit (EXIT_FAILURE);
}
if (inrate == outrate) {
// exit (EXIT_FAILURE);
}
if (inrate >= 65535 || outrate >= 65535) {
// exit (EXIT_FAILURE);
}
rate->opos = 0;
/* increment */
rate->opos_inc = (inrate * ((int64_t) UINT_MAX)) / outrate;
rate->ipos = 0;
rate->ilast.l = 0;
rate->ilast.r = 0;
return rate;
}
/*
* Processed signed long samples from ibuf to obuf.
* Return number of samples processed.
*/
void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
int *isamp, int *osamp)
{
rate_t rate = (rate_t) opaque;
st_sample_t *istart, *iend;
st_sample_t *ostart, *oend;
st_sample_t ilast, icur, out;
int64_t t;
ilast = rate->ilast;
istart = ibuf;
iend = ibuf + *isamp;
ostart = obuf;
oend = obuf + *osamp;
if (rate->opos_inc == 1ULL << 32) {
int i, n = *isamp > *osamp ? *osamp : *isamp;
for (i = 0; i < n; i++) {
obuf[i].l += ibuf[i].r;
obuf[i].r += ibuf[i].r;
}
*isamp = n;
*osamp = n;
return;
}
while (obuf < oend) {
/* Safety catch to make sure we have input samples. */
if (ibuf >= iend)
break;
/* read as many input samples so that ipos > opos */
while (rate->ipos <= (rate->opos >> 32)) {
ilast = *ibuf++;
rate->ipos++;
/* See if we finished the input buffer yet */
if (ibuf >= iend) goto the_end;
}
icur = *ibuf;
/* interpolate */
t = rate->opos & 0xffffffff;
out.l = (ilast.l * (INT_MAX - t) + icur.l * t) / INT_MAX;
out.r = (ilast.r * (INT_MAX - t) + icur.r * t) / INT_MAX;
/* output sample & increment position */
#if 0
*obuf++ = out;
#else
obuf->l += out.l;
obuf->r += out.r;
obuf += 1;
#endif
rate->opos += rate->opos_inc;
}
the_end:
*isamp = ibuf - istart;
*osamp = obuf - ostart;
rate->ilast = ilast;
}
void st_rate_stop (void *opaque)
{
qemu_free (opaque);
}