audio_convert.c
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997, 1998, 1999, 2000, 2001 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 of the License, or (at your option) any later version.
This library 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
Library General Public License for more details.
You should have received a copy of the GNU Library General Public
License along with this library; if not, write to the Free
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Sam Lantinga
slouken@devolution.com
*/
/*
* This file was derived from SDL's SDL_audiocvt.c and is an attempt to
* address the shortcomings of it.
*
* Perhaps we can adapt some good filters from SoX?
*/
#if HAVE_CONFIG_H
# include <config.h>
#endif
#include "SDL.h"
#include "SDL_sound.h"
#define __SDL_SOUND_INTERNAL__
#include "SDL_sound_internal.h"
/* Functions for audio drivers to perform runtime conversion of audio format */
/*
* Toggle endianness. This filter is, of course, only applied to 16-bit
* audio data.
*/
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static void Sound_ConvertEndian(Sound_AudioCVT *cvt, Uint16 *format)
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{
int i;
Uint8 *data, tmp;
/* SNDDBG(("Converting audio endianness\n")); */
data = cvt->buf;
for (i = cvt->len_cvt / 2; i; --i)
{
tmp = data[0];
data[0] = data[1];
data[1] = tmp;
data += 2;
} /* for */
*format = (*format ^ 0x1000);
} /* Sound_ConvertEndian */
/*
* Toggle signed/unsigned. Apparently this is done by toggling the most
* significant bit of each sample.
*/
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static void Sound_ConvertSign(Sound_AudioCVT *cvt, Uint16 *format)
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{
int i;
Uint8 *data;
/* SNDDBG(("Converting audio signedness\n")); */
data = cvt->buf;
/* 16-bit sound? */
if ((*format & 0xFF) == 16)
{
/* Little-endian? */
if ((*format & 0x1000) != 0x1000)
++data;
for (i = cvt->len_cvt / 2; i; --i)
{
*data ^= 0x80;
data += 2;
} /* for */
} /* if */
else
{
for (i = cvt->len_cvt; i; --i)
*data++ ^= 0x80;
} /* else */
*format = (*format ^ 0x8000);
} /* Sound_ConvertSign */
/*
* Convert 16-bit to 8-bit. This is done by taking the most significant byte
* of each 16-bit sample.
*/
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static void Sound_Convert8(Sound_AudioCVT *cvt, Uint16 *format)
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{
int i;
Uint8 *src, *dst;
/* SNDDBG(("Converting to 8-bit\n")); */
src = cvt->buf;
dst = cvt->buf;
/* Little-endian? */
if ((*format & 0x1000) != 0x1000)
++src;
for (i = cvt->len_cvt / 2; i; --i)
{
*dst = *src;
src += 2;
dst += 1;
} /* for */
*format = ((*format & ~0x9010) | AUDIO_U8);
cvt->len_cvt /= 2;
} /* Sound_Convert8 */
/* Convert 8-bit to 16-bit - LSB */
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static void Sound_Convert16LSB(Sound_AudioCVT *cvt, Uint16 *format)
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{
int i;
Uint8 *src, *dst;
/* SNDDBG(("Converting to 16-bit LSB\n")); */
src = cvt->buf + cvt->len_cvt;
dst = cvt->buf + cvt->len_cvt * 2;
for (i = cvt->len_cvt; i; --i)
{
src -= 1;
dst -= 2;
dst[1] = *src;
dst[0] = 0;
} /* for */
*format = ((*format & ~0x0008) | AUDIO_U16LSB);
cvt->len_cvt *= 2;
} /* Sound_Convert16LSB */
/* Convert 8-bit to 16-bit - MSB */
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static void Sound_Convert16MSB(Sound_AudioCVT *cvt, Uint16 *format)
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{
int i;
Uint8 *src, *dst;
/* SNDDBG(("Converting to 16-bit MSB\n")); */
src = cvt->buf + cvt->len_cvt;
dst = cvt->buf + cvt->len_cvt * 2;
for (i = cvt->len_cvt; i; --i)
{
src -= 1;
dst -= 2;
dst[0] = *src;
dst[1] = 0;
} /* for */
*format = ((*format & ~0x0008) | AUDIO_U16MSB);
cvt->len_cvt *= 2;
} /* Sound_Convert16MSB */
/* Duplicate a mono channel to both stereo channels */
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static void Sound_ConvertStereo(Sound_AudioCVT *cvt, Uint16 *format)
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{
int i;
/* SNDDBG(("Converting to stereo\n")); */
/* 16-bit sound? */
if ((*format & 0xFF) == 16)
{
Uint16 *src, *dst;
src = (Uint16 *) (cvt->buf + cvt->len_cvt);
dst = (Uint16 *) (cvt->buf + cvt->len_cvt * 2);
for (i = cvt->len_cvt/2; i; --i)
{
dst -= 2;
src -= 1;
dst[0] = src[0];
dst[1] = src[0];
} /* for */
} /* if */
else
{
Uint8 *src, *dst;
src = cvt->buf + cvt->len_cvt;
dst = cvt->buf + cvt->len_cvt * 2;
for (i = cvt->len_cvt; i; --i)
{
dst -= 2;
src -= 1;
dst[0] = src[0];
dst[1] = src[0];
} /* for */
} /* else */
cvt->len_cvt *= 2;
} /* Sound_ConvertStereo */
/* Effectively mix right and left channels into a single channel */
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static void Sound_ConvertMono(Sound_AudioCVT *cvt, Uint16 *format)
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{
int i;
Sint32 sample;
Uint8 *u_src, *u_dst;
Sint8 *s_src, *s_dst;
/* SNDDBG(("Converting to mono\n")); */
switch (*format)
{
case AUDIO_U8:
u_src = cvt->buf;
u_dst = cvt->buf;
for (i = cvt->len_cvt / 2; i; --i)
{
sample = u_src[0] + u_src[1];
*u_dst = (sample > 255) ? 255 : sample;
u_src += 2;
u_dst += 1;
} /* for */
break;
case AUDIO_S8:
s_src = (Sint8 *) cvt->buf;
s_dst = (Sint8 *) cvt->buf;
for (i = cvt->len_cvt / 2; i; --i)
{
sample = s_src[0] + s_src[1];
if (sample > 127)
*s_dst = 127;
else if (sample < -128)
*s_dst = -128;
else
*s_dst = sample;
s_src += 2;
s_dst += 1;
} /* for */
break;
case AUDIO_U16MSB:
u_src = cvt->buf;
u_dst = cvt->buf;
for (i = cvt->len_cvt / 4; i; --i)
{
sample = (Uint16) ((u_src[0] << 8) | u_src[1])
+ (Uint16) ((u_src[2] << 8) | u_src[3]);
if (sample > 65535)
{
u_dst[0] = 0xFF;
u_dst[1] = 0xFF;
} /* if */
else
{
u_dst[1] = (sample & 0xFF);
sample >>= 8;
u_dst[0] = (sample & 0xFF);
} /* else */
u_src += 4;
u_dst += 2;
} /* for */
break;
case AUDIO_U16LSB:
u_src = cvt->buf;
u_dst = cvt->buf;
for (i = cvt->len_cvt / 4; i; --i)
{
sample = (Uint16) ((u_src[1] << 8) | u_src[0])
+ (Uint16) ((u_src[3] << 8) | u_src[2]);
if (sample > 65535)
{
u_dst[0] = 0xFF;
u_dst[1] = 0xFF;
} /* if */
else
{
u_dst[0] = (sample & 0xFF);
sample >>= 8;
u_dst[1] = (sample & 0xFF);
} /* else */
u_src += 4;
u_dst += 2;
} /* for */
break;
case AUDIO_S16MSB:
u_src = cvt->buf;
u_dst = cvt->buf;
for (i = cvt->len_cvt / 4; i; --i)
{
sample = (Sint16) ((u_src[0] << 8) | u_src[1])
+ (Sint16) ((u_src[2] << 8) | u_src[3]);
if (sample > 32767)
{
u_dst[0] = 0x7F;
u_dst[1] = 0xFF;
} /* if */
else if (sample < -32768)
{
u_dst[0] = 0x80;
u_dst[1] = 0x00;
} /* else if */
else
{
u_dst[1] = (sample & 0xFF);
sample >>= 8;
u_dst[0] = (sample & 0xFF);
} /* else */
u_src += 4;
u_dst += 2;
} /* for */
break;
case AUDIO_S16LSB:
u_src = cvt->buf;
u_dst = cvt->buf;
for (i = cvt->len_cvt / 4; i; --i)
{
sample = (Sint16) ((u_src[1] << 8) | u_src[0])
+ (Sint16) ((u_src[3] << 8) | u_src[2]);
if (sample > 32767)
{
u_dst[1] = 0x7F;
u_dst[0] = 0xFF;
} /* if */
else if (sample < -32768)
{
u_dst[1] = 0x80;
u_dst[0] = 0x00;
} /* else if */
else
{
u_dst[0] = (sample & 0xFF);
sample >>= 8;
u_dst[1] = (sample & 0xFF);
} /* else */
u_src += 4;
u_dst += 2;
} /* for */
break;
} /* switch */
cvt->len_cvt /= 2;
} /* Sound_ConvertMono */
/* Convert rate up by multiple of 2 */
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static void Sound_RateMUL2(Sound_AudioCVT *cvt, Uint16 *format)
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{
int i;
Uint8 *src, *dst;
/* SNDDBG(("Converting audio rate * 2\n")); */
src = cvt->buf + cvt->len_cvt;
dst = cvt->buf + cvt->len_cvt*2;
/* 8- or 16-bit sound? */
switch (*format & 0xFF)
{
case 8:
for (i = cvt->len_cvt; i; --i)
{
src -= 1;
dst -= 2;
dst[0] = src[0];
dst[1] = src[0];
} /* for */
break;
case 16:
for (i = cvt->len_cvt / 2; i; --i)
{
src -= 2;
dst -= 4;
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[0];
dst[3] = src[1];
} /* for */
break;
} /* switch */
cvt->len_cvt *= 2;
} /* Sound_RateMUL2 */
/* Convert rate down by multiple of 2 */
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static void Sound_RateDIV2(Sound_AudioCVT *cvt, Uint16 *format)
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{
int i;
Uint8 *src, *dst;
/* SNDDBG(("Converting audio rate / 2\n")); */
src = cvt->buf;
dst = cvt->buf;
/* 8- or 16-bit sound? */
switch (*format & 0xFF)
{
case 8:
for (i = cvt->len_cvt / 2; i; --i)
{
dst[0] = src[0];
src += 2;
dst += 1;
} /* for */
break;
case 16:
for (i = cvt->len_cvt / 4; i; --i)
{
dst[0] = src[0];
dst[1] = src[1];
src += 4;
dst += 2;
}
break;
} /* switch */
cvt->len_cvt /= 2;
} /* Sound_RateDIV2 */
/* Very slow rate conversion routine */
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static void Sound_RateSLOW(Sound_AudioCVT *cvt, Uint16 *format)
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{
double ipos;
int i, clen;
Uint8 *output8;
Uint16 *output16;
/* SNDDBG(("Converting audio rate * %4.4f\n", 1.0/cvt->rate_incr)); */
clen = (int) ((double) cvt->len_cvt / cvt->rate_incr);
if (cvt->rate_incr > 1.0)
{
/* 8- or 16-bit sound? */
switch (*format & 0xFF)
{
case 8:
output8 = cvt->buf;
ipos = 0.0;
for (i = clen; i; --i)
{
*output8 = cvt->buf[(int) ipos];
ipos += cvt->rate_incr;
output8 += 1;
} /* for */
break;
case 16:
output16 = (Uint16 *) cvt->buf;
clen &= ~1;
ipos = 0.0;
for (i = clen / 2; i; --i)
{
*output16 = ((Uint16 *) cvt->buf)[(int) ipos];
ipos += cvt->rate_incr;
output16 += 1;
} /* for */
break;
} /* switch */
} /* if */
else
{
/* 8- or 16-bit sound */
switch (*format & 0xFF)
{
case 8:
output8 = cvt->buf + clen;
ipos = (double) cvt->len_cvt;
for (i = clen; i; --i)
{
ipos -= cvt->rate_incr;
output8 -= 1;
*output8 = cvt->buf[(int) ipos];
} /* for */
break;
case 16:
clen &= ~1;
output16 = (Uint16 *) (cvt->buf + clen);
ipos = (double) cvt->len_cvt / 2;
for (i = clen / 2; i; --i)
{
ipos -= cvt->rate_incr;
output16 -= 1;
*output16 = ((Uint16 *) cvt->buf)[(int) ipos];
} /* for */
break;
} /* switch */
} /* else */
cvt->len_cvt = clen;
} /* Sound_RateSLOW */
int Sound_ConvertAudio(Sound_AudioCVT *cvt)
{
Uint16 format;
/* Make sure there's data to convert */
if (cvt->buf == NULL)
{
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__Sound_SetError("No buffer allocated for conversion");
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return(-1);
} /* if */
/* Return okay if no conversion is necessary */
cvt->len_cvt = cvt->len;
if (cvt->filters[0] == NULL)
return(0);
/* Set up the conversion and go! */
format = cvt->src_format;
for (cvt->filter_index = 0; cvt->filters[cvt->filter_index];
cvt->filter_index++)
{
cvt->filters[cvt->filter_index](cvt, &format);
}
return(0);
} /* Sound_ConvertAudio */
/*
* Creates a set of audio filters to convert from one format to another.
* Returns -1 if the format conversion is not supported, or 1 if the
* audio filter is set up.
*/
int Sound_BuildAudioCVT(Sound_AudioCVT *cvt,
Uint16 src_format, Uint8 src_channels, Uint32 src_rate,
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Uint16 dst_format, Uint8 dst_channels, Uint32 dst_rate,
Uint32 dst_size)
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{
/* Start off with no conversion necessary */
cvt->needed = 0;
cvt->filter_index = 0;
cvt->filters[0] = NULL;
cvt->len_mult = 1;
cvt->len_ratio = 1.0;
/* First filter: Endian conversion from src to dst */
if ((src_format & 0x1000) != (dst_format & 0x1000) &&
((src_format & 0xff) != 8))
{
SNDDBG(("Adding filter: Sound_ConvertEndian\n"));
cvt->filters[cvt->filter_index++] = Sound_ConvertEndian;
} /* if */
/* Second filter: Sign conversion -- signed/unsigned */
if ((src_format & 0x8000) != (dst_format & 0x8000))
{
SNDDBG(("Adding filter: Sound_ConvertSign\n"));
cvt->filters[cvt->filter_index++] = Sound_ConvertSign;
} /* if */
/* Next filter: Convert 16 bit <--> 8 bit PCM. */
if ((src_format & 0xFF) != (dst_format & 0xFF))
{
switch (dst_format & 0x10FF)
{
case AUDIO_U8:
SNDDBG(("Adding filter: Sound_Convert8\n"));
cvt->filters[cvt->filter_index++] = Sound_Convert8;
cvt->len_ratio /= 2;
break;
case AUDIO_U16LSB:
SNDDBG(("Adding filter: Sound_Convert16LSB\n"));
cvt->filters[cvt->filter_index++] = Sound_Convert16LSB;
cvt->len_mult *= 2;
cvt->len_ratio *= 2;
break;
case AUDIO_U16MSB:
SNDDBG(("Adding filter: Sound_Convert16MSB\n"));
cvt->filters[cvt->filter_index++] = Sound_Convert16MSB;
cvt->len_mult *= 2;
cvt->len_ratio *= 2;
break;
} /* switch */
} /* if */
/* Next filter: Mono/Stereo conversion */
if (src_channels != dst_channels)
{
while ((src_channels * 2) <= dst_channels)
{
SNDDBG(("Adding filter: Sound_ConvertStereo\n"));
cvt->filters[cvt->filter_index++] = Sound_ConvertStereo;
cvt->len_mult *= 2;
src_channels *= 2;
cvt->len_ratio *= 2;
} /* while */
/* This assumes that 4 channel audio is in the format:
* Left {front/back} + Right {front/back}
* so converting to L/R stereo works properly.
*/
while (((src_channels % 2) == 0) &&
((src_channels / 2) >= dst_channels))
{
SNDDBG(("Adding filter: Sound_ConvertMono\n"));
cvt->filters[cvt->filter_index++] = Sound_ConvertMono;
src_channels /= 2;
cvt->len_ratio /= 2;
} /* while */
if ( src_channels != dst_channels ) {
/* Uh oh.. */;
} /* if */
} /* if */
/* Do rate conversion */
cvt->rate_incr = 0.0;
if ((src_rate / 100) != (dst_rate / 100))
{
Uint32 hi_rate, lo_rate;
int len_mult;
double len_ratio;
void (*rate_cvt)(Sound_AudioCVT *cvt, Uint16 *format);
if (src_rate > dst_rate)
{
hi_rate = src_rate;
lo_rate = dst_rate;
SNDDBG(("Adding filter: Sound_RateDIV2\n"));
rate_cvt = Sound_RateDIV2;
len_mult = 1;
len_ratio = 0.5;
} /* if */
else
{
hi_rate = dst_rate;
lo_rate = src_rate;
SNDDBG(("Adding filter: Sound_RateMUL2\n"));
rate_cvt = Sound_RateMUL2;
len_mult = 2;
len_ratio = 2.0;
} /* else */
/* If hi_rate = lo_rate*2^x then conversion is easy */
while (((lo_rate * 2) / 100) <= (hi_rate / 100))
{
cvt->filters[cvt->filter_index++] = rate_cvt;
cvt->len_mult *= len_mult;
lo_rate *= 2;
cvt->len_ratio *= len_ratio;
} /* while */
/* We may need a slow conversion here to finish up */
if ((lo_rate / 100) != (hi_rate / 100))
{
if (src_rate < dst_rate)
{
cvt->rate_incr = (double) lo_rate / hi_rate;
cvt->len_mult *= 2;
cvt->len_ratio /= cvt->rate_incr;
} /* if */
else
{
cvt->rate_incr = (double) hi_rate / lo_rate;
cvt->len_ratio *= cvt->rate_incr;
} /* else */
SNDDBG(("Adding filter: Sound_RateSLOW\n"));
cvt->filters[cvt->filter_index++] = Sound_RateSLOW;
} /* if */
} /* if */
/* Set up the filter information */
if (cvt->filter_index != 0)
{
cvt->needed = 1;
cvt->src_format = src_format;
cvt->dst_format = dst_format;
cvt->len = 0;
cvt->buf = NULL;
cvt->filters[cvt->filter_index] = NULL;
} /* if */
return(cvt->needed);
} /* Sound_BuildAudioCVT */