/*

SDL - Simple DirectMedia Layer

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

*/

#ifdef SAVE_RCSID

static char rcsid =

"@(#) $Id$";

#endif

/* Functions for audio drivers to perform runtime conversion of audio format */

#include <stdio.h>

#include "SDL_error.h"

#include "SDL_audio.h"

/* Effectively mix right and left channels into a single channel */

void SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format)

{

int i;

Sint32 sample;

#ifdef DEBUG_CONVERT

fprintf(stderr, "Converting to mono\n");

#endif

switch (format&0x8018) {

case AUDIO_U8: {

Uint8 *src, *dst;

src = cvt->buf;

dst = cvt->buf;

for ( i=cvt->len_cvt/2; i; --i ) {

sample = src[0] + src[1];

if ( sample > 255 ) {

*dst = 255;

} else {

*dst = sample;

}

src += 2;

dst += 1;

}

}

break;

case AUDIO_S8: {

Sint8 *src, *dst;

src = (Sint8 *)cvt->buf;

dst = (Sint8 *)cvt->buf;

for ( i=cvt->len_cvt/2; i; --i ) {

sample = src[0] + src[1];

if ( sample > 127 ) {

*dst = 127;

} else

if ( sample < -128 ) {

*dst = -128;

} else {

*dst = sample;

}

src += 2;

dst += 1;

}

}

break;

case AUDIO_U16: {

Uint8 *src, *dst;

src = cvt->buf;

dst = cvt->buf;

if ( (format & 0x1000) == 0x1000 ) {

for ( i=cvt->len_cvt/4; i; --i ) {

sample = (Uint16)((src[0]<<8)|src[1])+

(Uint16)((src[2]<<8)|src[3]);

if ( sample > 65535 ) {

dst[0] = 0xFF;

dst[1] = 0xFF;

} else {

dst[1] = (sample&0xFF);

sample >>= 8;

dst[0] = (sample&0xFF);

}

src += 4;

dst += 2;

}

} else {

for ( i=cvt->len_cvt/4; i; --i ) {

sample = (Uint16)((src[1]<<8)|src[0])+

(Uint16)((src[3]<<8)|src[2]);

if ( sample > 65535 ) {

dst[0] = 0xFF;

dst[1] = 0xFF;

} else {

dst[0] = (sample&0xFF);

sample >>= 8;

dst[1] = (sample&0xFF);

}

src += 4;

dst += 2;

}

}

}

break;

case AUDIO_S16: {

Uint8 *src, *dst;

src = cvt->buf;

dst = cvt->buf;

if ( (format & 0x1000) == 0x1000 ) {

for ( i=cvt->len_cvt/4; i; --i ) {

sample = (Sint16)((src[0]<<8)|src[1])+

(Sint16)((src[2]<<8)|src[3]);

if ( sample > 32767 ) {

dst[0] = 0x7F;

dst[1] = 0xFF;

} else

if ( sample < -32768 ) {

dst[0] = 0x80;

dst[1] = 0x00;

} else {

dst[1] = (sample&0xFF);

sample >>= 8;

dst[0] = (sample&0xFF);

}

src += 4;

dst += 2;

}

} else {

for ( i=cvt->len_cvt/4; i; --i ) {

sample = (Sint16)((src[1]<<8)|src[0])+

(Sint16)((src[3]<<8)|src[2]);

if ( sample > 32767 ) {

dst[1] = 0x7F;

dst[0] = 0xFF;

} else

if ( sample < -32768 ) {

dst[1] = 0x80;

dst[0] = 0x00;

} else {

dst[0] = (sample&0xFF);

sample >>= 8;

dst[1] = (sample&0xFF);

}

src += 4;

dst += 2;

}

}

}

break;

}

cvt->len_cvt /= 2;

if ( cvt->filters[++cvt->filter_index] ) {

cvt->filters[cvt->filter_index](cvt, format);

}

}

/* Duplicate a mono channel to both stereo channels */

void SDL_ConvertStereo(SDL_AudioCVT *cvt, Uint16 format)

{

int i;

#ifdef DEBUG_CONVERT

fprintf(stderr, "Converting to stereo\n");

#endif

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];

}

} 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];

}

}

cvt->len_cvt *= 2;

if ( cvt->filters[++cvt->filter_index] ) {

cvt->filters[cvt->filter_index](cvt, format);

}

}

/* Convert 8-bit to 16-bit - LSB */

void SDL_Convert16LSB(SDL_AudioCVT *cvt, Uint16 format)

{

int i;

Uint8 *src, *dst;

#ifdef DEBUG_CONVERT

fprintf(stderr, "Converting to 16-bit LSB\n");

#endif

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;

}

format = ((format & ~0x0008) | AUDIO_U16LSB);

cvt->len_cvt *= 2;

if ( cvt->filters[++cvt->filter_index] ) {

cvt->filters[cvt->filter_index](cvt, format);

}

}

/* Convert 8-bit to 16-bit - MSB */

void SDL_Convert16MSB(SDL_AudioCVT *cvt, Uint16 format)

{

int i;

Uint8 *src, *dst;

#ifdef DEBUG_CONVERT

fprintf(stderr, "Converting to 16-bit MSB\n");

#endif

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;

}

format = ((format & ~0x0008) | AUDIO_U16MSB);

cvt->len_cvt *= 2;

if ( cvt->filters[++cvt->filter_index] ) {

cvt->filters[cvt->filter_index](cvt, format);

}

}

/* Convert 16-bit to 8-bit */

void SDL_Convert8(SDL_AudioCVT *cvt, Uint16 format)

{

int i;

Uint8 *src, *dst;

#ifdef DEBUG_CONVERT

fprintf(stderr, "Converting to 8-bit\n");

#endif

src = cvt->buf;

dst = cvt->buf;

if ( (format & 0x1000) != 0x1000 ) { /* Little endian */

++src;

}

for ( i=cvt->len_cvt/2; i; --i ) {

*dst = *src;

src += 2;

dst += 1;

}

format = ((format & ~0x9010) | AUDIO_U8);

cvt->len_cvt /= 2;

if ( cvt->filters[++cvt->filter_index] ) {

cvt->filters[cvt->filter_index](cvt, format);

}

}

/* Toggle signed/unsigned */

void SDL_ConvertSign(SDL_AudioCVT *cvt, Uint16 format)

{

int i;

Uint8 *data;

#ifdef DEBUG_CONVERT

fprintf(stderr, "Converting audio signedness\n");

#endif

data = cvt->buf;

if ( (format & 0xFF) == 16 ) {

if ( (format & 0x1000) != 0x1000 ) { /* Little endian */

++data;

}

for ( i=cvt->len_cvt/2; i; --i ) {

*data ^= 0x80;

data += 2;

}

} else {

for ( i=cvt->len_cvt; i; --i ) {

*data++ ^= 0x80;

}

}

format = (format ^ 0x8000);

if ( cvt->filters[++cvt->filter_index] ) {

cvt->filters[cvt->filter_index](cvt, format);

}

}

/* Toggle endianness */

void SDL_ConvertEndian(SDL_AudioCVT *cvt, Uint16 format)

{

int i;

Uint8 *data, tmp;

#ifdef DEBUG_CONVERT

fprintf(stderr, "Converting audio endianness\n");

#endif

data = cvt->buf;

for ( i=cvt->len_cvt/2; i; --i ) {

tmp = data[0];

data[0] = data[1];

data[1] = tmp;

data += 2;

}

format = (format ^ 0x1000);

if ( cvt->filters[++cvt->filter_index] ) {

cvt->filters[cvt->filter_index](cvt, format);

}

}

/* Convert rate up by multiple of 2 */

void SDL_RateMUL2(SDL_AudioCVT *cvt, Uint16 format)

{

int i;

Uint8 *src, *dst;

#ifdef DEBUG_CONVERT

fprintf(stderr, "Converting audio rate * 2\n");

#endif

src = cvt->buf+cvt->len_cvt;

dst = cvt->buf+cvt->len_cvt*2;

switch (format & 0xFF) {

case 8:

for ( i=cvt->len_cvt; i; --i ) {

src -= 1;

dst -= 2;

dst[0] = src[0];

dst[1] = src[0];

}

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];

}

break;

}

cvt->len_cvt *= 2;

if ( cvt->filters[++cvt->filter_index] ) {

cvt->filters[cvt->filter_index](cvt, format);

}

}

/* Convert rate down by multiple of 2 */

void SDL_RateDIV2(SDL_AudioCVT *cvt, Uint16 format)

{

int i;

Uint8 *src, *dst;

#ifdef DEBUG_CONVERT

fprintf(stderr, "Converting audio rate / 2\n");

#endif

src = cvt->buf;

dst = cvt->buf;

switch (format & 0xFF) {

case 8:

for ( i=cvt->len_cvt/2; i; --i ) {

dst[0] = src[0];

src += 2;

dst += 1;

}

break;

case 16:

for ( i=cvt->len_cvt/4; i; --i ) {

dst[0] = src[0];

dst[1] = src[1];

src += 4;

dst += 2;

}

break;

}

cvt->len_cvt /= 2;

if ( cvt->filters[++cvt->filter_index] ) {

cvt->filters[cvt->filter_index](cvt, format);

}

}

/* Very slow rate conversion routine */

void SDL_RateSLOW(SDL_AudioCVT *cvt, Uint16 format)

{

double ipos;

int i, clen;

#ifdef DEBUG_CONVERT

fprintf(stderr, "Converting audio rate * %4.4f\n", 1.0/cvt->rate_incr);

#endif

clen = (int)((double)cvt->len_cvt / cvt->rate_incr);

if ( cvt->rate_incr > 1.0 ) {

switch (format & 0xFF) {

case 8: {

Uint8 *output;

output = cvt->buf;

ipos = 0.0;

for ( i=clen; i; --i ) {

*output = cvt->buf[(int)ipos];

ipos += cvt->rate_incr;

output += 1;

}

}

break;

case 16: {

Uint16 *output;

clen &= ~1;

output = (Uint16 *)cvt->buf;

ipos = 0.0;

for ( i=clen/2; i; --i ) {

*output=((Uint16 *)cvt->buf)[(int)ipos];

ipos += cvt->rate_incr;

output += 1;

}

}

break;

}

} else {

switch (format & 0xFF) {

case 8: {

Uint8 *output;

output = cvt->buf+clen;

ipos = (double)cvt->len_cvt;

for ( i=clen; i; --i ) {

ipos -= cvt->rate_incr;

output -= 1;

*output = cvt->buf[(int)ipos];

}

}

break;

case 16: {

Uint16 *output;

clen &= ~1;

output = (Uint16 *)(cvt->buf+clen);

ipos = (double)cvt->len_cvt/2;

for ( i=clen/2; i; --i ) {

ipos -= cvt->rate_incr;

output -= 1;

*output=((Uint16 *)cvt->buf)[(int)ipos];

}

}

break;

}

}

cvt->len_cvt = clen;

if ( cvt->filters[++cvt->filter_index] ) {

cvt->filters[cvt->filter_index](cvt, format);

}

}

int SDL_ConvertAudio(SDL_AudioCVT *cvt)

{

/* Make sure there's data to convert */

if ( cvt->buf == NULL ) {

SDL_SetError("No buffer allocated for conversion");

return(-1);

}

/* 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! */

cvt->filter_index = 0;

cvt->filters[0](cvt, cvt->src_format);

return(0);

}

/* 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 SDL_BuildAudioCVT(SDL_AudioCVT *cvt,

Uint16 src_format, Uint8 src_channels, int src_rate,

Uint16 dst_format, Uint8 dst_channels, int dst_rate)

{

/* 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) ) {

cvt->filters[cvt->filter_index++] = SDL_ConvertEndian;

}

/* Second filter: Sign conversion -- signed/unsigned */

if ( (src_format & 0x8000) != (dst_format & 0x8000) ) {

cvt->filters[cvt->filter_index++] = SDL_ConvertSign;

}

/* Next filter: Convert 16 bit <--> 8 bit PCM */

if ( (src_format & 0xFF) != (dst_format & 0xFF) ) {

switch (dst_format&0x10FF) {

case AUDIO_U8:

cvt->filters[cvt->filter_index++] =

SDL_Convert8;

cvt->len_ratio /= 2;

break;

case AUDIO_U16LSB:

cvt->filters[cvt->filter_index++] =

SDL_Convert16LSB;

cvt->len_mult *= 2;

cvt->len_ratio *= 2;

break;

case AUDIO_U16MSB:

cvt->filters[cvt->filter_index++] =

SDL_Convert16MSB;

cvt->len_mult *= 2;

cvt->len_ratio *= 2;

break;

}

}

/* Last filter: Mono/Stereo conversion */

if ( src_channels != dst_channels ) {

while ( (src_channels*2) <= dst_channels ) {

cvt->filters[cvt->filter_index++] =

SDL_ConvertStereo;

cvt->len_mult *= 2;

src_channels *= 2;

cvt->len_ratio *= 2;

}

/* 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) ) {

cvt->filters[cvt->filter_index++] =

SDL_ConvertMono;

src_channels /= 2;

cvt->len_ratio /= 2;

}

if ( src_channels != dst_channels ) {

/* Uh oh.. */;

}

}

/* 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)(SDL_AudioCVT *cvt, Uint16 format);

if ( src_rate > dst_rate ) {

hi_rate = src_rate;

lo_rate = dst_rate;

rate_cvt = SDL_RateDIV2;

len_mult = 1;

len_ratio = 0.5;

} else {

hi_rate = dst_rate;

lo_rate = src_rate;

rate_cvt = SDL_RateMUL2;

len_mult = 2;

len_ratio = 2.0;

}

/* 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;

}

/* We may need a slow conversion here to finish up */

if ( (lo_rate/100) != (hi_rate/100) ) {

#if 1

/* The problem with this is that if the input buffer is

say 1K, and the conversion rate is say 1.1, then the

output buffer is 1.1K, which may not be an acceptable

buffer size for the audio driver (not a power of 2)

*/

/* For now, punt and hope the rate distortion isn't great.

*/

#else

if ( src_rate < dst_rate ) {

cvt->rate_incr = (double)lo_rate/hi_rate;

cvt->len_mult *= 2;

cvt->len_ratio /= cvt->rate_incr;

} else {

cvt->rate_incr = (double)hi_rate/lo_rate;

cvt->len_ratio *= cvt->rate_incr;

}

cvt->filters[cvt->filter_index++] = SDL_RateSLOW;

#endif

}

}

/* 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;

}

return(cvt->needed);

}