127b85086b
- Added sound! - Added Linux makefile - Added _KOLIBRI definition - Removed not working parameters from --help in KolibriOS git-svn-id: svn://kolibrios.org@8645 a494cfbc-eb01-0410-851d-a64ba20cac60
643 lines
14 KiB
C
643 lines
14 KiB
C
/*
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SDL - Simple DirectMedia Layer
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Copyright (C) 1997, 1998, 1999, 2000, 2001 Sam Lantinga
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This library is free software; you can redistribute it and/or
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modify it under the terms of the GNU Library General Public
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License as published by the Free Software Foundation; either
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version 2 of the License, or (at your option) any later version.
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This library is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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Library General Public License for more details.
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You should have received a copy of the GNU Library General Public
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License along with this library; if not, write to the Free
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Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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Sam Lantinga
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slouken@devolution.com
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*/
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#ifdef SAVE_RCSID
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static char rcsid =
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"@(#) $Id: SDL_audiocvt.c,v 1.2 2001/04/26 16:50:17 hercules Exp $";
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#endif
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/* Functions for audio drivers to perform runtime conversion of audio format */
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#include <stdio.h>
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#include "SDL_error.h"
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#include "SDL_audio.h"
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/* Effectively mix right and left channels into a single channel */
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void SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format)
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{
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int i;
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Sint32 sample;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting to mono\n");
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#endif
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switch (format&0x8018) {
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case AUDIO_U8: {
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Uint8 *src, *dst;
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src = cvt->buf;
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dst = cvt->buf;
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for ( i=cvt->len_cvt/2; i; --i ) {
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sample = src[0] + src[1];
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if ( sample > 255 ) {
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*dst = 255;
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} else {
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*dst = sample;
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}
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src += 2;
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dst += 1;
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}
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}
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break;
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case AUDIO_S8: {
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Sint8 *src, *dst;
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src = (Sint8 *)cvt->buf;
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dst = (Sint8 *)cvt->buf;
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for ( i=cvt->len_cvt/2; i; --i ) {
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sample = src[0] + src[1];
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if ( sample > 127 ) {
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*dst = 127;
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} else
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if ( sample < -128 ) {
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*dst = -128;
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} else {
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*dst = sample;
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}
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src += 2;
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dst += 1;
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}
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}
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break;
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case AUDIO_U16: {
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Uint8 *src, *dst;
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src = cvt->buf;
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dst = cvt->buf;
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if ( (format & 0x1000) == 0x1000 ) {
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for ( i=cvt->len_cvt/4; i; --i ) {
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sample = (Uint16)((src[0]<<8)|src[1])+
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(Uint16)((src[2]<<8)|src[3]);
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if ( sample > 65535 ) {
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dst[0] = 0xFF;
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dst[1] = 0xFF;
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} else {
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dst[1] = (sample&0xFF);
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sample >>= 8;
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dst[0] = (sample&0xFF);
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}
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src += 4;
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dst += 2;
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}
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} else {
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for ( i=cvt->len_cvt/4; i; --i ) {
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sample = (Uint16)((src[1]<<8)|src[0])+
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(Uint16)((src[3]<<8)|src[2]);
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if ( sample > 65535 ) {
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dst[0] = 0xFF;
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dst[1] = 0xFF;
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} else {
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dst[0] = (sample&0xFF);
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sample >>= 8;
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dst[1] = (sample&0xFF);
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}
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src += 4;
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dst += 2;
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}
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}
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}
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break;
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case AUDIO_S16: {
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Uint8 *src, *dst;
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src = cvt->buf;
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dst = cvt->buf;
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if ( (format & 0x1000) == 0x1000 ) {
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for ( i=cvt->len_cvt/4; i; --i ) {
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sample = (Sint16)((src[0]<<8)|src[1])+
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(Sint16)((src[2]<<8)|src[3]);
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if ( sample > 32767 ) {
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dst[0] = 0x7F;
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dst[1] = 0xFF;
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} else
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if ( sample < -32768 ) {
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dst[0] = 0x80;
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dst[1] = 0x00;
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} else {
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dst[1] = (sample&0xFF);
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sample >>= 8;
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dst[0] = (sample&0xFF);
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}
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src += 4;
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dst += 2;
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}
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} else {
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for ( i=cvt->len_cvt/4; i; --i ) {
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sample = (Sint16)((src[1]<<8)|src[0])+
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(Sint16)((src[3]<<8)|src[2]);
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if ( sample > 32767 ) {
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dst[1] = 0x7F;
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dst[0] = 0xFF;
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} else
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if ( sample < -32768 ) {
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dst[1] = 0x80;
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dst[0] = 0x00;
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} else {
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dst[0] = (sample&0xFF);
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sample >>= 8;
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dst[1] = (sample&0xFF);
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}
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src += 4;
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dst += 2;
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}
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}
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}
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break;
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}
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cvt->len_cvt /= 2;
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if ( cvt->filters[++cvt->filter_index] ) {
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cvt->filters[cvt->filter_index](cvt, format);
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}
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}
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/* Duplicate a mono channel to both stereo channels */
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void SDL_ConvertStereo(SDL_AudioCVT *cvt, Uint16 format)
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{
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int i;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting to stereo\n");
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#endif
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if ( (format & 0xFF) == 16 ) {
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Uint16 *src, *dst;
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src = (Uint16 *)(cvt->buf+cvt->len_cvt);
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dst = (Uint16 *)(cvt->buf+cvt->len_cvt*2);
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for ( i=cvt->len_cvt/2; i; --i ) {
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dst -= 2;
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src -= 1;
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dst[0] = src[0];
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dst[1] = src[0];
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}
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} else {
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Uint8 *src, *dst;
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src = cvt->buf+cvt->len_cvt;
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dst = cvt->buf+cvt->len_cvt*2;
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for ( i=cvt->len_cvt; i; --i ) {
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dst -= 2;
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src -= 1;
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dst[0] = src[0];
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dst[1] = src[0];
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}
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}
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cvt->len_cvt *= 2;
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if ( cvt->filters[++cvt->filter_index] ) {
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cvt->filters[cvt->filter_index](cvt, format);
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}
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}
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/* Convert 8-bit to 16-bit - LSB */
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void SDL_Convert16LSB(SDL_AudioCVT *cvt, Uint16 format)
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{
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int i;
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Uint8 *src, *dst;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting to 16-bit LSB\n");
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#endif
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src = cvt->buf+cvt->len_cvt;
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dst = cvt->buf+cvt->len_cvt*2;
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for ( i=cvt->len_cvt; i; --i ) {
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src -= 1;
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dst -= 2;
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dst[1] = *src;
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dst[0] = 0;
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}
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format = ((format & ~0x0008) | AUDIO_U16LSB);
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cvt->len_cvt *= 2;
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if ( cvt->filters[++cvt->filter_index] ) {
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cvt->filters[cvt->filter_index](cvt, format);
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}
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}
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/* Convert 8-bit to 16-bit - MSB */
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void SDL_Convert16MSB(SDL_AudioCVT *cvt, Uint16 format)
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{
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int i;
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Uint8 *src, *dst;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting to 16-bit MSB\n");
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#endif
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src = cvt->buf+cvt->len_cvt;
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dst = cvt->buf+cvt->len_cvt*2;
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for ( i=cvt->len_cvt; i; --i ) {
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src -= 1;
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dst -= 2;
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dst[0] = *src;
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dst[1] = 0;
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}
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format = ((format & ~0x0008) | AUDIO_U16MSB);
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cvt->len_cvt *= 2;
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if ( cvt->filters[++cvt->filter_index] ) {
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cvt->filters[cvt->filter_index](cvt, format);
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}
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}
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/* Convert 16-bit to 8-bit */
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void SDL_Convert8(SDL_AudioCVT *cvt, Uint16 format)
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{
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int i;
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Uint8 *src, *dst;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting to 8-bit\n");
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#endif
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src = cvt->buf;
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dst = cvt->buf;
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if ( (format & 0x1000) != 0x1000 ) { /* Little endian */
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++src;
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}
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for ( i=cvt->len_cvt/2; i; --i ) {
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*dst = *src;
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src += 2;
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dst += 1;
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}
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format = ((format & ~0x9010) | AUDIO_U8);
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cvt->len_cvt /= 2;
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if ( cvt->filters[++cvt->filter_index] ) {
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cvt->filters[cvt->filter_index](cvt, format);
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}
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}
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/* Toggle signed/unsigned */
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void SDL_ConvertSign(SDL_AudioCVT *cvt, Uint16 format)
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{
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int i;
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Uint8 *data;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting audio signedness\n");
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#endif
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data = cvt->buf;
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if ( (format & 0xFF) == 16 ) {
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if ( (format & 0x1000) != 0x1000 ) { /* Little endian */
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++data;
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}
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for ( i=cvt->len_cvt/2; i; --i ) {
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*data ^= 0x80;
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data += 2;
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}
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} else {
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for ( i=cvt->len_cvt; i; --i ) {
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*data++ ^= 0x80;
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}
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}
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format = (format ^ 0x8000);
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if ( cvt->filters[++cvt->filter_index] ) {
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cvt->filters[cvt->filter_index](cvt, format);
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}
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}
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/* Toggle endianness */
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void SDL_ConvertEndian(SDL_AudioCVT *cvt, Uint16 format)
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{
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int i;
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Uint8 *data, tmp;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting audio endianness\n");
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#endif
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data = cvt->buf;
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for ( i=cvt->len_cvt/2; i; --i ) {
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tmp = data[0];
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data[0] = data[1];
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data[1] = tmp;
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data += 2;
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}
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format = (format ^ 0x1000);
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if ( cvt->filters[++cvt->filter_index] ) {
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cvt->filters[cvt->filter_index](cvt, format);
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}
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}
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/* Convert rate up by multiple of 2 */
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void SDL_RateMUL2(SDL_AudioCVT *cvt, Uint16 format)
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{
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int i;
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Uint8 *src, *dst;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting audio rate * 2\n");
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#endif
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src = cvt->buf+cvt->len_cvt;
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dst = cvt->buf+cvt->len_cvt*2;
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switch (format & 0xFF) {
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case 8:
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for ( i=cvt->len_cvt; i; --i ) {
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src -= 1;
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dst -= 2;
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dst[0] = src[0];
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dst[1] = src[0];
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}
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break;
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case 16:
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for ( i=cvt->len_cvt/2; i; --i ) {
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src -= 2;
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dst -= 4;
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dst[0] = src[0];
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dst[1] = src[1];
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dst[2] = src[0];
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dst[3] = src[1];
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}
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break;
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}
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cvt->len_cvt *= 2;
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if ( cvt->filters[++cvt->filter_index] ) {
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cvt->filters[cvt->filter_index](cvt, format);
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}
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}
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/* Convert rate down by multiple of 2 */
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void SDL_RateDIV2(SDL_AudioCVT *cvt, Uint16 format)
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{
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int i;
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Uint8 *src, *dst;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting audio rate / 2\n");
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#endif
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src = cvt->buf;
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dst = cvt->buf;
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switch (format & 0xFF) {
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case 8:
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for ( i=cvt->len_cvt/2; i; --i ) {
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dst[0] = src[0];
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src += 2;
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dst += 1;
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}
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break;
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case 16:
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for ( i=cvt->len_cvt/4; i; --i ) {
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dst[0] = src[0];
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dst[1] = src[1];
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src += 4;
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dst += 2;
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}
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break;
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}
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cvt->len_cvt /= 2;
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if ( cvt->filters[++cvt->filter_index] ) {
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cvt->filters[cvt->filter_index](cvt, format);
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}
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}
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/* Very slow rate conversion routine */
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void SDL_RateSLOW(SDL_AudioCVT *cvt, Uint16 format)
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{
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double ipos;
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int i, clen;
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#ifdef DEBUG_CONVERT
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fprintf(stderr, "Converting audio rate * %4.4f\n", 1.0/cvt->rate_incr);
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#endif
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clen = (int)((double)cvt->len_cvt / cvt->rate_incr);
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if ( cvt->rate_incr > 1.0 ) {
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switch (format & 0xFF) {
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case 8: {
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Uint8 *output;
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output = cvt->buf;
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ipos = 0.0;
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for ( i=clen; i; --i ) {
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*output = cvt->buf[(int)ipos];
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ipos += cvt->rate_incr;
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output += 1;
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}
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}
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break;
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case 16: {
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Uint16 *output;
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clen &= ~1;
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output = (Uint16 *)cvt->buf;
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ipos = 0.0;
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for ( i=clen/2; i; --i ) {
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*output=((Uint16 *)cvt->buf)[(int)ipos];
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ipos += cvt->rate_incr;
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output += 1;
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}
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}
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break;
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}
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} else {
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switch (format & 0xFF) {
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case 8: {
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Uint8 *output;
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output = cvt->buf+clen;
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ipos = (double)cvt->len_cvt;
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for ( i=clen; i; --i ) {
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ipos -= cvt->rate_incr;
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output -= 1;
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*output = cvt->buf[(int)ipos];
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}
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}
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break;
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case 16: {
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Uint16 *output;
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clen &= ~1;
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output = (Uint16 *)(cvt->buf+clen);
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ipos = (double)cvt->len_cvt/2;
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for ( i=clen/2; i; --i ) {
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ipos -= cvt->rate_incr;
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output -= 1;
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*output=((Uint16 *)cvt->buf)[(int)ipos];
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}
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}
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break;
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}
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}
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cvt->len_cvt = clen;
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if ( cvt->filters[++cvt->filter_index] ) {
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cvt->filters[cvt->filter_index](cvt, format);
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}
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}
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int SDL_ConvertAudio(SDL_AudioCVT *cvt)
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{
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/* Make sure there's data to convert */
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if ( cvt->buf == NULL ) {
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SDL_SetError("No buffer allocated for conversion");
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return(-1);
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}
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/* Return okay if no conversion is necessary */
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cvt->len_cvt = cvt->len;
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if ( cvt->filters[0] == NULL ) {
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return(0);
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}
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/* Set up the conversion and go! */
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cvt->filter_index = 0;
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cvt->filters[0](cvt, cvt->src_format);
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return(0);
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}
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/* Creates a set of audio filters to convert from one format to another.
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Returns -1 if the format conversion is not supported, or 1 if the
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audio filter is set up.
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*/
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int SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
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Uint16 src_format, Uint8 src_channels, int src_rate,
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Uint16 dst_format, Uint8 dst_channels, int dst_rate)
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{
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/* Start off with no conversion necessary */
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cvt->needed = 0;
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cvt->filter_index = 0;
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cvt->filters[0] = NULL;
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cvt->len_mult = 1;
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cvt->len_ratio = 1.0;
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/* First filter: Endian conversion from src to dst */
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if ( (src_format & 0x1000) != (dst_format & 0x1000)
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&& ((src_format & 0xff) != 8) ) {
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cvt->filters[cvt->filter_index++] = SDL_ConvertEndian;
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}
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/* Second filter: Sign conversion -- signed/unsigned */
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if ( (src_format & 0x8000) != (dst_format & 0x8000) ) {
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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);
|
|
}
|