forked from KolibriOS/kolibrios
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
251 lines
6.3 KiB
C
251 lines
6.3 KiB
C
/*
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SDL_mixer: An audio mixer library based on the SDL library
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Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
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This software is provided 'as-is', without any express or implied
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warranty. In no event will the authors be held liable for any damages
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arising from the use of this software.
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Permission is granted to anyone to use this software for any purpose,
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including commercial applications, and to alter it and redistribute it
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freely, subject to the following restrictions:
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1. The origin of this software must not be misrepresented; you must not
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claim that you wrote the original software. If you use this software
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in a product, an acknowledgment in the product documentation would be
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appreciated but is not required.
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2. Altered source versions must be plainly marked as such, and must not be
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misrepresented as being the original software.
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3. This notice may not be removed or altered from any source distribution.
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This is the source needed to decode an AIFF file into a waveform.
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It's pretty straightforward once you get going. The only
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externally-callable function is Mix_LoadAIFF_RW(), which is meant to
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act as identically to SDL_LoadWAV_RW() as possible.
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This file by Torbjörn Andersson (torbjorn.andersson@eurotime.se)
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8SVX file support added by Marc Le Douarain (mavati@club-internet.fr)
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in december 2002.
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*/
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/* $Id$ */
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#include <stdlib.h>
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#include <string.h>
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#include "SDL_endian.h"
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#include "SDL_mixer.h"
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#include "load_aiff.h"
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/*********************************************/
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/* Define values for AIFF (IFF audio) format */
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/*********************************************/
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#define FORM 0x4d524f46 /* "FORM" */
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#define AIFF 0x46464941 /* "AIFF" */
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#define SSND 0x444e5353 /* "SSND" */
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#define COMM 0x4d4d4f43 /* "COMM" */
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#define _8SVX 0x58565338 /* "8SVX" */
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#define VHDR 0x52444856 /* "VHDR" */
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#define BODY 0x59444F42 /* "BODY" */
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/* This function was taken from libsndfile. I don't pretend to fully
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* understand it.
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*/
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static Uint32 SANE_to_Uint32 (Uint8 *sanebuf)
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{
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/* Is the frequency outside of what we can represent with Uint32? */
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if ( (sanebuf[0] & 0x80) || (sanebuf[0] <= 0x3F) || (sanebuf[0] > 0x40)
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|| (sanebuf[0] == 0x40 && sanebuf[1] > 0x1C) )
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return 0;
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return ((sanebuf[2] << 23) | (sanebuf[3] << 15) | (sanebuf[4] << 7)
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| (sanebuf[5] >> 1)) >> (29 - sanebuf[1]);
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}
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/* This function is based on SDL_LoadWAV_RW(). */
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SDL_AudioSpec *Mix_LoadAIFF_RW (SDL_RWops *src, int freesrc,
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SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
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{
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int was_error;
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int found_SSND;
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int found_COMM;
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int found_VHDR;
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int found_BODY;
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long start = 0;
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Uint32 chunk_type;
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Uint32 chunk_length;
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long next_chunk;
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/* AIFF magic header */
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Uint32 FORMchunk;
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Uint32 AIFFmagic;
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/* SSND chunk */
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Uint32 offset;
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Uint32 blocksize;
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/* COMM format chunk */
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Uint16 channels = 0;
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Uint32 numsamples = 0;
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Uint16 samplesize = 0;
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Uint8 sane_freq[10];
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Uint32 frequency = 0;
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/* Make sure we are passed a valid data source */
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was_error = 0;
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if ( src == NULL ) {
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was_error = 1;
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goto done;
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}
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FORMchunk = SDL_ReadLE32(src);
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chunk_length = SDL_ReadBE32(src);
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if ( chunk_length == AIFF ) { /* The FORMchunk has already been read */
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AIFFmagic = chunk_length;
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chunk_length = FORMchunk;
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FORMchunk = FORM;
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} else {
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AIFFmagic = SDL_ReadLE32(src);
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}
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if ( (FORMchunk != FORM) || ( (AIFFmagic != AIFF) && (AIFFmagic != _8SVX) ) ) {
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SDL_SetError("Unrecognized file type (not AIFF nor 8SVX)");
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was_error = 1;
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goto done;
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}
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/* TODO: Better santity-checking. */
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found_SSND = 0;
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found_COMM = 0;
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found_VHDR = 0;
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found_BODY = 0;
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do {
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chunk_type = SDL_ReadLE32(src);
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chunk_length = SDL_ReadBE32(src);
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next_chunk = SDL_RWtell(src) + chunk_length;
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/* Paranoia to avoid infinite loops */
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if (chunk_length == 0)
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break;
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switch (chunk_type) {
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case SSND:
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found_SSND = 1;
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offset = SDL_ReadBE32(src);
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blocksize = SDL_ReadBE32(src);
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start = SDL_RWtell(src) + offset;
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break;
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case COMM:
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found_COMM = 1;
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channels = SDL_ReadBE16(src);
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numsamples = SDL_ReadBE32(src);
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samplesize = SDL_ReadBE16(src);
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SDL_RWread(src, sane_freq, sizeof(sane_freq), 1);
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frequency = SANE_to_Uint32(sane_freq);
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if (frequency == 0) {
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SDL_SetError("Bad AIFF sample frequency");
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was_error = 1;
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goto done;
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}
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break;
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case VHDR:
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found_VHDR = 1;
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SDL_ReadBE32(src);
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SDL_ReadBE32(src);
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SDL_ReadBE32(src);
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frequency = SDL_ReadBE16(src);
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channels = 1;
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samplesize = 8;
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break;
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case BODY:
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found_BODY = 1;
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numsamples = chunk_length;
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start = SDL_RWtell(src);
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break;
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default:
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break;
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}
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/* a 0 pad byte can be stored for any odd-length chunk */
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if (chunk_length&1)
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next_chunk++;
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} while ( ( ( (AIFFmagic == AIFF) && ( !found_SSND || !found_COMM ) )
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|| ( (AIFFmagic == _8SVX ) && ( !found_VHDR || !found_BODY ) ) )
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&& SDL_RWseek(src, next_chunk, RW_SEEK_SET) != 1 );
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if ( (AIFFmagic == AIFF) && !found_SSND ) {
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SDL_SetError("Bad AIFF (no SSND chunk)");
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was_error = 1;
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goto done;
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}
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if ( (AIFFmagic == AIFF) && !found_COMM ) {
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SDL_SetError("Bad AIFF (no COMM chunk)");
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was_error = 1;
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goto done;
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}
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if ( (AIFFmagic == _8SVX) && !found_VHDR ) {
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SDL_SetError("Bad 8SVX (no VHDR chunk)");
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was_error = 1;
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goto done;
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}
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if ( (AIFFmagic == _8SVX) && !found_BODY ) {
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SDL_SetError("Bad 8SVX (no BODY chunk)");
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was_error = 1;
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goto done;
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}
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/* Decode the audio data format */
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memset(spec, 0, sizeof(*spec));
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spec->freq = frequency;
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switch (samplesize) {
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case 8:
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spec->format = AUDIO_S8;
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break;
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case 16:
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spec->format = AUDIO_S16MSB;
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break;
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default:
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SDL_SetError("Unsupported AIFF samplesize");
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was_error = 1;
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goto done;
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}
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spec->channels = (Uint8) channels;
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spec->samples = 4096; /* Good default buffer size */
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*audio_len = channels * numsamples * (samplesize / 8);
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*audio_buf = (Uint8 *)SDL_malloc(*audio_len);
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if ( *audio_buf == NULL ) {
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SDL_SetError("Out of memory");
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return(NULL);
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}
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SDL_RWseek(src, start, RW_SEEK_SET);
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if ( SDL_RWread(src, *audio_buf, *audio_len, 1) != 1 ) {
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SDL_SetError("Unable to read audio data");
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return(NULL);
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}
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/* Don't return a buffer that isn't a multiple of samplesize */
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*audio_len &= ~((samplesize / 8) - 1);
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done:
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if ( freesrc && src ) {
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SDL_RWclose(src);
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}
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if ( was_error ) {
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spec = NULL;
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}
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return(spec);
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}
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