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wavestream.c
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/*
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SDL_mixer: An audio mixer library based on the SDL library
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Copyright (C) 1997-2004 Sam Lantinga
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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
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slouken@libsdl.org
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*/
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/* $Id$ */
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/* This file supports streaming WAV files, without volume adjustment */
#include <stdlib.h>
#include <string.h>
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#include "SDL_audio.h"
#include "SDL_mutex.h"
#include "SDL_rwops.h"
#include "SDL_endian.h"
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#include "SDL_mixer.h"
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#include "wavestream.h"
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/*
Taken with permission from SDL_wave.h, part of the SDL library,
available at: http://www.libsdl.org/
and placed under the same license as this mixer library.
*/
/* WAVE files are little-endian */
/*******************************************/
/* Define values for Microsoft WAVE format */
/*******************************************/
#define RIFF 0x46464952 /* "RIFF" */
#define WAVE 0x45564157 /* "WAVE" */
#define FACT 0x74636166 /* "fact" */
#define LIST 0x5453494c /* "LIST" */
#define FMT 0x20746D66 /* "fmt " */
#define DATA 0x61746164 /* "data" */
#define PCM_CODE 1
#define ADPCM_CODE 2
#define WAVE_MONO 1
#define WAVE_STEREO 2
/* Normally, these three chunks come consecutively in a WAVE file */
typedef struct WaveFMT {
/* Not saved in the chunk we read:
Uint32 FMTchunk;
Uint32 fmtlen;
*/
Uint16 encoding;
Uint16 channels; /* 1 = mono, 2 = stereo */
Uint32 frequency; /* One of 11025, 22050, or 44100 Hz */
Uint32 byterate; /* Average bytes per second */
Uint16 blockalign; /* Bytes per sample block */
Uint16 bitspersample; /* One of 8, 12, 16, or 4 for ADPCM */
} WaveFMT;
/* The general chunk found in the WAVE file */
typedef struct Chunk {
Uint32 magic;
Uint32 length;
Uint8 *data; /* Data includes magic and length */
} Chunk;
/*********************************************/
/* Define values for AIFF (IFF audio) format */
/*********************************************/
#define FORM 0x4d524f46 /* "FORM" */
#define AIFF 0x46464941 /* "AIFF" */
#define SSND 0x444e5353 /* "SSND" */
#define COMM 0x4d4d4f43 /* "COMM" */
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/* Currently we only support a single stream at a time */
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static WAVStream *music = NULL;
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/* This is the format of the audio mixer data */
static SDL_AudioSpec mixer;
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static int wavestream_volume = MIX_MAX_VOLUME;
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/* Function to load the WAV/AIFF stream */
static FILE *LoadWAVStream (const char *file, SDL_AudioSpec *spec,
long *start, long *stop);
static FILE *LoadAIFFStream (const char *file, SDL_AudioSpec *spec,
long *start, long *stop);
/* Initialize the WAVStream player, with the given mixer settings
This function returns 0, or -1 if there was an error.
*/
int WAVStream_Init(SDL_AudioSpec *mixerfmt)
{
mixer = *mixerfmt;
return(0);
}
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void WAVStream_SetVolume(int volume)
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{
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wavestream_volume = volume;
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}
/* Load a WAV stream from the given file */
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WAVStream *WAVStream_LoadSong(const char *file, const char *magic)
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{
WAVStream *wave;
SDL_AudioSpec wavespec;
if ( ! mixer.format ) {
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Mix_SetError("WAV music output not started");
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return(NULL);
}
wave = (WAVStream *)malloc(sizeof *wave);
if ( wave ) {
memset(wave, 0, (sizeof *wave));
if ( strcmp(magic, "RIFF") == 0 ) {
wave->wavefp = LoadWAVStream(file, &wavespec,
&wave->start, &wave->stop);
} else
if ( strcmp(magic, "FORM") == 0 ) {
wave->wavefp = LoadAIFFStream(file, &wavespec,
&wave->start, &wave->stop);
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} else {
Mix_SetError("Unknown WAVE format");
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}
if ( wave->wavefp == NULL ) {
free(wave);
return(NULL);
}
SDL_BuildAudioCVT(&wave->cvt,
wavespec.format, wavespec.channels, wavespec.freq,
mixer.format, mixer.channels, mixer.freq);
}
return(wave);
}
/* Start playback of a given WAV stream */
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void WAVStream_Start(WAVStream *wave)
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{
clearerr(wave->wavefp);
fseek(wave->wavefp, wave->start, SEEK_SET);
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music = wave;
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}
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/* Play some of a stream previously started with WAVStream_Start() */
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void WAVStream_PlaySome(Uint8 *stream, int len)
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{
long pos;
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if ( music && ((pos=ftell(music->wavefp)) < music->stop) ) {
if ( music->cvt.needed ) {
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int original_len;
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original_len=(int)((double)len/music->cvt.len_ratio);
if ( music->cvt.len != original_len ) {
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int worksize;
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if ( music->cvt.buf != NULL ) {
free(music->cvt.buf);
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}
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worksize = original_len*music->cvt.len_mult;
music->cvt.buf=(Uint8 *)malloc(worksize);
if ( music->cvt.buf == NULL ) {
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return;
}
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music->cvt.len = original_len;
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}
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if ( (music->stop - pos) < original_len ) {
original_len = (music->stop - pos);
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}
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original_len = fread(music->cvt.buf,1,original_len,music->wavefp);
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/* At least at the time of writing, SDL_ConvertAudio()
does byte-order swapping starting at the end of the
buffer. Thus, if we are reading 16-bit samples, we
had better make damn sure that we get an even
number of bytes, or we'll get garbage.
*/
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if ( (music->cvt.src_format & 0x0010) && (original_len & 1) ) {
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original_len--;
}
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music->cvt.len = original_len;
SDL_ConvertAudio(&music->cvt);
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SDL_MixAudio(stream, music->cvt.buf, music->cvt.len_cvt, wavestream_volume);
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} else {
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Uint8 *data;
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if ( (music->stop - pos) < len ) {
len = (music->stop - pos);
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}
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data = SDL_stack_alloc(Uint8, len);
if (data)
{
fread(data, len, 1, music->wavefp);
SDL_MixAudio(stream, data, len, wavestream_volume);
SDL_stack_free(data);
}
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}
}
}
/* Stop playback of a stream previously started with WAVStream_Start() */
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void WAVStream_Stop(void)
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{
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music = NULL;
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}
/* Close the given WAV stream */
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void WAVStream_FreeSong(WAVStream *wave)
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{
if ( wave ) {
/* Clean up associated data */
if ( wave->wavefp ) {
fclose(wave->wavefp);
}
if ( wave->cvt.buf ) {
free(wave->cvt.buf);
}
free(wave);
}
}
/* Return non-zero if a stream is currently playing */
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int WAVStream_Active(void)
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{
int active;
active = 0;
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if ( music && (ftell(music->wavefp) < music->stop) ) {
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active = 1;
}
return(active);
}
static int ReadChunk(SDL_RWops *src, Chunk *chunk, int read_data)
{
chunk->magic = SDL_ReadLE32(src);
chunk->length = SDL_ReadLE32(src);
if ( read_data ) {
chunk->data = (Uint8 *)malloc(chunk->length);
if ( chunk->data == NULL ) {
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Mix_SetError("Out of memory");
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return(-1);
}
if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) {
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Mix_SetError("Couldn't read chunk");
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free(chunk->data);
return(-1);
}
} else {
SDL_RWseek(src, chunk->length, SEEK_CUR);
}
return(chunk->length);
}
static FILE *LoadWAVStream (const char *file, SDL_AudioSpec *spec,
long *start, long *stop)
{
int was_error;
FILE *wavefp;
SDL_RWops *src;
Chunk chunk;
int lenread;
/* WAV magic header */
Uint32 RIFFchunk;
Uint32 wavelen;
Uint32 WAVEmagic;
/* FMT chunk */
WaveFMT *format = NULL;
/* Make sure we are passed a valid data source */
was_error = 0;
wavefp = fopen(file, "rb");
src = NULL;
if ( wavefp ) {
src = SDL_RWFromFP(wavefp, 0);
}
if ( src == NULL ) {
was_error = 1;
goto done;
}
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/* Check the magic header */
RIFFchunk = SDL_ReadLE32(src);
wavelen = SDL_ReadLE32(src);
WAVEmagic = SDL_ReadLE32(src);
if ( (RIFFchunk != RIFF) || (WAVEmagic != WAVE) ) {
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Mix_SetError("Unrecognized file type (not WAVE)");
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was_error = 1;
goto done;
}
/* Read the audio data format chunk */
chunk.data = NULL;
do {
/* FIXME! Add this logic to SDL_LoadWAV_RW() */
if ( chunk.data ) {
free(chunk.data);
}
lenread = ReadChunk(src, &chunk, 1);
if ( lenread < 0 ) {
was_error = 1;
goto done;
}
} while ( (chunk.magic == FACT) || (chunk.magic == LIST) );
/* Decode the audio data format */
format = (WaveFMT *)chunk.data;
if ( chunk.magic != FMT ) {
free(chunk.data);
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Mix_SetError("Complex WAVE files not supported");
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was_error = 1;
goto done;
}
switch (SDL_SwapLE16(format->encoding)) {
case PCM_CODE:
/* We can understand this */
break;
default:
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Mix_SetError("Unknown WAVE data format");
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was_error = 1;
goto done;
}
memset(spec, 0, (sizeof *spec));
spec->freq = SDL_SwapLE32(format->frequency);
switch (SDL_SwapLE16(format->bitspersample)) {
case 8:
spec->format = AUDIO_U8;
break;
case 16:
spec->format = AUDIO_S16;
break;
default:
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Mix_SetError("Unknown PCM data format");
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was_error = 1;
goto done;
}
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spec->channels = (Uint8) SDL_SwapLE16(format->channels);
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spec->samples = 4096; /* Good default buffer size */
/* Set the file offset to the DATA chunk data */
chunk.data = NULL;
do {
*start = SDL_RWtell(src) + 2*sizeof(Uint32);
lenread = ReadChunk(src, &chunk, 0);
if ( lenread < 0 ) {
was_error = 1;
goto done;
}
} while ( chunk.magic != DATA );
*stop = SDL_RWtell(src);
done:
if ( format != NULL ) {
free(format);
}
if ( src ) {
SDL_RWclose(src);
}
if ( was_error ) {
if ( wavefp ) {
fclose(wavefp);
wavefp = NULL;
}
}
return(wavefp);
}
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/* I couldn't get SANE_to_double() to work, so I stole this from libsndfile.
* I don't pretend to fully understand it.
*/
static Uint32 SANE_to_Uint32 (Uint8 *sanebuf)
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{
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/* Negative number? */
if (sanebuf[0] & 0x80)
return 0;
/* Less than 1? */
if (sanebuf[0] <= 0x3F)
return 1;
/* Way too big? */
if (sanebuf[0] > 0x40)
return 0x4000000;
/* Still too big? */
if (sanebuf[0] == 0x40 && sanebuf[1] > 0x1C)
return 800000000;
return ((sanebuf[2] << 23) | (sanebuf[3] << 15) | (sanebuf[4] << 7)
| (sanebuf[5] >> 1)) >> (29 - sanebuf[1]);
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}
static FILE *LoadAIFFStream (const char *file, SDL_AudioSpec *spec,
long *start, long *stop)
{
int was_error;
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int found_SSND;
int found_COMM;
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FILE *wavefp;
SDL_RWops *src;
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Uint32 chunk_type;
Uint32 chunk_length;
long next_chunk;
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/* AIFF magic header */
Uint32 FORMchunk;
Uint32 AIFFmagic;
/* SSND chunk */
Uint32 offset;
Uint32 blocksize;
/* COMM format chunk */
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Uint16 channels = 0;
Uint32 numsamples = 0;
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 */
was_error = 0;
wavefp = fopen(file, "rb");
src = NULL;
if ( wavefp ) {
src = SDL_RWFromFP(wavefp, 0);
}
if ( src == NULL ) {
was_error = 1;
goto done;
}
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/* Check the magic header */
FORMchunk = SDL_ReadLE32(src);
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chunk_length = SDL_ReadBE32(src);
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AIFFmagic = SDL_ReadLE32(src);
if ( (FORMchunk != FORM) || (AIFFmagic != AIFF) ) {
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Mix_SetError("Unrecognized file type (not AIFF)");
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was_error = 1;
goto done;
}
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/* From what I understand of the specification, chunks may appear in
* any order, and we should just ignore unknown ones.
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*
* TODO: Better sanity-checking. E.g. what happens if the AIFF file
* contains compressed sound data?
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*/
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found_SSND = 0;
found_COMM = 0;
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do {
chunk_type = SDL_ReadLE32(src);
chunk_length = SDL_ReadBE32(src);
next_chunk = SDL_RWtell(src) + chunk_length;
/* Paranoia to avoid infinite loops */
if (chunk_length == 0)
break;
switch (chunk_type) {
case SSND:
found_SSND = 1;
offset = SDL_ReadBE32(src);
blocksize = SDL_ReadBE32(src);
*start = SDL_RWtell(src) + offset;
break;
case COMM:
found_COMM = 1;
/* Read the audio data format chunk */
channels = SDL_ReadBE16(src);
numsamples = SDL_ReadBE32(src);
samplesize = SDL_ReadBE16(src);
SDL_RWread(src, sane_freq, sizeof(sane_freq), 1);
frequency = SANE_to_Uint32(sane_freq);
break;
default:
break;
}
} while ((!found_SSND || !found_COMM)
&& SDL_RWseek(src, next_chunk, SEEK_SET) != -1);
if (!found_SSND) {
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Mix_SetError("Bad AIFF file (no SSND chunk)");
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was_error = 1;
goto done;
}
if (!found_COMM) {
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Mix_SetError("Bad AIFF file (no COMM chunk)");
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was_error = 1;
goto done;
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}
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*stop = *start + channels * numsamples * (samplesize / 8);
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/* Decode the audio data format */
memset(spec, 0, (sizeof *spec));
spec->freq = frequency;
switch (samplesize) {
case 8:
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spec->format = AUDIO_S8;
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break;
case 16:
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spec->format = AUDIO_S16MSB;
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break;
default:
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Mix_SetError("Unknown samplesize in data format");
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was_error = 1;
goto done;
}
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spec->channels = (Uint8) channels;
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spec->samples = 4096; /* Good default buffer size */
done:
if ( src ) {
SDL_RWclose(src);
}
if ( was_error ) {
if ( wavefp ) {
fclose(wavefp);
wavefp = NULL;
}
}
return(wavefp);
}