This software is provided 'as-is', without any express or implied

warranty. In no event will the authors be held liable for any damages

arising from the use of this software.

Permission is granted to anyone to use this software for any purpose,

including commercial applications, and to alter it and redistribute it

freely, subject to the following restrictions:

1. The origin of this software must not be misrepresented; you must not

claim that you wrote the original software. If you use this software

in a product, an acknowledgment in the product documentation would be

appreciated but is not required.

2. Altered source versions must be plainly marked as such, and must not be

misrepresented as being the original software.

3. This notice may not be removed or altered from any source distribution.

*/

/* This file supports streaming WAV files, without volume adjustment */

#include <stdlib.h>

#include <string.h>

#include "SDL_audio.h"

#include "SDL_mutex.h"

#include "SDL_rwops.h"

#include "SDL_endian.h"

#include "wavestream.h"

/*

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 PCM_CODE 1

#define ADPCM_CODE 2

#define WAVE_MONO 1

#define WAVE_STEREO 2

Uint32 chunkID;

Uint32 chunkLen;

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;

typedef struct {

Uint32 identifier;

Uint32 type;

Uint32 start;

Uint32 end;

Uint32 fraction;

Uint32 play_count;

} SampleLoop;

typedef struct {

/* Not saved in the chunk we read:

Uint32 chunkID;

Uint32 chunkLen;

*/

Uint32 manufacturer;

Uint32 product;

Uint32 sample_period;

Uint32 MIDI_unity_note;

Uint32 MIDI_pitch_fraction;

Uint32 SMTPE_format;

Uint32 SMTPE_offset;

Uint32 sample_loops;

Uint32 sampler_data;

SampleLoop loops[];

} SamplerChunk;

/*********************************************/

/* Define values for AIFF (IFF audio) format */

/*********************************************/

#define FORM 0x4d524f46 /* "FORM" */

#define AIFF 0x46464941 /* "AIFF" */

#define SSND 0x444e5353 /* "SSND" */

#define COMM 0x4d4d4f43 /* "COMM" */

/* This is the format of the audio mixer data */

static SDL_AudioSpec mixer;

/* Function to load the WAV/AIFF stream */

static SDL_bool LoadWAVStream(WAVStream *wave);

static SDL_bool LoadAIFFStream(WAVStream *wave);

/* 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);

}

}

Mix_SetError("WAV music output not started");

return(NULL);

}

Uint32 magic;

SDL_zerop(wave);

wave->freesrc = freesrc;

magic = SDL_ReadLE32(src);

if (magic == RIFF || magic == WAVE) {

loaded = LoadWAVStream(wave);

} else if (magic == FORM) {

loaded = LoadAIFFStream(wave);

} else {

Mix_SetError("Unknown WAVE format");

}

if (!loaded) {

WAVStream_FreeSong(wave);

return(NULL);

}

SDL_BuildAudioCVT(&wave->cvt,

mixer.format, mixer.channels, mixer.freq);

} else {

SDL_OutOfMemory();

return(NULL);

}

return(wave);

}

/* Start playback of a given WAV stream */

int i;

for (i = 0; i < wave->numloops; ++i) {

WAVLoopPoint *loop = &wave->loops[i];

loop->active = SDL_TRUE;

loop->current_play_count = loop->initial_play_count;

}

SDL_RWseek(wave->src, wave->start, RW_SEEK_SET);

}

Sint64 pos, stop;

WAVLoopPoint *loop;

Sint64 loop_start;

Sint64 loop_stop;

int i;

int consumed;

pos = SDL_RWtell(music->src);

stop = music->stop;

loop = NULL;

for (i = 0; i < music->numloops; ++i) {

loop = &music->loops[i];

if (loop->active) {

const int bytes_per_sample = (SDL_AUDIO_BITSIZE(music->spec.format) / 8) * music->spec.channels;

loop_start = music->start + loop->start * bytes_per_sample;

loop_stop = music->start + (loop->stop + 1) * bytes_per_sample;

if (pos >= loop_start && pos < loop_stop)

{

stop = loop_stop;

break;

}

loop = NULL;

}

if (music->cvt.needed) {

int original_len;

original_len = (int)((double)len/music->cvt.len_ratio);

if (music->cvt.len != original_len) {

int worksize;

if (music->cvt.buf != NULL) {

SDL_free(music->cvt.buf);

worksize = original_len*music->cvt.len_mult;

music->cvt.buf=(Uint8 *)SDL_malloc(worksize);

if (music->cvt.buf == NULL) {

return 0;

}

music->cvt.len = original_len;

}

if ((stop - pos) < original_len) {

original_len = (int)(stop - pos);

}

original_len = SDL_RWread(music->src, music->cvt.buf, 1, original_len);

/* 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.

*/

if ((music->cvt.src_format & 0x0010) && (original_len & 1)) {

original_len--;

}

music->cvt.len = original_len;

SDL_ConvertAudio(&music->cvt);

consumed = music->cvt.len_cvt;

} else {

Uint8 *data;

if ((stop - pos) < len) {

len = (int)(stop - pos);

}

data = SDL_stack_alloc(Uint8, len);

if (data) {

len = SDL_RWread(music->src, data, 1, len);

SDL_stack_free(data);

}

consumed = len;

}

if (loop && SDL_RWtell(music->src) >= stop) {

if (loop->current_play_count == 1) {

loop->active = SDL_FALSE;

if (loop->current_play_count > 0) {

--loop->current_play_count;

}

}

return consumed;

}

int WAVStream_PlaySome(Uint8 *stream, int len)

{

if (!music)

return 0;

while ((SDL_RWtell(music->src) < music->stop) && (len > 0)) {

int consumed = PlaySome(stream, len);

if (!consumed)

break;

stream += consumed;

len -= consumed;

}

return len;

}

/* Stop playback of a stream previously started with WAVStream_Start() */

}

/* Close the given WAV stream */

if (wave->loops) {

SDL_free(wave->loops);

}

if (wave->cvt.buf) {

SDL_free(wave->cvt.buf);

}

}

SDL_free(wave);

}

}

/* Return non-zero if a stream is currently playing */

active = 1;

}

return(active);

}

SDL_RWops *src = wave->src;

SDL_AudioSpec *spec = &wave->spec;

WaveFMT *format;

Uint8 *data;

SDL_bool loaded = SDL_FALSE;

if (chunk_length < sizeof(*format)) {

Mix_SetError("Wave format chunk too small");

return SDL_FALSE;

}

data = (Uint8 *)SDL_malloc(chunk_length);

if (!data) {

Mix_SetError("Out of memory");

return SDL_FALSE;

}

if (!SDL_RWread(wave->src, data, chunk_length, 1)) {

Mix_SetError("Couldn't read %d bytes from WAV file", chunk_length);

return SDL_FALSE;

}

format = (WaveFMT *)data;

/* Decode the audio data format */

switch (SDL_SwapLE16(format->encoding)) {

case PCM_CODE:

/* We can understand this */

break;

default:

Mix_SetError("Unknown WAVE data format");

goto done;

}

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:

Mix_SetError("Unknown PCM data format");

goto done;

}

spec->channels = (Uint8) SDL_SwapLE16(format->channels);

spec->samples = 4096; /* Good default buffer size */

loaded = SDL_TRUE;

done:

SDL_free(data);

return loaded;

}

static SDL_bool ParseDATA(WAVStream *wave, Uint32 chunk_length)

{

wave->start = SDL_RWtell(wave->src);

wave->stop = wave->start + chunk_length;

SDL_RWseek(wave->src, chunk_length, RW_SEEK_CUR);

return SDL_TRUE;

}

static SDL_bool AddLoopPoint(WAVStream *wave, Uint32 play_count, Uint32 start, Uint32 stop)

{

WAVLoopPoint *loop;

WAVLoopPoint *loops = SDL_realloc(wave->loops, (wave->numloops + 1)*sizeof(*wave->loops));

if (!loops) {

Mix_SetError("Out of memory");

return SDL_FALSE;

}

loop = &loops[ wave->numloops ];

loop->start = start;

loop->stop = stop;

loop->initial_play_count = play_count;

loop->current_play_count = play_count;

wave->loops = loops;

++wave->numloops;

return SDL_TRUE;

}

static SDL_bool ParseSMPL(WAVStream *wave, Uint32 chunk_length)

{

SamplerChunk *chunk;

Uint8 *data;

int i;

SDL_bool loaded = SDL_FALSE;

data = (Uint8 *)SDL_malloc(chunk_length);

if (!data) {

Mix_SetError("Out of memory");

return SDL_FALSE;

}

if (!SDL_RWread(wave->src, data, chunk_length, 1)) {

Mix_SetError("Couldn't read %d bytes from WAV file", chunk_length);

return SDL_FALSE;

}

chunk = (SamplerChunk *)data;

for (i = 0; i < SDL_SwapLE32(chunk->sample_loops); ++i) {

const Uint32 LOOP_TYPE_FORWARD = 0;

Uint32 loop_type = SDL_SwapLE32(chunk->loops[i].type);

if (loop_type == LOOP_TYPE_FORWARD) {

AddLoopPoint(wave, SDL_SwapLE32(chunk->loops[i].play_count), SDL_SwapLE32(chunk->loops[i].start), SDL_SwapLE32(chunk->loops[i].end));

}

loaded = SDL_TRUE;

done:

SDL_free(data);

return loaded;

}

static SDL_bool LoadWAVStream(WAVStream *wave)

{

SDL_RWops *src = wave->src;

Uint32 chunk_type;

Uint32 chunk_length;

SDL_bool found_FMT = SDL_FALSE;

SDL_bool found_DATA = SDL_FALSE;

/* WAV magic header */

Uint32 wavelen;

Uint32 WAVEmagic;

/* Check the magic header */

wavelen = SDL_ReadLE32(src);

WAVEmagic = SDL_ReadLE32(src);

/* Read the chunks */

for (; ;) {

chunk_type = SDL_ReadLE32(src);

chunk_length = SDL_ReadLE32(src);

if (chunk_length == 0)

break;

switch (chunk_type)

{

case FMT:

found_FMT = SDL_TRUE;

if (!ParseFMT(wave, chunk_length))

return SDL_FALSE;

break;

case DATA:

found_DATA = SDL_TRUE;

if (!ParseDATA(wave, chunk_length))

return SDL_FALSE;

break;

case SMPL:

if (!ParseSMPL(wave, chunk_length))

return SDL_FALSE;

break;

default:

SDL_RWseek(src, chunk_length, RW_SEEK_CUR);

break;

}

if (!found_FMT) {

Mix_SetError("Bad WAV file (no FMT chunk)");

return SDL_FALSE;

if (!found_DATA) {

Mix_SetError("Bad WAV file (no DATA chunk)");

return SDL_FALSE;

}

return SDL_TRUE;

}

/* 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)

/* 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]);

}

SDL_RWops *src = wave->src;

SDL_AudioSpec *spec = &wave->spec;

SDL_bool found_SSND = SDL_FALSE;

SDL_bool found_COMM = SDL_FALSE;

Uint32 chunk_type;

Uint32 chunk_length;

/* AIFF magic header */

Uint32 AIFFmagic;

/* SSND chunk */

Uint32 offset;

Uint32 blocksize;

/* COMM format chunk */

Uint16 channels = 0;

Uint32 numsamples = 0;

Uint16 samplesize = 0;

Uint8 sane_freq[10];

Uint32 frequency = 0;

/* Check the magic header */

chunk_length = SDL_ReadBE32(src);

AIFFmagic = SDL_ReadLE32(src);

if (AIFFmagic != AIFF) {

}

/* From what I understand of the specification, chunks may appear in

*

* TODO: Better sanity-checking. E.g. what happens if the AIFF file

* contains compressed sound data?

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)

found_SSND = SDL_TRUE;

offset = SDL_ReadBE32(src);

blocksize = SDL_ReadBE32(src);

wave->start = SDL_RWtell(src) + offset;

break;

case COMM:

/* Read the audio data format chunk */

channels = SDL_ReadBE16(src);

numsamples = SDL_ReadBE32(src);

samplesize = SDL_ReadBE16(src);

break;

default:

break;

}

} while ((!found_SSND || !found_COMM)

&& SDL_RWseek(src, next_chunk, RW_SEEK_SET) != -1);

if (!found_SSND) {

Mix_SetError("Bad AIFF file (no SSND chunk)");

}

if (!found_COMM) {

Mix_SetError("Bad AIFF file (no COMM chunk)");

}

/* Decode the audio data format */

spec->freq = frequency;

switch (samplesize) {

case 8:

spec->format = AUDIO_S8;

break;

case 16:

spec->format = AUDIO_S16MSB;

break;

default:

Mix_SetError("Unknown samplesize in data format");

}

spec->channels = (Uint8) channels;

spec->samples = 4096; /* Good default buffer size */