/*

SDL - Simple DirectMedia Layer

Copyright (C) 1997-2004 Sam Lantinga

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

slouken@libsdl.org

*/

#ifdef SAVE_RCSID

static char rcsid =

"@(#) $Id$";

#endif

#ifndef DISABLE_FILE

/* Microsoft WAVE file loading routines */

#include <stdlib.h>

#include <string.h>

#include "SDL_error.h"

#include "SDL_audio.h"

#include "SDL_wave.h"

#include "SDL_endian.h"

#ifndef NELEMS

#define NELEMS(array) ((sizeof array)/(sizeof array[0]))

#endif

static int ReadChunk(SDL_RWops *src, Chunk *chunk);

struct MS_ADPCM_decodestate {

};

static struct MS_ADPCM_decoder {

} MS_ADPCM_state;

static int InitMS_ADPCM(WaveFMT *format)

{

Uint8 *rogue_feel;

Uint16 extra_info;

int i;

/* Set the rogue pointer to the MS_ADPCM specific data */

MS_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);

MS_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);

MS_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);

MS_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);

MS_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);

MS_ADPCM_state.wavefmt.bitspersample =

SDL_SwapLE16(format->bitspersample);

rogue_feel = (Uint8 *)format+sizeof(*format);

if ( sizeof(*format) == 16 ) {

extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]);

rogue_feel += sizeof(Uint16);

}

MS_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]);

rogue_feel += sizeof(Uint16);

MS_ADPCM_state.wNumCoef = ((rogue_feel[1]<<8)|rogue_feel[0]);

rogue_feel += sizeof(Uint16);

if ( MS_ADPCM_state.wNumCoef != 7 ) {

SDL_SetError("Unknown set of MS_ADPCM coefficients");

return(-1);

}

for ( i=0; i<MS_ADPCM_state.wNumCoef; ++i ) {

MS_ADPCM_state.aCoeff[i][0] = ((rogue_feel[1]<<8)|rogue_feel[0]);

rogue_feel += sizeof(Uint16);

MS_ADPCM_state.aCoeff[i][1] = ((rogue_feel[1]<<8)|rogue_feel[0]);

rogue_feel += sizeof(Uint16);

}

return(0);

}

static Sint32 MS_ADPCM_nibble(struct MS_ADPCM_decodestate *state,

Uint8 nybble, Sint16 *coeff)

{

const Sint32 max_audioval = ((1<<(16-1))-1);

const Sint32 min_audioval = -(1<<(16-1));

const Sint32 adaptive[] = {

230, 230, 230, 230, 307, 409, 512, 614,

768, 614, 512, 409, 307, 230, 230, 230

};

Sint32 new_sample, delta;

new_sample = ((state->iSamp1 * coeff[0]) +

(state->iSamp2 * coeff[1]))/256;

if ( nybble & 0x08 ) {

new_sample += state->iDelta * (nybble-0x10);

} else {

new_sample += state->iDelta * nybble;

}

if ( new_sample < min_audioval ) {

new_sample = min_audioval;

} else

if ( new_sample > max_audioval ) {

new_sample = max_audioval;

}

delta = ((Sint32)state->iDelta * adaptive[nybble])/256;

if ( delta < 16 ) {

delta = 16;

}

state->iDelta = delta;

state->iSamp2 = state->iSamp1;

state->iSamp1 = new_sample;

return(new_sample);

}

static int MS_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len)

{

struct MS_ADPCM_decodestate *state[2];

Uint8 *freeable, *encoded, *decoded;

Sint32 encoded_len, samplesleft;

Sint8 nybble, stereo;

Sint16 *coeff[2];

Sint32 new_sample;

/* Allocate the proper sized output buffer */

encoded_len = *audio_len;

encoded = *audio_buf;

freeable = *audio_buf;

*audio_len = (encoded_len/MS_ADPCM_state.wavefmt.blockalign) *

MS_ADPCM_state.wSamplesPerBlock*

MS_ADPCM_state.wavefmt.channels*sizeof(Sint16);

*audio_buf = (Uint8 *)malloc(*audio_len);

if ( *audio_buf == NULL ) {

SDL_Error(SDL_ENOMEM);

return(-1);

}

decoded = *audio_buf;

/* Get ready... Go! */

stereo = (MS_ADPCM_state.wavefmt.channels == 2);

state[0] = &MS_ADPCM_state.state[0];

state[1] = &MS_ADPCM_state.state[stereo];

while ( encoded_len >= MS_ADPCM_state.wavefmt.blockalign ) {

/* Grab the initial information for this block */

state[0]->hPredictor = *encoded++;

if ( stereo ) {

state[1]->hPredictor = *encoded++;

}

state[0]->iDelta = ((encoded[1]<<8)|encoded[0]);

encoded += sizeof(Sint16);

if ( stereo ) {

state[1]->iDelta = ((encoded[1]<<8)|encoded[0]);

encoded += sizeof(Sint16);

}

state[0]->iSamp1 = ((encoded[1]<<8)|encoded[0]);

encoded += sizeof(Sint16);

if ( stereo ) {

state[1]->iSamp1 = ((encoded[1]<<8)|encoded[0]);

encoded += sizeof(Sint16);

}

state[0]->iSamp2 = ((encoded[1]<<8)|encoded[0]);

encoded += sizeof(Sint16);

if ( stereo ) {

state[1]->iSamp2 = ((encoded[1]<<8)|encoded[0]);

encoded += sizeof(Sint16);

}

coeff[0] = MS_ADPCM_state.aCoeff[state[0]->hPredictor];

coeff[1] = MS_ADPCM_state.aCoeff[state[1]->hPredictor];

/* Store the two initial samples we start with */

decoded[0] = state[0]->iSamp2&0xFF;

decoded[1] = state[0]->iSamp2>>8;

decoded += 2;

if ( stereo ) {

decoded[0] = state[1]->iSamp2&0xFF;

decoded[1] = state[1]->iSamp2>>8;

decoded += 2;

}

decoded[0] = state[0]->iSamp1&0xFF;

decoded[1] = state[0]->iSamp1>>8;

decoded += 2;

if ( stereo ) {

decoded[0] = state[1]->iSamp1&0xFF;

decoded[1] = state[1]->iSamp1>>8;

decoded += 2;

}

/* Decode and store the other samples in this block */

samplesleft = (MS_ADPCM_state.wSamplesPerBlock-2)*

MS_ADPCM_state.wavefmt.channels;

while ( samplesleft > 0 ) {

nybble = (*encoded)>>4;

new_sample = MS_ADPCM_nibble(state[0],nybble,coeff[0]);

decoded[0] = new_sample&0xFF;

new_sample >>= 8;

decoded[1] = new_sample&0xFF;

decoded += 2;

nybble = (*encoded)&0x0F;

new_sample = MS_ADPCM_nibble(state[1],nybble,coeff[1]);

decoded[0] = new_sample&0xFF;

new_sample >>= 8;

decoded[1] = new_sample&0xFF;

decoded += 2;

++encoded;

samplesleft -= 2;

}

encoded_len -= MS_ADPCM_state.wavefmt.blockalign;

}

free(freeable);

return(0);

}

struct IMA_ADPCM_decodestate {

Sint32 sample;

Sint8 index;

};

static struct IMA_ADPCM_decoder {

} IMA_ADPCM_state;

static int InitIMA_ADPCM(WaveFMT *format)

{

Uint8 *rogue_feel;

Uint16 extra_info;

/* Set the rogue pointer to the IMA_ADPCM specific data */

IMA_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);

IMA_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);

IMA_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);

IMA_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);

IMA_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);

IMA_ADPCM_state.wavefmt.bitspersample =

SDL_SwapLE16(format->bitspersample);

rogue_feel = (Uint8 *)format+sizeof(*format);

if ( sizeof(*format) == 16 ) {

extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]);

rogue_feel += sizeof(Uint16);

}

IMA_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]);

return(0);

}

static Sint32 IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state,Uint8 nybble)

{

const Sint32 max_audioval = ((1<<(16-1))-1);

const Sint32 min_audioval = -(1<<(16-1));

const int index_table[16] = {

-1, -1, -1, -1,

2, 4, 6, 8,

-1, -1, -1, -1,

2, 4, 6, 8

};

const Sint32 step_table[89] = {

7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31,

34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130,

143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408,

449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282,

1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,

3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630,

9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350,

22385, 24623, 27086, 29794, 32767

};

Sint32 delta, step;

/* Compute difference and new sample value */

step = step_table[state->index];

delta = step >> 3;

if ( nybble & 0x04 ) delta += step;

if ( nybble & 0x02 ) delta += (step >> 1);

if ( nybble & 0x01 ) delta += (step >> 2);

if ( nybble & 0x08 ) delta = -delta;

state->sample += delta;

/* Update index value */

state->index += index_table[nybble];

if ( state->index > 88 ) {

state->index = 88;

} else

if ( state->index < 0 ) {

state->index = 0;

}

/* Clamp output sample */

if ( state->sample > max_audioval ) {

state->sample = max_audioval;

} else

if ( state->sample < min_audioval ) {

state->sample = min_audioval;

}

return(state->sample);

}

/* Fill the decode buffer with a channel block of data (8 samples) */

static void Fill_IMA_ADPCM_block(Uint8 *decoded, Uint8 *encoded,

int channel, int numchannels, struct IMA_ADPCM_decodestate *state)

{

int i;

Sint8 nybble;

Sint32 new_sample;

decoded += (channel * 2);

for ( i=0; i<4; ++i ) {

nybble = (*encoded)&0x0F;

new_sample = IMA_ADPCM_nibble(state, nybble);

decoded[0] = new_sample&0xFF;

new_sample >>= 8;

decoded[1] = new_sample&0xFF;

decoded += 2 * numchannels;

nybble = (*encoded)>>4;

new_sample = IMA_ADPCM_nibble(state, nybble);

decoded[0] = new_sample&0xFF;

new_sample >>= 8;

decoded[1] = new_sample&0xFF;

decoded += 2 * numchannels;

++encoded;

}

}

static int IMA_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len)

{

struct IMA_ADPCM_decodestate *state;

Uint8 *freeable, *encoded, *decoded;

Sint32 encoded_len, samplesleft;

int c, channels;

/* Check to make sure we have enough variables in the state array */

channels = IMA_ADPCM_state.wavefmt.channels;

if ( channels > NELEMS(IMA_ADPCM_state.state) ) {

SDL_SetError("IMA ADPCM decoder can only handle %d channels",

NELEMS(IMA_ADPCM_state.state));

return(-1);

}

state = IMA_ADPCM_state.state;

/* Allocate the proper sized output buffer */

encoded_len = *audio_len;

encoded = *audio_buf;

freeable = *audio_buf;

*audio_len = (encoded_len/IMA_ADPCM_state.wavefmt.blockalign) *

IMA_ADPCM_state.wSamplesPerBlock*

IMA_ADPCM_state.wavefmt.channels*sizeof(Sint16);

*audio_buf = (Uint8 *)malloc(*audio_len);

if ( *audio_buf == NULL ) {

SDL_Error(SDL_ENOMEM);

return(-1);

}

decoded = *audio_buf;

/* Get ready... Go! */

while ( encoded_len >= IMA_ADPCM_state.wavefmt.blockalign ) {

/* Grab the initial information for this block */

for ( c=0; c<channels; ++c ) {

/* Fill the state information for this block */

state[c].sample = ((encoded[1]<<8)|encoded[0]);

encoded += 2;

if ( state[c].sample & 0x8000 ) {

state[c].sample -= 0x10000;

}

state[c].index = *encoded++;

/* Reserved byte in buffer header, should be 0 */

if ( *encoded++ != 0 ) {

/* Uh oh, corrupt data? Buggy code? */;

}

/* Store the initial sample we start with */

decoded[0] = state[c].sample&0xFF;

decoded[1] = state[c].sample>>8;

decoded += 2;

}

/* Decode and store the other samples in this block */

samplesleft = (IMA_ADPCM_state.wSamplesPerBlock-1)*channels;

while ( samplesleft > 0 ) {

for ( c=0; c<channels; ++c ) {

Fill_IMA_ADPCM_block(decoded, encoded,

c, channels, &state[c]);

encoded += 4;

samplesleft -= 8;

}

decoded += (channels * 8 * 2);

}

encoded_len -= IMA_ADPCM_state.wavefmt.blockalign;

}

free(freeable);

return(0);

}

SDL_AudioSpec * SDL_LoadWAV_RW (SDL_RWops *src, int freesrc,

SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)

{

int was_error;

Chunk chunk;

int lenread;

int MS_ADPCM_encoded, IMA_ADPCM_encoded;

int samplesize;

/* 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;

if ( src == NULL ) {

was_error = 1;

goto done;

}

/* Check the magic header */

RIFFchunk = SDL_ReadLE32(src);

wavelen = SDL_ReadLE32(src);

if ( wavelen == WAVE ) { /* The RIFFchunk has already been read */

WAVEmagic = wavelen;

wavelen = RIFFchunk;

RIFFchunk = RIFF;

} else {

WAVEmagic = SDL_ReadLE32(src);

}

if ( (RIFFchunk != RIFF) || (WAVEmagic != WAVE) ) {

SDL_SetError("Unrecognized file type (not WAVE)");

was_error = 1;

goto done;

}

/* Read the audio data format chunk */

chunk.data = NULL;

do {

if ( chunk.data != NULL ) {

free(chunk.data);

}

lenread = ReadChunk(src, &chunk);

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 ) {

SDL_SetError("Complex WAVE files not supported");

was_error = 1;

goto done;

}

MS_ADPCM_encoded = IMA_ADPCM_encoded = 0;

switch (SDL_SwapLE16(format->encoding)) {

case PCM_CODE:

/* We can understand this */

break;

case MS_ADPCM_CODE:

/* Try to understand this */

if ( InitMS_ADPCM(format) < 0 ) {

was_error = 1;

goto done;

}

MS_ADPCM_encoded = 1;

break;

case IMA_ADPCM_CODE:

/* Try to understand this */

if ( InitIMA_ADPCM(format) < 0 ) {

was_error = 1;

goto done;

}

IMA_ADPCM_encoded = 1;

break;

default:

SDL_SetError("Unknown WAVE data format: 0x%.4x",

SDL_SwapLE16(format->encoding));

was_error = 1;

goto done;

}

memset(spec, 0, (sizeof *spec));

spec->freq = SDL_SwapLE32(format->frequency);

switch (SDL_SwapLE16(format->bitspersample)) {

case 4:

if ( MS_ADPCM_encoded || IMA_ADPCM_encoded ) {

spec->format = AUDIO_S16;

} else {

was_error = 1;

}

break;

case 8:

spec->format = AUDIO_U8;

break;

case 16:

spec->format = AUDIO_S16;

break;

default:

was_error = 1;

break;

}

if ( was_error ) {

SDL_SetError("Unknown %d-bit PCM data format",

SDL_SwapLE16(format->bitspersample));

goto done;

}

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

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

/* Read the audio data chunk */

*audio_buf = NULL;

do {

if ( *audio_buf != NULL ) {

free(*audio_buf);

}

lenread = ReadChunk(src, &chunk);

if ( lenread < 0 ) {

was_error = 1;

goto done;

}

*audio_len = lenread;

*audio_buf = chunk.data;

} while ( chunk.magic != DATA );

if ( MS_ADPCM_encoded ) {

if ( MS_ADPCM_decode(audio_buf, audio_len) < 0 ) {

was_error = 1;

goto done;

}

}

if ( IMA_ADPCM_encoded ) {

if ( IMA_ADPCM_decode(audio_buf, audio_len) < 0 ) {

was_error = 1;

goto done;

}

}

/* Don't return a buffer that isn't a multiple of samplesize */

samplesize = ((spec->format & 0xFF)/8)*spec->channels;

*audio_len &= ~(samplesize-1);

done:

if ( format != NULL ) {

free(format);

}

if ( freesrc && src ) {

SDL_RWclose(src);

}

if ( was_error ) {

spec = NULL;

}

return(spec);

}

/* Since the WAV memory is allocated in the shared library, it must also

be freed here. (Necessary under Win32, VC++)

*/

void SDL_FreeWAV(Uint8 *audio_buf)

{

if ( audio_buf != NULL ) {

free(audio_buf);

}

}

static int ReadChunk(SDL_RWops *src, Chunk *chunk)

{

chunk->magic = SDL_ReadLE32(src);

chunk->length = SDL_ReadLE32(src);

chunk->data = (Uint8 *)malloc(chunk->length);

if ( chunk->data == NULL ) {

SDL_Error(SDL_ENOMEM);

return(-1);

}

if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) {

SDL_Error(SDL_EFREAD);

free(chunk->data);

return(-1);

}

return(chunk->length);

}

#endif /* ENABLE_FILE */