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kolibrios-gitea/contrib/games/wolf3d/SDL/SDL_wave.c
turbocat 127b85086b Wolfenstein 3D: 
- 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
2021-03-23 06:51:33 +00:00

592 lines
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C
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997, 1998, 1999, 2000, 2001 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@devolution.com
*/
#ifdef SAVE_RCSID
static char rcsid =
"@(#) $Id: SDL_wave.c,v 1.2 2001/04/26 16:50:17 hercules Exp $";
#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 {
Uint8 hPredictor;
Uint16 iDelta;
Sint16 iSamp1;
Sint16 iSamp2;
};
static struct MS_ADPCM_decoder {
WaveFMT wavefmt;
Uint16 wSamplesPerBlock;
Uint16 wNumCoef;
Sint16 aCoeff[7][2];
/* * * */
struct MS_ADPCM_decodestate state[2];
} 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 {
WaveFMT wavefmt;
Uint16 wSamplesPerBlock;
/* * * */
struct IMA_ADPCM_decodestate state[2];
} 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);
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 */
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