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SDL_wave.c
636 lines (583 loc) · 19.2 KB
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/*
SDL - Simple DirectMedia Layer
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Copyright (C) 1997-2010 Sam Lantinga
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This library is free software; you can redistribute it and/or
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modify it under the terms of the GNU Lesser General Public
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License as published by the Free Software Foundation; either
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version 2.1 of the License, or (at your option) any later version.
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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
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Lesser General Public License for more details.
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You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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Sam Lantinga
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slouken@libsdl.org
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*/
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#include "SDL_config.h"
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/* Microsoft WAVE file loading routines */
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#include "SDL_audio.h"
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#include "SDL_wave.h"
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static int ReadChunk(SDL_RWops * src, Chunk * chunk);
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struct MS_ADPCM_decodestate
{
Uint8 hPredictor;
Uint16 iDelta;
Sint16 iSamp1;
Sint16 iSamp2;
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};
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static struct MS_ADPCM_decoder
{
WaveFMT wavefmt;
Uint16 wSamplesPerBlock;
Uint16 wNumCoef;
Sint16 aCoeff[7][2];
/* * * */
struct MS_ADPCM_decodestate state[2];
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} MS_ADPCM_state;
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static int
InitMS_ADPCM(WaveFMT * format)
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{
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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);
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}
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static Sint32
MS_ADPCM_nibble(struct MS_ADPCM_decodestate *state,
Uint8 nybble, Sint16 * coeff)
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{
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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 = (Uint16) delta;
state->iSamp2 = state->iSamp1;
state->iSamp1 = (Sint16) new_sample;
return (new_sample);
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}
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static int
MS_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
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{
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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 *) SDL_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;
}
SDL_free(freeable);
return (0);
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}
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struct IMA_ADPCM_decodestate
{
Sint32 sample;
Sint8 index;
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};
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static struct IMA_ADPCM_decoder
{
WaveFMT wavefmt;
Uint16 wSamplesPerBlock;
/* * * */
struct IMA_ADPCM_decodestate state[2];
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} IMA_ADPCM_state;
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static int
InitIMA_ADPCM(WaveFMT * format)
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{
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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);
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}
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static Sint32
IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state, Uint8 nybble)
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{
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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);
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}
/* Fill the decode buffer with a channel block of data (8 samples) */
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static void
Fill_IMA_ADPCM_block(Uint8 * decoded, Uint8 * encoded,
int channel, int numchannels,
struct IMA_ADPCM_decodestate *state)
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{
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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;
}
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}
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static int
IMA_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
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{
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struct IMA_ADPCM_decodestate *state;
Uint8 *freeable, *encoded, *decoded;
Sint32 encoded_len, samplesleft;
unsigned int c, channels;
/* Check to make sure we have enough variables in the state array */
channels = IMA_ADPCM_state.wavefmt.channels;
if (channels > SDL_arraysize(IMA_ADPCM_state.state)) {
SDL_SetError("IMA ADPCM decoder can only handle %d channels",
SDL_arraysize(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 *) SDL_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] = (Uint8) (state[c].sample & 0xFF);
decoded[1] = (Uint8) (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;
}
SDL_free(freeable);
return (0);
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}
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SDL_AudioSpec *
SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
SDL_AudioSpec * spec, Uint8 ** audio_buf, Uint32 * audio_len)
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{
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int was_error;
Chunk chunk;
int lenread;
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int IEEE_float_encoded, MS_ADPCM_encoded, IMA_ADPCM_encoded;
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int samplesize;
/* WAV magic header */
Uint32 RIFFchunk;
Uint32 wavelen = 0;
Uint32 WAVEmagic;
Uint32 headerDiff = 0;
/* 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;
}
headerDiff += sizeof(Uint32); /* for WAVE */
/* Read the audio data format chunk */
chunk.data = NULL;
do {
if (chunk.data != NULL) {
SDL_free(chunk.data);
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chunk.data = NULL;
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}
lenread = ReadChunk(src, &chunk);
if (lenread < 0) {
was_error = 1;
goto done;
}
/* 2 Uint32's for chunk header+len, plus the lenread */
headerDiff += lenread + 2 * sizeof(Uint32);
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} while ((chunk.magic == FACT) || (chunk.magic == LIST));
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/* 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;
}
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IEEE_float_encoded = MS_ADPCM_encoded = IMA_ADPCM_encoded = 0;
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switch (SDL_SwapLE16(format->encoding)) {
case PCM_CODE:
/* We can understand this */
break;
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case IEEE_FLOAT_CODE:
IEEE_float_encoded = 1;
/* We can understand this */
break;
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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;
case MP3_CODE:
SDL_SetError("MPEG Layer 3 data not supported",
SDL_SwapLE16(format->encoding));
was_error = 1;
goto done;
default:
SDL_SetError("Unknown WAVE data format: 0x%.4x",
SDL_SwapLE16(format->encoding));
was_error = 1;
goto done;
}
SDL_memset(spec, 0, (sizeof *spec));
spec->freq = SDL_SwapLE32(format->frequency);
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if (IEEE_float_encoded) {
if ((SDL_SwapLE16(format->bitspersample)) != 32) {
was_error = 1;
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} else {
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spec->format = AUDIO_F32;
}
} else {
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;
case 32:
spec->format = AUDIO_S32;
break;
default:
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was_error = 1;
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break;
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}
}
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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) {
SDL_free(*audio_buf);
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*audio_buf = NULL;
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}
lenread = ReadChunk(src, &chunk);
if (lenread < 0) {
was_error = 1;
goto done;
}
*audio_len = lenread;
*audio_buf = chunk.data;
if (chunk.magic != DATA)
headerDiff += lenread + 2 * sizeof(Uint32);
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} while (chunk.magic != DATA);
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headerDiff += 2 * sizeof(Uint32); /* for the data chunk and len */
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 */
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samplesize = ((SDL_AUDIO_BITSIZE(spec->format)) / 8) * spec->channels;
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*audio_len &= ~(samplesize - 1);
done:
if (format != NULL) {
SDL_free(format);
}
if (src) {
if (freesrc) {
SDL_RWclose(src);
} else {
/* seek to the end of the file (given by the RIFF chunk) */
SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR);
}
}
if (was_error) {
spec = NULL;
}
return (spec);
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608
}
/* Since the WAV memory is allocated in the shared library, it must also
be freed here. (Necessary under Win32, VC++)
*/
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void
SDL_FreeWAV(Uint8 * audio_buf)
611
{
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if (audio_buf != NULL) {
SDL_free(audio_buf);
}
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}
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static int
ReadChunk(SDL_RWops * src, Chunk * chunk)
619
{
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622
623
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629
chunk->magic = SDL_ReadLE32(src);
chunk->length = SDL_ReadLE32(src);
chunk->data = (Uint8 *) SDL_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);
SDL_free(chunk->data);
630
chunk->data = NULL;
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return (-1);
}
return (chunk->length);
634
}
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/* vi: set ts=4 sw=4 expandtab: */