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ADPCM decoder seems to be complete, other fixes too...
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@icculus
icculus committed Dec 27, 2001
1 parent 8a39608 commit 46692f7
Showing 1 changed file with 185 additions and 47 deletions.
232 changes: 185 additions & 47 deletions decoders/wav.c
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Expand Up @@ -101,6 +101,7 @@ static inline int read_uint8(SDL_RWops *rw, Uint8 *ui8)

#define riffID 0x46464952 /* "RIFF", in ascii. */
#define waveID 0x45564157 /* "WAVE", in ascii. */
#define factID 0x74636166 /* "fact", in ascii. */


/*****************************************************************************
Expand All @@ -114,8 +115,8 @@ static inline int read_uint8(SDL_RWops *rw, Uint8 *ui8)

typedef struct
{
Uint16 iCoef1;
Uint16 iCoef2;
Sint16 iCoef1;
Sint16 iCoef2;
} ADPCMCOEFSET;

typedef struct
Expand All @@ -137,6 +138,8 @@ typedef struct S_WAV_FMT_T
Uint16 wBlockAlign;
Uint16 wBitsPerSample;

Uint32 sample_frame_size;

void (*free)(struct S_WAV_FMT_T *fmt);
Uint32 (*read_sample)(Sound_Sample *sample);

Expand All @@ -149,6 +152,9 @@ typedef struct S_WAV_FMT_T
Uint16 wNumCoef;
ADPCMCOEFSET *aCoef;
ADPCMBLOCKHEADER *blockheaders;
Uint32 samples_left_in_block;
int nibble_state;
Sint8 nibble;
} adpcm;

/* put other format-specific data here... */
Expand Down Expand Up @@ -226,6 +232,9 @@ typedef struct
* Normal, uncompressed waveform handler... *
*****************************************************************************/

/*
* Sound_Decode() lands here for uncompressed WAVs...
*/
static Uint32 read_sample_fmt_normal(Sound_Sample *sample)
{
Uint32 retval;
Expand Down Expand Up @@ -274,12 +283,29 @@ static int read_fmt_normal(SDL_RWops *rw, fmt_t *fmt)
* ADPCM compression handler... *
*****************************************************************************/

static int read_adpcm_block_headers(SDL_RWops *rw, fmt_t *fmt)
#define FIXED_POINT_COEF_BASE 256
#define FIXED_POINT_ADAPTION_BASE 256
#define SMALLEST_ADPCM_DELTA 16


static inline int read_adpcm_block_headers(Sound_Sample *sample)
{
Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
SDL_RWops *rw = internal->rw;
wav_t *w = (wav_t *) internal->decoder_private;
fmt_t *fmt = w->fmt;
ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders;
int i;
int max = fmt->wChannels;

if (w->bytesLeft < fmt->wBlockAlign)
{
sample->flags |= SOUND_SAMPLEFLAG_EOF;
return(0);
} /* if */

w->bytesLeft -= fmt->wBlockAlign;

for (i = 0; i < max; i++)
BAIL_IF_MACRO(!read_uint8(rw, &headers[i].bPredictor), NULL, 0);

Expand All @@ -292,48 +318,164 @@ static int read_adpcm_block_headers(SDL_RWops *rw, fmt_t *fmt)
for (i = 0; i < max; i++)
BAIL_IF_MACRO(!read_le16(rw, &headers[i].iSamp2), NULL, 0);

fmt->fmt.adpcm.samples_left_in_block = fmt->fmt.adpcm.wSamplesPerBlock;
fmt->fmt.adpcm.nibble_state = 0;
return(1);
} /* read_adpcm_block_headers */


static int decode_adpcm_block(Sound_Sample *sample)
static inline void do_adpcm_nibble(Uint8 nib,
ADPCMBLOCKHEADER *header,
Sint32 lPredSamp)
{
static const Sint32 max_audioval = ((1<<(16-1))-1);
static const Sint32 min_audioval = -(1<<(16-1));
static const Sint32 AdaptionTable[] =
{
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};

Sint32 lNewSamp;
Sint32 delta;

if (nib & 0x08)
lNewSamp = lPredSamp + (header->iDelta * (nib - 0x10));
else
lNewSamp = lPredSamp + (header->iDelta * nib);

/* clamp value... */
if (lNewSamp < min_audioval)
lNewSamp = min_audioval;
else if (lNewSamp > max_audioval)
lNewSamp = max_audioval;

delta = ((Sint32) header->iDelta * AdaptionTable[nib]) /
FIXED_POINT_ADAPTION_BASE;

if (delta < SMALLEST_ADPCM_DELTA)
delta = SMALLEST_ADPCM_DELTA;

header->iDelta = delta;
header->iSamp2 = header->iSamp1;
header->iSamp1 = lNewSamp;
} /* do_adpcm_nibble */


static inline int decode_adpcm_sample_frame(Sound_Sample *sample)
{
Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
SDL_RWops *rw = internal->rw;
wav_t *w = (wav_t *) internal->decoder_private;
fmt_t *fmt = w->fmt;
ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders;
SDL_RWops *rw = internal->rw;
int i;
int max = fmt->wChannels;

if (!read_adpcm_block_headers(rw, fmt))
{
sample->flags |= SOUND_SAMPLEFLAG_ERROR;
return(0);
} /* if */
Sint32 delta;
Uint8 nib = fmt->fmt.adpcm.nibble;

for (i = 0; i < max; i++)
{
Uint16 iCoef1 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef1;
Uint16 iCoef2 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef2;
/*
Uint8 byte;
Sint16 iCoef1 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef1;
Sint16 iCoef2 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef2;
Sint32 lPredSamp = ((headers[i].iSamp1 * iCoef1) +
(headers[i].iSamp2 * iCoef2)) /
FIXED_POINT_COEF_BASE;
*/
FIXED_POINT_COEF_BASE;

if (fmt->fmt.adpcm.nibble_state == 0)
{
BAIL_IF_MACRO(!read_uint8(rw, &nib), NULL, 0);
fmt->fmt.adpcm.nibble_state = 1;
do_adpcm_nibble(nib >> 4, &headers[i], lPredSamp);
} /* if */
else
{
fmt->fmt.adpcm.nibble_state = 0;
do_adpcm_nibble(nib & 0x0F, &headers[i], lPredSamp);
} /* else */
} /* for */
} /* decode_adpcm_block */

fmt->fmt.adpcm.nibble = nib;
return(1);
} /* decode_adpcm_sample_frame */


static inline void put_adpcm_sample_frame1(Uint8 *_buf, fmt_t *fmt)
{
Uint16 *buf = (Uint16 *) _buf;
ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders;
int i;
for (i = 0; i < fmt->wChannels; i++)
*(buf++) = headers[i].iSamp1;
} /* put_adpcm_sample_frame1 */


static inline void put_adpcm_sample_frame2(Uint8 *_buf, fmt_t *fmt)
{
Uint16 *buf = (Uint16 *) _buf;
ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders;
int i;
for (i = 0; i < fmt->wChannels; i++)
*(buf++) = headers[i].iSamp2;
} /* put_adpcm_sample_frame2 */


/*
* Sound_Decode() lands here for ADPCM-encoded WAVs...
*/
static Uint32 read_sample_fmt_adpcm(Sound_Sample *sample)
{
/* !!! FIXME: Write this. */
Sound_SetError("WAV: Not implemented. Soon.");
sample->flags |= SOUND_SAMPLEFLAG_ERROR;
return(0);
Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
wav_t *w = (wav_t *) internal->decoder_private;
fmt_t *fmt = w->fmt;
Uint32 bw = 0;

while (bw < internal->buffer_size)
{
/* write ongoing sample frame before reading more data... */
switch (fmt->fmt.adpcm.samples_left_in_block)
{
case 0: /* need to read a new block... */
if (!read_adpcm_block_headers(sample))
{
if ((sample->flags & SOUND_SAMPLEFLAG_EOF) == 0)
sample->flags |= SOUND_SAMPLEFLAG_ERROR;
return(bw);
} /* if */

/* only write first sample frame for now. */
put_adpcm_sample_frame2(internal->buffer + bw, fmt);
fmt->fmt.adpcm.samples_left_in_block--;
bw += fmt->sample_frame_size;
break;

case 1: /* output last sample frame of block... */
put_adpcm_sample_frame1(internal->buffer + bw, fmt);
fmt->fmt.adpcm.samples_left_in_block--;
bw += fmt->sample_frame_size;
break;

default: /* output latest sample frame and read a new one... */
put_adpcm_sample_frame1(internal->buffer + bw, fmt);
fmt->fmt.adpcm.samples_left_in_block--;
bw += fmt->sample_frame_size;

if (!decode_adpcm_sample_frame(sample))
{
sample->flags |= SOUND_SAMPLEFLAG_ERROR;
return(bw);
} /* if */
} /* switch */
} /* while */

return(bw);
} /* read_sample_fmt_adpcm */


/*
* Sound_FreeSample() lands here for ADPCM-encoded WAVs...
*/
static void free_fmt_adpcm(fmt_t *fmt)
{
if (fmt->fmt.adpcm.aCoef != NULL)
Expand Down Expand Up @@ -361,37 +503,22 @@ static int read_fmt_adpcm(SDL_RWops *rw, fmt_t *fmt)
BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.wSamplesPerBlock), NULL, 0);
BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.wNumCoef), NULL, 0);

/*
* !!! FIXME: SDL seems nervous about this, so I'll make a debug
* !!! FIXME: note for the time being...
*/
if (fmt->fmt.adpcm.wNumCoef != 7)
{
SNDDBG(("WAV: adpcm's wNumCoef is NOT seven (it's %d)!\n",
fmt->fmt.adpcm.wNumCoef));
} /* if */

/* fmt->free() is always called, so these malloc()s will be cleaned up. */

i = sizeof (ADPCMCOEFSET) * fmt->fmt.adpcm.wNumCoef;
fmt->fmt.adpcm.aCoef = (ADPCMCOEFSET *) malloc(i);
BAIL_IF_MACRO(fmt->fmt.adpcm.aCoef == NULL, ERR_OUT_OF_MEMORY, 0);

i = sizeof (ADPCMBLOCKHEADER) * fmt->wChannels;
fmt->fmt.adpcm.blockheaders = (ADPCMBLOCKHEADER *) malloc(i);
BAIL_IF_MACRO(fmt->fmt.adpcm.blockheaders == NULL, ERR_OUT_OF_MEMORY, 0);

for (i = 0; i < fmt->fmt.adpcm.wNumCoef; i++)
{
int rc;
rc = SDL_RWread(rw, &fmt->fmt.adpcm.aCoef[i].iCoef1,
sizeof (fmt->fmt.adpcm.aCoef[i].iCoef1), 1);
BAIL_IF_MACRO(rc != 1, ERR_IO_ERROR, 0);

rc = SDL_RWread(rw, &fmt->fmt.adpcm.aCoef[i].iCoef2,
sizeof (fmt->fmt.adpcm.aCoef[i].iCoef2), 1);
BAIL_IF_MACRO(rc != 1, ERR_IO_ERROR, 0);
BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.aCoef[i].iCoef1), NULL, 0);
BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.aCoef[i].iCoef2), NULL, 0);
} /* for */

i = sizeof (ADPCMBLOCKHEADER) * fmt->wChannels;
fmt->fmt.adpcm.blockheaders = (ADPCMBLOCKHEADER *) malloc(i);
BAIL_IF_MACRO(fmt->fmt.adpcm.blockheaders == NULL, ERR_OUT_OF_MEMORY, 0);

return(1);
} /* read_fmt_adpcm */

Expand Down Expand Up @@ -447,6 +574,7 @@ static int find_chunk(SDL_RWops *rw, Uint32 id)
{
Sint32 siz = 0;
Uint32 _id = 0;
Uint32 pos = SDL_RWtell(rw);

while (1)
{
Expand All @@ -456,8 +584,10 @@ static int find_chunk(SDL_RWops *rw, Uint32 id)

/* skip ahead and see what next chunk is... */
BAIL_IF_MACRO(!read_le32(rw, &siz), NULL, 0);
assert(siz > 0);
BAIL_IF_MACRO(SDL_RWseek(rw, siz, SEEK_SET) != siz, NULL, 0);
assert(siz >= 0);
pos += (sizeof (Uint32) * 2) + siz;
if (siz > 0)
BAIL_IF_MACRO(SDL_RWseek(rw, pos, SEEK_SET) != pos, NULL, 0);
} /* while */

return(0); /* shouldn't hit this, but just in case... */
Expand All @@ -470,6 +600,7 @@ static int WAV_open_internal(Sound_Sample *sample, const char *ext, fmt_t *fmt)
SDL_RWops *rw = internal->rw;
data_t d;
wav_t *w;
Uint32 pos;

BAIL_IF_MACRO(SDL_ReadLE32(rw) != riffID, "WAV: Not a RIFF file.", 0);
SDL_ReadLE32(rw); /* throw the length away; we get this info later. */
Expand All @@ -479,9 +610,11 @@ static int WAV_open_internal(Sound_Sample *sample, const char *ext, fmt_t *fmt)

sample->actual.channels = (Uint8) fmt->wChannels;
sample->actual.rate = fmt->dwSamplesPerSec;
if (fmt->wBitsPerSample <= 8)
if (fmt->wBitsPerSample == 4)
sample->actual.format = AUDIO_S16SYS; /* !!! FIXME ? */
else if (fmt->wBitsPerSample == 8)
sample->actual.format = AUDIO_U8;
else if (fmt->wBitsPerSample <= 16)
else if (fmt->wBitsPerSample == 16)
sample->actual.format = AUDIO_S16LSB;
else
BAIL_MACRO("WAV: Unsupported sample size.", 0);
Expand All @@ -494,6 +627,11 @@ static int WAV_open_internal(Sound_Sample *sample, const char *ext, fmt_t *fmt)
BAIL_IF_MACRO(w == NULL, ERR_OUT_OF_MEMORY, 0);
w->fmt = fmt;
w->bytesLeft = d.chunkSize;

/* !!! FIXME: Move this to Sound_SampleInfo ? */
fmt->sample_frame_size = ( ((sample->actual.format & 0xFF) / 8) *
sample->actual.channels );

internal->decoder_private = (void *) w;

sample->flags = SOUND_SAMPLEFLAG_NONE;
Expand Down

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