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/*
MS ADPCM Decoder for MPlayer
by Mike Melanson
This file is responsible for decoding Microsoft ADPCM data.
Details about the data format can be found here:
http://www.pcisys.net/~melanson/codecs/
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include "config.h"
#include "bswap.h"
#include "ad_internal.h"
static ad_info_t info =
{
"MS ADPCM audio decoder",
"msadpcm",
AFM_MSADPCM,
"Nick Kurshev",
"Mike Melanson",
""
};
LIBAD_EXTERN(msadpcm)
static int ms_adapt_table[] =
{
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
static int ms_adapt_coeff1[] =
{
256, 512, 0, 192, 240, 460, 392
};
static int ms_adapt_coeff2[] =
{
0, -256, 0, 64, 0, -208, -232
};
#define MS_ADPCM_PREAMBLE_SIZE 6
#define LE_16(x) (le2me_16(*(unsigned short *)(x)))
#define LE_32(x) (le2me_32(*(unsigned int *)(x)))
// useful macros
// clamp a number between 0 and 88
#define CLAMP_0_TO_88(x) if (x < 0) x = 0; else if (x > 88) x = 88;
// clamp a number within a signed 16-bit range
#define CLAMP_S16(x) if (x < -32768) x = -32768; \
else if (x > 32767) x = 32767;
// clamp a number above 16
#define CLAMP_ABOVE_16(x) if (x < 16) x = 16;
// sign extend a 16-bit value
#define SE_16BIT(x) if (x & 0x8000) x -= 0x10000;
// sign extend a 4-bit value
#define SE_4BIT(x) if (x & 0x8) x -= 0x10;
static int preinit(sh_audio_t *sh_audio)
{
sh_audio->audio_out_minsize = sh_audio->wf->nBlockAlign * 4;
sh_audio->ds->ss_div =
(sh_audio->wf->nBlockAlign - MS_ADPCM_PREAMBLE_SIZE) * 2;
sh_audio->audio_in_minsize =
sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign;
return 1;
}
static int init(sh_audio_t *sh_audio)
{
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
sh_audio->i_bps = sh_audio->wf->nBlockAlign *
(sh_audio->channels*sh_audio->samplerate) / sh_audio->ds->ss_div;
return 1;
}
static void uninit(sh_audio_t *sh_audio)
{
}
static int control(sh_audio_t *sh_audio,int cmd,void* arg, ...)
{
if(cmd==ADCTRL_SKIP_FRAME){
demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,sh_audio->ds->ss_mul);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static int ms_adpcm_decode_block(unsigned short *output, unsigned char *input,
int channels, int block_size)
{
int current_channel = 0;
int idelta[2];
int sample1[2];
int sample2[2];
int coeff1[2];
int coeff2[2];
int stream_ptr = 0;
int out_ptr = 0;
int upper_nibble = 1;
int nibble;
int snibble; // signed nibble
int predictor;
// fetch the header information, in stereo if both channels are present
if (input[stream_ptr] > 6)
mp_msg(MSGT_DECAUDIO, MSGL_WARN,
"MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
input[stream_ptr]);
coeff1[0] = ms_adapt_coeff1[input[stream_ptr]];
coeff2[0] = ms_adapt_coeff2[input[stream_ptr]];
stream_ptr++;
if (channels == 2)
{
if (input[stream_ptr] > 6)
mp_msg(MSGT_DECAUDIO, MSGL_WARN,
"MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
input[stream_ptr]);
coeff1[1] = ms_adapt_coeff1[input[stream_ptr]];
coeff2[1] = ms_adapt_coeff2[input[stream_ptr]];
stream_ptr++;
}
idelta[0] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
SE_16BIT(idelta[0]);
if (channels == 2)
{
idelta[1] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
SE_16BIT(idelta[1]);
}
sample1[0] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
SE_16BIT(sample1[0]);
if (channels == 2)
{
sample1[1] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
SE_16BIT(sample1[1]);
}
sample2[0] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
SE_16BIT(sample2[0]);
if (channels == 2)
{
sample2[1] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
SE_16BIT(sample2[1]);
}
if (channels == 1)
{
output[out_ptr++] = sample2[0];
output[out_ptr++] = sample1[0];
} else {
output[out_ptr++] = sample2[0];
output[out_ptr++] = sample2[1];
output[out_ptr++] = sample1[0];
output[out_ptr++] = sample1[1];
}
while (stream_ptr < block_size)
{
// get the next nibble
if (upper_nibble)
nibble = snibble = input[stream_ptr] >> 4;
else
nibble = snibble = input[stream_ptr++] & 0x0F;
upper_nibble ^= 1;
SE_4BIT(snibble);
predictor = (
((sample1[current_channel] * coeff1[current_channel]) +
(sample2[current_channel] * coeff2[current_channel])) / 256) +
(snibble * idelta[current_channel]);
CLAMP_S16(predictor);
sample2[current_channel] = sample1[current_channel];
sample1[current_channel] = predictor;
output[out_ptr++] = predictor;
// compute the next adaptive scale factor (a.k.a. the variable idelta)
idelta[current_channel] =
(ms_adapt_table[nibble] * idelta[current_channel]) / 256;
CLAMP_ABOVE_16(idelta[current_channel]);
// toggle the channel
current_channel ^= channels - 1;
}
return (block_size - (MS_ADPCM_PREAMBLE_SIZE * channels)) * 2;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
if (demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,
sh_audio->ds->ss_mul) !=
sh_audio->ds->ss_mul)
return -1; /* EOF */
return 2 * ms_adpcm_decode_block(
(unsigned short*)buf, sh_audio->a_in_buffer,
sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign);
}
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