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/*
* MS ADPCM decoder
*
* This file is responsible for decoding Microsoft ADPCM data.
* Details about the data format can be found here:
* http://www.pcisys.net/~melanson/codecs/
*
* Copyright (c) 2002 Mike Melanson
*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include "config.h"
#include "libavutil/common.h"
#include "ffmpeg_files/intreadwrite.h"
#include "mpbswap.h"
#include "ad_internal.h"
static const ad_info_t info =
{
"MS ADPCM audio decoder",
"msadpcm",
"Nick Kurshev",
"Mike Melanson",
""
};
LIBAD_EXTERN(msadpcm)
static const int ms_adapt_table[] =
{
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
static const uint8_t ms_adapt_coeff1[] =
{
64, 128, 0, 48, 60, 115, 98
};
static const int8_t ms_adapt_coeff2[] =
{
0, -64, 0, 16, 0, -52, -58
};
#define MS_ADPCM_PREAMBLE_SIZE 6
#define LE_16(x) ((int16_t)AV_RL16(x))
// clamp a number between 0 and 88
#define CLAMP_0_TO_88(x) x = av_clip(x, 0, 88);
// clamp a number within a signed 16-bit range
#define CLAMP_S16(x) x = av_clip_int16(x);
// clamp a number above 16
#define CLAMP_ABOVE_16(x) if (x < 16) x = 16;
// 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;
sh_audio->samplesize=2;
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 inline int check_coeff(uint8_t c) {
if (c > 6) {
mp_msg(MSGT_DECAUDIO, MSGL_WARN,
"MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
c);
c = 6;
}
return c;
}
static int ms_adpcm_decode_block(unsigned short *output, unsigned char *input,
int channels, int block_size)
{
int current_channel = 0;
int coeff_idx;
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;
if (channels != 1) channels = 2;
if (block_size < 7 * channels)
return -1;
// fetch the header information, in stereo if both channels are present
coeff_idx = check_coeff(input[stream_ptr]);
coeff1[0] = ms_adapt_coeff1[coeff_idx];
coeff2[0] = ms_adapt_coeff2[coeff_idx];
stream_ptr++;
if (channels == 2)
{
coeff_idx = check_coeff(input[stream_ptr]);
coeff1[1] = ms_adapt_coeff1[coeff_idx];
coeff2[1] = ms_adapt_coeff2[coeff_idx];
stream_ptr++;
}
idelta[0] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
if (channels == 2)
{
idelta[1] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
}
sample1[0] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
if (channels == 2)
{
sample1[1] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
}
sample2[0] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
if (channels == 2)
{
sample2[1] = LE_16(&input[stream_ptr]);
stream_ptr += 2;
}
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);
// should this really be a division and not a shift?
// coefficients were originally scaled by for, which might have
// been an optimization for 8-bit CPUs _if_ a shift is correct
predictor = (
((sample1[current_channel] * coeff1[current_channel]) +
(sample2[current_channel] * coeff2[current_channel])) / 64) +
(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)
{
int res;
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 */
res = ms_adpcm_decode_block(
(unsigned short*)buf, sh_audio->a_in_buffer,
sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign);
return res < 0 ? res : 2 * res;
}
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