/*****************************************************************************
* adpcm.c : adpcm variant audio decoder
*****************************************************************************
* Copyright (C) 2001, 2002 VLC authors and VideoLAN
*
* Authors: Laurent Aimar <fenrir@via.ecp.fr>
* Rémi Denis-Courmont
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU Lesser General Public License as published by
* the Free Software Foundation; either version 2.1 of the License, or
* (at your option) any later version.
*
* This program 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 Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with this program; if not, write to the Free Software Foundation,
* Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*
* Documentation: http://www.pcisys.net/~melanson/codecs/adpcm.txt
*****************************************************************************/
#ifdef HAVE_CONFIG_H
# include
"config.h"
#endif
#include <vlc_common.h>
#include <vlc_plugin.h>
#include <vlc_codec.h>
/*****************************************************************************
* Module descriptor
*****************************************************************************/
static int OpenDecoder( vlc_object_t * );
static void CloseDecoder( vlc_object_t * );
static int DecodeAudio( decoder_t *, block_t * );
static void Flush( decoder_t * );
vlc_module_begin ()
set_description( N_(
"ADPCM audio decoder") )
set_capability(
"audio decoder",
50 )
set_subcategory( SUBCAT_INPUT_ACODEC )
set_callbacks( OpenDecoder, CloseDecoder )
vlc_module_end ()
/*****************************************************************************
* Local prototypes
*****************************************************************************/
enum adpcm_codec_e
{
ADPCM_IMA_QT,
ADPCM_IMA_WAV,
ADPCM_MS,
ADPCM_DK3,
ADPCM_DK4,
ADPCM_EA
};
typedef struct
{
enum adpcm_codec_e codec;
size_t i_block;
size_t i_samplesperblock;
date_t end_date;
int16_t *prev;
} decoder_sys_t;
static void DecodeAdpcmMs ( decoder_t *, int16_t *, uint8_t * );
static void DecodeAdpcmImaWav( decoder_t *, int16_t *, uint8_t * );
static void DecodeAdpcmImaQT ( decoder_t *, int16_t *, uint8_t * );
static void DecodeAdpcmDk4 ( decoder_t *, int16_t *, uint8_t * );
static void DecodeAdpcmDk3 ( decoder_t *, int16_t *, uint8_t * );
static void DecodeAdpcmEA ( decoder_t *, int16_t *, uint8_t * );
/* Various table from http://www.pcisys.net/~melanson/codecs/adpcm.txt */
static const int i_index_table[
16] =
{
-
1, -
1, -
1, -
1,
2,
4,
6,
8,
-
1, -
1, -
1, -
1,
2,
4,
6,
8
};
static const int i_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
};
static const int i_adaptation_table[
16] =
{
230,
230,
230,
230,
307,
409,
512,
614,
768,
614,
512,
409,
307,
230,
230,
230
};
static const int i_adaptation_coeff1[
7] =
{
256,
512,
0,
192,
240,
460,
392
};
static const int i_adaptation_coeff2[
7] =
{
0, -
256,
0,
64,
0, -
208, -
232
};
/*****************************************************************************
* OpenDecoder: probe the decoder and return score
*****************************************************************************/
static int OpenDecoder( vlc_object_t *p_this )
{
decoder_t *p_dec = (decoder_t*)p_this;
decoder_sys_t *p_sys;
switch( p_dec->fmt_in->i_codec )
{
case VLC_CODEC_ADPCM_IMA_QT:
case VLC_CODEC_ADPCM_IMA_WAV:
case VLC_CODEC_ADPCM_MS:
case VLC_CODEC_ADPCM_DK4:
case VLC_CODEC_ADPCM_DK3:
case VLC_CODEC_ADPCM_XA_EA:
break;
default:
return VLC_EGENERIC;
}
if( p_dec->fmt_in->audio.i_rate <=
0 )
{
msg_Err( p_dec,
"bad samplerate" );
return VLC_EGENERIC;
}
/* Allocate the memory needed to store the decoder's structure */
p_sys = malloc(
sizeof(*p_sys));
if( unlikely(p_sys == NULL) )
return VLC_ENOMEM;
p_sys->prev = NULL;
p_sys->i_samplesperblock =
0;
unsigned i_channels = p_dec->fmt_in->audio.i_channels;
uint8_t i_max_channels =
5;
switch( p_dec->fmt_in->i_codec )
{
case VLC_CODEC_ADPCM_IMA_QT:
/* IMA ADPCM */
p_sys->codec = ADPCM_IMA_QT;
i_max_channels =
2;
break;
case VLC_CODEC_ADPCM_IMA_WAV:
/* IMA ADPCM */
p_sys->codec = ADPCM_IMA_WAV;
i_max_channels =
2;
break;
case VLC_CODEC_ADPCM_MS:
/* MS ADPCM */
p_sys->codec = ADPCM_MS;
i_max_channels =
2;
break;
case VLC_CODEC_ADPCM_DK4:
/* Duck DK4 ADPCM */
p_sys->codec = ADPCM_DK4;
i_max_channels =
2;
break;
case VLC_CODEC_ADPCM_DK3:
/* Duck DK3 ADPCM */
p_sys->codec = ADPCM_DK3;
i_max_channels =
2;
break;
case VLC_CODEC_ADPCM_XA_EA:
/* EA ADPCM */
p_sys->codec = ADPCM_EA;
p_sys->prev = calloc(
2 * p_dec->fmt_in->audio.i_channels,
sizeof( int16_t ) );
if( unlikely(p_sys->prev == NULL) )
{
free( p_sys );
return VLC_ENOMEM;
}
break;
}
if (i_channels > i_max_channels || i_channels ==
0)
{
free(p_sys->prev);
free(p_sys);
msg_Err( p_dec,
"Invalid number of channels %i", p_dec->fmt_in->audio.i_channels );
return VLC_EGENERIC;
}
if( p_dec->fmt_in->audio.i_blockalign <=
0 )
{
p_sys->i_block = (p_sys->codec == ADPCM_IMA_QT) ?
34 * p_dec->fmt_in->audio.i_channels :
1024;
msg_Warn( p_dec,
"block size undefined, using %zu", p_sys->i_block );
}
else
{
p_sys->i_block = p_dec->fmt_in->audio.i_blockalign;
}
/* calculate samples per block */
switch( p_sys->codec )
{
case ADPCM_IMA_QT:
if( p_sys->i_block >=
34 * i_channels )
p_sys->i_samplesperblock = (p_sys->i_block / (
34 * i_channels) ) *
64;
break;
case ADPCM_IMA_WAV:
/* 4 bytes block/channel preamble */
/* samples: 4 bytes aligned nibble per channel. 2 samples per byte */
if( i_channels ==
2 && p_sys->i_block >= ((
4+
4) *
2) )
{
size_t padding = p_sys->i_block %
8;
p_sys->i_samplesperblock =
2 * ( p_sys->i_block -
4 * i_channels - padding )
/ i_channels;
}
else if( i_channels ==
1 && p_sys->i_block >
4 )
{
p_sys->i_samplesperblock =
2 * ( p_sys->i_block -
4 )
/ i_channels;
}
break;
case ADPCM_MS:
/* 7 bytes block/channel preamble outputs 2 samples */
if( p_sys->i_block >=
7 * i_channels )
{
p_sys->i_samplesperblock =
2 * (p_sys->i_block -
7 * i_channels) / i_channels +
2;
}
break;
case ADPCM_DK4:
/* 4 bytes block/channel preamble outputs 1 sample */
if( p_sys->i_block >=
4 * i_channels )
{
p_sys->i_samplesperblock =
2 * (p_sys->i_block -
4 * i_channels) / i_channels +
1;
}
break;
case ADPCM_DK3:
/* 16 bytes block (2 channels) preamble outputs 1 sample */
/* 3 ADPCM nibbles decodes to 4 16-bit PCM samples. Only stereo */
i_channels =
2;
if( p_sys->i_block >
16 )
{
size_t i_payload = p_sys->i_block -
16;
p_sys->i_samplesperblock =
4 * ( i_payload /
3 );
if( i_payload %
3 ==
2 )
p_sys->i_samplesperblock +=
2;
}
break;
case ADPCM_EA:
if( p_sys->i_block > i_channels )
{
p_sys->i_samplesperblock =
2 * (p_sys->i_block - i_channels) / i_channels;
}
}
msg_Dbg( p_dec,
"format: samplerate:%d Hz channels:%d bits/sample:%d "
"blockalign:%zu samplesperblock:%zu",
p_dec->fmt_in->audio.i_rate, i_channels,
p_dec->fmt_in->audio.i_bitspersample, p_sys->i_block,
p_sys->i_samplesperblock );
if (p_sys->i_samplesperblock ==
0)
{
free(p_sys->prev);
free(p_sys);
msg_Err( p_dec,
"Error computing number of samples per block");
return VLC_EGENERIC;
}
p_dec->p_sys = p_sys;
p_dec->fmt_out.i_codec = VLC_CODEC_S16N;
p_dec->fmt_out.audio.i_rate = p_dec->fmt_in->audio.i_rate;
p_dec->fmt_out.audio.i_channels = i_channels;
p_dec->fmt_out.audio.i_physical_channels = vlc_chan_maps[i_channels];
date_Init( &p_sys->end_date, p_dec->fmt_out.audio.i_rate,
1 );
p_dec->pf_decode = DecodeAudio;
p_dec->pf_flush = Flush;
return VLC_SUCCESS;
}
/*****************************************************************************
* Flush:
*****************************************************************************/
static void Flush( decoder_t *p_dec )
{
decoder_sys_t *p_sys = p_dec->p_sys;
date_Set( &p_sys->end_date, VLC_TICK_INVALID );
}
/*****************************************************************************
* DecodeBlock:
*****************************************************************************/
static block_t *DecodeBlock( decoder_t *p_dec, block_t **pp_block )
{
decoder_sys_t *p_sys = p_dec->p_sys;
block_t *p_block;
if( !*pp_block )
return NULL;
p_block = *pp_block;
if( p_block->i_flags & (BLOCK_FLAG_DISCONTINUITY|BLOCK_FLAG_CORRUPTED) )
{
Flush( p_dec );
if( p_block->i_flags & BLOCK_FLAG_CORRUPTED )
goto drop;
}
if( p_block->i_pts != VLC_TICK_INVALID &&
p_block->i_pts != date_Get( &p_sys->end_date ) )
{
date_Set( &p_sys->end_date, p_block->i_pts );
}
else if( date_Get( &p_sys->end_date ) == VLC_TICK_INVALID )
/* We've just started the stream, wait for the first PTS. */
goto drop;
/* Don't re-use the same pts twice */
p_block->i_pts = VLC_TICK_INVALID;
if( p_block->i_buffer >= p_sys->i_block )
{
block_t *p_out;
if( decoder_UpdateAudioFormat( p_dec ) )
goto drop;
p_out = decoder_NewAudioBuffer( p_dec, p_sys->i_samplesperblock );
if( p_out == NULL )
goto drop;
p_out->i_pts = date_Get( &p_sys->end_date );
p_out->i_length = date_Increment( &p_sys->end_date,
p_sys->i_samplesperblock ) - p_out->i_pts;
switch( p_sys->codec )
{
case ADPCM_IMA_QT:
DecodeAdpcmImaQT( p_dec, (int16_t*)p_out->p_buffer,
p_block->p_buffer );
break;
case ADPCM_IMA_WAV:
DecodeAdpcmImaWav( p_dec, (int16_t*)p_out->p_buffer,
p_block->p_buffer );
break;
case ADPCM_MS:
DecodeAdpcmMs( p_dec, (int16_t*)p_out->p_buffer,
p_block->p_buffer );
break;
case ADPCM_DK4:
DecodeAdpcmDk4( p_dec, (int16_t*)p_out->p_buffer,
p_block->p_buffer );
break;
case ADPCM_DK3:
DecodeAdpcmDk3( p_dec, (int16_t*)p_out->p_buffer,
p_block->p_buffer );
break;
case ADPCM_EA:
DecodeAdpcmEA( p_dec, (int16_t*)p_out->p_buffer,
p_block->p_buffer );
default:
break;
}
p_block->p_buffer += p_sys->i_block;
p_block->i_buffer -= p_sys->i_block;
return p_out;
}
drop:
block_Release( p_block );
*pp_block = NULL;
return NULL;
}
static int DecodeAudio( decoder_t *p_dec, block_t *p_block )
{
if( p_block == NULL )
/* No Drain */
return VLCDEC_SUCCESS;
block_t **pp_block = &p_block, *p_out;
while( ( p_out = DecodeBlock( p_dec, pp_block ) ) != NULL )
decoder_QueueAudio( p_dec, p_out );
return VLCDEC_SUCCESS;
}
/*****************************************************************************
* CloseDecoder:
*****************************************************************************/
static void CloseDecoder( vlc_object_t *p_this )
{
decoder_t *p_dec = (decoder_t *)p_this;
decoder_sys_t *p_sys = p_dec->p_sys;
free( p_sys->prev );
free( p_sys );
}
/*****************************************************************************
* Local functions
*****************************************************************************/
#define CLAMP( v, min, max ) \
if( (v) < (min) ) (v) = (min); \
if( (v) > (max) ) (v) = (max)
#define GetByte( v ) \
(v) = *p_buffer; p_buffer++;
#define GetWord( v ) \
(v) = *p_buffer; p_buffer++; \
(v) |= ( *p_buffer ) <<
8; p_buffer++; \
if( (v)&
0x8000 ) (v) -=
0x010000;
/*
* MS
*/
typedef struct adpcm_ms_channel_s
{
int i_idelta;
int i_sample1, i_sample2;
int i_coeff1, i_coeff2;
} adpcm_ms_channel_t;
static int AdpcmMsExpandNibble(adpcm_ms_channel_t *p_channel,
int i_nibble )
{
int i_predictor;
int i_snibble;
/* expand sign */
i_snibble = i_nibble - ( i_nibble&
0x08 ?
0x10 :
0 );
i_predictor = ( p_channel->i_sample1 * p_channel->i_coeff1 +
p_channel->i_sample2 * p_channel->i_coeff2 ) /
256 +
i_snibble * p_channel->i_idelta;
CLAMP( i_predictor, -
32768,
32767 );
p_channel->i_sample2 = p_channel->i_sample1;
p_channel->i_sample1 = i_predictor;
p_channel->i_idelta = ( i_adaptation_table[i_nibble] *
p_channel->i_idelta ) /
256;
if( p_channel->i_idelta <
16 )
{
p_channel->i_idelta =
16;
}
return( i_predictor );
}
static void DecodeAdpcmMs( decoder_t *p_dec, int16_t *p_sample,
uint8_t *p_buffer )
{
decoder_sys_t *p_sys = p_dec->p_sys;
adpcm_ms_channel_t channel[
2];
int b_stereo;
int i_block_predictor;
size_t i_total_samples = p_sys->i_samplesperblock;
if(i_total_samples <
2)
return;
b_stereo = p_dec->fmt_out.audio.i_channels ==
2 ?
1 :
0;
GetByte( i_block_predictor );
CLAMP( i_block_predictor,
0,
6 );
channel[
0].i_coeff1 = i_adaptation_coeff1[i_block_predictor];
channel[
0].i_coeff2 = i_adaptation_coeff2[i_block_predictor];
if( b_stereo )
{
GetByte( i_block_predictor );
CLAMP( i_block_predictor,
0,
6 );
channel[
1].i_coeff1 = i_adaptation_coeff1[i_block_predictor];
channel[
1].i_coeff2 = i_adaptation_coeff2[i_block_predictor];
}
GetWord( channel[
0].i_idelta );
if( b_stereo )
{
GetWord( channel[
1].i_idelta );
}
GetWord( channel[
0].i_sample1 );
if( b_stereo )
{
GetWord( channel[
1].i_sample1 );
}
GetWord( channel[
0].i_sample2 );
if( b_stereo )
{
GetWord( channel[
1].i_sample2 );
}
if( b_stereo )
{
*p_sample++ = channel[
0].i_sample2;
*p_sample++ = channel[
1].i_sample2;
*p_sample++ = channel[
0].i_sample1;
*p_sample++ = channel[
1].i_sample1;
}
else
{
*p_sample++ = channel[
0].i_sample2;
*p_sample++ = channel[
0].i_sample1;
}
for( i_total_samples -=
2; i_total_samples >=
2; i_total_samples -=
2, p_buffer++ )
{
*p_sample++ = AdpcmMsExpandNibble( &channel[
0], (*p_buffer) >>
4);
*p_sample++ = AdpcmMsExpandNibble( &channel[b_stereo ?
1 :
0],
(*p_buffer)&
0x0f);
}
}
/*
* IMA-WAV
*/
typedef struct adpcm_ima_wav_channel_s
{
int i_predictor;
int i_step_index;
} adpcm_ima_wav_channel_t;
static int AdpcmImaWavExpandNibble(adpcm_ima_wav_channel_t *p_channel,
int i_nibble )
{
int i_diff;
i_diff = i_step_table[p_channel->i_step_index] >>
3;
if( i_nibble&
0x04 ) i_diff += i_step_table[p_channel->i_step_
index];
if( i_nibble&0x02 ) i_diff += i_step_table[p_channel->i_step_index]>>1;
if( i_nibble&0x01 ) i_diff += i_step_table[p_channel->i_step_index]>>2;
if( i_nibble&0x08 )
p_channel->i_predictor -= i_diff;
else
p_channel->i_predictor += i_diff;
CLAMP( p_channel->i_predictor, -32768, 32767 );
p_channel->i_step_index += i_index_table[i_nibble];
CLAMP( p_channel->i_step_index, 0, 88 );
return( p_channel->i_predictor );
}
static void DecodeAdpcmImaWav( decoder_t *p_dec, int16_t *p_sample,
uint8_t *p_buffer )
{
decoder_sys_t *p_sys = p_dec->p_sys;
adpcm_ima_wav_channel_t channel[2];
int i_nibbles;
int b_stereo;
b_stereo = p_dec->fmt_out.audio.i_channels == 2 ? 1 : 0;
GetWord( channel[0].i_predictor );
GetByte( channel[0].i_step_index );
CLAMP( channel[0].i_step_index, 0, 88 );
p_buffer++;
if( b_stereo )
{
GetWord( channel[1].i_predictor );
GetByte( channel[1].i_step_index );
CLAMP( channel[1].i_step_index, 0, 88 );
p_buffer++;
}
if( b_stereo )
{
for( i_nibbles = 2 * (p_sys->i_block - 8);
i_nibbles > 15;
i_nibbles -= 16 )
{
int i;
for( i = 0; i < 4; i++ )
{
p_sample[i * 4] =
AdpcmImaWavExpandNibble(&channel[0],p_buffer[i]&0x0f);
p_sample[i * 4 + 2] =
AdpcmImaWavExpandNibble(&channel[0],p_buffer[i] >> 4);
}
p_buffer += 4;
for( i = 0; i < 4; i++ )
{
p_sample[i * 4 + 1] =
AdpcmImaWavExpandNibble(&channel[1],p_buffer[i]&0x0f);
p_sample[i * 4 + 3] =
AdpcmImaWavExpandNibble(&channel[1],p_buffer[i] >> 4);
}
p_buffer += 4;
p_sample += 16;
}
}
else
{
for( i_nibbles = 2 * (p_sys->i_block - 4);
i_nibbles > 0;
i_nibbles -= 2, p_buffer++ )
{
*p_sample++ =AdpcmImaWavExpandNibble( &channel[0], (*p_buffer)& style='color: green'>0x0f );
*p_sample++ =AdpcmImaWavExpandNibble( &channel[0], (*p_buffer) >> 4 );
}
}
}
/*
* Ima4 in QT file
*/
static void DecodeAdpcmImaQT( decoder_t *p_dec, int16_t *p_sample,
uint8_t *p_buffer )
{
decoder_sys_t *p_sys = p_dec->p_sys;
adpcm_ima_wav_channel_t channel[2];
int i_nibbles;
int i_ch;
int i_chans = p_dec->fmt_out.audio.i_channels;
for( unsigned i=0; i<p_sys->i_block/(34 * i_chans); i++)
{
for( i_ch = 0; i_ch < p_dec->fmt_out.audio.i_channels; i_ch++ )
{
int16_t *p_out = &p_sample[i_ch];
/* load preamble */
channel[i_ch].i_predictor = (int16_t)((( ( p_buffer[0] << 1 )|( p_buffer[1] >> 7 ) ))<<7);
channel[i_ch].i_step_index = p_buffer[1]&0x7f;
CLAMP( channel[i_ch].i_step_index, 0, 88 );
p_buffer += 2;
for( i_nibbles = 0; i_nibbles < 64; i_nibbles +=2 )
{
*p_out = AdpcmImaWavExpandNibble( &channel[i_ch], (*p_buffer)&0x0f);
p_out += i_chans;
*p_out = AdpcmImaWavExpandNibble( &channel[i_ch], (*p_buffer >> 4)&0x0f);
p_out += i_chans;
p_buffer++;
}
}
p_sample += 64 * i_chans;
}
}
/*
* Dk4
*/
static void DecodeAdpcmDk4( decoder_t *p_dec, int16_t *p_sample,
uint8_t *p_buffer )
{
decoder_sys_t *p_sys = p_dec->p_sys;
adpcm_ima_wav_channel_t channel[2];
size_t i_nibbles;
int b_stereo;
b_stereo = p_dec->fmt_out.audio.i_channels == 2 ? 1 : 0;
GetWord( channel[0].i_predictor );
GetByte( channel[0].i_step_index );
CLAMP( channel[0].i_step_index, 0, 88 );
p_buffer++;
if( b_stereo )
{
GetWord( channel[1].i_predictor );
GetByte( channel[1].i_step_index );
CLAMP( channel[1].i_step_index, 0, 88 );
p_buffer++;
}
/* first output predictor */
*p_sample++ = channel[0].i_predictor;
if( b_stereo )
{
*p_sample++ = channel[1].i_predictor;
}
for( i_nibbles = 0;
i_nibbles < p_sys->i_block - 4 * (b_stereo ? 2:1 );
i_nibbles++ )
{
*p_sample++ = AdpcmImaWavExpandNibble( &channel[0],
(*p_buffer) >> 4);
*p_sample++ = AdpcmImaWavExpandNibble( &channel[b_stereo ? 1 : 0],
(*p_buffer)&0x0f);
p_buffer++;
}
}
/*
* Dk3
*/
static void DecodeAdpcmDk3( decoder_t *p_dec, int16_t *p_sample,
uint8_t *p_buffer )
{
decoder_sys_t *p_sys = p_dec->p_sys;
uint8_t *p_end = &p_buffer[p_sys->i_block];
adpcm_ima_wav_channel_t sum;
adpcm_ima_wav_channel_t diff;
int i_diff_value;
p_buffer += 10;
GetWord( sum.i_predictor );
GetWord( diff.i_predictor );
GetByte( sum.i_step_index );
GetByte( diff.i_step_index );
CLAMP( sum.i_step_index, 0, 88 );
CLAMP( diff.i_step_index, 0, 88 );
i_diff_value = diff.i_predictor;
/* we process 6 nibbles at once */
while( p_buffer + 1 < p_end )
{
/* first 3 nibbles */
AdpcmImaWavExpandNibble( &sum,
(*p_buffer)&0x0f);
AdpcmImaWavExpandNibble( &diff,
(*p_buffer) >> 4 );
i_diff_value = ( i_diff_value + diff.i_predictor ) / 2;
*p_sample++ = sum.i_predictor + i_diff_value;
*p_sample++ = sum.i_predictor - i_diff_value;
p_buffer++;
AdpcmImaWavExpandNibble( &sum,
(*p_buffer)&0x0f);
*p_sample++ = sum.i_predictor + i_diff_value;
*p_sample++ = sum.i_predictor - i_diff_value;
/* now last 3 nibbles */
AdpcmImaWavExpandNibble( &sum,
(*p_buffer)>>4);
p_buffer++;
if( p_buffer < p_end )
{
AdpcmImaWavExpandNibble( &diff,
(*p_buffer)&0x0f );
i_diff_value = ( i_diff_value + diff.i_predictor ) / 2;
*p_sample++ = sum.i_predictor + i_diff_value;
*p_sample++ = sum.i_predictor - i_diff_value;
AdpcmImaWavExpandNibble( &sum,
(*p_buffer)>>4);
p_buffer++;
*p_sample++ = sum.i_predictor + i_diff_value;
*p_sample++ = sum.i_predictor - i_diff_value;
}
}
}
/*
* EA ADPCM
*/
#define MAX_CHAN 5
static void DecodeAdpcmEA( decoder_t *p_dec, int16_t *p_sample,
uint8_t *p_buffer )
{
static const int16_t EATable[]=
{
0x0000, 0x00F0, 0x01CC, 0x0188, 0x0000, 0x0000, 0xFF30, 0xFF24,
0x0000, 0x0001, 0x0003, 0x0004, 0x0007, 0x0008, 0x000A, 0x000B,
0x0000, 0xFFFF, 0xFFFD, 0xFFFC,
};
decoder_sys_t *p_sys = p_dec->p_sys;
int_fast32_t c1[MAX_CHAN], c2[MAX_CHAN];
int_fast8_t d[MAX_CHAN];
unsigned chans = p_dec->fmt_out.audio.i_channels;
const uint8_t *p_end = &p_buffer[p_sys->i_block];
int16_t *prev = p_sys->prev;
int16_t *cur = prev + chans;
for (unsigned c = 0; c < chans; c++)
{
uint8_t input = p_buffer[c];
c1[c] = EATable[input >> 4];
c2[c] = EATable[(input >> 4) + 4];
d[c] = (input & 0xf) + 8;
}
for (p_buffer += chans; p_buffer + chans <= p_end; p_buffer += chans)
{
union { uint32_t u; int32_t i; } spl;
for (unsigned c = 0; c < chans; c++)
{
spl.u = (p_buffer[c] & 0xf0u) << 24u;
spl.i >>= d[c];
spl.i = (spl.i + cur[c] * c1[c] + prev[c] * c2[c] + 0x80) >> 8;
CLAMP(spl.i, -32768, 32767);
prev[c] = cur[c];
cur[c] = spl.i;
*(p_sample++) = spl.i;
}
for (unsigned c = 0; c < chans; c++)
{
spl.u = (p_buffer[c] & 0x0fu) << 28u;
spl.i >>= d[c];
spl.i = (spl.i + cur[c] * c1[c] + prev[c] * c2[c] + 0x80) >> 8;
CLAMP(spl.i, -32768, 32767);
prev[c] = cur[c];
cur[c] = spl.i;
*(p_sample++) = spl.i;
}
}
}