/*****************************************************************************

 * 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)&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;

        }

    }

}