mirror of
https://code.videolan.org/videolan/vlc
synced 2024-10-03 01:31:53 +02:00
0549228196
Those multi-lines are still to be done (it's going to be fun...)
See also 5af4cc854b
.
794 lines
23 KiB
C
794 lines
23 KiB
C
/*****************************************************************************
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* adpcm.c : adpcm variant audio decoder
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*****************************************************************************
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* Copyright (C) 2001, 2002 the VideoLAN team
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* $Id$
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*
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* Authors: Laurent Aimar <fenrir@via.ecp.fr>
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* Rémi Denis-Courmont <rem # videolan.org>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
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*****************************************************************************/
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/*****************************************************************************
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* Preamble
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*
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* Documentation: http://www.pcisys.net/~melanson/codecs/adpcm.txt
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*****************************************************************************/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <vlc_common.h>
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#include <vlc_plugin.h>
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#include <vlc_aout.h>
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#include <vlc_codec.h>
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/*****************************************************************************
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* Module descriptor
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*****************************************************************************/
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static int OpenDecoder( vlc_object_t * );
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static void CloseDecoder( vlc_object_t * );
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static aout_buffer_t *DecodeBlock( decoder_t *, block_t ** );
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vlc_module_begin ()
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set_description( N_("ADPCM audio decoder") )
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set_capability( "decoder", 50 )
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set_category( CAT_INPUT )
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set_subcategory( SUBCAT_INPUT_ACODEC )
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set_callbacks( OpenDecoder, CloseDecoder )
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vlc_module_end ()
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/*****************************************************************************
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* Local prototypes
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*****************************************************************************/
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enum adpcm_codec_e
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{
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ADPCM_IMA_QT,
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ADPCM_IMA_WAV,
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ADPCM_MS,
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ADPCM_DK3,
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ADPCM_DK4,
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ADPCM_EA
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};
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struct decoder_sys_t
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{
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enum adpcm_codec_e codec;
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size_t i_block;
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size_t i_samplesperblock;
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audio_date_t end_date;
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};
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static void DecodeAdpcmMs ( decoder_t *, int16_t *, uint8_t * );
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static void DecodeAdpcmImaWav( decoder_t *, int16_t *, uint8_t * );
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static void DecodeAdpcmImaQT ( decoder_t *, int16_t *, uint8_t * );
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static void DecodeAdpcmDk4 ( decoder_t *, int16_t *, uint8_t * );
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static void DecodeAdpcmDk3 ( decoder_t *, int16_t *, uint8_t * );
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static void DecodeAdpcmEA ( decoder_t *, int16_t *, uint8_t * );
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static const int pi_channels_maps[6] =
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{
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0,
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AOUT_CHAN_CENTER,
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AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT,
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AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER,
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AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARLEFT,
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AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
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| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARLEFT
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};
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/* Various table from http://www.pcisys.net/~melanson/codecs/adpcm.txt */
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static const int i_index_table[16] =
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{
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-1, -1, -1, -1, 2, 4, 6, 8,
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-1, -1, -1, -1, 2, 4, 6, 8
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};
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static const int i_step_table[89] =
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{
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7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
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19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
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50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
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130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
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337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
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876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
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2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
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5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
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15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
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};
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static const int i_adaptation_table[16] =
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{
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230, 230, 230, 230, 307, 409, 512, 614,
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768, 614, 512, 409, 307, 230, 230, 230
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};
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static const int i_adaptation_coeff1[7] =
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{
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256, 512, 0, 192, 240, 460, 392
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};
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static const int i_adaptation_coeff2[7] =
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{
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0, -256, 0, 64, 0, -208, -232
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};
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/*****************************************************************************
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* OpenDecoder: probe the decoder and return score
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*****************************************************************************/
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static int OpenDecoder( vlc_object_t *p_this )
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{
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decoder_t *p_dec = (decoder_t*)p_this;
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decoder_sys_t *p_sys;
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switch( p_dec->fmt_in.i_codec )
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{
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case VLC_FOURCC('i','m','a', '4'): /* IMA ADPCM */
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case VLC_FOURCC('m','s',0x00,0x02): /* MS ADPCM */
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case VLC_FOURCC('m','s',0x00,0x11): /* IMA ADPCM */
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case VLC_FOURCC('m','s',0x00,0x61): /* Duck DK4 ADPCM */
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case VLC_FOURCC('m','s',0x00,0x62): /* Duck DK3 ADPCM */
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case VLC_FOURCC('X','A','J', 0): /* EA ADPCM */
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break;
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default:
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return VLC_EGENERIC;
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}
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if( p_dec->fmt_in.audio.i_channels <= 0 ||
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p_dec->fmt_in.audio.i_channels > 5 )
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{
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msg_Err( p_dec, "invalid number of channel (not between 1 and 5): %i",
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p_dec->fmt_in.audio.i_channels );
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return VLC_EGENERIC;
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}
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if( p_dec->fmt_in.audio.i_rate <= 0 )
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{
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msg_Err( p_dec, "bad samplerate" );
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return VLC_EGENERIC;
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}
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/* Allocate the memory needed to store the decoder's structure */
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if( ( p_dec->p_sys = p_sys =
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(decoder_sys_t *)malloc(sizeof(decoder_sys_t)) ) == NULL )
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return VLC_ENOMEM;
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switch( p_dec->fmt_in.i_codec )
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{
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case VLC_FOURCC('i','m','a', '4'): /* IMA ADPCM */
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p_sys->codec = ADPCM_IMA_QT;
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break;
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case VLC_FOURCC('m','s',0x00,0x11): /* IMA ADPCM */
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p_sys->codec = ADPCM_IMA_WAV;
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break;
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case VLC_FOURCC('m','s',0x00,0x02): /* MS ADPCM */
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p_sys->codec = ADPCM_MS;
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break;
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case VLC_FOURCC('m','s',0x00,0x61): /* Duck DK4 ADPCM */
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p_sys->codec = ADPCM_DK4;
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break;
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case VLC_FOURCC('m','s',0x00,0x62): /* Duck DK3 ADPCM */
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p_sys->codec = ADPCM_DK3;
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break;
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case VLC_FOURCC('X','A','J', 0): /* EA ADPCM */
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p_sys->codec = ADPCM_EA;
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p_dec->fmt_in.p_extra = calloc( 2 * p_dec->fmt_in.audio.i_channels,
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sizeof( int16_t ) );
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if( p_dec->fmt_in.p_extra == NULL )
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{
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free( p_sys );
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return VLC_ENOMEM;
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}
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break;
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}
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if( p_dec->fmt_in.audio.i_blockalign <= 0 )
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{
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p_sys->i_block = (p_sys->codec == ADPCM_IMA_QT) ?
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34 * p_dec->fmt_in.audio.i_channels : 1024;
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msg_Warn( p_dec, "block size undefined, using %zu", p_sys->i_block );
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}
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else
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{
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p_sys->i_block = p_dec->fmt_in.audio.i_blockalign;
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}
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/* calculate samples per block */
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switch( p_sys->codec )
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{
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case ADPCM_IMA_QT:
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p_sys->i_samplesperblock = 64;
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break;
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case ADPCM_IMA_WAV:
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p_sys->i_samplesperblock =
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2 * ( p_sys->i_block - 4 * p_dec->fmt_in.audio.i_channels ) /
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p_dec->fmt_in.audio.i_channels;
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break;
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case ADPCM_MS:
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p_sys->i_samplesperblock =
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2 * (p_sys->i_block - 7 * p_dec->fmt_in.audio.i_channels) /
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p_dec->fmt_in.audio.i_channels + 2;
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break;
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case ADPCM_DK4:
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p_sys->i_samplesperblock =
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2 * (p_sys->i_block - 4 * p_dec->fmt_in.audio.i_channels) /
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p_dec->fmt_in.audio.i_channels + 1;
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break;
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case ADPCM_DK3:
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p_dec->fmt_in.audio.i_channels = 2;
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p_sys->i_samplesperblock = ( 4 * ( p_sys->i_block - 16 ) + 2 )/ 3;
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break;
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case ADPCM_EA:
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p_sys->i_samplesperblock =
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2 * (p_sys->i_block - p_dec->fmt_in.audio.i_channels) /
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p_dec->fmt_in.audio.i_channels;
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}
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msg_Dbg( p_dec, "format: samplerate:%d Hz channels:%d bits/sample:%d "
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"blockalign:%zu samplesperblock:%zu",
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p_dec->fmt_in.audio.i_rate, p_dec->fmt_in.audio.i_channels,
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p_dec->fmt_in.audio.i_bitspersample, p_sys->i_block,
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p_sys->i_samplesperblock );
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p_dec->fmt_out.i_codec = AOUT_FMT_S16_NE;
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p_dec->fmt_out.audio.i_rate = p_dec->fmt_in.audio.i_rate;
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p_dec->fmt_out.audio.i_channels = p_dec->fmt_in.audio.i_channels;
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p_dec->fmt_out.audio.i_physical_channels =
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p_dec->fmt_out.audio.i_original_channels =
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pi_channels_maps[p_dec->fmt_in.audio.i_channels];
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aout_DateInit( &p_sys->end_date, p_dec->fmt_out.audio.i_rate );
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aout_DateSet( &p_sys->end_date, 0 );
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p_dec->pf_decode_audio = DecodeBlock;
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return VLC_SUCCESS;
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}
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/*****************************************************************************
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* DecodeBlock:
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*****************************************************************************/
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static aout_buffer_t *DecodeBlock( decoder_t *p_dec, block_t **pp_block )
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{
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decoder_sys_t *p_sys = p_dec->p_sys;
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block_t *p_block;
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if( !pp_block || !*pp_block ) return NULL;
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p_block = *pp_block;
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if( p_block->i_pts != 0 &&
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p_block->i_pts != aout_DateGet( &p_sys->end_date ) )
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{
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aout_DateSet( &p_sys->end_date, p_block->i_pts );
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}
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else if( !aout_DateGet( &p_sys->end_date ) )
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{
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/* We've just started the stream, wait for the first PTS. */
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block_Release( p_block );
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return NULL;
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}
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/* Don't re-use the same pts twice */
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p_block->i_pts = 0;
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if( p_block->i_buffer >= p_sys->i_block )
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{
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aout_buffer_t *p_out;
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p_out = decoder_NewAudioBuffer( p_dec, p_sys->i_samplesperblock );
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if( p_out == NULL )
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{
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block_Release( p_block );
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return NULL;
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}
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p_out->start_date = aout_DateGet( &p_sys->end_date );
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p_out->end_date =
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aout_DateIncrement( &p_sys->end_date, p_sys->i_samplesperblock );
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switch( p_sys->codec )
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{
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case ADPCM_IMA_QT:
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DecodeAdpcmImaQT( p_dec, (int16_t*)p_out->p_buffer,
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p_block->p_buffer );
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break;
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case ADPCM_IMA_WAV:
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DecodeAdpcmImaWav( p_dec, (int16_t*)p_out->p_buffer,
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p_block->p_buffer );
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break;
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case ADPCM_MS:
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DecodeAdpcmMs( p_dec, (int16_t*)p_out->p_buffer,
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p_block->p_buffer );
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break;
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case ADPCM_DK4:
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DecodeAdpcmDk4( p_dec, (int16_t*)p_out->p_buffer,
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p_block->p_buffer );
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break;
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case ADPCM_DK3:
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DecodeAdpcmDk3( p_dec, (int16_t*)p_out->p_buffer,
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p_block->p_buffer );
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break;
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case ADPCM_EA:
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DecodeAdpcmEA( p_dec, (int16_t*)p_out->p_buffer,
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p_block->p_buffer );
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default:
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break;
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}
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p_block->p_buffer += p_sys->i_block;
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p_block->i_buffer -= p_sys->i_block;
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return p_out;
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}
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block_Release( p_block );
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return NULL;
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}
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/*****************************************************************************
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* CloseDecoder:
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*****************************************************************************/
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static void CloseDecoder( vlc_object_t *p_this )
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{
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decoder_t *p_dec = (decoder_t *)p_this;
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decoder_sys_t *p_sys = p_dec->p_sys;
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if( p_sys->codec == ADPCM_EA )
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free( p_dec->fmt_in.p_extra );
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free( p_sys );
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}
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/*****************************************************************************
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* Local functions
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*****************************************************************************/
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#define CLAMP( v, min, max ) \
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if( (v) < (min) ) (v) = (min); \
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if( (v) > (max) ) (v) = (max)
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#define GetByte( v ) \
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(v) = *p_buffer; p_buffer++;
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#define GetWord( v ) \
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(v) = *p_buffer; p_buffer++; \
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(v) |= ( *p_buffer ) << 8; p_buffer++; \
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if( (v)&0x8000 ) (v) -= 0x010000;
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/*
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* MS
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*/
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typedef struct adpcm_ms_channel_s
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{
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int i_idelta;
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int i_sample1, i_sample2;
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int i_coeff1, i_coeff2;
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} adpcm_ms_channel_t;
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static int AdpcmMsExpandNibble(adpcm_ms_channel_t *p_channel,
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int i_nibble )
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{
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int i_predictor;
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int i_snibble;
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/* expand sign */
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i_snibble = i_nibble - ( i_nibble&0x08 ? 0x10 : 0 );
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i_predictor = ( p_channel->i_sample1 * p_channel->i_coeff1 +
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p_channel->i_sample2 * p_channel->i_coeff2 ) / 256 +
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i_snibble * p_channel->i_idelta;
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CLAMP( i_predictor, -32768, 32767 );
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p_channel->i_sample2 = p_channel->i_sample1;
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p_channel->i_sample1 = i_predictor;
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p_channel->i_idelta = ( i_adaptation_table[i_nibble] *
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p_channel->i_idelta ) / 256;
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if( p_channel->i_idelta < 16 )
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{
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p_channel->i_idelta = 16;
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}
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return( i_predictor );
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}
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static void DecodeAdpcmMs( decoder_t *p_dec, int16_t *p_sample,
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uint8_t *p_buffer )
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{
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decoder_sys_t *p_sys = p_dec->p_sys;
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adpcm_ms_channel_t channel[2];
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int i_nibbles;
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int b_stereo;
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int i_block_predictor;
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b_stereo = p_dec->fmt_in.audio.i_channels == 2 ? 1 : 0;
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GetByte( i_block_predictor );
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CLAMP( i_block_predictor, 0, 6 );
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channel[0].i_coeff1 = i_adaptation_coeff1[i_block_predictor];
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channel[0].i_coeff2 = i_adaptation_coeff2[i_block_predictor];
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if( b_stereo )
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{
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GetByte( i_block_predictor );
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CLAMP( i_block_predictor, 0, 6 );
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channel[1].i_coeff1 = i_adaptation_coeff1[i_block_predictor];
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channel[1].i_coeff2 = i_adaptation_coeff2[i_block_predictor];
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}
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GetWord( channel[0].i_idelta );
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if( b_stereo )
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{
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GetWord( channel[1].i_idelta );
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}
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GetWord( channel[0].i_sample1 );
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if( b_stereo )
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{
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GetWord( channel[1].i_sample1 );
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}
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GetWord( channel[0].i_sample2 );
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if( b_stereo )
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{
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GetWord( channel[1].i_sample2 );
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}
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if( b_stereo )
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{
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*p_sample++ = channel[0].i_sample2;
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*p_sample++ = channel[1].i_sample2;
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*p_sample++ = channel[0].i_sample1;
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*p_sample++ = channel[1].i_sample1;
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}
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else
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{
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*p_sample++ = channel[0].i_sample2;
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*p_sample++ = channel[0].i_sample1;
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}
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for( i_nibbles = 2 * (p_sys->i_block - 7 * p_dec->fmt_in.audio.i_channels);
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i_nibbles > 0; i_nibbles -= 2, p_buffer++ )
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{
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|
*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_in.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 > 0;
|
|
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 )
|
|
{
|
|
adpcm_ima_wav_channel_t channel[2];
|
|
int i_nibbles;
|
|
int i_ch;
|
|
int i_step;
|
|
|
|
i_step = p_dec->fmt_in.audio.i_channels;
|
|
|
|
for( i_ch = 0; i_ch < p_dec->fmt_in.audio.i_channels; i_ch++ )
|
|
{
|
|
/* load preambule */
|
|
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_sample = AdpcmImaWavExpandNibble( &channel[i_ch], (*p_buffer)&0x0f);
|
|
p_sample += i_step;
|
|
|
|
*p_sample = AdpcmImaWavExpandNibble( &channel[i_ch], (*p_buffer >> 4)&0x0f);
|
|
p_sample += i_step;
|
|
|
|
p_buffer++;
|
|
}
|
|
|
|
/* Next channel */
|
|
p_sample += 1 - 64 * i_step;
|
|
}
|
|
}
|
|
|
|
/*
|
|
* 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_in.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 );
|
|
|
|
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 uint32_t EATable[]=
|
|
{
|
|
0x00000000, 0x000000F0, 0x000001CC, 0x00000188,
|
|
0x00000000, 0x00000000, 0xFFFFFF30, 0xFFFFFF24,
|
|
0x00000000, 0x00000001, 0x00000003, 0x00000004,
|
|
0x00000007, 0x00000008, 0x0000000A, 0x0000000B,
|
|
0x00000000, 0xFFFFFFFF, 0xFFFFFFFD, 0xFFFFFFFC
|
|
};
|
|
decoder_sys_t *p_sys = p_dec->p_sys;
|
|
uint8_t *p_end;
|
|
unsigned i_channels, c;
|
|
int16_t *prev, *cur;
|
|
int32_t c1[MAX_CHAN], c2[MAX_CHAN];
|
|
int8_t d[MAX_CHAN];
|
|
|
|
i_channels = p_dec->fmt_in.audio.i_channels;
|
|
p_end = &p_buffer[p_sys->i_block];
|
|
|
|
prev = (int16_t *)p_dec->fmt_in.p_extra;
|
|
cur = prev + i_channels;
|
|
|
|
for (c = 0; c < i_channels; c++)
|
|
{
|
|
uint8_t input;
|
|
|
|
input = p_buffer[c];
|
|
c1[c] = EATable[input >> 4];
|
|
c2[c] = EATable[(input >> 4) + 4];
|
|
d[c] = (input & 0xf) + 8;
|
|
}
|
|
|
|
for( p_buffer += i_channels; p_buffer < p_end ; p_buffer += i_channels)
|
|
{
|
|
for (c = 0; c < i_channels; c++)
|
|
{
|
|
int32_t spl;
|
|
|
|
spl = (p_buffer[c] >> 4) & 0xf;
|
|
spl = (spl << 0x1c) >> d[c];
|
|
spl = (spl + cur[c] * c1[c] + prev[c] * c2[c] + 0x80) >> 8;
|
|
CLAMP( spl, -32768, 32767 );
|
|
prev[c] = cur[c];
|
|
cur[c] = spl;
|
|
|
|
*(p_sample++) = spl;
|
|
}
|
|
|
|
for (c = 0; c < i_channels; c++)
|
|
{
|
|
int32_t spl;
|
|
|
|
spl = p_buffer[c] & 0xf;
|
|
spl = (spl << 0x1c) >> d[c];
|
|
spl = (spl + cur[c] * c1[c] + prev[c] * c2[c] + 0x80) >> 8;
|
|
CLAMP( spl, -32768, 32767 );
|
|
prev[c] = cur[c];
|
|
cur[c] = spl;
|
|
|
|
*(p_sample++) = spl;
|
|
}
|
|
}
|
|
}
|