/*****************************************************************************
* adpcm.c : adpcm variant audio decoder
*****************************************************************************
- * Copyright (C) 2001, 2002 VideoLAN
- * $Id: adpcm.c,v 1.16 2003/11/16 21:07:30 gbazin Exp $
+ * Copyright (C) 2001, 2002 the VideoLAN team
+ * $Id$
*
* Authors: Laurent Aimar <fenrir@via.ecp.fr>
+ * Rémi Denis-Courmont <rem # videolan.org>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
*
* Documentation: http://www.pcisys.net/~melanson/codecs/adpcm.txt
*****************************************************************************/
-#include <stdlib.h> /* malloc(), free() */
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
-#include <vlc/vlc.h>
-#include <vlc/decoder.h>
-#include <vlc/input.h>
-
-#include "codecs.h"
+#include <vlc_common.h>
+#include <vlc_plugin.h>
+#include <vlc_aout.h>
+#include <vlc_codec.h>
/*****************************************************************************
* Module descriptor
static aout_buffer_t *DecodeBlock( decoder_t *, block_t ** );
-vlc_module_begin();
- set_description( _("ADPCM audio decoder") );
- set_capability( "decoder", 50 );
- set_callbacks( OpenDecoder, CloseDecoder );
-vlc_module_end();
+vlc_module_begin ()
+ set_description( N_("ADPCM audio decoder") )
+ set_capability( "decoder", 50 )
+ set_category( CAT_INPUT )
+ set_subcategory( SUBCAT_INPUT_ACODEC )
+ set_callbacks( OpenDecoder, CloseDecoder )
+vlc_module_end ()
/*****************************************************************************
* Local prototypes
ADPCM_IMA_WAV,
ADPCM_MS,
ADPCM_DK3,
- ADPCM_DK4
+ ADPCM_DK4,
+ ADPCM_EA
};
struct decoder_sys_t
{
enum adpcm_codec_e codec;
- int i_block;
- int i_samplesperblock;
+ size_t i_block;
+ size_t i_samplesperblock;
- audio_date_t end_date;
+ date_t end_date;
};
static void DecodeAdpcmMs ( 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 * );
-static int pi_channels_maps[6] =
+static const int pi_channels_maps[6] =
{
0,
AOUT_CHAN_CENTER,
};
/* Various table from http://www.pcisys.net/~melanson/codecs/adpcm.txt */
-static int i_index_table[16] =
+static const int i_index_table[16] =
{
-1, -1, -1, -1, 2, 4, 6, 8,
-1, -1, -1, -1, 2, 4, 6, 8
};
-static int i_step_table[89] =
+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,
15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
};
-static int i_adaptation_table[16] =
+static const int i_adaptation_table[16] =
{
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
-static int i_adaptation_coeff1[7] =
+static const int i_adaptation_coeff1[7] =
{
256, 512, 0, 192, 240, 460, 392
};
-static int i_adaptation_coeff2[7] =
+static const int i_adaptation_coeff2[7] =
{
0, -256, 0, 64, 0, -208, -232
};
case VLC_FOURCC('m','s',0x00,0x11): /* IMA ADPCM */
case VLC_FOURCC('m','s',0x00,0x61): /* Duck DK4 ADPCM */
case VLC_FOURCC('m','s',0x00,0x62): /* Duck DK3 ADPCM */
+ case VLC_FOURCC('X','A','J', 0): /* EA ADPCM */
break;
default:
return VLC_EGENERIC;
}
- /* Allocate the memory needed to store the decoder's structure */
- if( ( p_dec->p_sys = p_sys =
- (decoder_sys_t *)malloc(sizeof(decoder_sys_t)) ) == NULL )
- {
- msg_Err( p_dec, "out of memory" );
- return VLC_EGENERIC;
- }
-
if( p_dec->fmt_in.audio.i_channels <= 0 ||
p_dec->fmt_in.audio.i_channels > 5 )
{
- msg_Err( p_dec, "bad channels count(1-5)" );
+ msg_Err( p_dec, "invalid number of channel (not between 1 and 5): %i",
+ p_dec->fmt_in.audio.i_channels );
return VLC_EGENERIC;
}
return VLC_EGENERIC;
}
+ /* Allocate the memory needed to store the decoder's structure */
+ if( ( p_dec->p_sys = p_sys =
+ (decoder_sys_t *)malloc(sizeof(decoder_sys_t)) ) == NULL )
+ return VLC_ENOMEM;
+
switch( p_dec->fmt_in.i_codec )
{
case VLC_FOURCC('i','m','a', '4'): /* IMA ADPCM */
p_sys->codec = ADPCM_IMA_QT;
break;
- case VLC_FOURCC('m','s',0x00,0x11): /* IMA ADPCM */
+ case VLC_CODEC_ADPCM_IMA_WAV: /* IMA ADPCM */
p_sys->codec = ADPCM_IMA_WAV;
break;
- case VLC_FOURCC('m','s',0x00,0x02): /* MS ADPCM */
+ case VLC_CODEC_ADPCM_MS: /* MS ADPCM */
p_sys->codec = ADPCM_MS;
break;
case VLC_FOURCC('m','s',0x00,0x61): /* Duck DK4 ADPCM */
case VLC_FOURCC('m','s',0x00,0x62): /* Duck DK3 ADPCM */
p_sys->codec = ADPCM_DK3;
break;
+ case VLC_FOURCC('X','A','J', 0): /* EA ADPCM */
+ p_sys->codec = ADPCM_EA;
+ p_dec->fmt_in.p_extra = calloc( 2 * p_dec->fmt_in.audio.i_channels,
+ sizeof( int16_t ) );
+ if( p_dec->fmt_in.p_extra == NULL )
+ {
+ free( p_sys );
+ return VLC_ENOMEM;
+ }
+ break;
}
- if( 1 )//( p_sys->i_block = p_wf->nBlockAlign ) <= 0 )
+ 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 %d", p_sys->i_block );
+ 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 */
p_dec->fmt_in.audio.i_channels = 2;
p_sys->i_samplesperblock = ( 4 * ( p_sys->i_block - 16 ) + 2 )/ 3;
break;
+ case ADPCM_EA:
+ p_sys->i_samplesperblock =
+ 2 * (p_sys->i_block - p_dec->fmt_in.audio.i_channels) /
+ p_dec->fmt_in.audio.i_channels;
}
- msg_Dbg( p_dec, "format: samplerate:%dHz channels:%d bits/sample:%d "
- "blockalign:%d samplesperblock %d",
+ msg_Dbg( p_dec, "format: samplerate:%d Hz channels:%d bits/sample:%d "
+ "blockalign:%zu samplesperblock:%zu",
p_dec->fmt_in.audio.i_rate, p_dec->fmt_in.audio.i_channels,
p_dec->fmt_in.audio.i_bitspersample, p_sys->i_block,
p_sys->i_samplesperblock );
- p_dec->fmt_out.i_codec = AOUT_FMT_S16_NE;
+ p_dec->fmt_out.i_cat = AUDIO_ES;
+ 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 = p_dec->fmt_in.audio.i_channels;
p_dec->fmt_out.audio.i_physical_channels =
p_dec->fmt_out.audio.i_original_channels =
pi_channels_maps[p_dec->fmt_in.audio.i_channels];
- aout_DateInit( &p_sys->end_date, p_dec->fmt_out.audio.i_rate );
- aout_DateSet( &p_sys->end_date, 0 );
+ date_Init( &p_sys->end_date, p_dec->fmt_out.audio.i_rate, 1 );
+ date_Set( &p_sys->end_date, 0 );
p_dec->pf_decode_audio = DecodeBlock;
p_block = *pp_block;
- if( p_block->i_pts != 0 &&
- p_block->i_pts != aout_DateGet( &p_sys->end_date ) )
+ if( p_block->i_pts > VLC_TS_INVALID &&
+ p_block->i_pts != date_Get( &p_sys->end_date ) )
{
- aout_DateSet( &p_sys->end_date, p_block->i_pts );
+ date_Set( &p_sys->end_date, p_block->i_pts );
}
- else if( !aout_DateGet( &p_sys->end_date ) )
+ else if( !date_Get( &p_sys->end_date ) )
{
/* We've just started the stream, wait for the first PTS. */
block_Release( p_block );
}
/* Don't re-use the same pts twice */
- p_block->i_pts = 0;
+ p_block->i_pts = VLC_TS_INVALID;
if( p_block->i_buffer >= p_sys->i_block )
{
aout_buffer_t *p_out;
- p_out = p_dec->pf_aout_buffer_new( p_dec, p_sys->i_samplesperblock );
+ p_out = decoder_NewAudioBuffer( p_dec, p_sys->i_samplesperblock );
if( p_out == NULL )
{
block_Release( p_block );
return NULL;
}
- p_out->start_date = aout_DateGet( &p_sys->end_date );
- p_out->end_date =
- aout_DateIncrement( &p_sys->end_date, p_sys->i_samplesperblock );
+ 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 )
{
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_out->i_nb_bytes;
- p_block->i_buffer -= p_out->i_nb_bytes;
+ p_block->p_buffer += p_sys->i_block;
+ p_block->i_buffer -= p_sys->i_block;
return p_out;
}
decoder_t *p_dec = (decoder_t *)p_this;
decoder_sys_t *p_sys = p_dec->p_sys;
+ if( p_sys->codec == ADPCM_EA )
+ free( p_dec->fmt_in.p_extra );
free( p_sys );
}
{
decoder_sys_t *p_sys = p_dec->p_sys;
adpcm_ima_wav_channel_t channel[2];
- int i_nibbles;
+ size_t i_nibbles;
int b_stereo;
b_stereo = p_dec->fmt_in.audio.i_channels == 2 ? 1 : 0;
}
}
}
+
+
+/*
+ * 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;
+ }
+ }
+}
projects / vlc / blobdiff