Commit 6a24a618 authored by Laurent Aimar's avatar Laurent Aimar

Splitted dv audio helpers into rawdv.h

It will allow to share them with the AVI demuxer.
parent 4fbbd7cd
......@@ -3,7 +3,7 @@ SOURCES_flacsys = flac.c
SOURCES_ogg = ogg.c ogg.h oggseek.c oggseek.h vorbis.h kate_categories.c \
kate_categories.h xiph.h
SOURCES_demuxdump = demuxdump.c
SOURCES_rawdv = rawdv.c
SOURCES_rawdv = rawdv.c rawdv.h
SOURCES_rawvid = rawvid.c
SOURCES_au = au.c
SOURCES_rawaud = rawaud.c
......
......@@ -34,6 +34,8 @@
#include <vlc_plugin.h>
#include <vlc_demux.h>
#include "rawdv.h"
/*****************************************************************************
* Module descriptor
*****************************************************************************/
......@@ -70,13 +72,6 @@ vlc_module_end ()
*****************************************************************************/
/*****************************************************************************
* Constants
*****************************************************************************/
#define DV_PAL_FRAME_SIZE (12 * 150 * 80)
#define DV_NTSC_FRAME_SIZE (10 * 150 * 80)
/*****************************************************************************
* Definitions of structures used by this plugin
*****************************************************************************/
......@@ -123,9 +118,6 @@ struct demux_sys_t
static int Demux( demux_t * );
static int Control( demux_t *, int i_query, va_list args );
static block_t *dv_extract_audio( demux_t *p_demux,
block_t* p_frame_block );
/*****************************************************************************
* Open: initializes raw DV demux structures
*****************************************************************************/
......@@ -240,25 +232,7 @@ static int Open( vlc_object_t * p_this )
p_peek = p_peek_backup + 80*6+80*16*3 + 3; /* beginning of AAUX pack */
if( *p_peek == 0x50 )
{
/* 12 bits non-linear will be converted to 16 bits linear */
es_format_Init( &p_sys->fmt_audio, AUDIO_ES, VLC_CODEC_S16L );
p_sys->fmt_audio.audio.i_bitspersample = 16;
p_sys->fmt_audio.audio.i_channels = 2;
switch( (p_peek[4] >> 3) & 0x07 )
{
case 0:
p_sys->fmt_audio.audio.i_rate = 48000;
break;
case 1:
p_sys->fmt_audio.audio.i_rate = 44100;
break;
case 2:
default:
p_sys->fmt_audio.audio.i_rate = 32000;
break;
}
dv_get_audio_format( &p_sys->fmt_audio, &p_peek[1] );
p_sys->p_es_audio = es_out_Add( p_demux->out, &p_sys->fmt_audio );
}
......@@ -314,13 +288,9 @@ static int Demux( demux_t *p_demux )
if( b_audio )
{
block_t *p_audio_block = dv_extract_audio( p_demux, p_block );
block_t *p_audio_block = dv_extract_audio( p_block );
if( p_audio_block )
{
p_audio_block->i_pts =
p_audio_block->i_dts = VLC_TS_0 + p_sys->i_pcr;
es_out_Send( p_demux->out, p_sys->p_es_audio, p_audio_block );
}
}
es_out_Send( p_demux->out, p_sys->p_es_video, p_block );
......@@ -347,147 +317,3 @@ static int Control( demux_t *p_demux, int i_query, va_list args )
p_sys->frame_size, i_query, args );
}
static const uint16_t dv_audio_shuffle525[10][9] = {
{ 0, 30, 60, 20, 50, 80, 10, 40, 70 }, /* 1st channel */
{ 6, 36, 66, 26, 56, 86, 16, 46, 76 },
{ 12, 42, 72, 2, 32, 62, 22, 52, 82 },
{ 18, 48, 78, 8, 38, 68, 28, 58, 88 },
{ 24, 54, 84, 14, 44, 74, 4, 34, 64 },
{ 1, 31, 61, 21, 51, 81, 11, 41, 71 }, /* 2nd channel */
{ 7, 37, 67, 27, 57, 87, 17, 47, 77 },
{ 13, 43, 73, 3, 33, 63, 23, 53, 83 },
{ 19, 49, 79, 9, 39, 69, 29, 59, 89 },
{ 25, 55, 85, 15, 45, 75, 5, 35, 65 },
};
static const uint16_t dv_audio_shuffle625[12][9] = {
{ 0, 36, 72, 26, 62, 98, 16, 52, 88}, /* 1st channel */
{ 6, 42, 78, 32, 68, 104, 22, 58, 94},
{ 12, 48, 84, 2, 38, 74, 28, 64, 100},
{ 18, 54, 90, 8, 44, 80, 34, 70, 106},
{ 24, 60, 96, 14, 50, 86, 4, 40, 76},
{ 30, 66, 102, 20, 56, 92, 10, 46, 82},
{ 1, 37, 73, 27, 63, 99, 17, 53, 89}, /* 2nd channel */
{ 7, 43, 79, 33, 69, 105, 23, 59, 95},
{ 13, 49, 85, 3, 39, 75, 29, 65, 101},
{ 19, 55, 91, 9, 45, 81, 35, 71, 107},
{ 25, 61, 97, 15, 51, 87, 5, 41, 77},
{ 31, 67, 103, 21, 57, 93, 11, 47, 83},
};
static inline uint16_t dv_audio_12to16( uint16_t sample )
{
uint16_t shift, result;
sample = (sample < 0x800) ? sample : sample | 0xf000;
shift = (sample & 0xf00) >> 8;
if (shift < 0x2 || shift > 0xd) {
result = sample;
} else if (shift < 0x8) {
shift--;
result = (sample - (256 * shift)) << shift;
} else {
shift = 0xe - shift;
result = ((sample + ((256 * shift) + 1)) << shift) - 1;
}
return result;
}
static block_t *dv_extract_audio( demux_t *p_demux,
block_t* p_frame_block )
{
demux_sys_t *p_sys = p_demux->p_sys;
block_t *p_block;
uint8_t *p_frame, *p_buf;
int i_audio_quant, i_samples, i_size, i_half_ch;
const uint16_t (*audio_shuffle)[9];
int i, j, d, of;
uint16_t lc;
/* Beginning of AAUX pack */
p_buf = p_frame_block->p_buffer + 80*6+80*16*3 + 3;
if( *p_buf != 0x50 ) return NULL;
i_audio_quant = p_buf[4] & 0x07; /* 0 - 16bit, 1 - 12bit */
if( i_audio_quant > 1 )
{
msg_Dbg( p_demux, "unsupported quantization for DV audio");
return NULL;
}
i_samples = p_buf[1] & 0x3f; /* samples in this frame - min samples */
switch( (p_buf[4] >> 3) & 0x07 )
{
case 0:
i_size = p_sys->i_dsf ? 1896 : 1580;
break;
case 1:
i_size = p_sys->i_dsf ? 1742 : 1452;
break;
case 2:
default:
i_size = p_sys->i_dsf ? 1264 : 1053;
break;
}
i_size = (i_size + i_samples) * 4; /* 2ch, 2bytes */
p_block = block_New( p_demux, i_size );
/* for each DIF segment */
p_frame = p_frame_block->p_buffer;
audio_shuffle = p_sys->i_dsf ? dv_audio_shuffle625 : dv_audio_shuffle525;
i_half_ch = (p_sys->i_dsf ? 12 : 10)/2;
for( i = 0; i < (p_sys->i_dsf ? 12 : 10); i++ )
{
p_frame += 6 * 80; /* skip DIF segment header */
if( i_audio_quant == 1 && i == i_half_ch ) break;
for( j = 0; j < 9; j++ )
{
for( d = 8; d < 80; d += 2 )
{
if( i_audio_quant == 0 )
{
/* 16bit quantization */
of = audio_shuffle[i][j] + (d - 8) / 2 *
(p_sys->i_dsf ? 108 : 90);
if( of * 2 >= i_size ) continue;
/* big endian */
p_block->p_buffer[of*2] = p_frame[d+1];
p_block->p_buffer[of*2+1] = p_frame[d];
if( p_block->p_buffer[of*2+1] == 0x80 &&
p_block->p_buffer[of*2] == 0x00 )
p_block->p_buffer[of*2+1] = 0;
}
else
{
/* 12bit quantization */
lc = ((uint16_t)p_frame[d] << 4) |
((uint16_t)p_frame[d+2] >> 4);
lc = (lc == 0x800 ? 0 : dv_audio_12to16(lc));
of = audio_shuffle[i][j] + (d - 8) / 3 *
(p_sys->i_dsf ? 108 : 90);
if( of*2 >= i_size ) continue;
/* big endian */
p_block->p_buffer[of*2] = lc & 0xff;
p_block->p_buffer[of*2+1] = lc >> 8;
++d;
}
}
p_frame += 16 * 80; /* 15 Video DIFs + 1 Audio DIF */
}
}
return p_block;
}
/*****************************************************************************
* rawdv.h : raw DV helpers
*****************************************************************************
* Copyright (C) 2001-2011 the VideoLAN team
* $Id$
*
* Authors: Gildas Bazin <gbazin@netcourrier.com>
* Paul Corke <paul dot corke at datatote dot co dot uk>
*
* 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
* the Free Software Foundation; either version 2 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 General Public License for more details.
*
* 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., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
#define DV_PAL_FRAME_SIZE (12 * 150 * 80)
#define DV_NTSC_FRAME_SIZE (10 * 150 * 80)
static const uint16_t dv_audio_shuffle525[10][9] = {
{ 0, 30, 60, 20, 50, 80, 10, 40, 70 }, /* 1st channel */
{ 6, 36, 66, 26, 56, 86, 16, 46, 76 },
{ 12, 42, 72, 2, 32, 62, 22, 52, 82 },
{ 18, 48, 78, 8, 38, 68, 28, 58, 88 },
{ 24, 54, 84, 14, 44, 74, 4, 34, 64 },
{ 1, 31, 61, 21, 51, 81, 11, 41, 71 }, /* 2nd channel */
{ 7, 37, 67, 27, 57, 87, 17, 47, 77 },
{ 13, 43, 73, 3, 33, 63, 23, 53, 83 },
{ 19, 49, 79, 9, 39, 69, 29, 59, 89 },
{ 25, 55, 85, 15, 45, 75, 5, 35, 65 },
};
static const uint16_t dv_audio_shuffle625[12][9] = {
{ 0, 36, 72, 26, 62, 98, 16, 52, 88}, /* 1st channel */
{ 6, 42, 78, 32, 68, 104, 22, 58, 94},
{ 12, 48, 84, 2, 38, 74, 28, 64, 100},
{ 18, 54, 90, 8, 44, 80, 34, 70, 106},
{ 24, 60, 96, 14, 50, 86, 4, 40, 76},
{ 30, 66, 102, 20, 56, 92, 10, 46, 82},
{ 1, 37, 73, 27, 63, 99, 17, 53, 89}, /* 2nd channel */
{ 7, 43, 79, 33, 69, 105, 23, 59, 95},
{ 13, 49, 85, 3, 39, 75, 29, 65, 101},
{ 19, 55, 91, 9, 45, 81, 35, 71, 107},
{ 25, 61, 97, 15, 51, 87, 5, 41, 77},
{ 31, 67, 103, 21, 57, 93, 11, 47, 83},
};
static inline uint16_t dv_audio_12to16( uint16_t sample )
{
uint16_t shift, result;
sample = (sample < 0x800) ? sample : sample | 0xf000;
shift = (sample & 0xf00) >> 8;
if (shift < 0x2 || shift > 0xd) {
result = sample;
} else if (shift < 0x8) {
shift--;
result = (sample - (256 * shift)) << shift;
} else {
shift = 0xe - shift;
result = ((sample + ((256 * shift) + 1)) << shift) - 1;
}
return result;
}
static void dv_get_audio_format( es_format_t *p_fmt, const uint8_t *p_aaux_src )
{
/* 12 bits non-linear will be converted to 16 bits linear */
es_format_Init( p_fmt, AUDIO_ES, VLC_CODEC_S16L );
p_fmt->audio.i_bitspersample = 16;
p_fmt->audio.i_channels = 2;
switch( (p_aaux_src[4-1] >> 3) & 0x07 )
{
case 0:
p_fmt->audio.i_rate = 48000;
break;
case 1:
p_fmt->audio.i_rate = 44100;
break;
case 2:
default:
p_fmt->audio.i_rate = 32000;
break;
}
}
static block_t *dv_extract_audio( block_t *p_frame_block )
{
block_t *p_block;
uint8_t *p_frame, *p_buf;
int i_audio_quant, i_samples, i_size, i_half_ch;
const uint16_t (*audio_shuffle)[9];
int i, j, d, of;
uint16_t lc;
if( p_frame_block->i_buffer < 4 )
return NULL;
const int i_dsf = (p_frame_block->p_buffer[3] & 0x80) >> 7;
if( p_frame_block->i_buffer < (i_dsf ? DV_PAL_FRAME_SIZE
: DV_NTSC_FRAME_SIZE ) )
return NULL;
/* Beginning of AAUX pack */
p_buf = p_frame_block->p_buffer + 80*6+80*16*3 + 3;
if( *p_buf != 0x50 ) return NULL;
i_audio_quant = p_buf[4] & 0x07; /* 0 - 16bit, 1 - 12bit */
if( i_audio_quant > 1 )
return NULL;
i_samples = p_buf[1] & 0x3f; /* samples in this frame - min samples */
switch( (p_buf[4] >> 3) & 0x07 )
{
case 0:
i_size = i_dsf ? 1896 : 1580;
break;
case 1:
i_size = i_dsf ? 1742 : 1452;
break;
case 2:
default:
i_size = i_dsf ? 1264 : 1053;
break;
}
i_size = (i_size + i_samples) * 4; /* 2ch, 2bytes */
p_block = block_New( p_demux, i_size );
/* for each DIF segment */
p_frame = p_frame_block->p_buffer;
audio_shuffle = i_dsf ? dv_audio_shuffle625 : dv_audio_shuffle525;
i_half_ch = (i_dsf ? 12 : 10)/2;
for( i = 0; i < (i_dsf ? 12 : 10); i++ )
{
p_frame += 6 * 80; /* skip DIF segment header */
if( i_audio_quant == 1 && i == i_half_ch ) break;
for( j = 0; j < 9; j++ )
{
for( d = 8; d < 80; d += 2 )
{
if( i_audio_quant == 0 )
{
/* 16bit quantization */
of = audio_shuffle[i][j] + (d - 8) / 2 *
(i_dsf ? 108 : 90);
if( of * 2 >= i_size ) continue;
/* big endian */
p_block->p_buffer[of*2] = p_frame[d+1];
p_block->p_buffer[of*2+1] = p_frame[d];
if( p_block->p_buffer[of*2+1] == 0x80 &&
p_block->p_buffer[of*2] == 0x00 )
p_block->p_buffer[of*2+1] = 0;
}
else
{
/* 12bit quantization */
lc = ((uint16_t)p_frame[d] << 4) |
((uint16_t)p_frame[d+2] >> 4);
lc = (lc == 0x800 ? 0 : dv_audio_12to16(lc));
of = audio_shuffle[i][j] + (d - 8) / 3 *
(i_dsf ? 108 : 90);
if( of*2 >= i_size ) continue;
/* big endian */
p_block->p_buffer[of*2] = lc & 0xff;
p_block->p_buffer[of*2+1] = lc >> 8;
++d;
}
}
p_frame += 16 * 80; /* 15 Video DIFs + 1 Audio DIF */
}
}
p_block->i_pts = p_frame_block->i_pts > VLC_TS_INVALID ? p_frame_block->i_pts
: p_frame_block->i_dts;
p_block->i_dts = p_frame_block->i_dts;
return p_block;
}
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