modules/demux/rawdv.h

   1 /*****************************************************************************
   2  * rawdv.h : raw DV helpers
   3  *****************************************************************************
   4  * Copyright (C) 2001-2011 VLC authors and VideoLAN
   5  * $Id$
   6  *
   7  * Authors: Gildas Bazin <gbazin@netcourrier.com>
   8  *          Paul Corke <paul dot corke at datatote dot co dot uk>
   9  *
  10  * This program is free software; you can redistribute it and/or modify it
  11  * under the terms of the GNU Lesser General Public License as published by
  12  * the Free Software Foundation; either version 2.1 of the License, or
  13  * (at your option) any later version.
  14  *
  15  * This program is distributed in the hope that it will be useful,
  16  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  17  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
  18  * GNU Lesser General Public License for more details.
  19  *
  20  * You should have received a copy of the GNU Lesser General Public License
  21  * along with this program; if not, write to the Free Software Foundation,
  22  * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
  23  *****************************************************************************/
  24
  25 #define DV_PAL_FRAME_SIZE  (12 * 150 * 80)
  26 #define DV_NTSC_FRAME_SIZE (10 * 150 * 80)
  27
  28
  29 static const uint16_t dv_audio_shuffle525[10][9] = {
  30   {  0, 30, 60, 20, 50, 80, 10, 40, 70 }, /* 1st channel */
  31   {  6, 36, 66, 26, 56, 86, 16, 46, 76 },
  32   { 12, 42, 72,  2, 32, 62, 22, 52, 82 },
  33   { 18, 48, 78,  8, 38, 68, 28, 58, 88 },
  34   { 24, 54, 84, 14, 44, 74,  4, 34, 64 },
  35
  36   {  1, 31, 61, 21, 51, 81, 11, 41, 71 }, /* 2nd channel */
  37   {  7, 37, 67, 27, 57, 87, 17, 47, 77 },
  38   { 13, 43, 73,  3, 33, 63, 23, 53, 83 },
  39   { 19, 49, 79,  9, 39, 69, 29, 59, 89 },
  40   { 25, 55, 85, 15, 45, 75,  5, 35, 65 },
  41 };
  42
  43 static const uint16_t dv_audio_shuffle625[12][9] = {
  44   {   0,  36,  72,  26,  62,  98,  16,  52,  88}, /* 1st channel */
  45   {   6,  42,  78,  32,  68, 104,  22,  58,  94},
  46   {  12,  48,  84,   2,  38,  74,  28,  64, 100},
  47   {  18,  54,  90,   8,  44,  80,  34,  70, 106},
  48   {  24,  60,  96,  14,  50,  86,   4,  40,  76},
  49   {  30,  66, 102,  20,  56,  92,  10,  46,  82},
  50
  51   {   1,  37,  73,  27,  63,  99,  17,  53,  89}, /* 2nd channel */
  52   {   7,  43,  79,  33,  69, 105,  23,  59,  95},
  53   {  13,  49,  85,   3,  39,  75,  29,  65, 101},
  54   {  19,  55,  91,   9,  45,  81,  35,  71, 107},
  55   {  25,  61,  97,  15,  51,  87,   5,  41,  77},
  56   {  31,  67, 103,  21,  57,  93,  11,  47,  83},
  57 };
  58
  59 static inline uint16_t dv_audio_12to16( uint16_t sample )
  60 {
  61     uint16_t shift, result;
  62
  63     sample = (sample < 0x800) ? sample : sample | 0xf000;
  64     shift = (sample & 0xf00) >> 8;
  65
  66     if (shift < 0x2 || shift > 0xd) {
  67         result = sample;
  68     } else if (shift < 0x8) {
  69         shift--;
  70         result = (sample - (256 * shift)) << shift;
  71     } else {
  72         shift = 0xe - shift;
  73         result = ((sample + ((256 * shift) + 1)) << shift) - 1;
  74     }
  75
  76     return result;
  77 }
  78
  79 static inline void dv_get_audio_format( es_format_t *p_fmt,
  80                                         const uint8_t *p_aaux_src )
  81 {
  82     /* 12 bits non-linear will be converted to 16 bits linear */
  83     es_format_Init( p_fmt, AUDIO_ES, VLC_CODEC_S16L );
  84
  85     p_fmt->audio.i_bitspersample = 16;
  86     p_fmt->audio.i_channels = 2;
  87     switch( (p_aaux_src[4-1] >> 3) & 0x07 )
  88     {
  89     case 0:
  90         p_fmt->audio.i_rate = 48000;
  91         break;
  92     case 1:
  93         p_fmt->audio.i_rate = 44100;
  94         break;
  95     case 2:
  96     default:
  97         p_fmt->audio.i_rate = 32000;
  98         break;
  99     }
 100 }
 101
 102 static inline int dv_get_audio_sample_count( const uint8_t *p_buffer, int i_dsf )
 103 {
 104     int i_samples = p_buffer[0] & 0x3f; /* samples in this frame - min samples */
 105     switch( (p_buffer[3] >> 3) & 0x07 )
 106     {
 107     case 0:
 108         return i_samples + (i_dsf ? 1896 : 1580);
 109     case 1:
 110         return i_samples + (i_dsf ? 1742 : 1452);
 111     case 2:
 112     default:
 113         return i_samples + (i_dsf ? 1264 : 1053);
 114     }
 115 }
 116
 117 static inline block_t *dv_extract_audio( block_t *p_frame_block )
 118 {
 119     block_t *p_block;
 120     uint8_t *p_frame, *p_buf;
 121     int i_audio_quant, i_samples, i_half_ch;
 122     const uint16_t (*audio_shuffle)[9];
 123     int i, j, d, of;
 124
 125     if( p_frame_block->i_buffer < 4 )
 126         return NULL;
 127     const int i_dsf = (p_frame_block->p_buffer[3] & 0x80) >> 7;
 128     if( p_frame_block->i_buffer < (i_dsf ? DV_PAL_FRAME_SIZE
 129                                          : DV_NTSC_FRAME_SIZE ) )
 130         return NULL;
 131
 132     /* Beginning of AAUX pack */
 133     p_buf = p_frame_block->p_buffer + 80*6+80*16*3 + 3;
 134     if( *p_buf != 0x50 ) return NULL;
 135
 136     i_audio_quant = p_buf[4] & 0x07; /* 0 - 16bit, 1 - 12bit */
 137     if( i_audio_quant > 1 )
 138         return NULL;
 139
 140     i_samples = dv_get_audio_sample_count( &p_buf[1], i_dsf );
 141
 142     p_block = block_Alloc( 4 * i_samples );
 143
 144     /* for each DIF segment */
 145     p_frame = p_frame_block->p_buffer;
 146     audio_shuffle = i_dsf ? dv_audio_shuffle625 : dv_audio_shuffle525;
 147     i_half_ch = (i_dsf ? 12 : 10)/2;
 148     for( i = 0; i < (i_dsf ? 12 : 10); i++ )
 149     {
 150         p_frame += 6 * 80; /* skip DIF segment header */
 151
 152         if( i_audio_quant == 1 && i == i_half_ch ) break;
 153
 154         for( j = 0; j < 9; j++ )
 155         {
 156             for( d = 8; d < 80; d += 2 )
 157             {
 158                 if( i_audio_quant == 0 )
 159                 {
 160                     /* 16bit quantization */
 161                     of = audio_shuffle[i][j] + (d - 8) / 2 *
 162                            (i_dsf ? 108 : 90);
 163
 164                     if( of * 2 >= 4 * i_samples ) continue;
 165
 166                     /* big endian */
 167                     p_block->p_buffer[of*2] = p_frame[d+1];
 168                     p_block->p_buffer[of*2+1] = p_frame[d];
 169
 170                     if( p_block->p_buffer[of*2+1] == 0x80 &&
 171                         p_block->p_buffer[of*2] == 0x00 )
 172                         p_block->p_buffer[of*2+1] = 0;
 173                 }
 174                 else
 175                 {
 176                     /* 12bit quantization */
 177                     uint16_t lc = (p_frame[d+0] << 4) | (p_frame[d+2] >> 4);
 178                     uint16_t rc = (p_frame[d+1] << 4) | (p_frame[d+2] & 0x0f);
 179
 180                     lc = lc == 0x800 ? 0 : dv_audio_12to16(lc);
 181                     rc = rc == 0x800 ? 0 : dv_audio_12to16(rc);
 182
 183                     of = audio_shuffle[i][j] + (d - 8) / 3 * (i_dsf ? 108 : 90);
 184                     if( of*2 >= 4 * i_samples )
 185                         continue;
 186                     p_block->p_buffer[of*2+0] = lc & 0xff;
 187                     p_block->p_buffer[of*2+1] = lc >> 8;
 188
 189                     of = audio_shuffle[i + i_half_ch][j] + (d - 8) / 3 * (i_dsf ? 108 : 90);
 190                     if( of*2 >= 4 * i_samples )
 191                         continue;
 192                     p_block->p_buffer[of*2+0] = rc & 0xff;
 193                     p_block->p_buffer[of*2+1] = rc >> 8;
 194
 195                     ++d;
 196                 }
 197             }
 198
 199             p_frame += 16 * 80; /* 15 Video DIFs + 1 Audio DIF */
 200         }
 201     }
 202
 203     p_block->i_pts = p_frame_block->i_pts > VLC_TS_INVALID ? p_frame_block->i_pts
 204                                                            : p_frame_block->i_dts;
 205     p_block->i_dts = p_frame_block->i_dts;
 206     return p_block;
 207 }
 208