Splitted dv audio helpers into rawdv.h

It will allow to share them with the AVI demuxer.

modules/demux/Modules.am

@@ -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

modules/demux/rawdv.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 );
     }
 
@@ -314,14 +288,10 @@ 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;
-}

modules/demux/rawdv.h

+/*****************************************************************************
+ * 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;
+}
+