Commit 070b6a3a authored by Gildas Bazin's avatar Gildas Bazin

* modules/demux/rawdv.c: DV audio support was removed from libavcodec so reimplemented it here.

parent 23c72b3d
......@@ -2,7 +2,7 @@
* rawdv.c : raw dv input module for vlc
*****************************************************************************
* Copyright (C) 2001-2003 VideoLAN
* $Id: rawdv.c,v 1.14 2004/01/25 20:05:28 hartman Exp $
* $Id: rawdv.c,v 1.15 2004/02/29 19:01:22 gbazin Exp $
*
* Authors: Gildas Bazin <gbazin@netcourrier.com>
*
......@@ -35,6 +35,9 @@
static int Open ( vlc_object_t * );
static void Close( vlc_object_t * );
static block_t *dv_extract_audio( input_thread_t * p_input,
block_t* p_frame_block );
vlc_module_begin();
set_description( _("raw dv demuxer") );
set_capability( "demux", 2 );
......@@ -95,6 +98,7 @@ struct demux_sys_t
es_out_id_t *p_es_audio;
es_format_t fmt_audio;
int i_dsf;
double f_rate;
int i_bitrate;
......@@ -135,7 +139,8 @@ static int Open( vlc_object_t * p_this )
return VLC_EGENERIC;
}
if( stream_Peek( p_input->s, &p_peek, DV_PAL_FRAME_SIZE ) < DV_NTSC_FRAME_SIZE )
if( stream_Peek( p_input->s, &p_peek, DV_PAL_FRAME_SIZE ) <
DV_NTSC_FRAME_SIZE )
{
/* Stream too short ... */
msg_Err( p_input, "cannot peek()" );
......@@ -193,6 +198,7 @@ static int Open( vlc_object_t * p_this )
p_input->pf_demux_control = demux_vaControlDefault;
p_input->p_demux_data = p_sys = malloc( sizeof( demux_sys_t ) );
p_sys->i_dsf = dv_header.dsf;
p_sys->frame_size = dv_header.dsf ? 12 * 150 * 80 : 10 * 150 * 80;
p_sys->f_rate = dv_header.dsf ? 25 : 29.97;
......@@ -202,27 +208,10 @@ static int Open( vlc_object_t * p_this )
p_sys->i_bitrate = 0;
es_format_Init( &p_sys->fmt_video, VIDEO_ES, VLC_FOURCC( 'd','v','s','d' ) );
es_format_Init( &p_sys->fmt_video, VIDEO_ES, VLC_FOURCC('d','v','s','d') );
p_sys->fmt_video.video.i_width = 720;
p_sys->fmt_video.video.i_height= dv_header.dsf ? 576 : 480;;
/* Audio stuff */
#if 0
p_peek = p_peek_backup + 80*6+80*16*3 + 3; /* beginning of AAUX pack */
if( *p_peek != 0x50 || *p_peek != 0x51 )
{
msg_Err( p_input, "AAUX should begin with 0x50" );
}
#endif
es_format_Init( &p_sys->fmt_audio, AUDIO_ES, VLC_FOURCC( 'd','v','a','u' ) );
p_sys->fmt_audio.audio.i_channels = 2;
p_sys->fmt_audio.audio.i_rate = 44100; /* FIXME */
p_sys->fmt_audio.audio.i_bitspersample = 16;
p_sys->fmt_audio.audio.i_blockalign = p_sys->frame_size; /* ??? */
p_sys->fmt_audio.i_bitrate = p_sys->f_rate * p_sys->frame_size; /* ??? */
/* necessary because input_SplitBuffer() will only get
* INPUT_DEFAULT_BUFSIZE bytes at a time. */
p_input->i_bufsize = p_sys->frame_size;
......@@ -239,7 +228,34 @@ static int Open( vlc_object_t * p_this )
vlc_mutex_unlock( &p_input->stream.stream_lock );
p_sys->p_es_video = es_out_Add( p_input->p_es_out, &p_sys->fmt_video );
p_sys->p_es_audio = es_out_Add( p_input->p_es_out, &p_sys->fmt_audio );
/* Audio stuff */
p_peek = p_peek_backup + 80*6+80*16*3 + 3; /* beginning of AAUX pack */
if( *p_peek == 0x50 )
{
es_format_Init( &p_sys->fmt_audio, AUDIO_ES,
VLC_FOURCC('a','r','a','w') );
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;
}
/* 12 bits non-linear will be converted to 16 bits linear */
p_sys->fmt_audio.audio.i_bitspersample = 16;
p_sys->p_es_audio = es_out_Add( p_input->p_es_out, &p_sys->fmt_audio );
}
return VLC_SUCCESS;
}
......@@ -264,8 +280,7 @@ static int Demux( input_thread_t * p_input )
{
demux_sys_t *p_sys = p_input->p_demux_data;
block_t *p_block;
vlc_bool_t b_audio, b_video;
vlc_bool_t b_audio = VLC_FALSE, b_video = VLC_FALSE;
if( p_input->stream.p_selected_program->i_synchro_state == SYNCHRO_REINIT )
{
......@@ -297,40 +312,180 @@ static int Demux( input_thread_t * p_input )
return 0;
}
es_out_Control( p_input->p_es_out, ES_OUT_GET_ES_STATE,
p_sys->p_es_audio, &b_audio );
es_out_Control( p_input->p_es_out, ES_OUT_GET_ES_STATE,
p_sys->p_es_video, &b_video );
if( p_sys->p_es_audio )
{
es_out_Control( p_input->p_es_out, ES_OUT_GET_ES_STATE,
p_sys->p_es_audio, &b_audio );
}
p_block->i_dts =
p_block->i_pts = input_ClockGetTS( p_input,
p_input->stream.p_selected_program,
p_sys->i_pcr );
if( b_audio && b_video )
if( b_audio )
{
block_t *p_dup = block_Duplicate( p_block );
es_out_Send( p_input->p_es_out, p_sys->p_es_video, p_block );
if( p_dup )
block_t *p_audio_block = dv_extract_audio( p_input, p_block );
if( p_audio_block )
{
es_out_Send( p_input->p_es_out, p_sys->p_es_video, p_dup );
p_audio_block->i_pts = p_audio_block->i_dts = p_block->i_dts;
es_out_Send( p_input->p_es_out, p_sys->p_es_audio, p_audio_block );
}
}
else if( b_audio )
{
es_out_Send( p_input->p_es_out, p_sys->p_es_audio, p_block );
}
else if( b_video )
{
if( b_video )
es_out_Send( p_input->p_es_out, p_sys->p_es_video, p_block );
}
else
{
block_Release( p_block );
}
p_sys->i_pcr += ( 90000 / p_sys->f_rate );
return 1;
}
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( input_thread_t * p_input,
block_t* p_frame_block )
{
demux_sys_t *p_sys = p_input->p_demux_data;
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_input, "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_input, 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;
}
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