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Commit 1d5118e0 authored by Jon Lech Johansen's avatar Jon Lech Johansen
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* DTS S/PDIF support.

parent 03fc820b
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Showing with 553 additions and 2 deletions
configure.ac.in
View file @ 1d5118e0
...@@ -801,9 +801,9 @@ dnl ...@@ -801,9 +801,9 @@ dnl
BUILTINS="${BUILTINS} mpeg_video idct idctclassic motion" BUILTINS="${BUILTINS} mpeg_video idct idctclassic motion"
PLUGINS="${PLUGINS} dummy rc logger gestures access_file memcpy" PLUGINS="${PLUGINS} dummy rc logger gestures access_file memcpy"
PLUGINS="${PLUGINS} es audio m4v mpeg_system ps ts avi asf aac mp4 rawdv" PLUGINS="${PLUGINS} es audio m4v mpeg_system ps ts avi asf aac mp4 rawdv"
PLUGINS="${PLUGINS} spudec mpeg_audio lpcm a52 cinepak" PLUGINS="${PLUGINS} spudec mpeg_audio lpcm a52 dts cinepak"
PLUGINS="${PLUGINS} deinterlace invert adjust wall transform distort clone crop motionblur" PLUGINS="${PLUGINS} deinterlace invert adjust wall transform distort clone crop motionblur"
PLUGINS="${PLUGINS} float32tos16 float32tos8 float32tou16 float32tou8 a52tospdif fixed32tofloat32 fixed32tos16 s16tofloat32 s16tofloat32swab s8tofloat32 u8tofixed32 u8tofloat32" PLUGINS="${PLUGINS} float32tos16 float32tos8 float32tou16 float32tou8 a52tospdif dtstospdif fixed32tofloat32 fixed32tos16 s16tofloat32 s16tofloat32swab s8tofloat32 u8tofixed32 u8tofloat32"
PLUGINS="${PLUGINS} trivial_resampler ugly_resampler linear_resampler bandlimited_resampler" PLUGINS="${PLUGINS} trivial_resampler ugly_resampler linear_resampler bandlimited_resampler"
PLUGINS="${PLUGINS} trivial_channel_mixer headphone_channel_mixer" PLUGINS="${PLUGINS} trivial_channel_mixer headphone_channel_mixer"
PLUGINS="${PLUGINS} trivial_mixer spdif_mixer float32_mixer" PLUGINS="${PLUGINS} trivial_mixer spdif_mixer float32_mixer"
......
modules/audio_filter/converter/Modules.am
View file @ 1d5118e0
...@@ -4,6 +4,7 @@ SOURCES_float32tou16 = modules/audio_filter/converter/float32tou16.c ...@@ -4,6 +4,7 @@ SOURCES_float32tou16 = modules/audio_filter/converter/float32tou16.c
SOURCES_float32tou8 = modules/audio_filter/converter/float32tou8.c SOURCES_float32tou8 = modules/audio_filter/converter/float32tou8.c
SOURCES_a52tospdif = modules/audio_filter/converter/a52tospdif.c SOURCES_a52tospdif = modules/audio_filter/converter/a52tospdif.c
SOURCES_a52tofloat32 = modules/audio_filter/converter/a52tofloat32.c SOURCES_a52tofloat32 = modules/audio_filter/converter/a52tofloat32.c
SOURCES_dtstospdif = modules/audio_filter/converter/dtstospdif.c
SOURCES_fixed32tos16 = modules/audio_filter/converter/fixed32tos16.c SOURCES_fixed32tos16 = modules/audio_filter/converter/fixed32tos16.c
SOURCES_fixed32tofloat32 = modules/audio_filter/converter/fixed32tofloat32.c SOURCES_fixed32tofloat32 = modules/audio_filter/converter/fixed32tofloat32.c
SOURCES_s16tofloat32 = modules/audio_filter/converter/s16tofloat32.c SOURCES_s16tofloat32 = modules/audio_filter/converter/s16tofloat32.c
......
modules/audio_filter/converter/dtstospdif.c 0 → 100644
View file @ 1d5118e0
/*****************************************************************************
* dtstospdif.c : encapsulates DTS frames into S/PDIF packets
*****************************************************************************
* Copyright (C) 2003 VideoLAN
* $Id: dtstospdif.c,v 1.1 2003/03/09 20:07:47 jlj Exp $
*
* Authors: Jon Lech Johansen <jon-vl@nanocrew.net>
*
* 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., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
#include <stdlib.h> /* malloc(), free() */
#include <string.h>
#include <vlc/vlc.h>
#ifdef HAVE_UNISTD_H
# include <unistd.h>
#endif
#include "audio_output.h"
#include "aout_internal.h"
/*****************************************************************************
* Local prototypes
*****************************************************************************/
static int Create ( vlc_object_t * );
static void DoWork ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
aout_buffer_t * );
/*****************************************************************************
* Module descriptor
*****************************************************************************/
vlc_module_begin();
set_description( _("audio filter for DTS->S/PDIF encapsulation") );
set_capability( "audio filter", 10 );
set_callbacks( Create, NULL );
vlc_module_end();
/*****************************************************************************
* Create:
*****************************************************************************/
static int Create( vlc_object_t *p_this )
{
aout_filter_t * p_filter = (aout_filter_t *)p_this;
if ( p_filter->input.i_format != VLC_FOURCC('d','t','s',' ')
|| p_filter->output.i_format != VLC_FOURCC('s','p','d','i') )
{
return -1;
}
p_filter->pf_do_work = DoWork;
p_filter->b_in_place = 0;
return 0;
}
/*****************************************************************************
* DoWork: convert a buffer
*****************************************************************************/
static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
{
uint16_t i_fz = (p_in_buf->i_nb_samples / 3) * 4;
uint16_t i_frame, i_length = p_in_buf->i_nb_bytes / 3;
static const uint8_t p_sync[6] = { 0x72, 0xF8, 0x1F, 0x4E, 0x00, 0x00 };
for( i_frame = 0; i_frame < 3; i_frame++ )
{
#ifndef HAVE_SWAB
uint16_t i;
byte_t * p_tmp;
#endif
byte_t * p_out = p_out_buf->p_buffer + (i_frame * i_fz);
byte_t * p_in = p_in_buf->p_buffer + (i_frame * i_length);
/* Copy the S/PDIF headers. */
memcpy( p_out, p_sync, 6 );
switch( p_in_buf->i_nb_samples / 3 )
{
case 512: *(p_out + 4) = 0x0B; break;
case 1024: *(p_out + 4) = 0x0C; break;
case 2048: *(p_out + 4) = 0x0D; break;
}
*(p_out + 6) = (i_length * 8) & 0xff;
*(p_out + 7) = (i_length * 8) >> 8;
#ifdef HAVE_SWAB
swab( p_in, p_out + 8, i_length );
#else
p_tmp = p_out + 8;
for( i = i_length / 2 ; i-- ; )
{
p_tmp[0] = p_in[1];
p_tmp[1] = p_in[0];
p_tmp += 2; p_in += 2;
}
#endif
p_filter->p_vlc->pf_memset( p_out + 8 + i_length, 0,
i_fz - i_length - 8 );
}
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples * 4;
}
modules/codec/Modules.am
View file @ 1d5118e0
SOURCES_a52 = modules/codec/a52.c SOURCES_a52 = modules/codec/a52.c
SOURCES_dts = modules/codec/dts.c
SOURCES_flacdec = modules/codec/flacdec.c SOURCES_flacdec = modules/codec/flacdec.c
SOURCES_lpcm = modules/codec/lpcm.c SOURCES_lpcm = modules/codec/lpcm.c
SOURCES_araw = modules/codec/araw.c SOURCES_araw = modules/codec/araw.c
......
modules/codec/dts.c 0 → 100644
View file @ 1d5118e0
/*****************************************************************************
* dts.c: DTS basic parser
*****************************************************************************
* Copyright (C) 2003 VideoLAN
* $Id: dts.c,v 1.1 2003/03/09 20:07:47 jlj Exp $
*
* Authors: Jon Lech Johansen <jon-vl@nanocrew.net>
*
* 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., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
#include <stdio.h>
#include <stdlib.h>
#include <string.h> /* memcpy() */
#include <fcntl.h>
#include <vlc/vlc.h>
#include <vlc/decoder.h>
#include <vlc/aout.h>
#ifdef HAVE_UNISTD_H
# include <unistd.h>
#endif
/*****************************************************************************
* dec_thread_t : decoder thread descriptor
*****************************************************************************/
typedef struct dec_thread_t
{
/*
* Thread properties
*/
vlc_thread_t thread_id; /* id for thread functions */
/*
* Input properties
*/
decoder_fifo_t * p_fifo; /* stores the PES stream data */
bit_stream_t bit_stream;
/*
* Output properties
*/
aout_instance_t * p_aout; /* opaque */
aout_input_t * p_aout_input; /* opaque */
audio_sample_format_t output_format;
} dec_thread_t;
/****************************************************************************
* Local prototypes
****************************************************************************/
static int OpenDecoder ( vlc_object_t * );
static int RunDecoder ( decoder_fifo_t * );
static void EndThread ( dec_thread_t * );
static int SyncInfo ( const byte_t *, unsigned int *,
unsigned int *, unsigned int *,
unsigned int * );
/*****************************************************************************
* Module descriptor
*****************************************************************************/
vlc_module_begin();
set_description( _("DTS parser") );
set_capability( "decoder", 100 );
set_callbacks( OpenDecoder, NULL );
vlc_module_end();
/*****************************************************************************
* OpenDecoder: probe the decoder and return score
*****************************************************************************/
static int OpenDecoder( vlc_object_t *p_this )
{
decoder_fifo_t *p_fifo = (decoder_fifo_t*) p_this;
if( p_fifo->i_fourcc != VLC_FOURCC('d','t','s',' ')
&& p_fifo->i_fourcc != VLC_FOURCC('d','t','s','b') )
{
return VLC_EGENERIC;
}
p_fifo->pf_run = RunDecoder;
return VLC_SUCCESS;
}
/****************************************************************************
* RunDecoder: the whole thing
****************************************************************************
* This function is called just after the thread is launched.
****************************************************************************/
static int RunDecoder( decoder_fifo_t *p_fifo )
{
dec_thread_t * p_dec;
audio_date_t end_date;
/* Allocate the memory needed to store the thread's structure */
p_dec = malloc( sizeof(dec_thread_t) );
if( p_dec == NULL )
{
msg_Err( p_fifo, "out of memory" );
DecoderError( p_fifo );
return -1;
}
/* Initialize the thread properties */
p_dec->p_aout = NULL;
p_dec->p_aout_input = NULL;
p_dec->p_fifo = p_fifo;
p_dec->output_format.i_format = VLC_FOURCC('d','t','s',' ');
aout_DateSet( &end_date, 0 );
/* Init the bitstream */
if( InitBitstream( &p_dec->bit_stream, p_dec->p_fifo,
NULL, NULL ) != VLC_SUCCESS )
{
msg_Err( p_fifo, "cannot initialize bitstream" );
DecoderError( p_fifo );
free( p_dec );
return -1;
}
/* Decoder thread's main loop */
while( !p_dec->p_fifo->b_die && !p_dec->p_fifo->b_error )
{
int i;
mtime_t pts;
byte_t p_header[10];
unsigned int i_rate;
unsigned int i_bit_rate;
unsigned int i_frame_size;
unsigned int i_frame_length;
unsigned int i_original_channels;
aout_buffer_t * p_buffer = NULL;
for( i = 0; i < 3; i++ )
{
RealignBits( &p_dec->bit_stream );
while( (ShowBits( &p_dec->bit_stream, 32 ) ) != 0x7ffe8001 &&
(!p_dec->p_fifo->b_die) && (!p_dec->p_fifo->b_error) )
{
RemoveBits( &p_dec->bit_stream, 8 );
}
if( p_dec->p_fifo->b_die || p_dec->p_fifo->b_error ) break;
if( i == 0 )
{
/* Set the Presentation Time Stamp */
NextPTS( &p_dec->bit_stream, &pts, NULL );
if( pts != 0 && pts != aout_DateGet( &end_date ) )
{
aout_DateSet( &end_date, pts );
}
}
/* Get DTS frame header */
GetChunk( &p_dec->bit_stream, p_header, 10 );
if( p_dec->p_fifo->b_die || p_dec->p_fifo->b_error ) break;
i_frame_size = SyncInfo( p_header, &i_original_channels, &i_rate,
&i_bit_rate, &i_frame_length );
if( !i_frame_size )
{
msg_Warn( p_dec->p_fifo, "dts_syncinfo failed" );
i--; continue;
}
if( i == 0 )
{
if( (p_dec->p_aout_input != NULL) &&
( (p_dec->output_format.i_rate != i_rate)
|| (p_dec->output_format.i_original_channels
!= i_original_channels)
|| (p_dec->output_format.i_bytes_per_frame
!= i_frame_size * 3) ) )
{
/* Parameters changed - this should not happen. */
aout_DecDelete( p_dec->p_aout, p_dec->p_aout_input );
p_dec->p_aout_input = NULL;
}
/* Creating the audio input if not created yet. */
if( p_dec->p_aout_input == NULL )
{
p_dec->output_format.i_rate = i_rate;
p_dec->output_format.i_original_channels
= i_original_channels;
p_dec->output_format.i_physical_channels
= i_original_channels & AOUT_CHAN_PHYSMASK;
p_dec->output_format.i_bytes_per_frame = i_frame_size * 3;
p_dec->output_format.i_frame_length = i_frame_length * 3;
aout_DateInit( &end_date, i_rate );
p_dec->p_aout_input = aout_DecNew( p_dec->p_fifo,
&p_dec->p_aout,
&p_dec->output_format );
if( p_dec->p_aout_input == NULL )
{
p_dec->p_fifo->b_error = 1;
break;
}
}
}
if( !aout_DateGet( &end_date ) )
{
byte_t p_junk[ i_frame_size ];
/* We've just started the stream, wait for the first PTS. */
GetChunk( &p_dec->bit_stream, p_junk, i_frame_size - 10 );
i--; continue;
}
if( i == 0 )
{
p_buffer = aout_DecNewBuffer( p_dec->p_aout,
p_dec->p_aout_input,
i_frame_length * 3 );
if( p_buffer == NULL )
{
p_dec->p_fifo->b_error = 1;
break;
}
p_buffer->start_date = aout_DateGet( &end_date );
p_buffer->end_date = aout_DateIncrement( &end_date,
i_frame_length * 3 );
}
/* Get the whole frame. */
memcpy( p_buffer->p_buffer + (i * i_frame_size), p_header, 10 );
GetChunk( &p_dec->bit_stream,
p_buffer->p_buffer + (i * i_frame_size) + 10,
i_frame_size - 10 );
if( p_dec->p_fifo->b_die ) break;
}
if( p_dec->p_fifo->b_die || p_dec->p_fifo->b_error )
{
if( p_buffer != NULL )
{
aout_DecDeleteBuffer( p_dec->p_aout, p_dec->p_aout_input,
p_buffer );
}
break;
}
/* Send the buffer to the aout core. */
aout_DecPlay( p_dec->p_aout, p_dec->p_aout_input, p_buffer );
}
if( p_dec->p_fifo->b_error )
{
DecoderError( p_dec->p_fifo );
}
EndThread( p_dec );
return 0;
}
/*****************************************************************************
* EndThread : thread destruction
*****************************************************************************/
static void EndThread( dec_thread_t * p_dec )
{
if ( p_dec->p_aout_input != NULL )
{
aout_DecDelete( p_dec->p_aout, p_dec->p_aout_input );
}
CloseBitstream( &p_dec->bit_stream );
free( p_dec );
}
/*****************************************************************************
* SyncInfo: parse DTS sync info
*****************************************************************************/
static int SyncInfo( const byte_t * p_buf, unsigned int * pi_channels,
unsigned int * pi_sample_rate,
unsigned int * pi_bit_rate,
unsigned int * pi_frame_length )
{
unsigned int i_bit_rate;
unsigned int i_audio_mode;
unsigned int i_sample_rate;
unsigned int i_frame_size;
unsigned int i_frame_length;
static const unsigned int ppi_dts_samplerate[] =
{
0, 8000, 16000, 32000, 64000, 128000,
11025, 22050, 44010, 88020, 176400,
12000, 24000, 48000, 96000, 192000
};
static const unsigned int ppi_dts_bitrate[] =
{
32000, 56000, 64000, 96000, 112000, 128000,
192000, 224000, 256000, 320000, 384000,
448000, 512000, 576000, 640000, 768000,
896000, 1024000, 1152000, 1280000, 1344000,
1408000, 1411200, 1472000, 1536000, 1920000,
2048000, 3072000, 3840000, 4096000, 0, 0
};
if( ((uint32_t*)p_buf)[0] != 0x7ffe8001 )
{
return( 0 );
}
i_frame_length = (p_buf[4] & 0x01) << 6 | (p_buf[5] >> 2);
i_frame_size = (p_buf[5] & 0x03) << 12 | (p_buf[6] << 4) |
(p_buf[7] >> 4);
i_audio_mode = (p_buf[7] & 0x0f) << 2 | (p_buf[8] >> 6);
i_sample_rate = (p_buf[8] >> 2) & 0x0f;
i_bit_rate = (p_buf[8] & 0x03) << 3 | ((p_buf[9] >> 5) & 0x07);
switch( i_audio_mode )
{
case 0x0:
/* Mono */
*pi_channels = AOUT_CHAN_CENTER;
break;
case 0x1:
/* Dual-mono = stereo + dual-mono */
*pi_channels = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT |
AOUT_CHAN_DUALMONO;
break;
case 0x2:
case 0x3:
case 0x4:
/* Stereo */
*pi_channels = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT;
break;
case 0x5:
/* 3F */
*pi_channels = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER;
break;
case 0x6:
/* 2F/LFE */
*pi_channels = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_LFE;
break;
case 0x7:
/* 3F/LFE */
*pi_channels = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT |
AOUT_CHAN_CENTER | AOUT_CHAN_LFE;
break;
case 0x8:
/* 2F2R */
*pi_channels = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT |
AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT;
break;
case 0x9:
/* 3F2R */
*pi_channels = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT |
AOUT_CHAN_CENTER | AOUT_CHAN_REARLEFT |
AOUT_CHAN_REARRIGHT;
break;
case 0xA:
case 0xB:
/* 2F2M2R */
*pi_channels = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT |
AOUT_CHAN_MIDDLELEFT | AOUT_CHAN_MIDDLERIGHT |
AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT;
break;
case 0xC:
/* 3F2M2R */
*pi_channels = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT |
AOUT_CHAN_CENTER | AOUT_CHAN_MIDDLELEFT |
AOUT_CHAN_MIDDLERIGHT | AOUT_CHAN_REARLEFT |
AOUT_CHAN_REARRIGHT;
break;
case 0xD:
case 0xE:
/* 3F2M2R/LFE */
*pi_channels = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT |
AOUT_CHAN_CENTER | AOUT_CHAN_MIDDLELEFT |
AOUT_CHAN_MIDDLERIGHT | AOUT_CHAN_REARLEFT |
AOUT_CHAN_REARRIGHT | AOUT_CHAN_LFE;
break;
default:
return( 0 );
}
if( i_sample_rate >= sizeof( ppi_dts_samplerate ) /
sizeof( ppi_dts_samplerate[0] ) )
{
return( 0 );
}
*pi_sample_rate = ppi_dts_samplerate[ i_sample_rate ];
if( i_bit_rate >= sizeof( ppi_dts_bitrate ) /
sizeof( ppi_dts_bitrate[0] ) )
{
return( 0 );
}
*pi_bit_rate = ppi_dts_bitrate[ i_bit_rate ];
*pi_frame_length = (i_frame_length + 1) * 32;
return( i_frame_size + 1 );
}
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