* This plugin makes use of liba52 to decode A/52 audio
* (http://liba52.sf.net/).
*****************************************************************************
- * Copyright (C) 2001, 2002 the VideoLAN team
+ * Copyright (C) 2001-2009 the VideoLAN team
* $Id$
*
* Authors: Gildas Bazin <gbazin@videolan.org>
#include <vlc_common.h>
#include <vlc_plugin.h>
+#include <vlc_cpu.h>
#include <stdint.h> /* int16_t .. */
-#ifdef HAVE_UNISTD_H
-# include <unistd.h>
+#if !HAVE_FPU
+# define LIBA52_FIXED
#endif
-
#ifdef USE_A52DEC_TREE /* liba52 header file */
# include "include/a52.h"
#else
/*****************************************************************************
* Local prototypes
*****************************************************************************/
-static int Create ( vlc_object_t * );
-static void Destroy ( vlc_object_t * );
-static void DoWork ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
- aout_buffer_t * );
-static int Open ( vlc_object_t *, filter_sys_t *,
- audio_format_t, audio_format_t );
-
static int OpenFilter ( vlc_object_t * );
static void CloseFilter( vlc_object_t * );
static block_t *Convert( filter_t *, block_t * );
set_description( N_("ATSC A/52 (AC-3) audio decoder") )
set_category( CAT_INPUT )
set_subcategory( SUBCAT_INPUT_ACODEC )
- add_bool( "a52-dynrng", 1, NULL, DYNRNG_TEXT, DYNRNG_LONGTEXT, false )
- add_bool( "a52-upmix", 0, NULL, UPMIX_TEXT, UPMIX_LONGTEXT, true )
+ add_bool( "a52-dynrng", true, NULL, DYNRNG_TEXT, DYNRNG_LONGTEXT, false )
+ add_bool( "a52-upmix", false, NULL, UPMIX_TEXT, UPMIX_LONGTEXT, true )
set_capability( "audio filter", 100 )
- set_callbacks( Create, Destroy )
-
- add_submodule ()
- set_description( N_("ATSC A/52 (AC-3) audio decoder") )
- set_capability( "audio filter2", 100 )
set_callbacks( OpenFilter, CloseFilter )
vlc_module_end ()
-/*****************************************************************************
- * Create:
- *****************************************************************************/
-static int Create( vlc_object_t *p_this )
-{
- aout_filter_t *p_filter = (aout_filter_t *)p_this;
- filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
- int i_ret;
-
- if ( p_filter->input.i_format != VLC_CODEC_A52
-#ifdef LIBA52_FIXED
- || p_filter->output.i_format != VLC_CODEC_FI32 )
-#else
- || p_filter->output.i_format != VLC_CODEC_FL32 )
-#endif
- {
- return -1;
- }
-
- if ( p_filter->input.i_rate != p_filter->output.i_rate )
- {
- return -1;
- }
-
- /* Allocate the memory needed to store the module's structure */
- p_sys = malloc( sizeof(filter_sys_t) );
- p_filter->p_sys = (struct aout_filter_sys_t *)p_sys;
- if( p_sys == NULL )
- return -1;
-
- i_ret = Open( VLC_OBJECT(p_filter), p_sys,
- p_filter->input, p_filter->output );
-
- p_filter->pf_do_work = DoWork;
- p_filter->b_in_place = 0;
-
- return i_ret;
-}
-
/*****************************************************************************
* Open:
*****************************************************************************/
/*****************************************************************************
* Interleave: helper function to interleave channels
*****************************************************************************/
-static void Interleave( float * p_out, const float * p_in, int i_nb_channels,
- int *pi_chan_table )
+static void Interleave( sample_t * p_out, const sample_t * p_in,
+ int i_nb_channels, int *pi_chan_table )
{
/* We do not only have to interleave, but also reorder the channels */
/*****************************************************************************
* Duplicate: helper function to duplicate a unique channel
*****************************************************************************/
-static void Duplicate( float * p_out, const float * p_in )
+static void Duplicate( sample_t * p_out, const sample_t * p_in )
{
int i;
/*****************************************************************************
* Exchange: helper function to exchange left & right channels
*****************************************************************************/
-static void Exchange( float * p_out, const float * p_in )
+static void Exchange( sample_t * p_out, const sample_t * p_in )
{
int i;
- const float * p_first = p_in + 256;
- const float * p_second = p_in;
+ const sample_t * p_first = p_in + 256;
+ const sample_t * p_second = p_in;
for ( i = 0; i < 256; i++ )
{
/*****************************************************************************
* DoWork: decode an ATSC A/52 frame.
*****************************************************************************/
-static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
+static void DoWork( filter_t * p_filter,
aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
{
- filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
+ filter_sys_t *p_sys = p_filter->p_sys;
#ifdef LIBA52_FIXED
sample_t i_sample_level = (1 << 24);
#else
#endif
int i_flags = p_sys->i_flags;
int i_bytes_per_block = 256 * p_sys->i_nb_channels
- * sizeof(float);
+ * sizeof(sample_t);
int i;
/* Do the actual decoding now. */
if ( (i_flags & A52_CHANNEL_MASK) != (p_sys->i_flags & A52_CHANNEL_MASK)
&& !p_sys->b_dontwarn )
{
- msg_Warn( p_aout,
+ msg_Warn( p_filter,
"liba52 couldn't do the requested downmix 0x%x->0x%x",
p_sys->i_flags & A52_CHANNEL_MASK,
i_flags & A52_CHANNEL_MASK );
if( a52_block( p_sys->p_liba52 ) )
{
- msg_Warn( p_aout, "a52_block failed for block %d", i );
+ msg_Warn( p_filter, "a52_block failed for block %d", i );
}
p_samples = a52_samples( p_sys->p_liba52 );
if ( ((p_sys->i_flags & A52_CHANNEL_MASK) == A52_CHANNEL1
|| (p_sys->i_flags & A52_CHANNEL_MASK) == A52_CHANNEL2
|| (p_sys->i_flags & A52_CHANNEL_MASK) == A52_MONO)
- && (p_filter->output.i_physical_channels
+ && (p_filter->fmt_out.audio.i_physical_channels
& (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT)) )
{
- Duplicate( (float *)(p_out_buf->p_buffer + i * i_bytes_per_block),
+ Duplicate( (sample_t *)(p_out_buf->p_buffer + i * i_bytes_per_block),
p_samples );
}
- else if ( p_filter->output.i_original_channels
+ else if ( p_filter->fmt_out.audio.i_original_channels
& AOUT_CHAN_REVERSESTEREO )
{
- Exchange( (float *)(p_out_buf->p_buffer + i * i_bytes_per_block),
+ Exchange( (sample_t *)(p_out_buf->p_buffer + i * i_bytes_per_block),
p_samples );
}
else
{
/* Interleave the *$£%ù samples. */
- Interleave( (float *)(p_out_buf->p_buffer + i * i_bytes_per_block),
+ Interleave( (sample_t *)(p_out_buf->p_buffer + i * i_bytes_per_block),
p_samples, p_sys->i_nb_channels, p_sys->pi_chan_table);
}
}
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
- p_out_buf->i_nb_bytes = i_bytes_per_block * 6;
-}
-
-/*****************************************************************************
- * Destroy : deallocate data structures
- *****************************************************************************/
-static void Destroy( vlc_object_t *p_this )
-{
- aout_filter_t *p_filter = (aout_filter_t *)p_this;
- filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
-
- a52_free( p_sys->p_liba52 );
- free( p_sys );
+ p_out_buf->i_buffer = i_bytes_per_block * 6;
}
/*****************************************************************************
static block_t *Convert( filter_t *p_filter, block_t *p_block )
{
- aout_filter_t aout_filter;
- aout_buffer_t in_buf, out_buf;
- block_t *p_out;
- int i_out_size;
-
- if( !p_block || !p_block->i_samples )
+ if( !p_block || !p_block->i_nb_samples )
{
if( p_block )
block_Release( p_block );
return NULL;
}
- i_out_size = p_block->i_samples *
+ size_t i_out_size = p_block->i_nb_samples *
p_filter->fmt_out.audio.i_bitspersample *
p_filter->fmt_out.audio.i_channels / 8;
- p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
+ block_t *p_out = filter_NewAudioBuffer( p_filter, i_out_size );
if( !p_out )
{
msg_Warn( p_filter, "can't get output buffer" );
return NULL;
}
- p_out->i_samples = p_block->i_samples;
+ p_out->i_nb_samples = p_block->i_nb_samples;
p_out->i_dts = p_block->i_dts;
p_out->i_pts = p_block->i_pts;
p_out->i_length = p_block->i_length;
- aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
- aout_filter.input = p_filter->fmt_in.audio;
- aout_filter.input.i_format = p_filter->fmt_in.i_codec;
- aout_filter.output = p_filter->fmt_out.audio;
- aout_filter.output.i_format = p_filter->fmt_out.i_codec;
-
- in_buf.p_buffer = p_block->p_buffer;
- in_buf.i_nb_bytes = p_block->i_buffer;
- in_buf.i_nb_samples = p_block->i_samples;
- out_buf.p_buffer = p_out->p_buffer;
- out_buf.i_nb_bytes = p_out->i_buffer;
- out_buf.i_nb_samples = p_out->i_samples;
-
- DoWork( (aout_instance_t *)p_filter, &aout_filter, &in_buf, &out_buf );
-
- p_out->i_buffer = out_buf.i_nb_bytes;
- p_out->i_samples = out_buf.i_nb_samples;
+ DoWork( p_filter, p_block, p_out );
block_Release( p_block );
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