Commit e83b92d7 authored by Rémi Denis-Courmont's avatar Rémi Denis-Courmont

Remove the trivial "resampler"

The ugly resampler is almost as fast yet way better.
parent 5027135e
......@@ -331,7 +331,6 @@ $Id$
* tremor: a vorbis audio decoder using the libvorbisidec (aka tremor) library
* trivial_channel_mixer: Simple channel mixer plugin
* trivial_mixer: Trivial audio mixer plugin
* trivial_resampler: Simple audio resampler
* ts: MPEG-TS demuxer
* tta: Lossless True Audio parser
* twolame: a mp1 mp2 audio encoder based on twolame
......
SOURCES_trivial_resampler = trivial.c
SOURCES_ugly_resampler = ugly.c
SOURCES_linear_resampler = linear.c
SOURCES_bandlimited_resampler = bandlimited.c bandlimited.h
......
/*****************************************************************************
* trivial.c : trivial resampler (skips samples or pads with zeroes)
*****************************************************************************
* Copyright (C) 2002-2009 the VideoLAN team
* $Id$
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
* 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.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <vlc_common.h>
#include <vlc_plugin.h>
#include <vlc_aout.h>
#include <vlc_filter.h>
/*****************************************************************************
* Local prototypes
*****************************************************************************/
static int Create ( vlc_object_t * );
static block_t *DoWork( filter_t *, block_t * );
/*****************************************************************************
* Module descriptor
*****************************************************************************/
vlc_module_begin ()
set_description( N_("Audio filter for trivial resampling") )
set_capability( "audio filter", 1 )
set_category( CAT_AUDIO )
set_subcategory( SUBCAT_AUDIO_MISC )
set_callbacks( Create, NULL )
vlc_module_end ()
/*****************************************************************************
* Create: allocate trivial resampler
*****************************************************************************/
static int Create( vlc_object_t *p_this )
{
filter_t * p_filter = (filter_t *)p_this;
if ( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
|| p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
|| p_filter->fmt_in.audio.i_physical_channels
!= p_filter->fmt_out.audio.i_physical_channels
|| p_filter->fmt_in.audio.i_original_channels
!= p_filter->fmt_out.audio.i_original_channels
|| (p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32
&& p_filter->fmt_in.audio.i_format != VLC_CODEC_FI32) )
{
return VLC_EGENERIC;
}
p_filter->pf_audio_filter = DoWork;
return VLC_SUCCESS;
}
/*****************************************************************************
* DoWork: convert a buffer
*****************************************************************************/
static block_t *DoWork( filter_t * p_filter, block_t * p_block )
{
/* Check if we really need to run the resampler */
if( p_filter->fmt_out.audio.i_rate == p_filter->fmt_in.audio.i_rate )
return p_block;
int i_in_nb = p_block->i_nb_samples;
int i_out_nb = i_in_nb * p_filter->fmt_out.audio.i_rate
/ p_filter->fmt_in.audio.i_rate;
int i_sample_bytes = aout_FormatNbChannels( &p_filter->fmt_in.audio )
* sizeof(int32_t);
p_block = block_Realloc( p_block, 0, i_out_nb * i_sample_bytes );
if( !p_block )
return NULL;
if( i_out_nb > i_in_nb )
{
/* Pad with zeroes. */
memset( p_block->p_buffer + i_in_nb * i_sample_bytes,
0, (i_out_nb - i_in_nb) * i_sample_bytes );
}
p_block->i_nb_samples = i_out_nb;
p_block->i_length = p_block->i_nb_samples *
CLOCK_FREQ / p_filter->fmt_out.audio.i_rate;
return p_block;
}
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