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Commit 18aed9af authored by Rémi Denis-Courmont's avatar Rémi Denis-Courmont
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Remove linear resampler 

It is computationally more expensive than the ugly one (and has higher
priority). Yet, in my opinion, it sounds even worse.
parent e83b92d7
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modules/LIST
View file @ 18aed9af
...@@ -161,7 +161,6 @@ $Id$ ...@@ -161,7 +161,6 @@ $Id$
* kate: kate text bitstream decoder * kate: kate text bitstream decoder
* libass: Subtitle renderers using libass * libass: Subtitle renderers using libass
* libmpeg2: Mpeg2 video decoder using libmpeg2 * libmpeg2: Mpeg2 video decoder using libmpeg2
* linear_resampler: linear audio resampler
* lirc: Linux infrared control module * lirc: Linux infrared control module
* live555: rtp demux based on liveMedia (live555.com) * live555: rtp demux based on liveMedia (live555.com)
* logger: file logger plugin * logger: file logger plugin
......
modules/audio_filter/resampler/Modules.am
View file @ 18aed9af
SOURCES_ugly_resampler = ugly.c SOURCES_ugly_resampler = ugly.c
SOURCES_linear_resampler = linear.c
SOURCES_bandlimited_resampler = bandlimited.c bandlimited.h SOURCES_bandlimited_resampler = bandlimited.c bandlimited.h
libvlc_LTLIBRARIES += \ libvlc_LTLIBRARIES += \
......
modules/audio_filter/resampler/linear.c deleted 100644 → 0
View file @ e83b92d7
/*****************************************************************************
* linear.c : linear interpolation resampler
*****************************************************************************
* Copyright (C) 2002, 2006 the VideoLAN team
* $Id$
*
* Authors: Gildas Bazin <gbazin@netcourrier.com>
* Sigmund Augdal Helberg <dnumgis@videolan.org>
*
* 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>
#include <vlc_block.h>
#include <vlc_cpu.h>
/*****************************************************************************
* Local prototypes
*****************************************************************************/
static int OpenFilter ( vlc_object_t * );
static void CloseFilter( vlc_object_t * );
static block_t *Resample( filter_t *, block_t * );
#if HAVE_FPU
typedef float sample_t;
# define VLC_CODEC_NATIVE VLC_CODEC_FL32
#else
typedef int32_t sample_t;
# define VLC_CODEC_NATIVE VLC_CODEC_FI32
#endif
/*****************************************************************************
* Local structures
*****************************************************************************/
struct filter_sys_t
{
sample_t *p_prev_sample; /* this filter introduces a 1 sample delay */
unsigned int i_remainder; /* remainder of previous sample */
date_t end_date;
};
/*****************************************************************************
* Module descriptor
*****************************************************************************/
vlc_module_begin ()
set_description( N_("Audio filter for linear interpolation resampling") )
set_category( CAT_AUDIO )
set_subcategory( SUBCAT_AUDIO_MISC )
set_capability( "audio filter", 5 )
set_callbacks( OpenFilter, CloseFilter )
vlc_module_end ()
/*****************************************************************************
* Resample: convert a buffer
*****************************************************************************/
static block_t *Resample( filter_t *p_filter, block_t *p_in_buf )
{
if( !p_in_buf || !p_in_buf->i_nb_samples )
{
if( p_in_buf )
block_Release( p_in_buf );
return NULL;
}
filter_sys_t *p_sys = p_filter->p_sys;
unsigned i_nb_channels = p_filter->fmt_in.audio.i_channels;
sample_t *p_prev_sample = p_sys->p_prev_sample;
/* Check if we really need to run the resampler */
if( p_filter->fmt_out.audio.i_rate == p_filter->fmt_in.audio.i_rate )
{
if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) )
{
p_in_buf = block_Realloc( p_in_buf,
sizeof(sample_t) * i_nb_channels,
p_in_buf->i_buffer );
if( !p_in_buf )
return NULL;
memcpy( p_in_buf->p_buffer, p_prev_sample,
i_nb_channels * sizeof(sample_t) );
}
return p_in_buf;
}
unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
p_filter->fmt_out.audio.i_bitspersample / 8;
size_t i_out_size = i_bytes_per_frame * (1 + (p_in_buf->i_nb_samples *
p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate));
block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
if( !p_out_buf )
goto out;
sample_t *p_out = (sample_t *)p_out_buf->p_buffer;
unsigned i_in_nb = p_in_buf->i_nb_samples;
unsigned i_out = 0;
const sample_t *p_in = (sample_t *)p_in_buf->p_buffer;
/* Take care of the previous input sample (if any) */
if( p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY )
{
p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
p_sys->i_remainder = 0;
date_Init( &p_sys->end_date, p_filter->fmt_out.audio.i_rate, 1 );
}
else
{
while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
{
for( unsigned i = 0; i < i_nb_channels ; i++ )
{
p_out[i] = p_prev_sample[i];
#if HAVE_FPU
p_out[i] += (p_in[i] - p_prev_sample[i])
#else
p_out[i] += (int64_t)(p_in[i] - p_prev_sample[i])
#endif
* p_sys->i_remainder / p_filter->fmt_out.audio.i_rate;
}
p_out += i_nb_channels;
i_out++;
p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
}
p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
}
/* Take care of the current input samples (minus last one) */
for( unsigned i_in = 0; i_in < i_in_nb - 1; i_in++ )
{
while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
{
for( unsigned i = 0; i < i_nb_channels ; i++ )
{
p_out[i] = p_in[i];
#if HAVE_FPU
p_out[i] += (p_in[i + i_nb_channels] - p_in[i])
#else
p_out[i] += (int64_t)(p_in[i + i_nb_channels] - p_in[i])
#endif
* p_sys->i_remainder / p_filter->fmt_out.audio.i_rate;
}
p_out += i_nb_channels;
i_out++;
p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
}
p_in += i_nb_channels;
p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
}
/* Backup the last input sample for next time */
memcpy( p_prev_sample, p_in, i_nb_channels * sizeof(sample_t) );
p_out_buf->i_nb_samples = i_out;
p_out_buf->i_pts = p_in_buf->i_pts;
if( p_in_buf->i_pts !=
date_Get( &p_sys->end_date ) )
{
date_Set( &p_sys->end_date, p_in_buf->i_pts );
}
p_out_buf->i_length = date_Increment( &p_sys->end_date,
p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
p_out_buf->i_buffer = p_out_buf->i_nb_samples *
i_nb_channels * sizeof(sample_t);
out:
block_Release( p_in_buf );
return p_out_buf;
}
/*****************************************************************************
* OpenFilter:
*****************************************************************************/
static int OpenFilter( vlc_object_t *p_this )
{
filter_t *p_filter = (filter_t *)p_this;
filter_sys_t *p_sys;
int i_out_rate = p_filter->fmt_out.audio.i_rate;
if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
p_filter->fmt_in.i_codec != VLC_CODEC_NATIVE )
{
return VLC_EGENERIC;
}
/* Allocate the memory needed to store the module's structure */
p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
if( p_sys == NULL )
return VLC_ENOMEM;
p_sys->p_prev_sample = malloc(
p_filter->fmt_in.audio.i_channels * sizeof(sample_t) );
if( p_sys->p_prev_sample == NULL )
{
free( p_sys );
return VLC_ENOMEM;
}
date_Init( &p_sys->end_date, p_filter->fmt_in.audio.i_rate, 1 );
p_sys->i_remainder = 0;
p_filter->pf_audio_filter = Resample;
msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
(char *)&p_filter->fmt_in.i_codec,
p_filter->fmt_in.audio.i_rate,
p_filter->fmt_in.audio.i_channels,
(char *)&p_filter->fmt_out.i_codec,
p_filter->fmt_out.audio.i_rate,
p_filter->fmt_out.audio.i_channels);
p_filter->fmt_out = p_filter->fmt_in;
p_filter->fmt_out.audio.i_rate = i_out_rate;
return 0;
}
/*****************************************************************************
* CloseFilter : deallocate data structures
*****************************************************************************/
static void CloseFilter( vlc_object_t *p_this )
{
filter_t *p_filter = (filter_t *)p_this;
free( p_filter->p_sys->p_prev_sample );
free( p_filter->p_sys );
}
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