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vlc/modules/audio_filter/resampler/bandlimited.c
Clément Stenac a90a19a6b0 Improvements to preferences 
* Each module can declare a "human-readable short name" with set_name
* Modules are sorted by category (set_category, set_subcategory).
  Modules configs can be separated by set_section()
* Separated audio-filter and audio-visual
* Separated extraintf and control
* New command and widget : add_module_list() for comma-separated modules
* Vfilters now use "," as separator
2004-12-11 14:45:46 +00:00

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/*****************************************************************************
* bandlimited.c : band-limited interpolation resampler
*****************************************************************************
* Copyright (C) 2002 VideoLAN
* $Id$
*
* Authors: Gildas Bazin <gbazin@netcourrier.com>
*
* 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:
*
* This implementation of the band-limited interpolationis based on the
* following paper:
* http://ccrma-www.stanford.edu/~jos/resample/resample.html
*
* It uses a Kaiser-windowed sinc-function low-pass filter and the width of the
* filter is 13 samples.
*
*****************************************************************************/
#include <stdlib.h> /* malloc(), free() */
#include <string.h>
#include <vlc/vlc.h>
#include "audio_output.h"
#include "aout_internal.h"
#include "bandlimited.h"
/*****************************************************************************
* Local prototypes
*****************************************************************************/
static int Create ( vlc_object_t * );
static void Close ( vlc_object_t * );
static void DoWork ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
aout_buffer_t * );
static void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing,
float *f_in, float *f_out, uint32_t ui_remainder,
uint32_t ui_output_rate, int16_t Inc,
int i_nb_channels );
static void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing,
float *f_in, float *f_out, uint32_t ui_remainder,
uint32_t ui_output_rate, uint32_t ui_input_rate,
int16_t Inc, int i_nb_channels );
/*****************************************************************************
* Local structures
*****************************************************************************/
struct aout_filter_sys_t
{
int32_t *p_buf; /* this filter introduces a delay */
int i_buf_size;
int i_old_rate;
double d_old_factor;
int i_old_wing;
unsigned int i_remainder; /* remainder of previous sample */
audio_date_t end_date;
};
/*****************************************************************************
* Module descriptor
*****************************************************************************/
vlc_module_begin();
set_category( CAT_AUDIO );
set_subcategory( SUBCAT_AUDIO_MISC );
set_description( _("audio filter for band-limited interpolation resampling") );
set_capability( "audio filter", 20 );
set_callbacks( Create, Close );
vlc_module_end();
/*****************************************************************************
* Create: allocate linear resampler
*****************************************************************************/
static int Create( vlc_object_t *p_this )
{
aout_filter_t * p_filter = (aout_filter_t *)p_this;
double d_factor;
int i_filter_wing;
if ( p_filter->input.i_rate == p_filter->output.i_rate
|| p_filter->input.i_format != p_filter->output.i_format
|| p_filter->input.i_physical_channels
!= p_filter->output.i_physical_channels
|| p_filter->input.i_original_channels
!= p_filter->output.i_original_channels
|| p_filter->input.i_format != VLC_FOURCC('f','l','3','2') )
{
return VLC_EGENERIC;
}
#if !defined( SYS_DARWIN )
if( !config_GetInt( p_this, "hq-resampling" ) )
{
return VLC_EGENERIC;
}
#endif
/* Allocate the memory needed to store the module's structure */
p_filter->p_sys = malloc( sizeof(struct aout_filter_sys_t) );
if( p_filter->p_sys == NULL )
{
msg_Err( p_filter, "out of memory" );
return VLC_ENOMEM;
}
/* Calculate worst case for the length of the filter wing */
d_factor = (double)p_filter->output.i_rate
/ p_filter->input.i_rate;
i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
* __MAX(1.0, 1.0/d_factor) + 10;
p_filter->p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) *
sizeof(int32_t) * 2 * i_filter_wing;
/* Allocate enough memory to buffer previous samples */
p_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size );
if( p_filter->p_sys->p_buf == NULL )
{
msg_Err( p_filter, "out of memory" );
return VLC_ENOMEM;
}
p_filter->p_sys->i_old_wing = 0;
p_filter->pf_do_work = DoWork;
/* We don't want a new buffer to be created because we're not sure we'll
* actually need to resample anything. */
p_filter->b_in_place = VLC_TRUE;
return VLC_SUCCESS;
}
/*****************************************************************************
* Close: free our resources
*****************************************************************************/
static void Close( vlc_object_t * p_this )
{
aout_filter_t * p_filter = (aout_filter_t *)p_this;
free( p_filter->p_sys->p_buf );
free( p_filter->p_sys );
}
/*****************************************************************************
* 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 )
{
float *p_in, *p_in_orig, *p_out = (float *)p_out_buf->p_buffer;
int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
int i_in_nb = p_in_buf->i_nb_samples;
int i_in, i_out = 0;
double d_factor, d_scale_factor, d_old_scale_factor;
int i_filter_wing;
#if 0
int i;
#endif
/* Check if we really need to run the resampler */
if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
{
if( //p_filter->b_continuity && /* What difference does it make ? :) */
p_filter->p_sys->i_old_wing &&
p_in_buf->i_size >=
p_in_buf->i_nb_bytes + p_filter->p_sys->i_old_wing *
p_filter->input.i_bytes_per_frame )
{
/* output the whole thing with the samples from last time */
memmove( ((float *)(p_in_buf->p_buffer)) +
i_nb_channels * p_filter->p_sys->i_old_wing,
p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
memcpy( p_in_buf->p_buffer, p_filter->p_sys->p_buf +
i_nb_channels * p_filter->p_sys->i_old_wing,
p_filter->p_sys->i_old_wing *
p_filter->input.i_bytes_per_frame );
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
p_filter->p_sys->i_old_wing;
p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
p_out_buf->end_date =
aout_DateIncrement( &p_filter->p_sys->end_date,
p_out_buf->i_nb_samples );
p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
p_filter->input.i_bytes_per_frame;
}
p_filter->b_continuity = VLC_FALSE;
p_filter->p_sys->i_old_wing = 0;
return;
}
if( !p_filter->b_continuity )
{
/* Continuity in sound samples has been broken, we'd better reset
* everything. */
p_filter->b_continuity = VLC_TRUE;
p_filter->p_sys->i_remainder = 0;
aout_DateInit( &p_filter->p_sys->end_date, p_filter->output.i_rate );
aout_DateSet( &p_filter->p_sys->end_date, p_in_buf->start_date );
p_filter->p_sys->i_old_rate = p_filter->input.i_rate;
p_filter->p_sys->d_old_factor = 1;
p_filter->p_sys->i_old_wing = 0;
}
#if 0
msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
p_filter->p_sys->i_old_rate, p_filter->p_sys->d_old_factor,
p_filter->p_sys->i_old_wing, i_in_nb );
#endif
/* Prepare the source buffer */
i_in_nb += (p_filter->p_sys->i_old_wing * 2);
#ifdef HAVE_ALLOCA
p_in = p_in_orig = (float *)alloca( i_in_nb *
p_filter->input.i_bytes_per_frame );
#else
p_in = p_in_orig = (float *)malloc( i_in_nb *
p_filter->input.i_bytes_per_frame );
#endif
if( p_in == NULL )
{
return;
}
/* Copy all our samples in p_in */
if( p_filter->p_sys->i_old_wing )
{
p_aout->p_vlc->pf_memcpy( p_in, p_filter->p_sys->p_buf,
p_filter->p_sys->i_old_wing * 2 *
p_filter->input.i_bytes_per_frame );
}
p_aout->p_vlc->pf_memcpy( p_in + p_filter->p_sys->i_old_wing * 2 *
i_nb_channels, p_in_buf->p_buffer,
p_in_buf->i_nb_samples *
p_filter->input.i_bytes_per_frame );
/* Make sure the output buffer is reset */
memset( p_out, 0, p_out_buf->i_size );
/* Calculate the new length of the filter wing */
d_factor = (double)p_aout->mixer.mixer.i_rate / p_filter->input.i_rate;
i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
/* Account for increased filter gain when using factors less than 1 */
d_old_scale_factor = SMALL_FILTER_SCALE *
p_filter->p_sys->d_old_factor + 0.5;
d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
/* Apply the old rate until we have enough samples for the new one */
i_in = p_filter->p_sys->i_old_wing;
p_in += p_filter->p_sys->i_old_wing * i_nb_channels;
for( ; i_in < i_filter_wing &&
(i_in + p_filter->p_sys->i_old_wing) < i_in_nb; i_in++ )
{
if( p_filter->p_sys->d_old_factor == 1 )
{
/* Just copy the samples */
memcpy( p_out, p_in,
p_filter->input.i_bytes_per_frame );
p_in += i_nb_channels;
p_out += i_nb_channels;
i_out++;
continue;
}
while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
{
if( p_filter->p_sys->d_old_factor >= 1 )
{
/* FilterFloatUP() is faster if we can use it */
/* Perform left-wing inner product */
FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in, p_out,
p_filter->p_sys->i_remainder,
p_filter->output.i_rate,
-1, i_nb_channels );
/* Perform right-wing inner product */
FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
p_filter->output.i_rate -
p_filter->p_sys->i_remainder,
p_filter->output.i_rate,
1, i_nb_channels );
#if 0
/* Normalize for unity filter gain */
for( i = 0; i < i_nb_channels; i++ )
{
*(p_out+i) *= d_old_scale_factor;
}
#endif
/* Sanity check */
if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
<= (unsigned int)i_out+1 )
{
p_out += i_nb_channels;
i_out++;
p_filter->p_sys->i_remainder += p_filter->input.i_rate;
break;
}
}
else
{
/* Perform left-wing inner product */
FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in, p_out,
p_filter->p_sys->i_remainder,
p_filter->output.i_rate, p_filter->input.i_rate,
-1, i_nb_channels );
/* Perform right-wing inner product */
FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
p_filter->output.i_rate -
p_filter->p_sys->i_remainder,
p_filter->output.i_rate, p_filter->input.i_rate,
1, i_nb_channels );
}
p_out += i_nb_channels;
i_out++;
p_filter->p_sys->i_remainder += p_filter->input.i_rate;
}
p_in += i_nb_channels;
p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
}
/* Apply the new rate for the rest of the samples */
if( i_in < i_in_nb - i_filter_wing )
{
p_filter->p_sys->i_old_rate = p_filter->input.i_rate;
p_filter->p_sys->d_old_factor = d_factor;
p_filter->p_sys->i_old_wing = i_filter_wing;
}
for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
{
while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
{
if( d_factor >= 1 )
{
/* FilterFloatUP() is faster if we can use it */
/* Perform left-wing inner product */
FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in, p_out,
p_filter->p_sys->i_remainder,
p_filter->output.i_rate,
-1, i_nb_channels );
/* Perform right-wing inner product */
FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
p_filter->output.i_rate -
p_filter->p_sys->i_remainder,
p_filter->output.i_rate,
1, i_nb_channels );
#if 0
/* Normalize for unity filter gain */
for( i = 0; i < i_nb_channels; i++ )
{
*(p_out+i) *= d_old_scale_factor;
}
#endif
/* Sanity check */
if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
<= (unsigned int)i_out+1 )
{
p_out += i_nb_channels;
i_out++;
p_filter->p_sys->i_remainder += p_filter->input.i_rate;
break;
}
}
else
{
/* Perform left-wing inner product */
FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in, p_out,
p_filter->p_sys->i_remainder,
p_filter->output.i_rate, p_filter->input.i_rate,
-1, i_nb_channels );
/* Perform right-wing inner product */
FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
p_filter->output.i_rate -
p_filter->p_sys->i_remainder,
p_filter->output.i_rate, p_filter->input.i_rate,
1, i_nb_channels );
}
p_out += i_nb_channels;
i_out++;
p_filter->p_sys->i_remainder += p_filter->input.i_rate;
}
p_in += i_nb_channels;
p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
}
/* Buffer i_filter_wing * 2 samples for next time */
if( p_filter->p_sys->i_old_wing )
{
memcpy( p_filter->p_sys->p_buf,
p_in_orig + (i_in_nb - 2 * p_filter->p_sys->i_old_wing) *
i_nb_channels, (2 * p_filter->p_sys->i_old_wing) *
p_filter->input.i_bytes_per_frame );
}
#if 0
msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
i_out * p_filter->input.i_bytes_per_frame );
#endif
/* Free the temp buffer */
#ifndef HAVE_ALLOCA
free( p_in_orig );
#endif
/* Finalize aout buffer */
p_out_buf->i_nb_samples = i_out;
p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
p_out_buf->end_date = aout_DateIncrement( &p_filter->p_sys->end_date,
p_out_buf->i_nb_samples );
p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
i_nb_channels * sizeof(int32_t);
}
void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
float *p_out, uint32_t ui_remainder,
uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
{
float *Hp, *Hdp, *End;
float t, temp;
uint32_t ui_linear_remainder;
int i;
Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
End = &Imp[Nwing];
ui_linear_remainder = (ui_remainder<<Nhc) -
(ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
if (Inc == 1) /* If doing right wing... */
{ /* ...drop extra coeff, so when Ph is */
End--; /* 0.5, we don't do too many mult's */
if (ui_remainder == 0) /* If the phase is zero... */
{ /* ...then we've already skipped the */
Hp += Npc; /* first sample, so we must also */
Hdp += Npc; /* skip ahead in Imp[] and ImpD[] */
}
}
while (Hp < End) {
t = *Hp; /* Get filter coeff */
/* t is now interp'd filter coeff */
t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
for( i = 0; i < i_nb_channels; i++ )
{
temp = t;
temp *= *(p_in+i); /* Mult coeff by input sample */
*(p_out+i) += temp; /* The filter output */
}
Hdp += Npc; /* Filter coeff differences step */
Hp += Npc; /* Filter coeff step */
p_in += (Inc * i_nb_channels); /* Input signal step */
}
}
void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
float *p_out, uint32_t ui_remainder,
uint32_t ui_output_rate, uint32_t ui_input_rate,
int16_t Inc, int i_nb_channels )
{
float *Hp, *Hdp, *End;
float t, temp;
uint32_t ui_linear_remainder;
int i, ui_counter = 0;
Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
Hdp = ImpD + (ui_remainder<<Nhc) / ui_input_rate;
End = &Imp[Nwing];
if (Inc == 1) /* If doing right wing... */
{ /* ...drop extra coeff, so when Ph is */
End--; /* 0.5, we don't do too many mult's */
if (ui_remainder == 0) /* If the phase is zero... */
{ /* ...then we've already skipped the */
Hp = Imp + /* first sample, so we must also */
(ui_output_rate << Nhc) / ui_input_rate;
Hdp = ImpD + /* skip ahead in Imp[] and ImpD[] */
(ui_output_rate << Nhc) / ui_input_rate;
ui_counter++;
}
}
while (Hp < End) {
t = *Hp; /* Get filter coeff */
/* t is now interp'd filter coeff */
ui_linear_remainder =
((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
ui_input_rate * ui_input_rate;
t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
for( i = 0; i < i_nb_channels; i++ )
{
temp = t;
temp *= *(p_in+i); /* Mult coeff by input sample */
*(p_out+i) += temp; /* The filter output */
}
ui_counter++;
/* Filter coeff step */
Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
/ ui_input_rate;
/* Filter coeff differences step */
Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
/ ui_input_rate;
p_in += (Inc * i_nb_channels); /* Input signal step */
}
}
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