/*****************************************************************************
- * bandlimited.c : bandlimited interpolation resampler
+ * bandlimited.c : band-limited interpolation resampler
*****************************************************************************
- * Copyright (C) 2002 VideoLAN
- * $Id: bandlimited.c,v 1.5 2003/03/05 22:37:05 gbazin Exp $
+ * Copyright (C) 2002, 2006 the VideoLAN team
+ * $Id$
*
* Authors: Gildas Bazin <gbazin@netcourrier.com>
*
* 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
*
* 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.
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble:
*
- * This implementation of the bandlimited interpolationis based on the
+ * This implementation of the band-limited interpolationis based on the
* following paper:
* http://ccrma-www.stanford.edu/~jos/resample/resample.html
*
* filter is 13 samples.
*
*****************************************************************************/
-#include <stdlib.h> /* malloc(), free() */
-#include <string.h>
-#include <vlc/vlc.h>
-#include "audio_output.h"
-#include "aout_internal.h"
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <vlc_common.h>
+#include <vlc_plugin.h>
+#include <vlc_aout.h>
+
#include "bandlimited.h"
/*****************************************************************************
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,
+static void FilterFloatUP( const float Imp[], const 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,
+static void FilterFloatUD( const float Imp[], const 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 );
* Module descriptor
*****************************************************************************/
vlc_module_begin();
- set_description( _("audio filter for bandlimited interpolation resampling") );
+ set_category( CAT_AUDIO );
+ set_subcategory( SUBCAT_AUDIO_MISC );
+ set_description( N_("Audio filter for band-limited interpolation resampling") );
set_capability( "audio filter", 20 );
set_callbacks( Create, Close );
vlc_module_end();
return VLC_EGENERIC;
}
+#if !defined( __APPLE__ )
+ 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;
+ / p_filter->input.i_rate / AOUT_MAX_INPUT_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 ) *
/* 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;
+ p_filter->b_in_place = true;
return VLC_SUCCESS;
}
/* 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 ? :) */
+ 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_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->b_continuity = false;
p_filter->p_sys->i_old_wing = 0;
return;
}
{
/* Continuity in sound samples has been broken, we'd better reset
* everything. */
- p_filter->b_continuity = VLC_TRUE;
+ p_filter->b_continuity = 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 );
/* 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 );
+ vlc_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 );
+ vlc_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 );
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 );
+ memcpy( p_out, p_in,
+ p_filter->input.i_bytes_per_frame );
p_in += i_nb_channels;
p_out += i_nb_channels;
i_out++;
}
-void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
+void FilterFloatUP( const float Imp[], const 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;
+ const float *Hp, *Hdp, *End;
float t, temp;
uint32_t ui_linear_remainder;
int i;
}
}
-void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
+void FilterFloatUD( const float Imp[], const 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;
+ const float *Hp, *Hdp, *End;
float t, temp;
uint32_t ui_linear_remainder;
int i, ui_counter = 0;