/*****************************************************************************
- * bandlimited.c : bandlimited interpolation resampler
+ * bandlimited.c : band-limited interpolation resampler
*****************************************************************************
- * Copyright (C) 2002 VideoLAN
- * $Id: bandlimited.c,v 1.1 2003/03/04 03:27:40 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"
+#include <vlc_aout.h>
+
#include "bandlimited.h"
/*****************************************************************************
int i_buf_size;
int i_old_rate;
- int d_old_factor;
+ double d_old_factor;
int i_old_wing;
unsigned int i_remainder; /* remainder of previous sample */
* Module descriptor
*****************************************************************************/
vlc_module_begin();
- set_description( _("audio filter for bandlimited interpolation resampling") );
- set_capability( "audio filter", 4 );
+ 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();
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 )
/* 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 ) *
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
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 = (double)p_aout->mixer.mixer.i_rate
- / p_filter->input.i_rate;
- int i_filter_wing, i_left_over;
+ 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 &&
+ 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 + sizeof(float) * i_nb_channels )
+ p_in_buf->i_nb_bytes + p_filter->p_sys->i_old_wing *
+ p_filter->input.i_bytes_per_frame )
{
- if( p_filter->p_sys->i_old_wing )
- {
- /* 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,
- i_nb_channels * 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;
-
- aout_DateSet( &p_filter->p_sys->end_date,
- p_in_buf->start_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);
- }
+ /* 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;
}
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: %i, old wing: %i, i_in: %i",
+ 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
- /* Calculate the 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;
-
- /* Check if we have enough buffered data to start with the new rate. */
- i_left_over = i_filter_wing - p_filter->p_sys->i_old_wing;
-
/* Prepare the source buffer */
i_in_nb += (p_filter->p_sys->i_old_wing * 2);
#ifdef HAVE_ALLOCA
/* 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_aout->p_libvlc->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 *
+ p_aout->p_libvlc->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 );
-#if 0
+ /* 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 */
- if( d_factor < 1 )
- {
- LpScl = SMALL_FILTER_SCALE * d_factor + 0.5;
- }
-#endif
+ 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 */
- for( i_in = p_filter->p_sys->i_old_wing; i_in < i_left_over; i_in++ )
+ 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_buf->p_buffer, 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++;
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
{
1, i_nb_channels );
}
-#if 0
- v *= LpScl; /* Normalize for unity filter gain */
-#endif
-
p_out += i_nb_channels;
i_out++;
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
{
1, i_nb_channels );
}
-#if 0
- v *= LpScl; /* Normalize for unity filter gain */
-#endif
-
p_out += i_nb_channels;
i_out++;
}
#if 0
- msg_Err( p_filter, "pout size: %i, nb_samples out: %i", p_out_buf->i_size,
+ 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
/* Finalize aout buffer */
p_out_buf->i_nb_samples = i_out;
- p_out_buf->start_date = p_in_buf->start_date;
-
- if( p_in_buf->start_date !=
- aout_DateGet( &p_filter->p_sys->end_date ) )
- {
- aout_DateSet( &p_filter->p_sys->end_date, p_in_buf->start_date );
- }
-
+ 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 );
((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
ui_input_rate * ui_input_rate;
- //(ui_remainder<<Nhc)* ui_output_rate/ui_input_rate -
- //(ui_remainder<<Nhc) / ui_input_rate * ui_output_rate;
t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
for( i = 0; i < i_nb_channels; i++ )
{
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