/*****************************************************************************
- * 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"
+#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 "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,
+/* audio filter2 */
+static int OpenFilter ( vlc_object_t * );
+static void CloseFilter( vlc_object_t * );
+static block_t *Resample( filter_t *, block_t * );
+
+
+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 );
/*****************************************************************************
* Local structures
*****************************************************************************/
-struct aout_filter_sys_t
+struct filter_sys_t
{
int32_t *p_buf; /* this filter introduces a delay */
- int i_buf_size;
+ size_t 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 */
+ bool b_first;
- audio_date_t end_date;
+ date_t end_date;
};
/*****************************************************************************
* Module descriptor
*****************************************************************************/
-vlc_module_begin();
- set_description( _("audio filter for bandlimited interpolation resampling") );
- set_capability( "audio filter", 4 );
- set_callbacks( Create, Close );
-vlc_module_end();
+vlc_module_begin ()
+ set_category( CAT_AUDIO )
+ set_subcategory( SUBCAT_AUDIO_MISC )
+ set_description( N_("Audio filter for band-limited interpolation resampling") )
+ set_capability( "audio filter2", 20 )
+ set_callbacks( OpenFilter, CloseFilter )
+vlc_module_end ()
/*****************************************************************************
- * Create: allocate linear resampler
+ * Resample: convert a buffer
*****************************************************************************/
-static int Create( vlc_object_t *p_this )
+static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
{
- 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;
- }
-
- /* 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 )
+ if( !p_in_buf || !p_in_buf->i_nb_samples )
{
- msg_Err( p_filter, "out of memory" );
- return VLC_ENOMEM;
+ if( p_in_buf )
+ block_Release( p_in_buf );
+ return NULL;
}
- /* 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->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 = (double)p_aout->mixer.mixer.i_rate
- / p_filter->input.i_rate;
- int i_filter_wing, i_left_over;
+ filter_sys_t *p_sys = p_filter->p_sys;
+ unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate;
+ int i_nb_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
/* Check if we really need to run the resampler */
- if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
+ if( i_out_rate == p_filter->fmt_in.audio.i_rate )
{
- if( p_filter->b_continuity &&
- p_in_buf->i_size >=
- p_in_buf->i_nb_bytes + sizeof(float) * i_nb_channels )
+ if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) &&
+ p_sys->i_old_wing )
{
- 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 */
+ p_in_buf = block_Realloc( p_in_buf,
+ p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame,
+ p_in_buf->i_buffer );
+ if( !p_in_buf )
+ return NULL;
+ memcpy( p_in_buf->p_buffer, p_sys->p_buf +
+ i_nb_channels * p_sys->i_old_wing,
+ p_sys->i_old_wing *
+ p_filter->fmt_in.audio.i_bytes_per_frame );
+
+ p_in_buf->i_nb_samples += p_sys->i_old_wing;
+
+ p_in_buf->i_pts = date_Get( &p_sys->end_date );
+ p_in_buf->i_length =
+ date_Increment( &p_sys->end_date,
+ p_in_buf->i_nb_samples ) - p_in_buf->i_pts;
}
- p_filter->b_continuity = VLC_FALSE;
- return;
+ p_sys->i_old_wing = 0;
+ p_sys->b_first = true;
+ return p_in_buf;
}
- if( !p_filter->b_continuity )
+ 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) )
+ + p_filter->p_sys->i_buf_size;
+ block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
+ if( !p_out_buf )
+ return NULL;
+ float *p_out = (float *)p_out_buf->p_buffer;
+
+ if( (p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) || p_sys->b_first )
{
/* 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 );
-
- 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;
+ p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
+ p_sys->i_remainder = 0;
+ date_Init( &p_sys->end_date, i_out_rate, 1 );
+ date_Set( &p_sys->end_date, p_in_buf->i_pts );
+ p_sys->d_old_factor = 1;
+ p_sys->i_old_wing = 0;
+ p_sys->b_first = false;
}
+ 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
- msg_Err( p_filter, "old rate: %i, old factor: %i, 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 );
+ msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
+ p_sys->i_old_rate, p_sys->d_old_factor,
+ 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
- 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;
- }
+ i_in_nb += (p_sys->i_old_wing * 2);
+
+ float p_in_orig[i_in_nb * p_filter->fmt_in.audio.i_bytes_per_frame / 4],
+ *p_in = p_in_orig;
/* Copy all our samples in p_in */
- if( p_filter->p_sys->i_old_wing )
+ if( 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_sys->p_buf,
+ p_sys->i_old_wing * 2 *
+ p_filter->fmt_in.audio.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 );
+ /* XXX: why i_nb_channels instead of i_bytes_per_frame??? */
+ vlc_memcpy( p_in + p_sys->i_old_wing * 2 * i_nb_channels,
+ p_in_buf->p_buffer,
+ p_in_buf->i_nb_samples * p_filter->fmt_in.audio.i_bytes_per_frame );
+ block_Release( p_in_buf );
/* Make sure the output buffer is reset */
- memset( p_out, 0, p_out_buf->i_size );
+ memset( p_out, 0, p_out_buf->i_buffer );
+
+ /* Calculate the new length of the filter wing */
+ d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
+ i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
-#if 0
/* 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_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_sys->i_old_wing;
+ p_in += p_sys->i_old_wing * i_nb_channels;
+ for( ; i_in < i_filter_wing &&
+ (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
{
- if( p_filter->p_sys->d_old_factor == 1 )
+ if( 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->fmt_in.audio.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 )
+ while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
{
- if( p_filter->p_sys->d_old_factor >= 1 )
+ if( 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,
+ p_sys->i_remainder,
+ p_filter->fmt_out.audio.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,
+ p_filter->fmt_out.audio.i_rate -
+ p_sys->i_remainder,
+ p_filter->fmt_out.audio.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_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
+ <= (unsigned int)i_out+1 )
+ {
+ p_out += i_nb_channels;
+ i_out++;
+ p_sys->i_remainder += p_filter->fmt_in.audio.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,
+ p_sys->i_remainder,
+ p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.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,
+ p_filter->fmt_out.audio.i_rate -
+ p_sys->i_remainder,
+ p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
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->input.i_rate;
+ p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
}
p_in += i_nb_channels;
- p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
+ p_sys->i_remainder -= p_filter->fmt_out.audio.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;
+ p_sys->d_old_factor = d_factor;
+ 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 )
+ while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
{
if( d_factor >= 1 )
/* 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,
+ p_sys->i_remainder,
+ p_filter->fmt_out.audio.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,
+ p_filter->fmt_out.audio.i_rate -
+ p_sys->i_remainder,
+ p_filter->fmt_out.audio.i_rate,
1, i_nb_channels );
+
+#if 0
+ /* Normalize for unity filter gain */
+ for( int i = 0; i < i_nb_channels; i++ )
+ {
+ *(p_out+i) *= d_old_scale_factor;
+ }
+#endif
+ /* Sanity check */
+ if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
+ <= (unsigned int)i_out+1 )
+ {
+ p_out += i_nb_channels;
+ i_out++;
+ p_sys->i_remainder += p_filter->fmt_in.audio.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,
+ p_sys->i_remainder,
+ p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.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,
+ p_filter->fmt_out.audio.i_rate -
+ p_sys->i_remainder,
+ p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
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->input.i_rate;
+ p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
}
p_in += i_nb_channels;
- p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
+ p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
}
+ /* Finalize aout buffer */
+ p_out_buf->i_nb_samples = i_out;
+ p_out_buf->i_pts = date_Get( &p_sys->end_date );
+ 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(int32_t);
+
/* Buffer i_filter_wing * 2 samples for next time */
- if( p_filter->p_sys->i_old_wing )
+ if( 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 );
+ size_t newsize = p_sys->i_old_wing * 2
+ * p_filter->fmt_in.audio.i_bytes_per_frame;
+ if( newsize > p_sys->i_buf_size )
+ {
+ free( p_sys->p_buf );
+ p_sys->p_buf = malloc( newsize );
+ if( p_sys->p_buf != NULL )
+ p_sys->i_buf_size = newsize;
+ else
+ {
+ p_sys->i_buf_size = p_sys->i_old_wing = 0; /* oops! */
+ return p_out_buf;
+ }
+ }
+ memcpy( p_sys->p_buf,
+ p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
+ i_nb_channels, (2 * p_sys->i_old_wing) *
+ p_filter->fmt_in.audio.i_bytes_per_frame );
}
#if 0
- msg_Err( p_filter, "pout size: %i, nb_samples out: %i", p_out_buf->i_size,
- i_out * p_filter->input.i_bytes_per_frame );
+ msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_buffer,
+ i_out * p_filter->fmt_in.audio.i_bytes_per_frame );
#endif
- /* Free the temp buffer */
-#ifndef HAVE_ALLOCA
- free( p_in_orig );
-#endif
+ return p_out_buf;
+}
- /* Finalize aout buffer */
- p_out_buf->i_nb_samples = i_out;
- p_out_buf->start_date = p_in_buf->start_date;
+/*****************************************************************************
+ * OpenFilter:
+ *****************************************************************************/
+static int OpenFilter( vlc_object_t *p_this )
+{
+ filter_t *p_filter = (filter_t *)p_this;
+ filter_sys_t *p_sys;
+ unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate;
- if( p_in_buf->start_date !=
- aout_DateGet( &p_filter->p_sys->end_date ) )
+ if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
+ p_filter->fmt_in.i_codec != VLC_CODEC_FL32 )
{
- aout_DateSet( &p_filter->p_sys->end_date, p_in_buf->start_date );
+ return VLC_EGENERIC;
}
- p_out_buf->end_date = aout_DateIncrement( &p_filter->p_sys->end_date,
- p_out_buf->i_nb_samples );
+#if !defined( SYS_DARWIN )
+ if( !config_GetInt( p_this, "hq-resampling" ) )
+ {
+ return VLC_EGENERIC;
+ }
+#endif
- p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
- i_nb_channels * sizeof(int32_t);
+ /* 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_buf = NULL;
+ p_sys->i_buf_size = 0;
+
+ p_sys->i_old_wing = 0;
+ p_sys->b_first = true;
+ 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_buf );
+ free( p_filter->p_sys );
}
-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;
((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++ )
{
projects / vlc / blobdiff