--- /dev/null
+/*****************************************************************************
+ * bandlimited.c : bandlimited interpolation resampler
+ *****************************************************************************
+ * Copyright (C) 2002 VideoLAN
+ * $Id: bandlimited.c,v 1.1 2003/03/04 03:27:40 gbazin Exp $
+ *
+ * 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 bandlimited 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;
+ int 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_description( _("audio filter for bandlimited interpolation resampling") );
+ set_capability( "audio filter", 4 );
+ 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;
+ }
+
+ /* 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->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;
+
+ /* 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 &&
+ p_in_buf->i_size >=
+ p_in_buf->i_nb_bytes + sizeof(float) * i_nb_channels )
+ {
+ 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);
+ }
+ }
+ p_filter->b_continuity = VLC_FALSE;
+ 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 );
+
+ 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",
+ 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
+ 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 );
+
+#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
+
+ /* 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++ )
+ {
+ 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 );
+ 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 );
+
+ }
+ 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 );
+ }
+
+#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_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 );
+ }
+ 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 );
+ }
+
+#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_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, "pout size: %i, nb_samples 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 = 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->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;
+ //(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++ )
+ {
+ 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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