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![Clément Stenac](/assets/img/avatar_default.png)
* 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
554 lines
21 KiB
C
554 lines
21 KiB
C
/*****************************************************************************
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* bandlimited.c : band-limited interpolation resampler
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*****************************************************************************
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* Copyright (C) 2002 VideoLAN
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* $Id$
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*
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* Authors: Gildas Bazin <gbazin@netcourrier.com>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
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*****************************************************************************/
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/*****************************************************************************
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* Preamble:
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*
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* This implementation of the band-limited interpolationis based on the
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* following paper:
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* http://ccrma-www.stanford.edu/~jos/resample/resample.html
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*
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* It uses a Kaiser-windowed sinc-function low-pass filter and the width of the
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* filter is 13 samples.
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*
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*****************************************************************************/
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#include <stdlib.h> /* malloc(), free() */
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#include <string.h>
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#include <vlc/vlc.h>
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#include "audio_output.h"
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#include "aout_internal.h"
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#include "bandlimited.h"
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/*****************************************************************************
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* Local prototypes
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*****************************************************************************/
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static int Create ( vlc_object_t * );
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static void Close ( vlc_object_t * );
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static void DoWork ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
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aout_buffer_t * );
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static void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing,
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float *f_in, float *f_out, uint32_t ui_remainder,
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uint32_t ui_output_rate, int16_t Inc,
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int i_nb_channels );
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static void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing,
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float *f_in, float *f_out, uint32_t ui_remainder,
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uint32_t ui_output_rate, uint32_t ui_input_rate,
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int16_t Inc, int i_nb_channels );
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/*****************************************************************************
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* Local structures
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*****************************************************************************/
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struct aout_filter_sys_t
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{
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int32_t *p_buf; /* this filter introduces a delay */
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int i_buf_size;
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int i_old_rate;
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double d_old_factor;
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int i_old_wing;
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unsigned int i_remainder; /* remainder of previous sample */
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audio_date_t end_date;
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};
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/*****************************************************************************
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* Module descriptor
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*****************************************************************************/
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vlc_module_begin();
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set_category( CAT_AUDIO );
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set_subcategory( SUBCAT_AUDIO_MISC );
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set_description( _("audio filter for band-limited interpolation resampling") );
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set_capability( "audio filter", 20 );
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set_callbacks( Create, Close );
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vlc_module_end();
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/*****************************************************************************
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* Create: allocate linear resampler
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*****************************************************************************/
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static int Create( vlc_object_t *p_this )
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{
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aout_filter_t * p_filter = (aout_filter_t *)p_this;
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double d_factor;
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int i_filter_wing;
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if ( p_filter->input.i_rate == p_filter->output.i_rate
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|| p_filter->input.i_format != p_filter->output.i_format
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|| p_filter->input.i_physical_channels
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!= p_filter->output.i_physical_channels
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|| p_filter->input.i_original_channels
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!= p_filter->output.i_original_channels
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|| p_filter->input.i_format != VLC_FOURCC('f','l','3','2') )
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{
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return VLC_EGENERIC;
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}
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#if !defined( SYS_DARWIN )
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if( !config_GetInt( p_this, "hq-resampling" ) )
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{
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return VLC_EGENERIC;
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}
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#endif
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/* Allocate the memory needed to store the module's structure */
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p_filter->p_sys = malloc( sizeof(struct aout_filter_sys_t) );
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if( p_filter->p_sys == NULL )
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{
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msg_Err( p_filter, "out of memory" );
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return VLC_ENOMEM;
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}
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/* Calculate worst case for the length of the filter wing */
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d_factor = (double)p_filter->output.i_rate
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/ p_filter->input.i_rate;
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i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
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* __MAX(1.0, 1.0/d_factor) + 10;
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p_filter->p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) *
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sizeof(int32_t) * 2 * i_filter_wing;
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/* Allocate enough memory to buffer previous samples */
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p_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size );
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if( p_filter->p_sys->p_buf == NULL )
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{
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msg_Err( p_filter, "out of memory" );
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return VLC_ENOMEM;
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}
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p_filter->p_sys->i_old_wing = 0;
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p_filter->pf_do_work = DoWork;
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/* We don't want a new buffer to be created because we're not sure we'll
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* actually need to resample anything. */
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p_filter->b_in_place = VLC_TRUE;
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return VLC_SUCCESS;
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}
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/*****************************************************************************
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* Close: free our resources
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*****************************************************************************/
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static void Close( vlc_object_t * p_this )
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{
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aout_filter_t * p_filter = (aout_filter_t *)p_this;
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free( p_filter->p_sys->p_buf );
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free( p_filter->p_sys );
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}
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/*****************************************************************************
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* DoWork: convert a buffer
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*****************************************************************************/
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static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
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aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
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{
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float *p_in, *p_in_orig, *p_out = (float *)p_out_buf->p_buffer;
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int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
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int i_in_nb = p_in_buf->i_nb_samples;
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int i_in, i_out = 0;
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double d_factor, d_scale_factor, d_old_scale_factor;
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int i_filter_wing;
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#if 0
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int i;
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#endif
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/* Check if we really need to run the resampler */
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if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
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{
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if( //p_filter->b_continuity && /* What difference does it make ? :) */
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p_filter->p_sys->i_old_wing &&
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p_in_buf->i_size >=
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p_in_buf->i_nb_bytes + p_filter->p_sys->i_old_wing *
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p_filter->input.i_bytes_per_frame )
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{
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/* output the whole thing with the samples from last time */
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memmove( ((float *)(p_in_buf->p_buffer)) +
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i_nb_channels * p_filter->p_sys->i_old_wing,
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p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
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memcpy( p_in_buf->p_buffer, p_filter->p_sys->p_buf +
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i_nb_channels * p_filter->p_sys->i_old_wing,
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p_filter->p_sys->i_old_wing *
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p_filter->input.i_bytes_per_frame );
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p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
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p_filter->p_sys->i_old_wing;
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p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
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p_out_buf->end_date =
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aout_DateIncrement( &p_filter->p_sys->end_date,
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p_out_buf->i_nb_samples );
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p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
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p_filter->input.i_bytes_per_frame;
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}
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p_filter->b_continuity = VLC_FALSE;
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p_filter->p_sys->i_old_wing = 0;
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return;
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}
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if( !p_filter->b_continuity )
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{
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/* Continuity in sound samples has been broken, we'd better reset
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* everything. */
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p_filter->b_continuity = VLC_TRUE;
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p_filter->p_sys->i_remainder = 0;
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aout_DateInit( &p_filter->p_sys->end_date, p_filter->output.i_rate );
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aout_DateSet( &p_filter->p_sys->end_date, p_in_buf->start_date );
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p_filter->p_sys->i_old_rate = p_filter->input.i_rate;
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p_filter->p_sys->d_old_factor = 1;
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p_filter->p_sys->i_old_wing = 0;
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}
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#if 0
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msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
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p_filter->p_sys->i_old_rate, p_filter->p_sys->d_old_factor,
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p_filter->p_sys->i_old_wing, i_in_nb );
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#endif
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/* Prepare the source buffer */
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i_in_nb += (p_filter->p_sys->i_old_wing * 2);
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#ifdef HAVE_ALLOCA
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p_in = p_in_orig = (float *)alloca( i_in_nb *
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p_filter->input.i_bytes_per_frame );
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#else
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p_in = p_in_orig = (float *)malloc( i_in_nb *
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p_filter->input.i_bytes_per_frame );
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#endif
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if( p_in == NULL )
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{
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return;
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}
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/* Copy all our samples in p_in */
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if( p_filter->p_sys->i_old_wing )
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{
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p_aout->p_vlc->pf_memcpy( p_in, p_filter->p_sys->p_buf,
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p_filter->p_sys->i_old_wing * 2 *
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p_filter->input.i_bytes_per_frame );
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}
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p_aout->p_vlc->pf_memcpy( p_in + p_filter->p_sys->i_old_wing * 2 *
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i_nb_channels, p_in_buf->p_buffer,
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p_in_buf->i_nb_samples *
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p_filter->input.i_bytes_per_frame );
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/* Make sure the output buffer is reset */
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memset( p_out, 0, p_out_buf->i_size );
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/* Calculate the new length of the filter wing */
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d_factor = (double)p_aout->mixer.mixer.i_rate / p_filter->input.i_rate;
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i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
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/* Account for increased filter gain when using factors less than 1 */
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d_old_scale_factor = SMALL_FILTER_SCALE *
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p_filter->p_sys->d_old_factor + 0.5;
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d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
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/* Apply the old rate until we have enough samples for the new one */
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i_in = p_filter->p_sys->i_old_wing;
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p_in += p_filter->p_sys->i_old_wing * i_nb_channels;
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for( ; i_in < i_filter_wing &&
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(i_in + p_filter->p_sys->i_old_wing) < i_in_nb; i_in++ )
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{
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if( p_filter->p_sys->d_old_factor == 1 )
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{
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/* Just copy the samples */
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memcpy( p_out, p_in,
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p_filter->input.i_bytes_per_frame );
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p_in += i_nb_channels;
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p_out += i_nb_channels;
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i_out++;
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continue;
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}
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while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
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{
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if( p_filter->p_sys->d_old_factor >= 1 )
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{
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/* FilterFloatUP() is faster if we can use it */
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/* Perform left-wing inner product */
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FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
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SMALL_FILTER_NWING, p_in, p_out,
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p_filter->p_sys->i_remainder,
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p_filter->output.i_rate,
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-1, i_nb_channels );
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/* Perform right-wing inner product */
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FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
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SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
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p_filter->output.i_rate -
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p_filter->p_sys->i_remainder,
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p_filter->output.i_rate,
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1, i_nb_channels );
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#if 0
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/* Normalize for unity filter gain */
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for( i = 0; i < i_nb_channels; i++ )
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{
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*(p_out+i) *= d_old_scale_factor;
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}
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#endif
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/* Sanity check */
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if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
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<= (unsigned int)i_out+1 )
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{
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p_out += i_nb_channels;
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i_out++;
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p_filter->p_sys->i_remainder += p_filter->input.i_rate;
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break;
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}
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}
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else
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{
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/* Perform left-wing inner product */
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FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
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SMALL_FILTER_NWING, p_in, p_out,
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p_filter->p_sys->i_remainder,
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p_filter->output.i_rate, p_filter->input.i_rate,
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-1, i_nb_channels );
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/* Perform right-wing inner product */
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FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
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SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
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p_filter->output.i_rate -
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p_filter->p_sys->i_remainder,
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p_filter->output.i_rate, p_filter->input.i_rate,
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1, i_nb_channels );
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}
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p_out += i_nb_channels;
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i_out++;
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p_filter->p_sys->i_remainder += p_filter->input.i_rate;
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}
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p_in += i_nb_channels;
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p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
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}
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/* Apply the new rate for the rest of the samples */
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if( i_in < i_in_nb - i_filter_wing )
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{
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p_filter->p_sys->i_old_rate = p_filter->input.i_rate;
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p_filter->p_sys->d_old_factor = d_factor;
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p_filter->p_sys->i_old_wing = i_filter_wing;
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}
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for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
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{
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while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
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{
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if( d_factor >= 1 )
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{
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/* FilterFloatUP() is faster if we can use it */
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/* Perform left-wing inner product */
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FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
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SMALL_FILTER_NWING, p_in, p_out,
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p_filter->p_sys->i_remainder,
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p_filter->output.i_rate,
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-1, i_nb_channels );
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/* Perform right-wing inner product */
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FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
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SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
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p_filter->output.i_rate -
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p_filter->p_sys->i_remainder,
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p_filter->output.i_rate,
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1, i_nb_channels );
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#if 0
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/* Normalize for unity filter gain */
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for( i = 0; i < i_nb_channels; i++ )
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{
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*(p_out+i) *= d_old_scale_factor;
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}
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#endif
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/* Sanity check */
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if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
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<= (unsigned int)i_out+1 )
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{
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p_out += i_nb_channels;
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i_out++;
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p_filter->p_sys->i_remainder += p_filter->input.i_rate;
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break;
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}
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}
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else
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{
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/* Perform left-wing inner product */
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FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
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SMALL_FILTER_NWING, p_in, p_out,
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p_filter->p_sys->i_remainder,
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p_filter->output.i_rate, p_filter->input.i_rate,
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-1, i_nb_channels );
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/* Perform right-wing inner product */
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FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
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SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
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p_filter->output.i_rate -
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p_filter->p_sys->i_remainder,
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p_filter->output.i_rate, p_filter->input.i_rate,
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1, i_nb_channels );
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}
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p_out += i_nb_channels;
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i_out++;
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p_filter->p_sys->i_remainder += p_filter->input.i_rate;
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}
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p_in += i_nb_channels;
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p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
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}
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/* Buffer i_filter_wing * 2 samples for next time */
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if( p_filter->p_sys->i_old_wing )
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{
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memcpy( p_filter->p_sys->p_buf,
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p_in_orig + (i_in_nb - 2 * p_filter->p_sys->i_old_wing) *
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i_nb_channels, (2 * p_filter->p_sys->i_old_wing) *
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p_filter->input.i_bytes_per_frame );
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}
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#if 0
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msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
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i_out * p_filter->input.i_bytes_per_frame );
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#endif
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/* Free the temp buffer */
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#ifndef HAVE_ALLOCA
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free( p_in_orig );
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#endif
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/* Finalize aout buffer */
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p_out_buf->i_nb_samples = i_out;
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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 */
|
|
}
|
|
}
|