bandlimited.c

VLC Player Source Code

/*****************************************************************************
 * bandlimited.c : band-limited interpolation resampler
 *****************************************************************************
 * Copyright (C) 2002, 2006 the VideoLAN team
 * $Id$
 *
 * Authors: Gildas Bazin <[e-mail]>
 *
 * 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., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
 *****************************************************************************/

/*****************************************************************************
 * Preamble:
 *
 * This implementation of the band-limited 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.
 *
 *****************************************************************************/

#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 * );

/* 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( 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 filter_sys_t
{
    int32_t *p_buf;                        /* this filter introduces a delay */
    int i_buf_size;

    int i_old_rate;
    double d_old_factor;
    int i_old_wing;

    unsigned int i_remainder;                /* remainder of previous sample */

    audio_date_t end_date;

    bool b_first;
    bool b_filter2;
};

/*****************************************************************************
 * Module descriptor
 *****************************************************************************/
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 filter", 20 );
    set_callbacks( Create, Close );

    add_submodule();
    set_description( _("Audio filter for band-limited interpolation resampling") );
    set_capability( "audio filter2", 20 );
    set_callbacks( OpenFilter, CloseFilter );
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;
    struct filter_sys_t * p_sys;
    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;
    }

#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_sys = malloc( sizeof(filter_sys_t) );
    if( p_sys == NULL )
        return VLC_ENOMEM;
    p_filter->p_sys = (struct aout_filter_sys_t *)p_sys;

    /* Calculate worst case for the length of the filter wing */
    d_factor = (double)p_filter->output.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_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) *
        sizeof(int32_t) * 2 * i_filter_wing;

    /* Allocate enough memory to buffer previous samples */
    p_sys->p_buf = malloc( p_sys->i_buf_size );
    if( p_sys->p_buf == NULL )
    {
        free( p_sys );
        return VLC_ENOMEM;
    }

    p_sys->i_old_wing = 0;
    p_sys->b_filter2 = false;           /* It seams to be a good valuefor this module */
    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 = 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;
    filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
    free( p_sys->p_buf );
    free( 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 )
{
    filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
    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;
    unsigned int i_out_rate;
    double d_factor, d_scale_factor, d_old_scale_factor;
    int i_filter_wing;

    if( p_sys->b_filter2 )
        i_out_rate = p_filter->output.i_rate;
    else
        i_out_rate = p_aout->mixer.mixer.i_rate;

    /* Check if we really need to run the resampler */
    if( i_out_rate == p_filter->input.i_rate )
    {
        if( /*p_filter->b_continuity && /--* What difference does it make ? :) */
            p_sys->i_old_wing &&
            p_in_buf->i_size >=
              p_in_buf->i_nb_bytes + p_sys->i_old_wing *
              p_filter->input.i_bytes_per_frame )
        {
            /* output the whole thing with the samples from last time */
            memmove( ((float *)(p_in_buf->p_buffer)) +
                     i_nb_channels * p_sys->i_old_wing,
                     p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
            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->input.i_bytes_per_frame );

            p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
                p_sys->i_old_wing;

            p_out_buf->start_date = aout_DateGet( &p_sys->end_date );
            p_out_buf->end_date =
                aout_DateIncrement( &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 = false;
        p_sys->i_old_wing = 0;
        return;
    }

    if( !p_filter->b_continuity )
    {
        /* Continuity in sound samples has been broken, we'd better reset
         * everything. */
        p_filter->b_continuity = true;
        p_sys->i_remainder = 0;
        aout_DateInit( &p_sys->end_date, i_out_rate );
        aout_DateSet( &p_sys->end_date, p_in_buf->start_date );
        p_sys->i_old_rate   = p_filter->input.i_rate;
        p_sys->d_old_factor = 1;
        p_sys->i_old_wing   = 0;
    }

#if 0
    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

    /* Prepare the source buffer */
    i_in_nb += (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_sys->i_old_wing )
    {
        vlc_memcpy( p_in, p_sys->p_buf,
                    p_sys->i_old_wing * 2 *
                      p_filter->input.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->input.i_bytes_per_frame );

    /* Make sure the output buffer is reset */
    memset( p_out, 0, p_out_buf->i_size );

    /* Calculate the new length of the filter wing */
    d_factor = (double)i_out_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 */
    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 */
    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_sys->d_old_factor == 1 )
        {
            /* Just copy the samples */
            memcpy( p_out, p_in,
                    p_filter->input.i_bytes_per_frame );
            p_in += i_nb_channels;
            p_out += i_nb_channels;
            i_out++;
            continue;
        }

        while( p_sys->i_remainder < p_filter->output.i_rate )
        {

            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_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_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_sys->i_remainder += p_filter->input.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_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,