/***************************************************************************** * linear.c : linear interpolation resampler ***************************************************************************** * Copyright (C) 2002 VideoLAN * $Id: linear.c,v 1.11 2003/12/22 14:32:55 sam Exp $ * * Authors: Gildas Bazin <gbazin@netcourrier.com> *          Sigmund Augdal <sigmunau@idi.ntnu.no> * * 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 *****************************************************************************/#include <stdlib.h>                                      /* malloc(), free() */#include <string.h>#include <vlc/vlc.h>#include "audio_output.h"#include "aout_internal.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 * );/***************************************************************************** * Local structures *****************************************************************************/struct aout_filter_sys_t{    int32_t *p_prev_sample;       /* this filter introduces a 1 sample delay */    unsigned int i_remainder;                /* remainder of previous sample */    audio_date_t end_date;};/***************************************************************************** * Module descriptor *****************************************************************************/vlc_module_begin();    set_description( _("audio filter for linear interpolation resampling") );    set_capability( "audio filter", 2 );    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;    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;    }    p_filter->p_sys->p_prev_sample = malloc(        aout_FormatNbChannels( &p_filter->input ) * sizeof(int32_t) );    if( p_filter->p_sys->p_prev_sample == 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_prev_sample );    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_out = (float *)p_out_buf->p_buffer;    float *p_prev_sample = (float *)p_filter->p_sys->p_prev_sample;    int i_nb_channels = aout_FormatNbChannels( &p_filter->input );    int i_in_nb = p_in_buf->i_nb_samples;    int i_chan, i_in, i_out = 0;    /* 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 )        {            /* output the whole thing with the last sample from last time */            memmove( ((float *)(p_in_buf->p_buffer)) + i_nb_channels,                     p_in_buf->p_buffer, p_in_buf->i_nb_bytes );            memcpy( p_in_buf->p_buffer, p_prev_sample,                    i_nb_channels * sizeof(float) );        }        p_filter->b_continuity = VLC_FALSE;        return;    }#ifdef HAVE_ALLOCA    p_in = (float *)alloca( p_in_buf->i_nb_bytes );#else    p_in = (float *)malloc( p_in_buf->i_nb_bytes );#endif    if( p_in == NULL )    {        return;    }    p_aout->p_vlc->pf_memcpy( p_in, p_in_buf->p_buffer, p_in_buf->i_nb_bytes );    /* Take care of the previous input sample (if any) */    if( !p_filter->b_continuity )    {        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 );    }    else    {        while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )        {            for( i_chan = i_nb_channels ; i_chan ; )            {                i_chan--;                p_out[i_chan] = p_prev_sample[i_chan];                p_out[i_chan] += ( (p_prev_sample[i_chan] - p_in[i_chan])                                   * p_filter->p_sys->i_remainder                                   / p_filter->output.i_rate );            }            p_out += i_nb_channels;              i_out++;            p_filter->p_sys->i_remainder += p_filter->input.i_rate;        }        p_filter->p_sys->i_remainder -= p_filter->output.i_rate;    }    /* Take care of the current input samples (minus last one) */    for( i_in = 0; i_in < i_in_nb - 1; i_in++ )    {        while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )        {            for( i_chan = i_nb_channels ; i_chan ; )            {                i_chan--;                p_out[i_chan] = p_in[i_chan];                p_out[i_chan] += ( (p_in[i_chan] -                    p_in[i_chan + i_nb_channels])                    * p_filter->p_sys->i_remainder / p_filter->output.i_rate );            }            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;    }    /* Backup the last input sample for next time */    for( i_chan = i_nb_channels ; i_chan ; )    {        i_chan--;        p_prev_sample[i_chan] = p_in[i_chan];    }    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);}