non/nonlib/dsp.C

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/*******************************************************************************/
/* Copyright (C) 2008 Jonathan Moore Liles */
/* */
/* 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; see the file COPYING. If not,write to the Free Software */
/* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
/*******************************************************************************/
/* General DSP related functions. */
#include "dsp.h"
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#include "string.h" // for memset.
/* TODO: these functions are all targets for optimization (SSE?) */
void
buffer_apply_gain ( sample_t *buf, nframes_t nframes, float g )
{
if ( g != 1.0f )
while ( nframes-- )
*(buf++) *= g;
}
void
buffer_apply_gain_buffer ( sample_t *buf, const sample_t *gainbuf, nframes_t nframes )
{
while ( nframes-- )
*(buf++) *= *(gainbuf++);
}
void
buffer_copy_and_apply_gain_buffer ( sample_t *dst, const sample_t *src, const sample_t *gainbuf, nframes_t nframes )
{
while ( nframes-- )
*(dst++) = *(src++) * *(gainbuf++);
}
void
buffer_mix ( sample_t *dst, const sample_t *src, nframes_t nframes )
{
while ( nframes-- )
*(dst++) += *(src++);
}
void
buffer_mix_with_gain ( sample_t *dst, const sample_t *src, nframes_t nframes, float g )
{
while ( nframes-- )
*(dst++) += *(src++) * g;
}
void
buffer_interleave_one_channel ( sample_t *dst, const sample_t *src, int channel, int channels, nframes_t nframes )
{
dst += channel;
while ( nframes-- )
{
*dst = *(src++);
dst += channels;
}
}
void
buffer_interleave_one_channel_and_mix ( sample_t *dst, const sample_t *src, int channel, int channels, nframes_t nframes )
{
dst += channel;
while ( nframes-- )
{
*dst += *(src++);
dst += channels;
}
}
void
buffer_deinterleave_one_channel ( sample_t *dst, const sample_t *src, int channel, int channels, nframes_t nframes )
{
src += channel;
while ( nframes-- )
{
*(dst++) = *src;
src += channels;
}
}
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void
buffer_fill_with_silence ( sample_t *buf, nframes_t nframes )
{
memset( buf, 0, nframes * sizeof( sample_t ) );
}
bool
buffer_is_digital_black ( sample_t *buf, nframes_t nframes )
{
while ( nframes-- )
{
if ( 0 != buf[nframes] )
return false;
}
return true;
}
void
buffer_copy ( sample_t *dst, const sample_t *src, nframes_t nframes )
{
memcpy( dst, src, nframes * sizeof( sample_t ) );
}
void
buffer_copy_and_apply_gain ( sample_t *dst, const sample_t *src, nframes_t nframes, float gain )
{
memcpy( dst, src, nframes * sizeof( sample_t ) );
buffer_apply_gain( dst, nframes, gain );
}
void
Value_Smoothing_Filter::sample_rate ( nframes_t n )
{
const float FS = n;
const float T = 0.05f;
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w = _cutoff / (FS * T);
}
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bool
Value_Smoothing_Filter::apply( sample_t *dst, nframes_t nframes, float gt )
{
const float a = 0.07f;
const float b = 1 + a;
const float gm = b * gt;
float g1 = this->g1;
float g2 = this->g2;
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if ( target_reached(gt) )
return false;
for (nframes_t i = 0; i < nframes; i++)
{
g1 += w * (gm - g1 - a * g2);
g2 += w * (g1 - g2);
dst[i] = g2;
}
if ( fabsf( gt - g2 ) < 0.0001f )
g2 = gt;
this->g1 = g1;
this->g2 = g2;
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return true;
}