/*******************************************************************************/ /* 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" #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; } } 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; w = 10.0f / (FS * T); } void 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; 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; }