📄 param_eq.c
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/***************************************************************************** * param_eq.c: ***************************************************************************** * Copyright (C) 2006 the VideoLAN team * $Id: param_eq.c 17012 2006-10-09 22:11:32Z xtophe $ * * Authors: Antti Huovilainen * Sigmund A. Helberg <dnumgis@videolan.org> * * 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 *****************************************************************************/#include <stdlib.h> /* malloc(), free() */#include <string.h>#include <math.h>#include <vlc/vlc.h>#include <vlc/aout.h>#include "aout_internal.h"/***************************************************************************** * Module descriptor *****************************************************************************/static int Open ( vlc_object_t * );static void Close( vlc_object_t * );static void CalcPeakEQCoeffs( float, float, float, float, float * );static void CalcShelfEQCoeffs( float, float, float, int, float, float * );static void ProcessEQ( float *, float *, float *, int, int, float *, int );static void DoWork( aout_instance_t *, aout_filter_t *, aout_buffer_t *, aout_buffer_t * );vlc_module_begin(); set_description( _("Parametric Equalizer") ); set_shortname( _("Parametric Equalizer" ) ); set_capability( "audio filter", 0 ); set_category( CAT_AUDIO ); set_subcategory( SUBCAT_AUDIO_AFILTER ); add_float( "param-eq-lowf", 100, NULL, N_("Low freq (Hz)"),"", VLC_FALSE ); /// \bug Db -> dB add_float_with_range( "param-eq-lowgain", 0, -20.0, 20.0, NULL, N_("Low freq gain (Db)"), "",VLC_FALSE ); add_float( "param-eq-highf", 10000, NULL, N_("High freq (Hz)"),"", VLC_FALSE ); /// \bug Db -> dB add_float_with_range( "param-eq-highgain", 0, -20.0, 20.0, NULL, N_("High freq gain (Db)"),"",VLC_FALSE ); add_float( "param-eq-f1", 300, NULL, N_("Freq 1 (Hz)"),"", VLC_FALSE ); /// \bug Db -> dB add_float_with_range( "param-eq-gain1", 0, -20.0, 20.0, NULL, N_("Freq 1 gain (Db)"), "",VLC_FALSE ); add_float_with_range( "param-eq-q1", 3, 0.1, 100.0, NULL, N_("Freq 1 Q"), "",VLC_FALSE ); add_float( "param-eq-f2", 1000, NULL, N_("Freq 2 (Hz)"),"", VLC_FALSE ); /// \bug Db -> dB add_float_with_range( "param-eq-gain2", 0, -20.0, 20.0, NULL, N_("Freq 2 gain (Db)"),"",VLC_FALSE ); add_float_with_range( "param-eq-q2", 3, 0.1, 100.0, NULL, N_("Freq 2 Q"),"",VLC_FALSE ); add_float( "param-eq-f3", 3000, NULL, N_("Freq 3 (Hz)"),"", VLC_FALSE ); /// \bug Db -> dB add_float_with_range( "param-eq-gain3", 0, -20.0, 20.0, NULL, N_("Freq 3 gain (Db)"),"",VLC_FALSE ); add_float_with_range( "param-eq-q3", 3, 0.1, 100.0, NULL, N_("Freq 3 Q"),"",VLC_FALSE ); set_callbacks( Open, Close );vlc_module_end();/***************************************************************************** * Local prototypes *****************************************************************************/typedef struct aout_filter_sys_t{ /* Filter static config */ float f_lowf, f_lowgain; float f_f1, f_Q1, f_gain1; float f_f2, f_Q2, f_gain2; float f_f3, f_Q3, f_gain3; float f_highf, f_highgain; /* Filter computed coeffs */ float coeffs[5*5]; /* State */ float *p_state; } aout_filter_sys_t;/***************************************************************************** * Open: *****************************************************************************/static int Open( vlc_object_t *p_this ){ aout_filter_t *p_filter = (aout_filter_t *)p_this; aout_filter_sys_t *p_sys; vlc_bool_t b_fit = VLC_TRUE; int i_samplerate; if( p_filter->input.i_format != VLC_FOURCC('f','l','3','2' ) || p_filter->output.i_format != VLC_FOURCC('f','l','3','2') ) { b_fit = VLC_FALSE; p_filter->input.i_format = VLC_FOURCC('f','l','3','2'); p_filter->output.i_format = VLC_FOURCC('f','l','3','2'); msg_Warn( p_filter, "bad input or output format" ); } if ( !AOUT_FMTS_SIMILAR( &p_filter->input, &p_filter->output ) ) { b_fit = VLC_FALSE; memcpy( &p_filter->output, &p_filter->input, sizeof(audio_sample_format_t) ); msg_Warn( p_filter, "input and output formats are not similar" ); } if ( ! b_fit ) { return VLC_EGENERIC; } p_filter->pf_do_work = DoWork; p_filter->b_in_place = VLC_TRUE; /* Allocate structure */ p_sys = p_filter->p_sys = malloc( sizeof( aout_filter_sys_t ) ); p_sys->f_lowf = config_GetFloat( p_this, "param-eq-lowf"); p_sys->f_lowgain = config_GetFloat( p_this, "param-eq-lowgain"); p_sys->f_highf = config_GetFloat( p_this, "param-eq-highf"); p_sys->f_highgain = config_GetFloat( p_this, "param-eq-highgain"); p_sys->f_f1 = config_GetFloat( p_this, "param-eq-f1"); p_sys->f_Q1 = config_GetFloat( p_this, "param-eq-q1"); p_sys->f_gain1 = config_GetFloat( p_this, "param-eq-gain1"); p_sys->f_f2 = config_GetFloat( p_this, "param-eq-f2"); p_sys->f_Q2 = config_GetFloat( p_this, "param-eq-q2"); p_sys->f_gain2 = config_GetFloat( p_this, "param-eq-gain2"); p_sys->f_f3 = config_GetFloat( p_this, "param-eq-f3"); p_sys->f_Q3 = config_GetFloat( p_this, "param-eq-q3"); p_sys->f_gain3 = config_GetFloat( p_this, "param-eq-gain3"); i_samplerate = p_filter->input.i_rate; CalcPeakEQCoeffs(p_sys->f_f1, p_sys->f_Q1, p_sys->f_gain1, i_samplerate, p_sys->coeffs+0*5); CalcPeakEQCoeffs(p_sys->f_f2, p_sys->f_Q2, p_sys->f_gain2, i_samplerate, p_sys->coeffs+1*5); CalcPeakEQCoeffs(p_sys->f_f3, p_sys->f_Q3, p_sys->f_gain3, i_samplerate, p_sys->coeffs+2*5); CalcShelfEQCoeffs(p_sys->f_lowf, 1, p_sys->f_lowgain, 0, i_samplerate, p_sys->coeffs+3*5); CalcShelfEQCoeffs(p_sys->f_highf, 1, p_sys->f_highgain, 0, i_samplerate, p_sys->coeffs+4*5); p_sys->p_state = (float*)calloc( p_filter->input.i_channels*5*4, sizeof(float) ); return VLC_SUCCESS;}static void Close( vlc_object_t *p_this ){ aout_filter_t *p_filter = (aout_filter_t *)p_this; free( p_filter->p_sys->p_state ); free( p_filter->p_sys );}/***************************************************************************** * DoWork: process samples 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 ){ p_out_buf->i_nb_samples = p_in_buf->i_nb_samples; p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes; ProcessEQ( (float*)p_in_buf->p_buffer, (float*)p_out_buf->p_buffer, p_filter->p_sys->p_state, p_filter->input.i_channels, p_in_buf->i_nb_samples, p_filter->p_sys->coeffs, 5 );}/* * Calculate direct form IIR coefficients for peaking EQ * coeffs[0] = b0 * coeffs[1] = b1 * coeffs[2] = b2 * coeffs[3] = a1 * coeffs[4] = a2 * * Equations taken from RBJ audio EQ cookbook * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt) */static void CalcPeakEQCoeffs( float f0, float Q, float gainDB, float Fs, float *coeffs ){ float A; float w0; float alpha; float b0, b1, b2; float a0, a1, a2; // Provide sane limits to avoid overflow if (Q < 0.1f) Q = 0.1f; if (Q > 100) Q = 100; if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f; if (gainDB < -40) gainDB = -40; if (gainDB > 40) gainDB = 40; A = pow(10, gainDB/40); w0 = 2*3.141593f*f0/Fs; alpha = sin(w0)/(2*Q); b0 = 1 + alpha*A; b1 = -2*cos(w0); b2 = 1 - alpha*A; a0 = 1 + alpha/A; a1 = -2*cos(w0); a2 = 1 - alpha/A; // Store values to coeffs and normalize by 1/a0 coeffs[0] = b0/a0; coeffs[1] = b1/a0; coeffs[2] = b2/a0; coeffs[3] = a1/a0; coeffs[4] = a2/a0;}/* * Calculate direct form IIR coefficients for low/high shelf EQ * coeffs[0] = b0 * coeffs[1] = b1 * coeffs[2] = b2 * coeffs[3] = a1 * coeffs[4] = a2 * * Equations taken from RBJ audio EQ cookbook * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt) */static void CalcShelfEQCoeffs( float f0, float slope, float gainDB, int high, float Fs, float *coeffs ){ float A; float w0; float alpha; float b0, b1, b2; float a0, a1, a2; // Provide sane limits to avoid overflow if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f; if (gainDB < -40) gainDB = -40; if (gainDB > 40) gainDB = 40; A = pow(10, gainDB/40); w0 = 2*3.141593f*f0/Fs; alpha = sin(w0)/2 * sqrt( (A + 1/A)*(1/slope - 1) + 2 ); if (high) { b0 = A*( (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha ); b1 = -2*A*( (A-1) + (A+1)*cos(w0) ); b2 = A*( (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha ); a0 = (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha; a1 = 2*( (A-1) - (A+1)*cos(w0) ); a2 = (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha; } else { b0 = A*( (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha ); b1 = 2*A*( (A-1) - (A+1)*cos(w0)); b2 = A*( (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha ); a0 = (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha; a1 = -2*( (A-1) + (A+1)*cos(w0)); a2 = (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha; } // Store values to coeffs and normalize by 1/a0 coeffs[0] = b0/a0; coeffs[1] = b1/a0; coeffs[2] = b2/a0; coeffs[3] = a1/a0; coeffs[4] = a2/a0;}/* src is assumed to be interleaved dest is assumed to be interleaved size of state is 4*channels*eqCount samples is not premultiplied by channels size of coeffs is 5*eqCount*/void ProcessEQ( float *src, float *dest, float *state, int channels, int samples, float *coeffs, int eqCount ){ int i, chn, eq; float b0, b1, b2, a1, a2; float x, y = 0; float *src1, *dest1; float *coeffs1, *state1; src1 = src; dest1 = dest; for (i = 0; i < samples; i++) { state1 = state; for (chn = 0; chn < channels; chn++) { coeffs1 = coeffs; x = *src1++; /* Direct form 1 IIRs */ for (eq = 0; eq < eqCount; eq++) { b0 = coeffs1[0]; b1 = coeffs1[1]; b2 = coeffs1[2]; a1 = coeffs1[3]; a2 = coeffs1[4]; coeffs1 += 5; y = x*b0 + state1[0]*b1 + state1[1]*b2 - state1[2]*a1 - state1[3]*a2; state1[1] = state1[0]; state1[0] = x; state1[3] = state1[2]; state1[2] = y; x = y; state1 += 4; } *dest1++ = y; } }}
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