From 13b03068c18b87aa201b53492d64244757875332 Mon Sep 17 00:00:00 2001 From: Sigmund Augdal Helberg Date: Mon, 13 Feb 2006 22:44:24 +0000 Subject: [PATCH] all: parametric equalizer by Antti Huovilainen (adapted for vlc by me) --- configure.ac | 4 +- modules/audio_filter/Modules.am | 1 + modules/audio_filter/param_eq.c | 344 ++++++++++++++++++++++++++++++++ 3 files changed, 347 insertions(+), 2 deletions(-) create mode 100644 modules/audio_filter/param_eq.c diff --git a/configure.ac b/configure.ac index acb5e0b4c2..121065c059 100644 --- a/configure.ac +++ b/configure.ac @@ -523,7 +523,7 @@ AC_CHECK_LIB(m,cos,[ VLC_ADD_LDFLAGS([adjust distort a52tofloat32 dtstofloat32 x264 goom],[-lm]) ]) AC_CHECK_LIB(m,pow,[ - VLC_ADD_LDFLAGS([ffmpeg ffmpegaltivec stream_out_transrate i420_rgb faad twolame equalizer vlc freetype mpc],[-lm]) + VLC_ADD_LDFLAGS([ffmpeg ffmpegaltivec stream_out_transrate i420_rgb faad twolame equalizer param_eq vlc freetype mpc],[-lm]) ]) AC_CHECK_LIB(m,sqrt,[ VLC_ADD_LDFLAGS([headphone_channel_mixer normvol speex],[-lm]) @@ -1069,7 +1069,7 @@ dnl VLC_ADD_PLUGINS([externrun]) VLC_ADD_PLUGINS([i420_yuy2 i422_yuy2 i420_ymga]) VLC_ADD_PLUGINS([aout_file linear_resampler bandlimited_resampler]) VLC_ADD_PLUGINS([float32_mixer spdif_mixer simple_channel_mixer]) - VLC_ADD_PLUGINS([dolby_surround_decoder headphone_channel_mixer normvol equalizer]) + VLC_ADD_PLUGINS([dolby_surround_decoder headphone_channel_mixer normvol equalizer param_eq]) VLC_ADD_PLUGINS([fixed32tofloat32 float32tos16 float32tos8 float32tou16 float32tou8 a52tospdif dtstospdif s16tofloat32 s16tofloat32swab s8tofloat32 u8tofloat32 audio_format]) fi diff --git a/modules/audio_filter/Modules.am b/modules/audio_filter/Modules.am index 052ca63b10..fa3dc4f6f0 100644 --- a/modules/audio_filter/Modules.am +++ b/modules/audio_filter/Modules.am @@ -1,3 +1,4 @@ SOURCES_equalizer = equalizer.c equalizer_presets.h SOURCES_normvol = normvol.c SOURCES_audio_format = format.c +SOURCES_param_eq = param_eq.c diff --git a/modules/audio_filter/param_eq.c b/modules/audio_filter/param_eq.c new file mode 100644 index 0000000000..de580e971d --- /dev/null +++ b/modules/audio_filter/param_eq.c @@ -0,0 +1,344 @@ +/***************************************************************************** + * param_eq.c: + ***************************************************************************** + * Copyright (C) 2006 the VideoLAN team + * $Id: equalizer.c 13905 2006-01-12 23:10:04Z dionoea $ + * + * Authors: Antti Huovilainen + * Sigmund A. Helberg + * + * 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 /* malloc(), free() */ +#include +#include + +#include + +#include +#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( N_("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)"),NULL, VLC_FALSE ); + add_float_with_range( "param-eq-lowgain", 0, -20.0, 20.0, NULL, + N_("Low freq gain (Db)"), NULL,VLC_FALSE ); + add_float( "param-eq-highf", 10000, NULL, N_("High freq (Hz)"),NULL, VLC_FALSE ); + add_float_with_range( "param-eq-highgain", 0, -20.0, 20.0, NULL, + N_("High freq gain (Db)"), NULL,VLC_FALSE ); + add_float( "param-eq-f1", 300, NULL, N_("Freq 1 (Hz)"),NULL, VLC_FALSE ); + add_float_with_range( "param-eq-gain1", 0, -20.0, 20.0, NULL, + N_("Freq 1 gain (Db)"), NULL,VLC_FALSE ); + add_float_with_range( "param-eq-q1", 3, 0.1, 100.0, NULL, + N_("Freq 1 Q"), NULL,VLC_FALSE ); + add_float( "param-eq-f2", 1000, NULL, N_("Freq 2 (Hz)"),NULL, VLC_FALSE ); + add_float_with_range( "param-eq-gain2", 0, -20.0, 20.0, NULL, + N_("Freq 2 gain (Db)"), NULL,VLC_FALSE ); + add_float_with_range( "param-eq-q2", 3, 0.1, 100.0, NULL, + N_("Freq 2 Q"), NULL,VLC_FALSE ); + add_float( "param-eq-f3", 3000, NULL, N_("Freq 3 (Hz)"),NULL, VLC_FALSE ); + add_float_with_range( "param-eq-gain3", 0, -20.0, 20.0, NULL, + N_("Freq 3 gain (Db)"), NULL,VLC_FALSE ); + add_float_with_range( "param-eq-q3", 3, 0.1, 100.0, NULL, + N_("Freq 3 Q"), NULL,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; + } + } +} + -- GitLab