ref: c35561001da2eccbf68c793c523db2fba1563fce
parent: e8d9e315c9ca2ec268363816a8f718fdb32913ec
author: rrt <rrt>
date: Mon Nov 13 20:52:27 EST 2006
Add equalizer effect from Pascal Giard
--- a/src/Makefile.dos
+++ b/src/Makefile.dos
@@ -19,15 +19,14 @@
raw.obj sf.obj smp.obj sndrtool.obj sphere.obj sunaudio.obj \
tx16w.obj voc.obj vorbis.obj vox.obj wav.obj wve.obj
-EOBJ = avg.obj band.obj bandpass.obj breject.obj btrworth.obj chorus.obj \
+EOBJ = avg.obj band.obj bandpass.obj biquad.obj breject.obj btrworth.obj chorus.obj \
compand.obj copy.obj dcshift.obj deemphas.obj earwax.o \
- echo.obj echos.obj fade.obj FFT.obj filter.obj flanger.obj \
- biquad.obj tone.obj \
+ echo.obj echos.obj equalizer.obj fade.obj FFT.obj filter.obj flanger.obj \
highp.obj highpass.obj lowp.obj lowpass.obj mask.obj mcompand.obj \
noiseprof.obj noisered.obj phaser.obj pitch.obj pan.obj \
polyphase.obj rate.obj repeat.obj resample.obj \
reverb.obj reverse.obj silence.obj speed.obj stat.obj \
- stretch.obj swap.obj synth.obj trim.obj vibro.obj vol.obj
+ stretch.obj swap.obj synth.obj tone.obj trim.obj vibro.obj vol.obj
LIBOBJS = $(FOBJ) $(EOBJ) handlers.obj misc.obj stio.obj getopt.obj \
getopt1.obj util.obj
--- a/src/Makefile.gcc
+++ b/src/Makefile.gcc
@@ -25,13 +25,12 @@
sndrtool.o sphere.o sunaudio.o tx16w.o voc.o vorbis.o \
vox.o wav.o wve.o
-EOBJ = avg.o band.o bandpass.o breject.o btrworth.o chorus.o compand.o \
- copy.o dcshift.o deemphas.o earwax.o echo.o echos.o fade.o FFT.o \
- biquad.o tone.o \
- filter.o flanger.o highp.o highpass.o lowp.o lowpass.o \
+EOBJ = avg.o band.o bandpass.o biquad.o breject.o btrworth.o chorus.o compand.o \
+ copy.o dcshift.o deemphas.o earwax.o echo.o echos.o equalizer.o \
+ fade.o FFT.o filter.o flanger.o highp.o highpass.o lowp.o lowpass.o \
mask.o mcompand.o noiseprof.o noisered.o pan.o phaser.o pitch.o \
polyphas.o rate.o repeat.o resample.o reverb.o reverse.o \
- silence.o speed.o stat.o stretch.o swap.o synth.o trim.o \
+ silence.o speed.o stat.o stretch.o swap.o synth.o tone.o trim.o \
vibro.o vol.o
SOUNDLIB = libst.a
--- a/src/Makefile.in
+++ b/src/Makefile.in
@@ -47,14 +47,13 @@
raw.o sf.o smp.o sndrtool.o sphere.o tx16w.o voc.o vorbis.o \
vox.o wav.o wve.o xa.o
-EOBJ = avg.o band.o bandpass.o breject.o btrworth.o chorus.o compand.o \
- copy.o dcshift.o deemphas.o earwax.o echo.o echos.o \
- biquad.o tone.o \
- fade.o FFT.o filter.o flanger.o highp.o highpass.o lowp.o \
+EOBJ = avg.o band.o bandpass.o biquad.o breject.o btrworth.o chorus.o \
+ compand.o copy.o dcshift.o deemphas.o earwax.o echo.o echos.o \
+ equalizer.o fade.o FFT.o filter.o flanger.o highp.o highpass.o lowp.o \
lowpass.o mask.o mcompand.o noiseprof.o noisered.o pan.o \
phaser.o pitch.o polyphas.o rabbit.o rate.o repeat.o resample.o \
reverb.o reverse.o silence.o speed.o stat.o \
- stretch.o swap.o synth.o trim.o vibro.o vol.o
+ stretch.o swap.o synth.o tone.o trim.o vibro.o vol.o
GSMOBJ_0 =
--- /dev/null
+++ b/src/equalizer.c
@@ -1,0 +1,184 @@
+/*
+ Equalizer filter effect file for SoX
+ Copyright (C) 2006 Pascal Giard <evilynux@gmail.com>
+
+ This library is free software; you can redistribute it and/or
+ modify it under the terms of the GNU Lesser General Public
+ License as published by the Free Software Foundation; either
+ version 2 of the License, or (at your option) any later version.
+
+ This library 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
+ Lesser General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public
+ License along with this library; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+
+ Code based on the biquad filters described in
+
+ Cookbook formulae for audio EQ biquad filter coefficients
+ by Robert Bristow-Johnson <rbj@audioimagination.com>
+
+ Theory:
+ y[n] = (a0/b0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2]
+ - (a1/a0)*y[n-1] - (a2/a0)*y[n-2]
+
+ Where:
+ w0 = 2*M_PI*cfreq/srate
+ A = 10^(gain/40)
+ alpha = sin(w0)/( 2*Q )
+
+ For a PeakingEQ filter:
+ b0 = 1 + alpha*A
+ b1 = -2*cos(w0)
+ b2 = 1 - alpha*A
+ a0 = 1 + alpha/A
+ a1 = -2*cos(w0)
+ a2 = 1 - alpha/A
+
+ Reminder:
+ Q = sqrt(2^n)/(2^n - 1) where n is bandwidth in octave
+ n = log2(bw) where bw is bandwidth in Hz
+
+ Transfer function is:
+ (a0/b0) + (b1/a0)z^-1 + (b2/a0)z^-2
+ H(z) = -----------------------------------
+ 1 + (a1/a0)z^-1 + (a2/a0)z^-2
+ */
+
+#include <math.h>
+#include "st_i.h"
+
+static st_effect_t st_equalizer_effect;
+
+/* Filter parameters */
+typedef struct filterparams {+ float rate; // Sample rate
+ float Q; // Q-factor
+ float cfreq; // Central frequency (Hz)
+ float gain; // Gain (dB)
+ double x[3]; // In where x[2] <=> x[ n - 2 ]
+ double y[3]; // Out
+ double b[3]; // From this point, equation constants...
+ double a[3];
+} *equalizer_t;
+
+int st_equalizer_getopts(eff_t effp, int n, char **argv)
+{+ equalizer_t eq = (equalizer_t) effp->priv;
+ int i;
+
+ if (n < 3)
+ {+ st_fail("Usage: equalizer center-freq Q gain");+ return (ST_EOF);
+ }
+
+ i = 0;
+ sscanf(argv[i++], "%f", &eq->cfreq);
+ sscanf(argv[i++], "%f", &eq->Q);
+ sscanf(argv[i++], "%f", &eq->gain);
+
+ // TODO: Would be nice to validate the params..
+
+ return (ST_SUCCESS);
+}
+
+// Set the filter constants
+int st_equalizer_start(eff_t effp)
+{+ equalizer_t eq = (equalizer_t) effp->priv;
+ double w0;
+ double amp;
+ double alpha;
+
+ // Sample rate
+ eq->rate = effp->ininfo.rate;
+
+ w0 = 2*M_PI*eq->cfreq/eq->rate;
+ amp = pow( 10, eq->gain/40 );
+ alpha = sin(w0)/( 2*eq->Q );
+
+ st_report("Debug: cfreq: %fHz", eq->cfreq);+ st_report("Debug: Q: %f", eq->Q);+ st_report("Debug: gain: %fdB", eq->gain);+ st_report("Debug: rate: %f", eq->rate);+ st_report("Debug: w0: %f", w0);+ st_report("Debug: amp: %f", amp);+ st_report("Debug: alpha: %f", alpha);+
+ // Initialisation
+ eq->b[0] = 1 + alpha*amp;
+ eq->b[1] = -2*cos(w0);
+ eq->b[2] = 1 - alpha*amp;
+ eq->a[0] = 1 + alpha/amp;
+ eq->a[1] = -2*cos(w0);
+ eq->a[2] = 1 - alpha/amp;
+
+ eq->x[0] = 0; // x[n]
+ eq->x[1] = 0; // x[n-1]
+ eq->x[2] = 0; // x[n-2]
+ eq->y[0] = 0; // y[n]
+ eq->y[1] = 0; // y[n-1]
+ eq->y[2] = 0; // y[n-2]
+
+ return (ST_SUCCESS);
+}
+
+int st_equalizer_flow(eff_t effp, st_sample_t *ibuf,
+ st_sample_t *obuf, st_size_t *isamp,
+ st_size_t *osamp)
+{+ equalizer_t eq = (equalizer_t) effp->priv;
+ st_size_t len, done;
+ double out;
+
+ len = ((*isamp > *osamp) ? *osamp : *isamp);
+
+ for(done = 0; done < len; done++) {+ eq->x[2] = eq->x[1];
+ eq->x[1] = eq->x[0];
+ eq->x[0] = *ibuf++;
+
+ eq->y[2] = eq->y[1];
+ eq->y[1] = eq->y[0];
+ out = (
+ (eq->b[0]/eq->a[0])*eq->x[0] +
+ (eq->b[1]/eq->a[0])*eq->x[1] +
+ (eq->b[2]/eq->a[0])*eq->x[2] -
+ (eq->a[1]/eq->a[0])*eq->y[1] -
+ (eq->a[2]/eq->a[0])*eq->y[2]
+ );
+ eq->y[0] = out;
+
+ if (out < ST_SAMPLE_MIN) {+ out = ST_SAMPLE_MIN;
+ }
+ else if (out > ST_SAMPLE_MAX) {+ out = ST_SAMPLE_MAX;
+ }
+
+ *obuf++ = out;
+ }
+
+ *isamp = len;
+ *osamp = len;
+
+ return (ST_SUCCESS);
+}
+
+static st_effect_t st_equalizer_effect = {+ "equalizer",
+ "Usage: equalizer central-freqency Q gain",
+ 0,
+ st_equalizer_getopts, st_equalizer_start,
+ st_equalizer_flow, st_effect_nothing_drain,
+ st_effect_nothing
+};
+
+const st_effect_t *st_equalizer_effect_fn(void)
+{+ return &st_equalizer_effect;
+}
--- a/src/handlers.c
+++ b/src/handlers.c
@@ -96,6 +96,7 @@
st_earwax_effect_fn,
st_echo_effect_fn,
st_echos_effect_fn,
+ st_equalizer_effect_fn,
st_fade_effect_fn,
st_filter_effect_fn,
st_flanger_effect_fn,
--- a/src/st_i.h
+++ b/src/st_i.h
@@ -246,6 +246,7 @@
extern const st_effect_t *st_earwax_effect_fn(void);
extern const st_effect_t *st_echo_effect_fn(void);
extern const st_effect_t *st_echos_effect_fn(void);
+extern const st_effect_t *st_equalizer_effect_fn(void);
extern const st_effect_t *st_fade_effect_fn(void);
extern const st_effect_t *st_filter_effect_fn(void);
extern const st_effect_t *st_flanger_effect_fn(void);