Signed-off-by: Paul B Mahol <one...@gmail.com>
---
doc/filters.texi | 49 ++++
libavfilter/Makefile | 1 +
libavfilter/af_anequalizer.c | 534 +++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 585 insertions(+)
create mode 100644 libavfilter/af_anequalizer.c
diff --git a/doc/filters.texi b/doc/filters.texi
index a55cad4..49921e2 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -992,6 +992,55 @@ stream ends. The default value is 2 seconds.
@end table
+@section anequalizer
+
+Parametric equalizer with unlimited number of bands for each channel.
+
+It accepts single option string in format:
+"c=@var{chn} f=@var{cf} w=@var{w} g=@var{g} t=@var{f} | ..."
+Each equalization filter is separated by '|'.
+
+@table @option
+@item chn
+Set channel number to which equalization will be applied.
+If input doesn't have that channel the entry is ignored.
+
+@item cf
+Set central frequency for band.
+If input doesn't have that frequency the entry is ignored.
+
+@item w
+Set band width in hertz
+
+@item g
+Set gain in dB
+
+@item f
+Set filter type, can be:
+
+@table @samp
+@item 0
+Butterworth.
+
+@item 1
+Chebyshev type 1.
+
+@item 2
+Chebyshev type 2.
+@end table
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Lower gain by 10 of central frequency 200Hz and width 100 Hz
+for first 2 channels using Chebyshev type 1 filter:
+@example
+anequalizer=args=c=0 f=200 w=100 g=-10 t=1|c=0 f=200 w=100 g=-10 t=1
+@end example
+@end itemize
+
@section anull
Pass the audio source unchanged to the output.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index dea012a..adbbc39 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -29,6 +29,7 @@ OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
+OBJS-$(CONFIG_ANEQUALIZER_FILTER) += af_anequalizer.o
OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
diff --git a/libavfilter/af_anequalizer.c b/libavfilter/af_anequalizer.c
new file mode 100644
index 0000000..31f4134
--- /dev/null
+++ b/libavfilter/af_anequalizer.c
@@ -0,0 +1,534 @@
+/*
+ * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald
Johansen and others
+ * Copyright (c) 2015 Paul B Mahol
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/avstring.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "internal.h"
+#include "audio.h"
+
+enum FilterType {
+ BUTTERWORTH,
+ CHEBYSHEV1,
+ CHEBYSHEV2,
+ NB_TYPES
+};
+
+typedef struct FoSection {
+ double a0, a1, a2, a3, a4;
+ double b0, b1, b2, b3, b4;
+
+ double num[4];
+ double denum[4];
+} FoSection;
+
+typedef struct EqualizatorFilter {
+ int ignore;
+ int channel;
+ int type;
+
+ double freq;
+ double gain;
+ double width;
+
+ FoSection section[2];
+} EqualizatorFilter;
+
+typedef struct AudioNEqualizerContext {
+ const AVClass *class;
+ char *args;
+
+ int nb_filters;
+ int nb_allocated;
+ EqualizatorFilter *filters;
+} AudioNEqualizerContext;
+
+#define OFFSET(x) offsetof(AudioNEqualizerContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption anequalizer_options[] = {
+ { "args", NULL, OFFSET(args), AV_OPT_TYPE_STRING, {.str=""}, 0, 0, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(anequalizer);
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBLP,
+ AV_SAMPLE_FMT_NONE
+ };
+ int ret;
+
+ layouts = ff_all_channel_counts();
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_all_samplerates();
+ if (!formats)
+ return AVERROR(ENOMEM);
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioNEqualizerContext *s = ctx->priv;
+
+ av_freep(&s->filters);
+ s->nb_filters = 0;
+ s->nb_allocated = 0;
+}
+
+static void butterworth_fo_section(FoSection *S, double beta,
+ double s, double g, double g0,
+ double D, double c0)
+{
+ S->b0 = (g*g*beta*beta + 2*g*g0*s*beta + g0*g0)/D;
+ S->b1 = -4*c0*(g0*g0 + g*g0*s*beta)/D;
+ S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - g*g*beta*beta)/D;
+ S->b3 = -4*c0*(g0*g0 - g*g0*s*beta)/D;
+ S->b4 = (g*g*beta*beta - 2*g*g0*s*beta + g0*g0)/D;
+
+ S->a0 = 1;
+ S->a1 = -4*c0*(1 + s*beta)/D;
+ S->a2 = 2*(1 + 2*c0*c0 - beta*beta)/D;
+ S->a3 = -4*c0*(1 - s*beta)/D;
+ S->a4 = (beta*beta - 2*s*beta + 1)/D;
+}
+
+static void butterworth_bp_filter(EqualizatorFilter *f,
+ int N, double w0, double wb,
+ double G, double Gb, double G0)
+{
+ double g, c0, g0, beta;
+ double epsilon;
+ int r = N % 2;
+ int L = (N - r) / 2;
+ int i;
+
+ if (G == 0 && G0 == 0) {
+ f->section[0].a0 = 1;
+ f->section[0].b0 = 1;
+ f->section[1].a0 = 1;
+ f->section[1].b0 = 1;
+ return;
+ }
+
+ G = pow(10, G/20);
+ Gb = pow(10, Gb/20);
+ G0 = pow(10, G0/20);
+
+ epsilon = sqrt((G * G - Gb * Gb) / (Gb * Gb - G0 * G0));
+ g = pow(G, 1.0 / (double)N);
+ g0 = pow(G0, 1.0 / (double)N);
+ beta = pow(epsilon, -1.0/(double)N) * tan(wb / 2.0);
+
+ c0 = cos(w0);
+ if (w0 == 0)
+ c0 = 1;
+ if (w0 == M_PI/2)
+ c0 = 0;
+ if (w0 == M_PI)
+ c0 = -1;
+
+ for (i = 1; i <= L; i++) {
+ double ui = (2.0 * i - 1) / N;
+ double si = sin(M_PI * ui / 2.0);
+ double Di = beta * beta + 2 * si * beta + 1;
+
+ butterworth_fo_section(&f->section[i - 1], beta, si, g, g0, Di, c0);
+ }
+}
+
+static void chebyshev1_fo_section(FoSection *S, double a,
+ double c, double tetta_b,
+ double g0, double s, double b,
+ double D, double c0)
+{
+ S->b0 = ((b*b + g0*g0*c*c)*tetta_b*tetta_b + 2*g0*b*s*tetta_b + g0*g0)/D;
+ S->b1 = -4*c0*(g0*g0 + g0*b*s*tetta_b)/D;
+ S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - (b*b + g0*g0*c*c)*tetta_b*tetta_b)/D;
+ S->b3 = -4*c0*(g0*g0 - g0*b*s*tetta_b)/D;
+ S->b4 = ((b*b + g0*g0*c*c)*tetta_b*tetta_b - 2*g0*b*s*tetta_b + g0*g0)/D;
+
+ S->a0 = 1;
+ S->a1 = -4*c0*(1 + a*s*tetta_b)/D;
+ S->a2 = 2*(1 + 2*c0*c0 - (a*a + c*c)*tetta_b*tetta_b)/D;
+ S->a3 = -4*c0*(1 - a*s*tetta_b)/D;
+ S->a4 = ((a*a + c*c)*tetta_b*tetta_b - 2*a*s*tetta_b + 1)/D;
+}
+
+static void chebyshev1_bp_filter(EqualizatorFilter *f,
+ int N, double w0, double wb,
+ double G, double Gb, double G0)
+{
+ double a, b, c0, g0, alfa, beta, tetta_b;
+ double epsilon;
+ int r = N % 2;
+ int L = (N - r) / 2;
+ int i;
+
+ if (G == 0 && G0 == 0) {
+ f->section[0].a0 = 1;
+ f->section[0].b0 = 1;
+ f->section[1].a0 = 1;
+ f->section[1].b0 = 1;
+ return;
+ }
+
+ G = pow(10, G/20);
+ Gb = pow(10, Gb/20);
+ G0 = pow(10, G0/20);
+
+ epsilon = sqrt((G*G - Gb*Gb) / (Gb*Gb - G0*G0));
+ g0 = pow(G0,1.0/N);
+ alfa = pow(1.0/epsilon + sqrt(1 + pow(epsilon,-2.0)), 1.0/N);
+ beta = pow(G/epsilon + Gb * sqrt(1 + pow(epsilon,-2.0)), 1.0/N);
+ a = 0.5 * (alfa - 1.0/alfa);
+ b = 0.5 * (beta - g0*g0*(1/beta));
+ tetta_b = tan(wb/2);
+
+ c0 = cos(w0);
+ if (w0 == 0)
+ c0 = 1;
+ if (w0 == M_PI/2)
+ c0 = 0;
+ if (w0 == M_PI)
+ c0 = -1;
+
+ for (i = 1; i <= L; i++) {
+ double ui = (2.0*i-1.0)/N;
+ double ci = cos(M_PI*ui/2.0);
+ double si = sin(M_PI*ui/2.0);
+ double Di = (a*a + ci*ci)*tetta_b*tetta_b + 2.0*a*si*tetta_b + 1;
+
+ chebyshev1_fo_section(&f->section[i -1], a, ci, tetta_b, g0, si, b,
Di, c0);
+ }
+}
+
+static void chebyshev2_fo_section(FoSection *S, double a,
+ double c, double tetta_b,
+ double g, double s, double b,
+ double D, double c0)
+{
+ S->b0 = (g*g*tetta_b*tetta_b + 2*g*b*s*tetta_b + b*b + g*g*c*c)/D;
+ S->b1 = -4*c0*(b*b + g*g*c*c + g*b*s*tetta_b)/D;
+ S->b2 = 2*((b*b + g*g*c*c)*(1 + 2*c0*c0) - g*g*tetta_b*tetta_b)/D;
+ S->b3 = -4*c0*(b*b + g*g*c*c - g*b*s*tetta_b)/D;
+ S->b4 = (g*g*tetta_b*tetta_b - 2*g*b*s*tetta_b + b*b + g*g*c*c)/D;
+
+ S->a0 = 1;
+ S->a1 = -4*c0*(a*a + c*c + a*s*tetta_b)/D;
+ S->a2 = 2*((a*a + c*c)*(1 + 2*c0*c0) - tetta_b*tetta_b)/D;
+ S->a3 = -4*c0*(a*a + c*c - a*s*tetta_b)/D;
+ S->a4 = (tetta_b*tetta_b - 2*a*s*tetta_b + a*a + c*c)/D;
+}
+
+static void chebyshev2_bp_filter(EqualizatorFilter *f,
+ int N, double w0, double wb,
+ double G, double Gb, double G0)
+{
+ double a, b, c0, tetta_b;
+ double epsilon, g, eu, ew;
+ int r = N % 2;
+ int L = (N - r) / 2;
+ int i;
+
+ if (G == 0 && G0 == 0) {
+ f->section[0].a0 = 1;
+ f->section[0].b0 = 1;
+ f->section[1].a0 = 1;
+ f->section[1].b0 = 1;
+ return;
+ }
+
+ G = pow(10, G/20);
+ Gb = pow(10, Gb/20);
+ G0 = pow(10, G0/20);
+
+ epsilon = sqrt((G*G - Gb*Gb) / (Gb*Gb - G0*G0));
+ g = pow(G, 1.0 / N);
+ eu = pow(epsilon + sqrt(1 + epsilon*epsilon), 1.0/N);
+ ew = pow(G0*epsilon + Gb*sqrt(1 + epsilon*epsilon), 1.0/N);
+ a = (eu - 1.0/eu)/2.0;
+ b = (ew - g*g/ew)/2.0;
+ tetta_b = tan(wb/2);
+
+ c0 = cos(w0);
+ if (w0 == 0)
+ c0 = 1;
+ if (w0 == M_PI/2)
+ c0 = 0;
+ if (w0 == M_PI)
+ c0 = -1;
+
+ for (i = 1; i <= L; i++) {
+ double ui = (2.0 * i - 1.0)/N;
+ double ci = cos(M_PI * ui / 2.0);
+ double si = sin(M_PI * ui / 2.0);
+ double Di = tetta_b*tetta_b + 2*a*si*tetta_b + a*a + ci*ci;
+
+ chebyshev2_fo_section(&f->section[i - 1], a, ci, tetta_b, g, si, b,
Di, c0);
+ }
+}
+
+static double butterworth_compute_bw_gain_db(double gain)
+{
+ double bw_gain = 0;
+
+ if (gain <= -6)
+ bw_gain = gain + 3;
+ else if(gain > -6 && gain < 6)
+ bw_gain = gain * 0.5;
+ else if(gain >= 6)
+ bw_gain = gain - 3;
+
+ return bw_gain;
+}
+
+static double chebyshev1_compute_bw_gain_db(double gain)
+{
+ double bw_gain = 0;
+
+ if (gain <= -6)
+ bw_gain = gain + 1;
+ else if(gain > -6 && gain < 6)
+ bw_gain = gain * 0.9;
+ else if(gain >= 6)
+ bw_gain = gain - 1;
+
+ return bw_gain;
+}
+
+static double chebyshev2_compute_bw_gain_db(double gain)
+{
+ double bw_gain = 0;
+
+ if (gain <= -6)
+ bw_gain = -3;
+ else if(gain > -6 && gain < 6)
+ bw_gain = gain * 0.3;
+ else if(gain >= 6)
+ bw_gain = 3;
+
+ return bw_gain;
+}
+
+static inline double hz_2_rad(double x, double fs)
+{
+ return 2 * M_PI * x / fs;
+}
+
+static void equalizer(EqualizatorFilter *f, double sample_rate)
+{
+ double w0 = hz_2_rad(f->freq, sample_rate);
+ double wb = hz_2_rad(f->width, sample_rate);
+ double bw_gain;
+
+ switch (f->type) {
+ case BUTTERWORTH:
+ bw_gain = butterworth_compute_bw_gain_db(f->gain);
+ butterworth_bp_filter(f, 4, w0, wb, f->gain, bw_gain, 0);
+ break;
+ case CHEBYSHEV1:
+ bw_gain = chebyshev1_compute_bw_gain_db(f->gain);
+ chebyshev1_bp_filter(f, 4, w0, wb, f->gain, bw_gain, 0);
+ break;
+ case CHEBYSHEV2:
+ bw_gain = chebyshev2_compute_bw_gain_db(f->gain);
+ chebyshev2_bp_filter(f, 4, w0, wb, f->gain, bw_gain, 0);
+ break;
+ }
+
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AudioNEqualizerContext *s = ctx->priv;
+ char *args = av_strdup(s->args);
+ char *saveptr = NULL;
+
+ if (!args)
+ return AVERROR(ENOMEM);
+
+ s->nb_allocated = 32 * inlink->channels;
+ s->filters = av_calloc(inlink->channels, 32 * sizeof(*s->filters));
+ if (!s->filters) {
+ s->nb_allocated = 0;
+ return AVERROR(ENOMEM);
+ }
+
+ while (1) {
+ char *arg = av_strtok(s->nb_filters == 0 ? args : NULL, "|", &saveptr);
+
+ if (!arg)
+ break;
+
+ s->filters[s->nb_filters].type = 0;
+ if (sscanf(arg, "c=%d f=%lf w=%lf g=%lf t=%d",
&s->filters[s->nb_filters].channel,
+
&s->filters[s->nb_filters].freq,
+
&s->filters[s->nb_filters].width,
+
&s->filters[s->nb_filters].gain,
+
&s->filters[s->nb_filters].type) != 5 &&
+ sscanf(arg, "c=%d f=%lf w=%lf g=%lf",
&s->filters[s->nb_filters].channel,
+
&s->filters[s->nb_filters].freq,
+
&s->filters[s->nb_filters].width,
+
&s->filters[s->nb_filters].gain) != 4 ) {
+ av_free(args);
+ return AVERROR(EINVAL);
+ }
+
+ if (s->filters[s->nb_filters].freq < 0 ||
+ s->filters[s->nb_filters].freq >= inlink->sample_rate / 2)
+ s->filters[s->nb_filters].ignore = 1;
+
+ if (s->filters[s->nb_filters].channel < 0 ||
+ s->filters[s->nb_filters].channel >= inlink->channels)
+ s->filters[s->nb_filters].ignore = 1;
+
+ av_clip(s->filters[s->nb_filters].type, 0, NB_TYPES - 1);
+ equalizer(&s->filters[s->nb_filters], inlink->sample_rate);
+ s->nb_filters++;
+ if (s->nb_filters >= s->nb_allocated) {
+ EqualizatorFilter *filters;
+
+ filters = av_calloc(s->nb_allocated, 2 * sizeof(*s->filters));
+ if (!filters) {
+ av_free(args);
+ return AVERROR(ENOMEM);
+ }
+ memcpy(filters, s->filters, sizeof(*s->filters) * s->nb_allocated);
+ av_free(s->filters);
+ s->filters = filters;
+ s->nb_allocated *= 2;
+ }
+ }
+
+ av_free(args);
+
+ return 0;
+}
+
+static inline double section_process(FoSection *S, double in)
+{
+ double out;
+
+ out = S->b0 * in;
+ out+= S->b1 * S->num[0] - S->denum[0] * S->a1;
+ out+= S->b2 * S->num[1] - S->denum[1] * S->a2;
+ out+= S->b3 * S->num[2] - S->denum[2] * S->a3;
+ out+= S->b4 * S->num[3] - S->denum[3] * S->a4;
+
+ S->num[3] = S->num[2];
+ S->num[2] = S->num[1];
+ S->num[1] = S->num[0];
+ S->num[0] = in;
+
+ S->denum[3] = S->denum[2];
+ S->denum[2] = S->denum[1];
+ S->denum[1] = S->denum[0];
+ S->denum[0] = out;
+
+ return out;
+}
+
+static double process_sample(FoSection *s1, FoSection *s2, double in)
+{
+ double p0 = in, p1;
+
+ p1 = section_process(s1, p0);
+ p1 = section_process(s2, p1);
+
+ return p1;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AudioNEqualizerContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ double *bptr;
+ int i, n;
+
+ for (i = 0; i < s->nb_filters; i++) {
+ EqualizatorFilter *f = &s->filters[i];
+
+ if (f->gain == 0. || f->ignore)
+ continue;
+
+ bptr = (double *)buf->extended_data[f->channel];
+ for (n = 0; n < buf->nb_samples; n++) {
+ double sample = bptr[n];
+
+ sample = process_sample(&f->section[0],
+ &f->section[1],
+ sample);
+ bptr[n] = sample;
+ }
+ }
+
+ return ff_filter_frame(outlink, buf);
+}
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_input,
+ .filter_frame = filter_frame,
+ .needs_writable = 1,
+ },
+ { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_anequalizer = {
+ .name = "anequalizer",
+ .description = NULL_IF_CONFIG_SMALL("Apply audio parametric N band
equalizer."),
+ .priv_size = sizeof(AudioNEqualizerContext),
+ .priv_class = &anequalizer_class,
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .inputs = inputs,
+ .outputs = outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 131e067..a039a39 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -59,6 +59,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(ALLPASS, allpass, af);
REGISTER_FILTER(AMERGE, amerge, af);
REGISTER_FILTER(AMIX, amix, af);
+ REGISTER_FILTER(ANEQUALIZER, anequalizer, af);
REGISTER_FILTER(ANULL, anull, af);
REGISTER_FILTER(APAD, apad, af);
REGISTER_FILTER(APERMS, aperms, af);
--
1.9.1
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