avfilter: add superequalizer filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>

Changelog

@@ -20,6 +20,7 @@ version <next>:
 - sofalizer filter switched to libmysofa
 - Gremlin Digital Video demuxer and decoder
 - headphone audio filter
+- superequalizer audio filter
 
 version 3.3:
 - CrystalHD decoder moved to new decode API

doc/filters.texi

@@ -3835,6 +3835,49 @@ channels. Default is 0.3.
 Set level of input signal of original channel. Default is 0.8.
 @end table
 
+@section superequalizer
+Apply 18th band equalizer.
+
+The filter accpets the following options:
+@table @option
+@item 1b
+Set 65Hz band gain.
+@item 2b
+Set 92Hz band gain.
+@item 3b
+Set 131Hz band gain.
+@item 4b
+Set 185Hz band gain.
+@item 5b
+Set 262Hz band gain.
+@item 6b
+Set 370Hz band gain.
+@item 7b
+Set 523Hz band gain.
+@item 8b
+Set 740Hz band gain.
+@item 9b
+Set 1047Hz band gain.
+@item 10b
+Set 1480Hz band gain.
+@item 11b
+Set 2093Hz band gain.
+@item 12b
+Set 2960Hz band gain.
+@item 13b
+Set 4186Hz band gain.
+@item 14b
+Set 5920Hz band gain.
+@item 15b
+Set 8372Hz band gain.
+@item 16b
+Set 11840Hz band gain.
+@item 17b
+Set 16744Hz band gain.
+@item 18b
+Set 20000Hz band gain.
+@end table
+
 @section surround
 Apply audio surround upmix filter.
 

libavfilter/Makefile

@@ -109,6 +109,7 @@ OBJS-$(CONFIG_SILENCEREMOVE_FILTER)          += af_silenceremove.o
 OBJS-$(CONFIG_SOFALIZER_FILTER)              += af_sofalizer.o
 OBJS-$(CONFIG_STEREOTOOLS_FILTER)            += af_stereotools.o
 OBJS-$(CONFIG_STEREOWIDEN_FILTER)            += af_stereowiden.o
+OBJS-$(CONFIG_SUPEREQUALIZER_FILTER)         += af_superequalizer.o
 OBJS-$(CONFIG_SURROUND_FILTER)               += af_surround.o
 OBJS-$(CONFIG_TREBLE_FILTER)                 += af_biquads.o
 OBJS-$(CONFIG_TREMOLO_FILTER)                += af_tremolo.o

libavfilter/af_superequalizer.c

+/*
+ * Copyright (c) 2002 Naoki Shibata
+ * Copyright (c) 2017 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/opt.h"
+
+#include "libavcodec/avfft.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+#define NBANDS 17
+#define M 15
+
+typedef struct EqParameter {
+    float lower, upper, gain;
+} EqParameter;
+
+typedef struct SuperEqualizerContext {
+    const AVClass *class;
+
+    EqParameter params[NBANDS + 1];
+
+    float gains[NBANDS + 1];
+
+    float fact[M + 1];
+    float aa;
+    float iza;
+    float *ires, *irest;
+    float *fsamples;
+    int winlen, tabsize;
+
+    AVFrame *in, *out;
+    RDFTContext *rdft, *irdft;
+} SuperEqualizerContext;
+
+static const float bands[] = {
+    65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023,
+    1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036
+};
+
+static float izero(SuperEqualizerContext *s, float x)
+{
+    float ret = 1;
+    int m;
+
+    for (m = 1; m <= M; m++) {
+        float t;
+
+        t = pow(x / 2, m) / s->fact[m];
+        ret += t*t;
+    }
+
+    return ret;
+}
+
+static float hn_lpf(int n, float f, float fs)
+{
+    float t = 1 / fs;
+    float omega = 2 * M_PI * f;
+
+    if (n * omega * t == 0)
+        return 2 * f * t;
+    return 2 * f * t * sinf(n * omega * t) / (n * omega * t);
+}
+
+static float hn_imp(int n)
+{
+    return n == 0 ? 1.f : 0.f;
+}
+
+static float hn(int n, EqParameter *param, float fs)
+{
+    float ret, lhn;
+    int i;
+
+    lhn = hn_lpf(n, param[0].upper, fs);
+    ret = param[0].gain*lhn;
+
+    for (i = 1; i < NBANDS + 1 && param[i].upper < fs / 2; i++) {
+        float lhn2 = hn_lpf(n, param[i].upper, fs);
+        ret += param[i].gain * (lhn2 - lhn);
+        lhn = lhn2;
+    }
+
+    ret += param[i].gain * (hn_imp(n) - lhn);
+
+    return ret;
+}
+
+static float alpha(float a)
+{
+    if (a <= 21)
+        return 0;
+    if (a <= 50)
+        return .5842f * pow(a - 21, 0.4f) + 0.07886f * (a - 21);
+    return .1102f * (a - 8.7f);
+}
+
+static float win(SuperEqualizerContext *s, float n, int N)
+{
+    return izero(s, alpha(s->aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza;
+}
+
+static void process_param(float *bc, EqParameter *param, float fs)
+{
+    int i;
+
+    for (i = 0; i <= NBANDS; i++) {
+        param[i].lower = i == 0 ? 0 : bands[i - 1];
+        param[i].upper = i == NBANDS - 1 ? fs : bands[i];
+        param[i].gain  = bc[i];
+    }
+}
+
+static int equ_init(SuperEqualizerContext *s, int wb)
+{
+    int i,j;
+
+    s->rdft  = av_rdft_init(wb, DFT_R2C);
+    s->irdft = av_rdft_init(wb, IDFT_C2R);
+    if (!s->rdft || !s->irdft)
+        return AVERROR(ENOMEM);
+
+    s->aa = 96;
+    s->winlen = (1 << (wb-1))-1;
+    s->tabsize  = 1 << wb;
+
+    s->ires     = av_calloc(s->tabsize, sizeof(float));
+    s->irest    = av_calloc(s->tabsize, sizeof(float));
+    s->fsamples = av_calloc(s->tabsize, sizeof(float));
+
+    for (i = 0; i <= M; i++) {
+        s->fact[i] = 1;
+        for (j = 1; j <= i; j++)
+            s->fact[i] *= j;
+    }
+
+    s->iza = izero(s, alpha(s->aa));
+
+    return 0;
+}
+
+static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
+{
+    const int winlen = s->winlen;
+    const int tabsize = s->tabsize;
+    float *nires;
+    int i;
+
+    if (fs <= 0)
+        return;
+
+    process_param(lbc, param, fs);
+    for (i = 0; i < winlen; i++)
+        s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
+    for (; i < tabsize; i++)
+        s->irest[i] = 0;
+
+    av_rdft_calc(s->rdft, s->irest);
+    nires = s->ires;
+    for (i = 0; i < tabsize; i++)
+        nires[i] = s->irest[i];
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    SuperEqualizerContext *s = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    const float *ires = s->ires;
+    float *fsamples = s->fsamples;
+    int ch, i;
+
+    AVFrame *out = ff_get_audio_buffer(outlink, s->winlen);
+    float *src, *dst, *ptr;
+
+    if (!out) {
+        av_frame_free(&in);
+        return AVERROR(ENOMEM);
+    }
+
+    for (ch = 0; ch < in->channels; ch++) {
+        ptr = (float *)out->extended_data[ch];
+        dst = (float *)s->out->extended_data[ch];
+        src = (float *)in->extended_data[ch];
+
+        for (i = 0; i < s->winlen; i++)
+            fsamples[i] = src[i];
+        for (; i < s->tabsize; i++)
+            fsamples[i] = 0;
+
+        av_rdft_calc(s->rdft, fsamples);
+
+        fsamples[0] = ires[0] * fsamples[0];
+        fsamples[1] = ires[1] * fsamples[1];
+        for (i = 1; i < s->tabsize / 2; i++) {
+            float re, im;
+
+            re = ires[i*2  ] * fsamples[i*2] - ires[i*2+1] * fsamples[i*2+1];
+            im = ires[i*2+1] * fsamples[i*2] + ires[i*2  ] * fsamples[i*2+1];
+
+            fsamples[i*2  ] = re;
+            fsamples[i*2+1] = im;
+        }
+
+        av_rdft_calc(s->irdft, fsamples);
+
+        for (i = 0; i < s->winlen; i++)
+            dst[i] += fsamples[i] / s->tabsize * 2;
+        for (i = s->winlen; i < s->tabsize; i++)
+            dst[i]  = fsamples[i] / s->tabsize * 2;
+        for (i = 0; i < s->winlen; i++)
+            ptr[i] = dst[i];
+        for (i = 0; i < s->winlen; i++)
+            dst[i] = dst[i+s->winlen];
+    }
+
+    out->pts = in->pts;
+    av_frame_free(&in);
+
+    return ff_filter_frame(outlink, out);
+}
+
+static av_cold int init(AVFilterContext *ctx)
+{
+    SuperEqualizerContext *s = ctx->priv;
+
+    return equ_init(s, 14);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats;
+    AVFilterChannelLayouts *layouts;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_FLTP,
+        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 ((ret = ff_set_common_formats(ctx, formats)) < 0)
+        return ret;
+
+    formats = ff_all_samplerates();
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    SuperEqualizerContext *s = ctx->priv;
+
+    inlink->partial_buf_size =
+    inlink->min_samples =
+    inlink->max_samples = s->winlen;
+
+    s->out = ff_get_audio_buffer(inlink, s->tabsize);
+    if (!s->out)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    SuperEqualizerContext *s = ctx->priv;
+
+    make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate);
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    SuperEqualizerContext *s = ctx->priv;
+
+    av_freep(&s->irest);
+    av_freep(&s->ires);
+    av_freep(&s->fsamples);
+    av_rdft_end(s->rdft);
+    av_rdft_end(s->irdft);
+}
+
+static const AVFilterPad superequalizer_inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+        .config_props = config_input,
+    },
+    { NULL }
+};
+
+static const AVFilterPad superequalizer_outputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .config_props = config_output,
+    },
+    { NULL }
+};
+
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define OFFSET(x) offsetof(SuperEqualizerContext, x)
+
+static const AVOption superequalizer_options[] = {
+    {  "1b", "set 65Hz band gain",    OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    {  "2b", "set 92Hz band gain",    OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    {  "3b", "set 131Hz band gain",   OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    {  "4b", "set 185Hz band gain",   OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    {  "5b", "set 262Hz band gain",   OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    {  "6b", "set 370Hz band gain",   OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    {  "7b", "set 523Hz band gain",   OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    {  "8b", "set 740Hz band gain",   OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    {  "9b", "set 1047Hz band gain",  OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    { "10b", "set 1480Hz band gain",  OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    { "11b", "set 2093Hz band gain",  OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    { "12b", "set 2960Hz band gain",  OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    { "13b", "set 4186Hz band gain",  OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    { "14b", "set 5920Hz band gain",  OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    { "15b", "set 8372Hz band gain",  OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    { "16b", "set 11840Hz band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    { "17b", "set 16744Hz band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    { "18b", "set 20000Hz band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(superequalizer);
+
+AVFilter ff_af_superequalizer = {
+    .name          = "superequalizer",
+    .description   = NULL_IF_CONFIG_SMALL("Apply 18-th band equalization filter."),
+    .priv_size     = sizeof(SuperEqualizerContext),
+    .priv_class    = &superequalizer_class,
+    .query_formats = query_formats,
+    .init          = init,
+    .uninit        = uninit,
+    .inputs        = superequalizer_inputs,
+    .outputs       = superequalizer_outputs,
+};

libavfilter/allfilters.c

@@ -122,6 +122,7 @@ static void register_all(void)
     REGISTER_FILTER(SOFALIZER,      sofalizer,      af);
     REGISTER_FILTER(STEREOTOOLS,    stereotools,    af);
     REGISTER_FILTER(STEREOWIDEN,    stereowiden,    af);
+    REGISTER_FILTER(SUPEREQUALIZER, superequalizer, af);
     REGISTER_FILTER(SURROUND,       surround,       af);
     REGISTER_FILTER(TREBLE,         treble,         af);
     REGISTER_FILTER(TREMOLO,        tremolo,        af);

libavfilter/version.h

@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   6
-#define LIBAVFILTER_VERSION_MINOR  92
+#define LIBAVFILTER_VERSION_MINOR  93
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \