avfilter: add audio emphasis filter

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

Changelog

@@ -40,6 +40,7 @@ version <next>:
 - apulsator filter
 - sidechaingate audio filter
 - mipsdspr1 option has been renamed to mipsdsp
+- aemphasis filter
 
 
 version 2.8:

configure

@@ -1051,6 +1051,21 @@ int main(void){ $func(); }
 EOF
 }
 
+check_complexfunc(){
+    log check_complexfunc "$@"
+    func=$1
+    narg=$2
+    shift 2
+    test $narg = 2 && args="f, g" || args="f * I"
+    disable $func
+    check_ld "cc" "$@" <<EOF && enable $func
+#include <complex.h>
+#include <math.h>
+float foo(complex float f, complex float g) { return $func($args); }
+int main(void){ return (int) foo; }
+EOF
+}
+
 check_mathfunc(){
     log check_mathfunc "$@"
     func=$1
@@ -1768,6 +1783,11 @@ INTRINSICS_LIST="
     intrinsics_neon
 "
 
+COMPLEX_FUNCS="
+    cabs
+    cexp
+"
+
 MATH_FUNCS="
     atanf
     atan2f
@@ -1903,6 +1923,7 @@ HAVE_LIST="
     $ARCH_FEATURES
     $ATOMICS_LIST
     $BUILTIN_LIST
+    $COMPLEX_FUNCS
     $HAVE_LIST_CMDLINE
     $HAVE_LIST_PUB
     $HEADERS_LIST
@@ -2785,6 +2806,7 @@ unix_protocol_deps="sys_un_h"
 unix_protocol_select="network"
 
 # filters
+aemphasis_filter_deps="cabs cexp"
 amovie_filter_deps="avcodec avformat"
 aresample_filter_deps="swresample"
 ass_filter_deps="libass"
@@ -5324,6 +5346,10 @@ for func in $MATH_FUNCS; do
     eval check_mathfunc $func \${${func}_args:-1}
 done
 
+for func in $COMPLEX_FUNCS; do
+    eval check_complexfunc $func \${${func}_args:-1}
+done
+
 # these are off by default, so fail if requested and not available
 enabled avfoundation_indev && { check_header_oc AVFoundation/AVFoundation.h || disable avfoundation_indev; }
 enabled avfoundation_indev && { check_lib2 CoreGraphics/CoreGraphics.h CGGetActiveDisplayList -framework CoreGraphics ||

doc/filters.texi

@@ -528,6 +528,52 @@ aecho=0.8:0.9:1000|1800:0.3|0.25
 @end example
 @end itemize
 
+@section aemphasis
+Audio emphasis filter creates or restores material directly taken from LPs or
+emphased CDs with different filter curves. E.g. to store music on vinyl the
+signal has to be altered by a filter first to even out the disadvantages of
+this recording medium.
+Once the material is played back the inverse filter has to be applied to
+restore the distortion of the frequency response.
+
+The filter accepts the following options:
+
+@table @option
+@item level_in
+Set input gain.
+
+@item level_out
+Set output gain.
+
+@item mode
+Set filter mode. For restoring material use @code{reproduction} mode, otherwise
+use @code{production} mode. Default is @code{reproduction} mode.
+
+@item type
+Set filter type. Selects medium. Can be one of the following:
+
+@table @option
+@item col
+select Columbia.
+@item emi
+select EMI.
+@item bsi
+select BSI (78RPM).
+@item riaa
+select RIAA.
+@item cd
+select Compact Disc (CD).
+@item 50fm
+select 50µs (FM).
+@item 75fm
+select 75µs (FM).
+@item 50kf
+select 50µs (FM-KF).
+@item 75kf
+select 75µs (FM-KF).
+@end table
+@end table
+
 @section aeval
 
 Modify an audio signal according to the specified expressions.

libavfilter/Makefile

@@ -27,6 +27,7 @@ OBJS-$(CONFIG_ACOMPRESSOR_FILTER)            += af_sidechaincompress.o
 OBJS-$(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_AEVAL_FILTER)                  += aeval.o
 OBJS-$(CONFIG_AFADE_FILTER)                  += af_afade.o
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o

libavfilter/af_aemphasis.c

@@ -0,0 +1,370 @@
+/*
+ * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit and others
+ *
+ * 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 <complex.h>
+
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "internal.h"
+#include "audio.h"
+
+typedef struct BiquadCoeffs {
+    double a0, a1, a2, b1, b2;
+} BiquadCoeffs;
+
+typedef struct BiquadD2 {
+    double a0, a1, a2, b1, b2, w1, w2;
+} BiquadD2;
+
+typedef struct RIAACurve {
+    BiquadD2 r1;
+    BiquadD2 brickw;
+    int use_brickw;
+} RIAACurve;
+
+typedef struct AudioEmphasisContext {
+    const AVClass *class;
+    int mode, type;
+    double level_in, level_out;
+
+    RIAACurve *rc;
+} AudioEmphasisContext;
+
+#define OFFSET(x) offsetof(AudioEmphasisContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption aemphasis_options[] = {
+    { "level_in",      "set input gain", OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
+    { "level_out",    "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
+    { "mode",         "set filter mode", OFFSET(mode), AV_OPT_TYPE_INT,   {.i64=0}, 0, 1, FLAGS, "mode" },
+    { "reproduction",              NULL,            0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" },
+    { "production",                NULL,            0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" },
+    { "type",         "set filter type", OFFSET(type), AV_OPT_TYPE_INT,   {.i64=4}, 0, 8, FLAGS, "type" },
+    { "col",                 "Columbia",            0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "type" },
+    { "emi",                      "EMI",            0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "type" },
+    { "bsi",              "BSI (78RPM)",            0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, "type" },
+    { "riaa",                    "RIAA",            0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, FLAGS, "type" },
+    { "cd",         "Compact Disc (CD)",            0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, FLAGS, "type" },
+    { "50fm",               "50µs (FM)",            0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, FLAGS, "type" },
+    { "75fm",               "75µs (FM)",            0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, FLAGS, "type" },
+    { "50kf",            "50µs (FM-KF)",            0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, FLAGS, "type" },
+    { "75kf",            "75µs (FM-KF)",            0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, FLAGS, "type" },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(aemphasis);
+
+static inline double biquad(BiquadD2 *bq, double in)
+{
+    double n = in;
+    double tmp = n - bq->w1 * bq->b1 - bq->w2 * bq->b2;
+    double out = tmp * bq->a0 + bq->w1 * bq->a1 + bq->w2 * bq->a2;
+
+    bq->w2 = bq->w1;
+    bq->w1 = tmp;
+
+    return out;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AudioEmphasisContext *s = ctx->priv;
+    const double *src = (const double *)in->data[0];
+    const double level_out = s->level_out;
+    const double level_in = s->level_in;
+    AVFrame *out;
+    double *dst;
+    int n, c;
+
+    if (av_frame_is_writable(in)) {
+        out = in;
+    } else {
+        out = ff_get_audio_buffer(inlink, in->nb_samples);
+        if (!out) {
+            av_frame_free(&in);
+            return AVERROR(ENOMEM);
+        }
+        av_frame_copy_props(out, in);
+    }
+    dst = (double *)out->data[0];
+
+    for (n = 0; n < in->nb_samples; n++) {
+        for (c = 0; c < inlink->channels; c++)
+            dst[c] = level_out * biquad(&s->rc[c].r1, s->rc[c].use_brickw ? biquad(&s->rc[c].brickw, src[c] * level_in) : src[c] * level_in);
+        dst += inlink->channels;
+        src += inlink->channels;
+    }
+
+    if (in != out)
+        av_frame_free(&in);
+    return ff_filter_frame(outlink, out);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterChannelLayouts *layouts;
+    AVFilterFormats *formats;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_DBL,
+        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 inline void set_highshelf_rbj(BiquadD2 *bq, double freq, double q, double peak, double sr)
+{
+    double A = sqrt(peak);
+    double w0 = freq * 2 * M_PI / sr;
+    double alpha = sin(w0) / (2 * q);
+    double cw0 = cos(w0);
+    double tmp = 2 * sqrt(A) * alpha;
+    double b0 = 0, ib0 = 0;
+
+    bq->a0 =    A*( (A+1) + (A-1)*cw0 + tmp);
+    bq->a1 = -2*A*( (A-1) + (A+1)*cw0);
+    bq->a2 =    A*( (A+1) + (A-1)*cw0 - tmp);
+        b0 =        (A+1) - (A-1)*cw0 + tmp;
+    bq->b1 =    2*( (A-1) - (A+1)*cw0);
+    bq->b2 =        (A+1) - (A-1)*cw0 - tmp;
+
+    ib0     = 1 / b0;
+    bq->b1 *= ib0;
+    bq->b2 *= ib0;
+    bq->a0 *= ib0;
+    bq->a1 *= ib0;
+    bq->a2 *= ib0;
+}
+
+static inline void set_lp_rbj(BiquadD2 *bq, double fc, double q, double sr, double gain)
+{
+    double omega = 2.0 * M_PI * fc / sr;
+    double sn = sin(omega);
+    double cs = cos(omega);
+    double alpha = sn/(2 * q);
+    double inv = 1.0/(1.0 + alpha);
+
+    bq->a2 = bq->a0 = gain * inv * (1.0 - cs) * 0.5;
+    bq->a1 = bq->a0 + bq->a0;
+    bq->b1 = (-2.0 * cs * inv);
+    bq->b2 = ((1.0 - alpha) * inv);
+}
+
+static double freq_gain(BiquadCoeffs *c, double freq, double sr)
+{
+    double complex z, w;
+
+    freq *= 2.0 * M_PI / sr;
+    w = 0 + I * freq;
+    z = 1.0 / cexp(w);
+
+    return cabs(((double complex)c->a0 + c->a1 * z + c->a2 * z*z) /
+                ((double complex)1.0 + c->b1 * z + c->b2 * z*z));
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    double i, j, k, g, t, a0, a1, a2, b1, b2, tau1, tau2, tau3;
+    double cutfreq, gain1kHz, gc, sr = inlink->sample_rate;
+    AVFilterContext *ctx = inlink->dst;
+    AudioEmphasisContext *s = ctx->priv;
+    BiquadCoeffs coeffs;
+    int ch;
+
+    s->rc = av_calloc(inlink->channels, sizeof(*s->rc));
+    if (!s->rc)
+        return AVERROR(ENOMEM);
+
+    switch (s->type) {
+    case 0: //"Columbia"
+        i = 100.;
+        j = 500.;
+        k = 1590.;
+        break;
+    case 1: //"EMI"
+        i = 70.;
+        j = 500.;
+        k = 2500.;
+        break;
+    case 2: //"BSI(78rpm)"
+        i = 50.;
+        j = 353.;
+        k = 3180.;
+        break;
+    case 3: //"RIAA"
+    default:
+        tau1 = 0.003180;
+        tau2 = 0.000318;
+        tau3 = 0.000075;
+        i = 1. / (2. * M_PI * tau1);
+        j = 1. / (2. * M_PI * tau2);
+        k = 1. / (2. * M_PI * tau3);
+        break;
+    case 4: //"CD Mastering"
+        tau1 = 0.000050;
+        tau2 = 0.000015;
+        tau3 = 0.0000001;// 1.6MHz out of audible range for null impact
+        i = 1. / (2. * M_PI * tau1);
+        j = 1. / (2. * M_PI * tau2);
+        k = 1. / (2. * M_PI * tau3);
+        break;
+    case 5: //"50µs FM (Europe)"
+        tau1 = 0.000050;
+        tau2 = tau1 / 20;// not used
+        tau3 = tau1 / 50;//
+        i = 1. / (2. * M_PI * tau1);
+        j = 1. / (2. * M_PI * tau2);
+        k = 1. / (2. * M_PI * tau3);
+        break;
+    case 6: //"75µs FM (US)"
+        tau1 = 0.000075;
+        tau2 = tau1 / 20;// not used
+        tau3 = tau1 / 50;//
+        i = 1. / (2. * M_PI * tau1);
+        j = 1. / (2. * M_PI * tau2);
+        k = 1. / (2. * M_PI * tau3);
+        break;
+    }
+
+    i *= 2 * M_PI;
+    j *= 2 * M_PI;
+    k *= 2 * M_PI;
+
+    t = 1. / sr;
+
+    //swap a1 b1, a2 b2
+    if (s->type == 7 || s->type == 8) {
+        s->rc[0].use_brickw = 0;
+        double tau = (s->type == 7 ? 0.000050 : 0.000075);
+        double f = 1.0 / (2 * M_PI * tau);
+        double nyq = sr * 0.5;
+        double gain = sqrt(1.0 + nyq * nyq / (f * f)); // gain at Nyquist
+        double cfreq = sqrt((gain - 1.0) * f * f); // frequency
+        double q = 1.0;
+
+        if (s->type == 8)
+            q = pow((sr / 3269.0) + 19.5, -0.25); // somewhat poor curve-fit
+        if (s->type == 7)
+            q = pow((sr / 4750.0) + 19.5, -0.25);
+        if (s->mode == 0)
+            set_highshelf_rbj(&s->rc[0].r1, cfreq, q, 1. / gain, sr);
+        else
+            set_highshelf_rbj(&s->rc[0].r1, cfreq, q, gain, sr);
+    } else {
+        s->rc[0].use_brickw = 1;
+        if (s->mode == 0) { // Reproduction
+            g  = 1. / (4.+2.*i*t+2.*k*t+i*k*t*t);
+            a0 = (2.*t+j*t*t)*g;
+            a1 = (2.*j*t*t)*g;
+            a2 = (-2.*t+j*t*t)*g;
+            b1 = (-8.+2.*i*k*t*t)*g;
+            b2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
+        } else {  // Production
+            g  = 1. / (2.*t+j*t*t);
+            a0 = (4.+2.*i*t+2.*k*t+i*k*t*t)*g;
+            a1 = (-8.+2.*i*k*t*t)*g;
+            a2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
+            b1 = (2.*j*t*t)*g;
+            b2 = (-2.*t+j*t*t)*g;
+        }
+
+        coeffs.a0 = a0;
+        coeffs.a1 = a1;
+        coeffs.a2 = a2;
+        coeffs.b1 = b1;
+        coeffs.b2 = b2;
+
+        // the coeffs above give non-normalized value, so it should be normalized to produce 0dB at 1 kHz
+        // find actual gain
+        // Note: for FM emphasis, use 100 Hz for normalization instead
+        gain1kHz = freq_gain(&coeffs, 1000.0, sr);
+        // divide one filter's x[n-m] coefficients by that value
+        gc = 1.0 / gain1kHz;
+        s->rc[0].r1.a0 = coeffs.a0 * gc;
+        s->rc[0].r1.a1 = coeffs.a1 * gc;
+        s->rc[0].r1.a2 = coeffs.a2 * gc;
+        s->rc[0].r1.b1 = coeffs.b1;
+        s->rc[0].r1.b2 = coeffs.b2;
+    }
+
+    cutfreq = FFMIN(0.45 * sr, 21000.);
+    set_lp_rbj(&s->rc[0].brickw, cutfreq, 0.707, sr, 1.);
+
+    for (ch = 1; ch < inlink->channels; ch++) {
+        memcpy(&s->rc[ch], &s->rc[0], sizeof(RIAACurve));
+    }
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AudioEmphasisContext *s = ctx->priv;
+    av_freep(&s->rc);
+}
+
+static const AVFilterPad avfilter_af_aemphasis_inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .config_props = config_input,
+        .filter_frame = filter_frame,
+    },
+    { NULL }
+};
+
+static const AVFilterPad avfilter_af_aemphasis_outputs[] = {
+    {
+        .name = "default",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_aemphasis = {
+    .name          = "aemphasis",
+    .description   = NULL_IF_CONFIG_SMALL("Audio emphasis."),
+    .priv_size     = sizeof(AudioEmphasisContext),
+    .priv_class    = &aemphasis_class,
+    .uninit        = uninit,
+    .query_formats = query_formats,
+    .inputs        = avfilter_af_aemphasis_inputs,
+    .outputs       = avfilter_af_aemphasis_outputs,
+};

libavfilter/allfilters.c

@@ -49,6 +49,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(ACROSSFADE,     acrossfade,     af);
     REGISTER_FILTER(ADELAY,         adelay,         af);
     REGISTER_FILTER(AECHO,          aecho,          af);
+    REGISTER_FILTER(AEMPHASIS,      aemphasis,      af);
     REGISTER_FILTER(AEVAL,          aeval,          af);
     REGISTER_FILTER(AFADE,          afade,          af);
     REGISTER_FILTER(AFORMAT,        aformat,        af);

libavfilter/version.h

@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   6
-#define LIBAVFILTER_VERSION_MINOR  19
+#define LIBAVFILTER_VERSION_MINOR  20
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \