@@ -40,6 +40,7 @@ version <next>:

 - apulsator filter

 - sidechaingate audio filter

 - mipsdspr1 option has been renamed to mipsdsp

+- aemphasis filter

 version 2.8:

@@ -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 ||

@@ -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.

@@ -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

new file mode 100644

@@ -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,

+};

@@ -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);

@@ -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, \