avfilter: add acrusher filter

Changelog

@@ -14,6 +14,7 @@ version <next>:
 - MediaCodec hwaccel
 - True Audio (TTA) muxer
 - crystalizer audio filter
+- acrusher audio filter
 
 
 version 3.1:

doc/filters.texi

@@ -441,6 +441,64 @@ ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c
 @end example
 @end itemize
 
+@section acrusher
+
+Reduce audio bit resolution.
+
+This filter is bit crusher with enhanced funcionality. A bit crusher
+is used to audibly reduce number of bits an audio signal is sampled
+with. This doesn't change the bit depth at all, it just produces the
+effect. Material reduced in bit depth sounds more harsh and "digital".
+This filter is able to even round to continous values instead of discrete
+bit depths.
+Additionally it has a D/C offset which results in different crushing of
+the lower and the upper half of the signal.
+An Anti-Aliasing setting is able to produce "softer" crushing sounds.
+
+Another feature of this filter is the logarithmic mode.
+This setting switches from linear distances between bits to logarithmic ones.
+The result is a much more "natural" sounding crusher which doesn't gate low
+signals for example. The human ear has a logarithmic perception, too
+so this kind of crushing is much more pleasant.
+Logarithmic crushing is also able to get anti-aliased.
+
+The filter accepts the following options:
+
+@table @option
+@item level_in
+Set level in.
+
+@item level_out
+Set level out.
+
+@item bits
+Set bit reduction.
+
+@item mix
+Set mixing ammount.
+
+@item mode
+Can be linear: @code{lin} or logarithmic: @code{log}.
+
+@item dc
+Set DC.
+
+@item aa
+Set anti-aliasing.
+
+@item samples
+Set sample reduction.
+
+@item lfo
+Enable LFO. By default disabled.
+
+@item lforange
+Set LFO range.
+
+@item lforate
+Set LFO rate.
+@end table
+
 @section adelay
 
 Delay one or more audio channels.

libavfilter/Makefile

@@ -30,6 +30,7 @@ OBJS-$(HAVE_THREADS)                         += pthread.o
 OBJS-$(CONFIG_ABENCH_FILTER)                 += f_bench.o
 OBJS-$(CONFIG_ACOMPRESSOR_FILTER)            += af_sidechaincompress.o
 OBJS-$(CONFIG_ACROSSFADE_FILTER)             += af_afade.o
+OBJS-$(CONFIG_ACRUSHER_FILTER)               += af_acrusher.o
 OBJS-$(CONFIG_ADELAY_FILTER)                 += af_adelay.o
 OBJS-$(CONFIG_AECHO_FILTER)                  += af_aecho.o
 OBJS-$(CONFIG_AEMPHASIS_FILTER)              += af_aemphasis.o

libavfilter/af_acrusher.c

+/*
+ * Copyright (c) Markus Schmidt and Christian Holschuh
+ *
+ * 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 "avfilter.h"
+#include "internal.h"
+#include "audio.h"
+
+typedef struct LFOContext {
+    double freq;
+    double offset;
+    int srate;
+    double amount;
+    double pwidth;
+    double phase;
+} LFOContext;
+
+typedef struct SRContext {
+    double target;
+    double real;
+    double samples;
+    double last;
+} SRContext;
+
+typedef struct ACrusherContext {
+    const AVClass *class;
+
+    double level_in;
+    double level_out;
+    double bits;
+    double mix;
+    int mode;
+    double dc;
+    double idc;
+    double aa;
+    double samples;
+    int is_lfo;
+    double lforange;
+    double lforate;
+
+    double sqr;
+    double aa1;
+    double coeff;
+    int    round;
+    double sov;
+    double smin;
+    double sdiff;
+
+    LFOContext lfo;
+    SRContext *sr;
+} ACrusherContext;
+
+#define OFFSET(x) offsetof(ACrusherContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption acrusher_options[] = {
+    { "level_in", "set level in",         OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1},    0.015625, 64, A },
+    { "level_out","set level out",        OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},    0.015625, 64, A },
+    { "bits",     "set bit reduction",    OFFSET(bits),      AV_OPT_TYPE_DOUBLE, {.dbl=8},    1,        64, A },
+    { "mix",      "set mix",              OFFSET(mix),       AV_OPT_TYPE_DOUBLE, {.dbl=.5},   0,         1, A },
+    { "mode",     "set mode",             OFFSET(mode),      AV_OPT_TYPE_INT,    {.i64=0},    0,         1, A, "mode" },
+    {   "lin",    "linear",               0,                 AV_OPT_TYPE_CONST,  {.i64=0},    0,         0, A, "mode" },
+    {   "log",    "logarithmic",          0,                 AV_OPT_TYPE_CONST,  {.i64=1},    0,         0, A, "mode" },
+    { "dc",       "set DC",               OFFSET(dc),        AV_OPT_TYPE_DOUBLE, {.dbl=1},  .25,         4, A },
+    { "aa",       "set anti-aliasing",    OFFSET(aa),        AV_OPT_TYPE_DOUBLE, {.dbl=.5},   0,         1, A },
+    { "samples",  "set sample reduction", OFFSET(samples),   AV_OPT_TYPE_DOUBLE, {.dbl=1},    1,       250, A },
+    { "lfo",      "enable LFO",           OFFSET(is_lfo),    AV_OPT_TYPE_BOOL,   {.i64=0},    0,         1, A },
+    { "lforange", "set LFO depth",        OFFSET(lforange),  AV_OPT_TYPE_DOUBLE, {.dbl=20},   1,       250, A },
+    { "lforate",  "set LFO rate",         OFFSET(lforate),   AV_OPT_TYPE_DOUBLE, {.dbl=.3}, .01,       200, A },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(acrusher);
+
+static double samplereduction(ACrusherContext *s, SRContext *sr, double in)
+{
+    sr->samples++;
+    if (sr->samples >= s->round) {
+        sr->target += s->samples;
+        sr->real += s->round;
+        if (sr->target + s->samples >= sr->real + 1) {
+            sr->last = in;
+            sr->target = 0;
+            sr->real   = 0;
+        }
+        sr->samples = 0;
+    }
+    return sr->last;
+}
+
+static double add_dc(double s, double dc, double idc)
+{
+    return s > 0 ? s * dc : s * idc;
+}
+
+static double remove_dc(double s, double dc, double idc)
+{
+    return s > 0 ? s * idc : s * dc;
+}
+
+static inline double factor(double y, double k, double aa1, double aa)
+{
+    return 0.5 * (sin(M_PI * (fabs(y - k) - aa1) / aa - M_PI_2) + 1);
+}
+
+static double bitreduction(ACrusherContext *s, double in)
+{
+    const double sqr = s->sqr;
+    const double coeff = s->coeff;
+    const double aa = s->aa;
+    const double aa1 = s->aa1;
+    double y, k;
+
+    // add dc
+    in = add_dc(in, s->dc, s->idc);
+
+    // main rounding calculation depending on mode
+
+    // the idea for anti-aliasing:
+    // you need a function f which brings you to the scale, where
+    // you want to round and the function f_b (with f(f_b)=id) which
+    // brings you back to your original scale.
+    //
+    // then you can use the logic below in the following way:
+    // y = f(in) and k = roundf(y)
+    // if (y > k + aa1)
+    //      k = f_b(k) + ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1)
+    // if (y < k + aa1)
+    //      k = f_b(k) - ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1)
+    //
+    // whereas x = (fabs(f(in) - k) - aa1) * PI / aa
+    // for both cases.
+
+    switch (s->mode) {
+    case 0:
+    default:
+        // linear
+        y = in * coeff;
+        k = roundf(y);
+        if (k - aa1 <= y && y <= k + aa1) {
+            k /= coeff;
+        } else if (y > k + aa1) {
+            k = k / coeff + ((k + 1) / coeff - k / coeff) *
+                factor(y, k, aa1, aa);
+        } else {
+            k = k / coeff - (k / coeff - (k - 1) / coeff) *
+                factor(y, k, aa1, aa);
+        }
+        break;
+    case 1:
+        // logarithmic
+        y = sqr * log(fabs(in)) + sqr * sqr;
+        k = roundf(y);
+        if(!in) {
+            k = 0;
+        } else if (k - aa1 <= y && y <= k + aa1) {
+            k = in / fabs(in) * exp(k / sqr - sqr);
+        } else if (y > k + aa1) {
+            double x = exp(k / sqr - sqr);
+            k = FFSIGN(in) * (x + (exp((k + 1) / sqr - sqr) - x) *
+                factor(y, k, aa1, aa));
+        } else {
+            double x = exp(k / sqr - sqr);
+            k = in / fabs(in) * (x - (x - exp((k - 1) / sqr - sqr)) *
+                factor(y, k, aa1, aa));
+        }
+        break;
+    }
+
+    // mix between dry and wet signal
+    k += (in - k) * s->mix;
+
+    // remove dc
+    k = remove_dc(k, s->dc, s->idc);
+
+    return k;
+}
+
+static double lfo_get(LFOContext *lfo)
+{
+    double phs = FFMIN(100., lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
+    double val;
+
+    if (phs > 1)
+        phs = fmod(phs, 1.);
+
+    val = sin((phs * 360.) * M_PI / 180);
+
+    return val * lfo->amount;
+}
+
+static void lfo_advance(LFOContext *lfo, unsigned count)
+{
+    lfo->phase = fabs(lfo->phase + count * lfo->freq * (1. / lfo->srate));
+    if (lfo->phase >= 1.)
+        lfo->phase = fmod(lfo->phase, 1.);
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    ACrusherContext *s = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AVFrame *out;
+    const double *src = (const double *)in->data[0];
+    double *dst;
+    const double level_in = s->level_in;
+    const double level_out = s->level_out;
+    const double mix = s->mix;
+    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++) {
+        if (s->is_lfo) {
+            s->samples = s->smin + s->sdiff * (lfo_get(&s->lfo) + 0.5);
+            s->round = round(s->samples);
+        }
+
+        for (c = 0; c < inlink->channels; c++) {
+            double sample = src[c] * level_in;
+
+            sample = mix * samplereduction(s, &s->sr[c], sample) + src[c] * (1. - mix) * level_in;
+            dst[c] = bitreduction(s, sample) * level_out;
+        }
+        src += c;
+        dst += c;
+
+        if (s->is_lfo)
+            lfo_advance(&s->lfo, 1);
+    }
+
+    if (in != out)
+        av_frame_free(&in);
+
+    return ff_filter_frame(outlink, out);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats;
+    AVFilterChannelLayouts *layouts;
+    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 av_cold void uninit(AVFilterContext *ctx)
+{
+    ACrusherContext *s = ctx->priv;
+
+    av_freep(&s->sr);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    ACrusherContext *s = ctx->priv;
+    double rad, sun, smax, sov;
+
+    s->idc = 1. / s->dc;
+    s->coeff = exp2(s->bits) - 1;
+    s->sqr = sqrt(s->coeff / 2);
+    s->aa1 = (1. - s->aa) / 2.;
+    s->round = round(s->samples);
+    rad = s->lforange / 2.;
+    s->smin = FFMAX(s->samples - rad, 1.);
+    sun = s->samples - rad - s->smin;
+    smax = FFMIN(s->samples + rad, 250.);
+    sov = s->samples + rad - smax;
+    smax -= sun;
+    s->smin -= sov;
+    s->sdiff = smax - s->smin;
+
+    s->lfo.freq = s->lforate;
+    s->lfo.pwidth = 1.;
+    s->lfo.srate = inlink->sample_rate;
+    s->lfo.amount = .5;
+
+    s->sr = av_calloc(inlink->channels, sizeof(*s->sr));
+    if (!s->sr)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static const AVFilterPad avfilter_af_acrusher_inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .config_props = config_input,
+        .filter_frame = filter_frame,
+    },
+    { NULL }
+};
+
+static const AVFilterPad avfilter_af_acrusher_outputs[] = {
+    {
+        .name = "default",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_acrusher = {
+    .name          = "acrusher",
+    .description   = NULL_IF_CONFIG_SMALL("Reduce audio bit resolution."),
+    .priv_size     = sizeof(ACrusherContext),
+    .priv_class    = &acrusher_class,
+    .uninit        = uninit,
+    .query_formats = query_formats,
+    .inputs        = avfilter_af_acrusher_inputs,
+    .outputs       = avfilter_af_acrusher_outputs,
+};

libavfilter/allfilters.c

@@ -48,6 +48,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(ABENCH,         abench,         af);
     REGISTER_FILTER(ACOMPRESSOR,    acompressor,    af);
     REGISTER_FILTER(ACROSSFADE,     acrossfade,     af);
+    REGISTER_FILTER(ACRUSHER,       acrusher,       af);
     REGISTER_FILTER(ADELAY,         adelay,         af);
     REGISTER_FILTER(AECHO,          aecho,          af);
     REGISTER_FILTER(AEMPHASIS,      aemphasis,      af);

libavfilter/version.h

@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   6
-#define LIBAVFILTER_VERSION_MINOR  50
+#define LIBAVFILTER_VERSION_MINOR  51
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \