@@ -14,6 +14,7 @@ version <next>:

 - MediaCodec hwaccel

 - True Audio (TTA) muxer

 - crystalizer audio filter

+- acrusher audio filter

 version 3.1:

@@ -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.

@@ -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

new file mode 100644

@@ -0,0 +1,362 @@

+/*

+ * 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,

+};

@@ -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);

@@ -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, \