@@ -24,6 +24,7 @@ version <next>:

 - Random filter

 - deband filter

 - AAC fixed-point decoding

+- sidechaincompress audio filter

 version 2.7:

@@ -622,6 +622,7 @@ slope

 Specify the band-width of a filter in width_type units.

 @end table

+@anchor{amerge}

 @section amerge

 Merge two or more audio streams into a single multi-channel stream.

@@ -2020,6 +2021,7 @@ Applies only to double-pole filter.

 The default is 0.707q and gives a Butterworth response.

 @end table

+@anchor{pan}

 @section pan

 Mix channels with specific gain levels. The filter accepts the output

@@ -2121,6 +2123,66 @@ At end of filtering it displays @code{track_gain} and @code{track_peak}.

 Convert the audio sample format, sample rate and channel layout. It is

 not meant to be used directly.

+@section sidechaincompress

+

+This filter acts like normal compressor but has the ability to compress

+detected signal using second input signal.

+It needs two input streams and returns one output stream.

+First input stream will be processed depending on second stream signal.

+The filtered signal then can be filtered with other filters in later stages of

+processing. See @ref{pan} and @ref{amerge} filter.

+

+The filter accepts the following options:

+

+@table @option

+@item threshold

+If a signal of second stream raises above this level it will affect the gain

+reduction of first stream.

+By default is 0.125. Range is between 0.00097563 and 1.

+

+@item ratio

+Set a ratio about which the signal is reduced. 1:2 means that if the level

+raised 4dB above the threshold, it will be only 2dB above after the reduction.

+Default is 2. Range is between 1 and 20.

+

+@item attack

+Amount of milliseconds the signal has to rise above the threshold before gain

+reduction starts. Default is 20. Range is between 0.01 and 2000.

+

+@item release

+Amount of milliseconds the signal has to fall bellow the threshold before

+reduction is decreased again. Default is 250. Range is between 0.01 and 9000.

+

+@item makeup

+Set the amount by how much signal will be amplified after processing.

+Default is 2. Range is from 1 and 64.

+

+@item knee

+Curve the sharp knee around the threshold to enter gain reduction more softly.

+Default is 2.82843. Range is between 1 and 8.

+

+@item link

+Choose if the @code{average} level between all channels of side-chain stream

+or the louder(@code{maximum}) channel of side-chain stream affects the

+reduction. Default is @code{average}.

+

+@item detection

+Should the exact signal be taken in case of @code{peak} or an RMS one in case

+of @code{rms}. Default is @code{rms} which is mainly smoother.

+@end table

+

+@subsection Examples

+

+@itemize

+@item

+Full ffmpeg example taking 2 audio inputs, 1st input to be compressed

+depending on the signal of 2nd input and later compressed signal to be

+merged with 2nd input:

+@example

+ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"

+@end example

+@end itemize

+

 @section silencedetect

 Detect silence in an audio stream.

@@ -80,6 +80,7 @@ OBJS-$(CONFIG_LOWPASS_FILTER)                += af_biquads.o

 OBJS-$(CONFIG_PAN_FILTER)                    += af_pan.o

 OBJS-$(CONFIG_REPLAYGAIN_FILTER)             += af_replaygain.o

 OBJS-$(CONFIG_RESAMPLE_FILTER)               += af_resample.o

+OBJS-$(CONFIG_SIDECHAINCOMPRESS_FILTER)      += af_sidechaincompress.o

 OBJS-$(CONFIG_SILENCEDETECT_FILTER)          += af_silencedetect.o

 OBJS-$(CONFIG_SILENCEREMOVE_FILTER)          += af_silenceremove.o

 OBJS-$(CONFIG_TREBLE_FILTER)                 += af_biquads.o

new file mode 100644

@@ -0,0 +1,338 @@

+/*

+ * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others

+ * Copyright (c) 2015 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

+ */

+

+/**

+ * @file

+ * Sidechain compressor filter

+ */

+

+#include "libavutil/avassert.h"

+#include "libavutil/channel_layout.h"

+#include "libavutil/common.h"

+#include "libavutil/opt.h"

+

+#include "audio.h"

+#include "avfilter.h"

+#include "formats.h"

+#include "internal.h"

+

+typedef struct SidechainCompressContext {

+    const AVClass *class;

+

+    double attack, attack_coeff;

+    double release, release_coeff;

+    double lin_slope;

+    double ratio;

+    double threshold;

+    double makeup;

+    double thres;

+    double knee;

+    double knee_start;

+    double knee_stop;

+    double lin_knee_start;

+    double compressed_knee_stop;

+    int link;

+    int detection;

+

+    AVFrame *input_frame[2];

+} SidechainCompressContext;

+

+#define OFFSET(x) offsetof(SidechainCompressContext, x)

+#define A AV_OPT_FLAG_AUDIO_PARAM

+#define F AV_OPT_FLAG_FILTERING_PARAM

+

+static const AVOption sidechaincompress_options[] = {

+    { "threshold", "set threshold",    OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125}, 0.000976563,    1, A|F },

+    { "ratio",     "set ratio",        OFFSET(ratio),     AV_OPT_TYPE_DOUBLE, {.dbl=2},               1,   20, A|F },

+    { "attack",    "set attack",       OFFSET(attack),    AV_OPT_TYPE_DOUBLE, {.dbl=20},           0.01, 2000, A|F },

+    { "release",   "set release",      OFFSET(release),   AV_OPT_TYPE_DOUBLE, {.dbl=250},          0.01, 9000, A|F },

+    { "makeup",    "set make up gain", OFFSET(makeup),    AV_OPT_TYPE_DOUBLE, {.dbl=2},               1,   64, A|F },

+    { "knee",      "set knee",         OFFSET(knee),      AV_OPT_TYPE_DOUBLE, {.dbl=2.82843},         1,    8, A|F },

+    { "link",      "set link type",    OFFSET(link),      AV_OPT_TYPE_INT,    {.i64=0},               0,    1, A|F, "link" },

+    {   "average", 0,                  0,                 AV_OPT_TYPE_CONST,  {.i64=0},               0,    0, A|F, "link" },

+    {   "maximum", 0,                  0,                 AV_OPT_TYPE_CONST,  {.i64=1},               0,    0, A|F, "link" },

+    { "detection", "set detection",    OFFSET(detection), AV_OPT_TYPE_INT,    {.i64=1},               0,    1, A|F, "detection" },

+    {   "peak",    0,                  0,                 AV_OPT_TYPE_CONST,  {.i64=0},               0,    0, A|F, "detection" },

+    {   "rms",     0,                  0,                 AV_OPT_TYPE_CONST,  {.i64=1},               0,    0, A|F, "detection" },

+    { NULL }

+};

+

+AVFILTER_DEFINE_CLASS(sidechaincompress);

+

+static av_cold int init(AVFilterContext *ctx)

+{

+    SidechainCompressContext *s = ctx->priv;

+

+    s->thres = log(s->threshold);

+    s->lin_knee_start = s->threshold / sqrt(s->knee);

+    s->knee_start = log(s->lin_knee_start);

+    s->knee_stop = log(s->threshold * sqrt(s->knee));

+    s->compressed_knee_stop = (s->knee_stop - s->thres) / s->ratio + s->thres;

+

+    return 0;

+}

+

+static inline float hermite_interpolation(float x, float x0, float x1,

+                                          float p0, float p1,

+                                          float m0, float m1)

+{

+    float width = x1 - x0;

+    float t = (x - x0) / width;

+    float t2, t3;

+    float ct0, ct1, ct2, ct3;

+

+    m0 *= width;

+    m1 *= width;

+

+    t2 = t*t;

+    t3 = t2*t;

+    ct0 = p0;

+    ct1 = m0;

+

+    ct2 = -3 * p0 - 2 * m0 + 3 * p1 - m1;

+    ct3 = 2 * p0 + m0  - 2 * p1 + m1;

+

+    return ct3 * t3 + ct2 * t2 + ct1 * t + ct0;

+}

+

+// A fake infinity value (because real infinity may break some hosts)

+#define FAKE_INFINITY (65536.0 * 65536.0)

+

+// Check for infinity (with appropriate-ish tolerance)

+#define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)

+

+static double output_gain(double lin_slope, double ratio, double thres,

+                          double knee, double knee_start, double knee_stop,

+                          double compressed_knee_stop, int detection)

+{

+    double slope = log(lin_slope);

+    double gain = 0.0;

+    double delta = 0.0;

+

+    if (detection)

+        slope *= 0.5;

+

+    if (IS_FAKE_INFINITY(ratio)) {

+        gain = thres;

+        delta = 0.0;

+    } else {

+        gain = (slope - thres) / ratio + thres;

+        delta = 1.0 / ratio;

+    }

+

+    if (knee > 1.0 && slope < knee_stop)

+        gain = hermite_interpolation(slope, knee_start, knee_stop,

+                                     knee_start, compressed_knee_stop,

+                                     1.0, delta);

+

+    return exp(gain - slope);

+}

+

+static int filter_frame(AVFilterLink *link, AVFrame *frame)

+{

+    AVFilterContext *ctx = link->dst;

+    SidechainCompressContext *s = ctx->priv;

+    AVFilterLink *sclink = ctx->inputs[1];

+    AVFilterLink *outlink = ctx->outputs[0];

+    const double makeup = s->makeup;

+    const double *scsrc;

+    double *sample;

+    int nb_samples;

+    int ret, i, c;

+

+    for (i = 0; i < 2; i++)

+        if (link == ctx->inputs[i])

+            break;

+    av_assert0(!s->input_frame[i]);

+    s->input_frame[i] = frame;

+

+    if (!s->input_frame[0] || !s->input_frame[1])

+        return 0;

+

+    nb_samples = FFMIN(s->input_frame[0]->nb_samples,

+                       s->input_frame[1]->nb_samples);

+

+    sample = (double *)s->input_frame[0]->data[0];

+    scsrc = (const double *)s->input_frame[1]->data[0];

+

+    for (i = 0; i < nb_samples; i++) {

+        double abs_sample, gain = 1.0;

+

+        abs_sample = FFABS(scsrc[0]);

+

+        if (s->link == 1) {

+            for (c = 1; c < sclink->channels; c++)

+                abs_sample = FFMAX(FFABS(scsrc[c]), abs_sample);

+        } else {

+            for (c = 1; c < sclink->channels; c++)

+                abs_sample += FFABS(scsrc[c]);

+

+            abs_sample /= sclink->channels;

+        }

+

+        if (s->detection)

+            abs_sample *= abs_sample;

+

+        s->lin_slope += (abs_sample - s->lin_slope) * (abs_sample > s->lin_slope ? s->attack_coeff : s->release_coeff);

+

+        if (s->lin_slope > 0.0 && s->lin_slope > s->lin_knee_start)

+            gain = output_gain(s->lin_slope, s->ratio, s->thres, s->knee,

+                               s->knee_start, s->knee_stop,

+                               s->compressed_knee_stop, s->detection);

+

+        for (c = 0; c < outlink->channels; c++)

+            sample[c] *= gain * makeup;

+

+        sample += outlink->channels;

+        scsrc += sclink->channels;

+    }

+

+    ret = ff_filter_frame(outlink, s->input_frame[0]);

+

+    s->input_frame[0] = NULL;

+    av_frame_free(&s->input_frame[1]);

+

+    return ret;

+}

+

+static int request_frame(AVFilterLink *outlink)

+{

+    AVFilterContext *ctx = outlink->src;

+    SidechainCompressContext *s = ctx->priv;

+    int i, ret;

+

+    /* get a frame on each input */

+    for (i = 0; i < 2; i++) {

+        AVFilterLink *inlink = ctx->inputs[i];

+        if (!s->input_frame[i] &&

+            (ret = ff_request_frame(inlink)) < 0)

+            return ret;

+

+        /* request the same number of samples on all inputs */

+        if (i == 0)

+            ctx->inputs[1]->request_samples = s->input_frame[0]->nb_samples;

+    }

+

+    return 0;

+}

+

+static int query_formats(AVFilterContext *ctx)

+{

+    AVFilterFormats *formats;

+    AVFilterChannelLayouts *layouts = NULL;

+    static const enum AVSampleFormat sample_fmts[] = {

+        AV_SAMPLE_FMT_DBL,

+        AV_SAMPLE_FMT_NONE

+    };

+    int ret, i;

+

+    if (!ctx->inputs[0]->in_channel_layouts ||

+        !ctx->inputs[0]->in_channel_layouts->nb_channel_layouts) {

+        av_log(ctx, AV_LOG_WARNING,

+               "No channel layout for input 1\n");

+            return AVERROR(EAGAIN);

+    }

+

+    ff_add_channel_layout(&layouts, ctx->inputs[0]->in_channel_layouts->channel_layouts[0]);

+    if (!layouts)

+        return AVERROR(ENOMEM);

+    ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts);

+

+    for (i = 0; i < 2; i++) {

+        layouts = ff_all_channel_layouts();

+        if (!layouts)

+            return AVERROR(ENOMEM);

+        ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts);

+    }

+

+    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 int config_output(AVFilterLink *outlink)

+{

+    AVFilterContext *ctx = outlink->src;

+    SidechainCompressContext *s = ctx->priv;

+

+    if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {

+        av_log(ctx, AV_LOG_ERROR,

+               "Inputs must have the same sample rate "

+               "%d for in0 vs %d for in1\n",

+               ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);

+        return AVERROR(EINVAL);

+    }

+

+    outlink->sample_rate = ctx->inputs[0]->sample_rate;

+    outlink->time_base   = ctx->inputs[0]->time_base;

+    outlink->channel_layout = ctx->inputs[0]->channel_layout;

+    outlink->channels = ctx->inputs[0]->channels;

+

+    s->attack_coeff = FFMIN(1.f, 1.f / (s->attack * outlink->sample_rate / 4000.f));

+    s->release_coeff = FFMIN(1.f, 1.f / (s->release * outlink->sample_rate / 4000.f));

+

+    return 0;

+}

+

+static const AVFilterPad sidechaincompress_inputs[] = {

+    {

+        .name           = "main",

+        .type           = AVMEDIA_TYPE_AUDIO,

+        .filter_frame   = filter_frame,

+        .needs_writable = 1,

+        .needs_fifo     = 1,

+    },{

+        .name           = "sidechain",

+        .type           = AVMEDIA_TYPE_AUDIO,

+        .filter_frame   = filter_frame,

+        .needs_fifo     = 1,

+    },

+    { NULL }

+};

+

+static const AVFilterPad sidechaincompress_outputs[] = {

+    {

+        .name          = "default",

+        .type          = AVMEDIA_TYPE_AUDIO,

+        .config_props  = config_output,

+        .request_frame = request_frame,

+    },

+    { NULL }

+};

+

+AVFilter ff_af_sidechaincompress = {

+    .name           = "sidechaincompress",

+    .description    = NULL_IF_CONFIG_SMALL("Sidechain compressor."),

+    .priv_size      = sizeof(SidechainCompressContext),

+    .priv_class     = &sidechaincompress_class,

+    .init           = init,

+    .query_formats  = query_formats,

+    .inputs         = sidechaincompress_inputs,

+    .outputs        = sidechaincompress_outputs,

+};

@@ -96,6 +96,7 @@ void avfilter_register_all(void)

     REGISTER_FILTER(PAN,            pan,            af);

     REGISTER_FILTER(REPLAYGAIN,     replaygain,     af);

     REGISTER_FILTER(RESAMPLE,       resample,       af);

+    REGISTER_FILTER(SIDECHAINCOMPRESS, sidechaincompress, af);

     REGISTER_FILTER(SILENCEDETECT,  silencedetect,  af);

     REGISTER_FILTER(SILENCEREMOVE,  silenceremove,  af);

     REGISTER_FILTER(TREBLE,         treble,         af);

@@ -30,7 +30,7 @@

 #include "libavutil/version.h"

 #define LIBAVFILTER_VERSION_MAJOR  5

-#define LIBAVFILTER_VERSION_MINOR  28

+#define LIBAVFILTER_VERSION_MINOR  29

 #define LIBAVFILTER_VERSION_MICRO 100

 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \