@@ -35,6 +35,7 @@ version <next>:

 - anoisesrc audio filter source

 - IVR demuxer

 - compensationdelay filter

+- acompressor filter

 version 2.8:

@@ -318,6 +318,78 @@ build.

 Below is a description of the currently available audio filters.

+@section acompressor

+

+A compressor is mainly used to reduce the dynamic range of a signal.

+Especially modern music is mostly compressed at a high ratio to

+improve the overall loudness. It's done to get the highest attention

+of a listener, "fatten" the sound and bring more "power" to the track.

+If a signal is compressed too much it may sound dull or "dead"

+afterwards or it may start to "pump" (which could be a powerful effect

+but can also destroy a track completely).

+The right compression is the key to reach a professional sound and is

+the high art of mixing and mastering. Because of its complex settings

+it may take a long time to get the right feeling for this kind of effect.

+

+Compression is done by detecting the volume above a chosen level

+@code{threshold} and dividing it by the factor set with @code{ratio}.

+So if you set the threshold to -12dB and your signal reaches -6dB a ratio

+of 2:1 will result in a signal at -9dB. Because an exact manipulation of

+the signal would cause distortion of the waveform the reduction can be

+levelled over the time. This is done by setting "Attack" and "Release".

+@code{attack} determines how long the signal has to rise above the threshold

+before any reduction will occur and @code{release} sets the time the signal

+has to fall below the threshold to reduce the reduction again. Shorter signals

+than the chosen attack time will be left untouched.

+The overall reduction of the signal can be made up afterwards with the

+@code{makeup} setting. So compressing the peaks of a signal about 6dB and

+raising the makeup to this level results in a signal twice as loud than the

+source. To gain a softer entry in the compression the @code{knee} flattens the

+hard edge at the threshold in the range of the chosen decibels.

+

+The filter accepts the following options:

+

+@table @option

+@item threshold

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

+reduction of the first stream.

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

+

+@item ratio

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

+rose 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 below 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 input stream

+or the louder(@code{maximum}) channel of input 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 mostly smoother.

+

+@item mix

+How much to use compressed signal in output. Default is 1.

+Range is between 0 and 1.

+@end table

+

 @section acrossfade

 Apply cross fade from one input audio stream to another input audio stream.

@@ -23,6 +23,7 @@ OBJS = allfilters.o                                                     \

        transform.o                                                      \

        video.o                                                          \

+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

@@ -21,7 +21,7 @@

 /**

  * @file

- * Sidechain compressor filter

+ * Audio (Sidechain) Compressor filter

  */

 #include "libavutil/avassert.h"

@@ -61,7 +61,7 @@ typedef struct SidechainCompressContext {

 #define A AV_OPT_FLAG_AUDIO_PARAM

 #define F AV_OPT_FLAG_FILTERING_PARAM

-static const AVOption sidechaincompress_options[] = {

+static const AVOption 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 },

@@ -78,6 +78,7 @@ static const AVOption sidechaincompress_options[] = {

     { NULL }

 };

+#define sidechaincompress_options options

 AVFILTER_DEFINE_CLASS(sidechaincompress);

 static av_cold int init(AVFilterContext *ctx)

@@ -126,33 +127,24 @@ static double output_gain(double lin_slope, double ratio, double thres,

     return exp(gain - slope);

 }

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

+static int compressor_config_output(AVFilterLink *outlink)

 {

-    AVFilterContext *ctx = link->dst;

+    AVFilterContext *ctx = outlink->src;

     SidechainCompressContext *s = ctx->priv;

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

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

-    const double makeup = s->makeup;

-    const double mix = s->mix;

-    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(i < 2 && !s->input_frame[i]);

-    s->input_frame[i] = frame;

-

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

-        return 0;

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

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

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

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

+    return 0;

+}

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

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

+static void compressor(SidechainCompressContext *s,

+                       double *sample, const double *scsrc, int nb_samples,

+                       AVFilterLink *inlink, AVFilterLink *sclink)

+{

+    const double makeup = s->makeup;

+    const double mix = s->mix;

+    int i, c;

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

         double abs_sample, gain = 1.0;

@@ -179,13 +171,42 @@ static int filter_frame(AVFilterLink *link, AVFrame *frame)

                                s->knee_start, s->knee_stop,

                                s->compressed_knee_stop, s->detection);

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

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

             sample[c] *= (gain * makeup * mix + (1. - mix));

-        sample += outlink->channels;

+        sample += inlink->channels;

         scsrc += sclink->channels;

     }

+}

+

+#if CONFIG_SIDECHAINCOMPRESS_FILTER

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

+{

+    AVFilterContext *ctx = link->dst;

+    SidechainCompressContext *s = ctx->priv;

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

+    const double *scsrc;

+    double *sample;

+    int nb_samples;

+    int ret, i;

+

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

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

+            break;

+    av_assert0(i < 2 && !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];

+    compressor(s, sample, scsrc, nb_samples,

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

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

     s->input_frame[0] = NULL;

@@ -253,7 +274,6 @@ static int query_formats(AVFilterContext *ctx)

 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,

@@ -268,8 +288,7 @@ static int config_output(AVFilterLink *outlink)

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

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

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

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

+    compressor_config_output(outlink);

     return 0;

 }

@@ -310,3 +329,83 @@ AVFilter ff_af_sidechaincompress = {

     .inputs         = sidechaincompress_inputs,

     .outputs        = sidechaincompress_outputs,

 };

+#endif  /* CONFIG_SIDECHAINCOMPRESS_FILTER */

+

+#if CONFIG_ACOMPRESSOR_FILTER

+static int acompressor_filter_frame(AVFilterLink *inlink, AVFrame *frame)

+{

+    AVFilterContext *ctx = inlink->dst;

+    SidechainCompressContext *s = ctx->priv;

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

+    double *sample;

+

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

+    compressor(s, sample, sample, frame->nb_samples,

+               inlink, inlink);

+

+    return ff_filter_frame(outlink, frame);

+}

+

+static int acompressor_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);

+}

+

+#define acompressor_options options

+AVFILTER_DEFINE_CLASS(acompressor);

+

+static const AVFilterPad acompressor_inputs[] = {

+    {

+        .name           = "default",

+        .type           = AVMEDIA_TYPE_AUDIO,

+        .filter_frame   = acompressor_filter_frame,

+        .needs_writable = 1,

+    },

+    { NULL }

+};

+

+static const AVFilterPad acompressor_outputs[] = {

+    {

+        .name          = "default",

+        .type          = AVMEDIA_TYPE_AUDIO,

+        .config_props  = compressor_config_output,

+    },

+    { NULL }

+};

+

+AVFilter ff_af_acompressor = {

+    .name           = "acompressor",

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

+    .priv_size      = sizeof(SidechainCompressContext),

+    .priv_class     = &acompressor_class,

+    .init           = init,

+    .query_formats  = acompressor_query_formats,

+    .inputs         = acompressor_inputs,

+    .outputs        = acompressor_outputs,

+};

+#endif  /* CONFIG_ACOMPRESSOR_FILTER */

@@ -45,6 +45,7 @@ void avfilter_register_all(void)

         return;

     initialized = 1;

+    REGISTER_FILTER(ACOMPRESSOR,    acompressor,    af);

     REGISTER_FILTER(ACROSSFADE,     acrossfade,     af);

     REGISTER_FILTER(ADELAY,         adelay,         af);

     REGISTER_FILTER(AECHO,          aecho,          af);

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

 #include "libavutil/version.h"

 #define LIBAVFILTER_VERSION_MAJOR   6

-#define LIBAVFILTER_VERSION_MINOR  16

+#define LIBAVFILTER_VERSION_MINOR  17

 #define LIBAVFILTER_VERSION_MICRO 100

 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \