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

 - aecho filter

 - perspective filter ported from libmpcodecs

 - ffprobe -show_programs option

+- compand filter

 version 2.0:

@@ -1176,6 +1176,83 @@ front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'

 side_right.wav

 @end example

+@section compand

+

+Compress or expand audio dynamic range.

+

+A description of the accepted options follows.

+

+@table @option

+@item attacks

+@item decays

+Set list of times in seconds for each channel over which the instantaneous

+level of the input signal is averaged to determine its volume.

+@option{attacks} refers to increase of volume and @option{decays} refers

+to decrease of volume.

+For most situations, the attack time (response to the audio getting louder)

+should be shorter than the decay time because the human ear is more sensitive

+to sudden loud audio than sudden soft audio.

+Typical value for attack is @code{0.3} seconds and for decay @code{0.8}

+seconds.

+

+@item points

+Set list of points for transfer function, specified in dB relative to maximum

+possible signal amplitude.

+Each key points list need to be defined using the following syntax:

+@code{x0/y0 x1/y1 x2/y2 ...}.

+

+The input values must be in strictly increasing order but the transfer

+function does not have to me monotonically rising.

+The point @code{0/0} is assumed but may be overridden (by @code{0/out-dBn}).

+Typical values for the transfer function are @code{-70/-70 -60/-20}.

+

+@item soft-knee

+Set amount for which the points at where adjacent line segments on the

+transfer function meet will be rounded. Defaults is @code{0.01}.

+

+@item gain

+Set additional gain in dB to be applied at all points on the transfer function

+and allows easy adjustment of the overall gain.

+Default is @code{0}.

+

+@item volume

+Set initial volume in dB to be assumed for each channel when filtering starts.

+This permits the user to supply a nominal level initially, so that,

+for example, a very large gain is not applied to initial signal levels before

+the companding has begun to operate. A typical value for audio which is

+initially quiet is -90 dB. Default is @code{0}.

+

+@item delay

+Set delay in seconds. Default is @code{0}. The input audio

+is analysed immediately, but audio is delayed before being fed to the

+volume adjuster. Specifying a delay approximately equal to the attack/decay

+times allows the filter to effectively operate in predictive rather than

+reactive mode.

+@end table

+

+@subsection Examples

+@itemize

+@item

+Make music with both quiet and loud passages suitable for listening

+in a noisy environment:

+@example

+compand=.3 .3:1 1:-90/-60 -60/-40 -40/-30 -20/-20:6:0:-90:0.2

+@end example

+

+@item

+Noise-gate for when the noise is at a lower level than the signal:

+@example

+compand=.1 .1:.2 .2:-900/-900 -50.1/-900 -50/-50:.01:0:-90:.1

+@end example

+

+@item

+Here is another noise-gate, this time for when the noise is at a higher level

+than the signal (making it, in some ways, similar to squelch):

+@example

+compand=.1 .1:.1 .1:-45.1/-45.1 -45/-900 0/-900:.01:45:-90:.1

+@end example

+@end itemize

+

 @section earwax

 Make audio easier to listen to on headphones.

@@ -84,6 +84,7 @@ OBJS-$(CONFIG_BASS_FILTER)                   += af_biquads.o

 OBJS-$(CONFIG_BIQUAD_FILTER)                 += af_biquads.o

 OBJS-$(CONFIG_CHANNELMAP_FILTER)             += af_channelmap.o

 OBJS-$(CONFIG_CHANNELSPLIT_FILTER)           += af_channelsplit.o

+OBJS-$(CONFIG_COMPAND_FILTER)                += af_compand.o

 OBJS-$(CONFIG_EARWAX_FILTER)                 += af_earwax.o

 OBJS-$(CONFIG_EBUR128_FILTER)                += f_ebur128.o

 OBJS-$(CONFIG_EQUALIZER_FILTER)              += af_biquads.o

new file mode 100644

@@ -0,0 +1,515 @@

+/*

+ * Copyright (c) 1999 Chris Bagwell

+ * Copyright (c) 1999 Nick Bailey

+ * Copyright (c) 2007 Rob Sykes <robs@users.sourceforge.net>

+ * Copyright (c) 2013 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

+ *

+ */

+

+#include "libavutil/avstring.h"

+#include "libavutil/opt.h"

+#include "libavutil/samplefmt.h"

+#include "avfilter.h"

+#include "audio.h"

+#include "internal.h"

+

+typedef struct ChanParam {

+    double attack;

+    double decay;

+    double volume;

+} ChanParam;

+

+typedef struct CompandSegment {

+    double x, y;

+    double a, b;

+} CompandSegment;

+

+typedef struct CompandContext {

+    const AVClass *class;

+    char *attacks, *decays, *points;

+    CompandSegment *segments;

+    ChanParam *channels;

+    double in_min_lin;

+    double out_min_lin;

+    double curve_dB;

+    double gain_dB;

+    double initial_volume;

+    double delay;

+    uint8_t **delayptrs;

+    int delay_samples;

+    int delay_count;

+    int delay_index;

+    int64_t pts;

+

+    int (*compand)(AVFilterContext *ctx, AVFrame *frame);

+} CompandContext;

+

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

+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

+

+static const AVOption compand_options[] = {

+    { "attacks", "set time over which increase of volume is determined", OFFSET(attacks), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },

+    { "decays", "set time over which decrease of volume is determined", OFFSET(decays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },

+    { "points", "set points of transfer function", OFFSET(points), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },

+    { "soft-knee", "set soft-knee", OFFSET(curve_dB), AV_OPT_TYPE_DOUBLE, {.dbl=0.01}, 0.01, 900, A },

+    { "gain", "set output gain", OFFSET(gain_dB), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, A },

+    { "volume", "set initial volume", OFFSET(initial_volume), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 0, A },

+    { "delay", "set delay for samples before sending them to volume adjuster", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 20, A },

+    { NULL },

+};

+

+AVFILTER_DEFINE_CLASS(compand);

+

+static av_cold int init(AVFilterContext *ctx)

+{

+    CompandContext *s = ctx->priv;

+

+    if (!s->attacks || !s->decays || !s->points) {

+        av_log(ctx, AV_LOG_ERROR, "Missing attacks and/or decays and/or points.\n");

+        return AVERROR(EINVAL);

+    }

+

+    return 0;

+}

+

+static av_cold void uninit(AVFilterContext *ctx)

+{

+    CompandContext *s = ctx->priv;

+

+    av_freep(&s->channels);

+    av_freep(&s->segments);

+    if (s->delayptrs)

+        av_freep(&s->delayptrs[0]);

+    av_freep(&s->delayptrs);

+}

+

+static int query_formats(AVFilterContext *ctx)

+{

+    AVFilterChannelLayouts *layouts;

+    AVFilterFormats *formats;

+    static const enum AVSampleFormat sample_fmts[] = {

+        AV_SAMPLE_FMT_DBLP,

+        AV_SAMPLE_FMT_NONE

+    };

+

+    layouts = ff_all_channel_layouts();

+    if (!layouts)

+        return AVERROR(ENOMEM);

+    ff_set_common_channel_layouts(ctx, layouts);

+

+    formats = ff_make_format_list(sample_fmts);

+    if (!formats)

+        return AVERROR(ENOMEM);

+    ff_set_common_formats(ctx, formats);

+

+    formats = ff_all_samplerates();

+    if (!formats)

+        return AVERROR(ENOMEM);

+    ff_set_common_samplerates(ctx, formats);

+

+    return 0;

+}

+

+static void count_items(char *item_str, int *nb_items)

+{

+    char *p;

+

+    *nb_items = 1;

+    for (p = item_str; *p; p++) {

+        if (*p == ' ')

+            (*nb_items)++;

+    }

+

+}

+

+static void update_volume(ChanParam *cp, double in)

+{

+    double delta = in - cp->volume;

+

+    if (delta > 0.0)

+        cp->volume += delta * cp->attack;

+    else

+        cp->volume += delta * cp->decay;

+}

+

+static double get_volume(CompandContext *s, double in_lin)

+{

+    CompandSegment *cs;

+    double in_log, out_log;

+    int i;

+

+    if (in_lin < s->in_min_lin)

+        return s->out_min_lin;

+

+    in_log = log(in_lin);

+

+    for (i = 1;; i++)

+        if (in_log <= s->segments[i + 1].x)

+            break;

+

+    cs = &s->segments[i];

+    in_log -= cs->x;

+    out_log = cs->y + in_log * (cs->a * in_log + cs->b);

+

+    return exp(out_log);

+}

+

+static int compand_nodelay(AVFilterContext *ctx, AVFrame *frame)

+{

+    CompandContext *s = ctx->priv;

+    AVFilterLink *inlink = ctx->inputs[0];

+    const int channels = inlink->channels;

+    const int nb_samples = frame->nb_samples;

+    AVFrame *out_frame;

+    int chan, i;

+

+    if (av_frame_is_writable(frame)) {

+        out_frame = frame;

+    } else {

+        out_frame = ff_get_audio_buffer(inlink, nb_samples);

+        if (!out_frame)

+            return AVERROR(ENOMEM);

+        av_frame_copy_props(out_frame, frame);

+    }

+

+    for (chan = 0; chan < channels; chan++) {

+        const double *src = (double *)frame->data[chan];

+        double *dst = (double *)out_frame->data[chan];

+        ChanParam *cp = &s->channels[chan];

+

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

+            update_volume(cp, fabs(src[i]));

+

+            dst[i] = av_clipd(src[i] * get_volume(s, cp->volume), -1, 1);

+        }

+    }

+

+    if (frame != out_frame)

+        av_frame_free(&frame);

+

+    return ff_filter_frame(ctx->outputs[0], out_frame);

+}

+

+#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))

+

+static int compand_delay(AVFilterContext *ctx, AVFrame *frame)

+{

+    CompandContext *s = ctx->priv;

+    AVFilterLink *inlink = ctx->inputs[0];

+    const int channels = inlink->channels;

+    const int nb_samples = frame->nb_samples;

+    int chan, i, dindex, oindex, count;

+    AVFrame *out_frame = NULL;

+

+    for (chan = 0; chan < channels; chan++) {

+        const double *src = (double *)frame->data[chan];

+        double *dbuf = (double *)s->delayptrs[chan];

+        ChanParam *cp = &s->channels[chan];

+        double *dst;

+

+        count  = s->delay_count;

+        dindex = s->delay_index;

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

+            const double in = src[i];

+            update_volume(cp, fabs(in));

+

+            if (count >= s->delay_samples) {

+                if (!out_frame) {

+                    out_frame = ff_get_audio_buffer(inlink, nb_samples - i);

+                    if (!out_frame)

+                        return AVERROR(ENOMEM);

+                    av_frame_copy_props(out_frame, frame);

+                    out_frame->pts = s->pts;

+                    s->pts += av_rescale_q(nb_samples - i, (AVRational){1, inlink->sample_rate}, inlink->time_base);

+                }

+

+                dst = (double *)out_frame->data[chan];

+                dst[oindex++] = av_clipd(dbuf[dindex] * get_volume(s, cp->volume), -1, 1);

+            } else {

+                count++;

+            }

+

+            dbuf[dindex] = in;

+            dindex = MOD(dindex + 1, s->delay_samples);

+        }

+    }

+

+    s->delay_count = count;

+    s->delay_index = dindex;

+

+    av_frame_free(&frame);

+    return out_frame ? ff_filter_frame(ctx->outputs[0], out_frame) : 0;

+}

+

+static int compand_drain(AVFilterLink *outlink)

+{

+    AVFilterContext *ctx = outlink->src;

+    CompandContext *s = ctx->priv;

+    const int channels = outlink->channels;

+    int chan, i, dindex;

+    AVFrame *frame = NULL;

+

+    frame = ff_get_audio_buffer(outlink, FFMIN(2048, s->delay_count));

+    if (!frame)

+        return AVERROR(ENOMEM);

+    frame->pts = s->pts;

+    s->pts += av_rescale_q(frame->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);

+

+    for (chan = 0; chan < channels; chan++) {

+        double *dbuf = (double *)s->delayptrs[chan];

+        double *dst = (double *)frame->data[chan];

+        ChanParam *cp = &s->channels[chan];

+

+        dindex = s->delay_index;

+        for (i = 0; i < frame->nb_samples; i++) {

+            dst[i] = av_clipd(dbuf[dindex] * get_volume(s, cp->volume), -1, 1);

+            dindex = MOD(dindex + 1, s->delay_samples);

+        }

+    }

+    s->delay_count -= frame->nb_samples;

+    s->delay_index = dindex;

+

+    return ff_filter_frame(outlink, frame);

+}

+

+static int config_output(AVFilterLink *outlink)

+{

+    AVFilterContext *ctx = outlink->src;

+    CompandContext *s = ctx->priv;

+    const int sample_rate = outlink->sample_rate;

+    double radius = s->curve_dB * M_LN10 / 20;

+    int nb_attacks, nb_decays, nb_points;

+    char *p, *saveptr = NULL;

+    int new_nb_items, num;

+    int i;

+

+    count_items(s->attacks, &nb_attacks);

+    count_items(s->decays, &nb_decays);

+    count_items(s->points, &nb_points);

+

+    if ((nb_attacks > outlink->channels) || (nb_decays > outlink->channels)) {

+        av_log(ctx, AV_LOG_ERROR, "Number of attacks/decays bigger than number of channels.\n");

+        return AVERROR(EINVAL);

+    }

+

+    uninit(ctx);

+

+    s->channels = av_mallocz_array(outlink->channels, sizeof(*s->channels));

+    s->segments = av_mallocz_array((nb_points + 4) * 2, sizeof(*s->segments));

+

+    if (!s->channels || !s->segments)

+        return AVERROR(ENOMEM);

+

+    p = s->attacks;

+    for (i = 0, new_nb_items = 0; i < nb_attacks; i++) {

+        char *tstr = av_strtok(p, " ", &saveptr);

+        p = NULL;

+        new_nb_items += sscanf(tstr, "%lf", &s->channels[i].attack) == 1;

+        if (s->channels[i].attack < 0)

+            return AVERROR(EINVAL);

+    }

+    nb_attacks = new_nb_items;

+

+    p = s->decays;

+    for (i = 0, new_nb_items = 0; i < nb_decays; i++) {

+        char *tstr = av_strtok(p, " ", &saveptr);

+        p = NULL;

+        new_nb_items += sscanf(tstr, "%lf", &s->channels[i].decay) == 1;

+        if (s->channels[i].decay < 0)

+            return AVERROR(EINVAL);

+    }

+    nb_decays = new_nb_items;

+

+    if (nb_attacks != nb_decays) {

+        av_log(ctx, AV_LOG_ERROR, "Number of attacks %d differs from number of decays %d.\n", nb_attacks, nb_decays);

+        return AVERROR(EINVAL);

+    }

+

+#define S(x) s->segments[2 * ((x) + 1)]

+    p = s->points;

+    for (i = 0, new_nb_items = 0; i < nb_points; i++) {

+        char *tstr = av_strtok(p, " ", &saveptr);

+        p = NULL;

+        if (sscanf(tstr, "%lf/%lf", &S(i).x, &S(i).y) != 2) {

+            av_log(ctx, AV_LOG_ERROR, "Invalid and/or missing input/output value.\n");

+            return AVERROR(EINVAL);

+        }

+        if (i && S(i - 1).x > S(i).x) {

+            av_log(ctx, AV_LOG_ERROR, "Transfer function input values must be increasing.\n");

+            return AVERROR(EINVAL);

+        }

+        S(i).y -= S(i).x;

+        av_log(ctx, AV_LOG_DEBUG, "%d: x=%lf y=%lf\n", i, S(i).x, S(i).y);

+        new_nb_items++;

+    }

+    num = new_nb_items;

+

+    /* Add 0,0 if necessary */

+    if (num == 0 || S(num - 1).x)

+        num++;

+

+#undef S

+#define S(x) s->segments[2 * (x)]

+    /* Add a tail off segment at the start */

+    S(0).x = S(1).x - 2 * s->curve_dB;

+    S(0).y = S(1).y;

+    num++;

+

+    /* Join adjacent colinear segments */

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

+        double g1 = (S(i - 1).y - S(i - 2).y) * (S(i - 0).x - S(i - 1).x);

+        double g2 = (S(i - 0).y - S(i - 1).y) * (S(i - 1).x - S(i - 2).x);

+        int j;

+

+        if (fabs(g1 - g2))

+            continue;

+        num--;

+        for (j = --i; j < num; j++)

+            S(j) = S(j + 1);

+    }

+

+    for (i = 0; !i || s->segments[i - 2].x; i += 2) {

+        s->segments[i].y += s->gain_dB;

+        s->segments[i].x *= M_LN10 / 20;

+        s->segments[i].y *= M_LN10 / 20;

+    }

+

+#define L(x) s->segments[i - (x)]

+    for (i = 4; s->segments[i - 2].x; i += 2) {

+        double x, y, cx, cy, in1, in2, out1, out2, theta, len, r;

+

+        L(4).a = 0;

+        L(4).b = (L(2).y - L(4).y) / (L(2).x - L(4).x);

+

+        L(2).a = 0;

+        L(2).b = (L(0).y - L(2).y) / (L(0).x - L(2).x);

+

+        theta = atan2(L(2).y - L(4).y, L(2).x - L(4).x);

+        len = sqrt(pow(L(2).x - L(4).x, 2.) + pow(L(2).y - L(4).y, 2.));

+        r = FFMIN(radius, len);

+        L(3).x = L(2).x - r * cos(theta);

+        L(3).y = L(2).y - r * sin(theta);

+

+        theta = atan2(L(0).y - L(2).y, L(0).x - L(2).x);

+        len = sqrt(pow(L(0).x - L(2).x, 2.) + pow(L(0).y - L(2).y, 2.));

+        r = FFMIN(radius, len / 2);

+        x = L(2).x + r * cos(theta);

+        y = L(2).y + r * sin(theta);

+

+        cx = (L(3).x + L(2).x + x) / 3;

+        cy = (L(3).y + L(2).y + y) / 3;

+

+        L(2).x = x;

+        L(2).y = y;

+

+        in1 = cx - L(3).x;

+        out1 = cy - L(3).y;

+        in2 = L(2).x - L(3).x;

+        out2 = L(2).y - L(3).y;

+        L(3).a = (out2 / in2 - out1 / in1) / (in2-in1);

+        L(3).b = out1 / in1 - L(3).a * in1;

+    }

+    L(3).x = 0;

+    L(3).y = L(2).y;

+

+    s->in_min_lin  = exp(s->segments[1].x);

+    s->out_min_lin = exp(s->segments[1].y);

+

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

+        ChanParam *cp = &s->channels[i];

+

+        if (cp->attack > 1.0 / sample_rate)

+            cp->attack = 1.0 - exp(-1.0 / (sample_rate * cp->attack));

+        else

+            cp->attack = 1.0;

+        if (cp->decay > 1.0 / sample_rate)

+            cp->decay = 1.0 - exp(-1.0 / (sample_rate * cp->decay));

+        else

+            cp->decay = 1.0;

+        cp->volume = pow(10.0, s->initial_volume / 20);

+    }

+

+    s->delay_samples = s->delay * sample_rate;

+    if (s->delay_samples > 0) {

+        int ret;

+        if ((ret = av_samples_alloc_array_and_samples(&s->delayptrs, NULL,

+                                                      outlink->channels,

+                                                      s->delay_samples,

+                                                      outlink->format, 0)) < 0)

+            return ret;

+        s->compand = compand_delay;

+        outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP;

+    } else {

+        s->compand = compand_nodelay;

+    }

+    return 0;

+}

+

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

+{

+    AVFilterContext *ctx = inlink->dst;

+    CompandContext *s = ctx->priv;

+

+    return s->compand(ctx, frame);

+}

+

+static int request_frame(AVFilterLink *outlink)

+{

+    AVFilterContext *ctx = outlink->src;

+    CompandContext *s = ctx->priv;

+    int ret;

+

+    ret = ff_request_frame(ctx->inputs[0]);

+

+    if (ret == AVERROR_EOF && !ctx->is_disabled && s->delay_count)

+        ret = compand_drain(outlink);

+

+    return ret;

+}

+

+static const AVFilterPad compand_inputs[] = {

+    {

+        .name         = "default",

+        .type         = AVMEDIA_TYPE_AUDIO,

+        .filter_frame = filter_frame,

+    },

+    { NULL },

+};

+

+static const AVFilterPad compand_outputs[] = {

+    {

+        .name          = "default",

+        .request_frame = request_frame,

+        .config_props  = config_output,

+        .type          = AVMEDIA_TYPE_AUDIO,

+    },

+    { NULL },

+};

+

+AVFilter avfilter_af_compand = {

+    .name          = "compand",

+    .description   = NULL_IF_CONFIG_SMALL("Compress or expand audio dynamic range."),

+    .query_formats = query_formats,

+    .priv_size     = sizeof(CompandContext),

+    .priv_class    = &compand_class,

+    .init          = init,

+    .uninit        = uninit,

+    .inputs        = compand_inputs,

+    .outputs       = compand_outputs,

+};

@@ -80,6 +80,7 @@ void avfilter_register_all(void)

     REGISTER_FILTER(BIQUAD,         biquad,         af);

     REGISTER_FILTER(CHANNELMAP,     channelmap,     af);

     REGISTER_FILTER(CHANNELSPLIT,   channelsplit,   af);

+    REGISTER_FILTER(COMPAND,        compand,        af);

     REGISTER_FILTER(EARWAX,         earwax,         af);

     REGISTER_FILTER(EBUR128,        ebur128,        af);

     REGISTER_FILTER(EQUALIZER,      equalizer,      af);

@@ -30,8 +30,8 @@

 #include "libavutil/avutil.h"

 #define LIBAVFILTER_VERSION_MAJOR  3

-#define LIBAVFILTER_VERSION_MINOR  81

-#define LIBAVFILTER_VERSION_MICRO 103

+#define LIBAVFILTER_VERSION_MINOR  82

+#define LIBAVFILTER_VERSION_MICRO 100

 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \

                                                LIBAVFILTER_VERSION_MINOR, \