avfilter: add audio limiter filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>

1 files changed, 361 insertions, 0 deletions

diff --git a/libavfilter/af_alimiter.c b/libavfilter/af_alimiter.c
new file mode 100644
index 0000000000..37839de7bf
--- /dev/null
+++ b/libavfilter/af_alimiter.c
@@ -0,0 +1,361 @@
+/*
+ * 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
+ * Lookahead limiter 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 AudioLimiterContext {
+    const AVClass *class;
+
+    double limit;
+    double attack;
+    double release;
+    double att;
+    int auto_release;
+    double asc;
+    int asc_c;
+    int asc_pos;
+    double asc_coeff;
+
+    double *buffer;
+    int buffer_size;
+    int pos;
+    int *nextpos;
+    double *nextdelta;
+
+    double delta;
+    int nextiter;
+    int nextlen;
+    int asc_changed;
+} AudioLimiterContext;
+
+#define OFFSET(x) offsetof(AudioLimiterContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM
+#define F AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption alimiter_options[] = {
+    { "limit",     "set limit",     OFFSET(limit),        AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625,    1, A|F },
+    { "attack",    "set attack",    OFFSET(attack),       AV_OPT_TYPE_DOUBLE, {.dbl=5},    0.1,   80, A|F },
+    { "release",   "set release",   OFFSET(release),      AV_OPT_TYPE_DOUBLE, {.dbl=50},     1, 8000, A|F },
+    { "asc",       "enable asc",    OFFSET(auto_release), AV_OPT_TYPE_BOOL,   {.i64=0},      0,    1, A|F },
+    { "asc_level", "set asc level", OFFSET(asc_coeff),    AV_OPT_TYPE_DOUBLE, {.dbl=0.5},    0,    1, A|F },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(alimiter);
+
+static av_cold int init(AVFilterContext *ctx)
+{
+    AudioLimiterContext *s = ctx->priv;
+
+    s->attack   /= 1000.;
+    s->release  /= 1000.;
+    s->att       = 1.;
+    s->asc_pos   = -1;
+    s->asc_coeff = pow(0.5, s->asc_coeff - 0.5) * 2 * -1;
+
+    return 0;
+}
+
+static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate,
+                         double peak, double limit, double patt, int asc)
+{
+    double rdelta = (1.0 - patt) / (sample_rate * release);
+
+    if (asc && s->auto_release && s->asc_c > 0) {
+        double a_att = limit / (s->asc_coeff * s->asc) * (double)s->asc_c;
+
+        if (a_att > patt) {
+            double delta = FFMAX((a_att - patt) / (sample_rate * release), rdelta / 10);
+
+            if (delta < rdelta)
+                rdelta = delta;
+        }
+    }
+
+    return rdelta;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AudioLimiterContext *s = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    const double *src = (const double *)in->data[0];
+    const int channels = inlink->channels;
+    const int buffer_size = s->buffer_size;
+    double *dst, *buffer = s->buffer;
+    const double release = s->release;
+    const double limit = s->limit;
+    double *nextdelta = s->nextdelta;
+    int *nextpos = s->nextpos;
+    AVFrame *out;
+    double *buf;
+    int n, c, i;
+
+    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++) {
+        double peak = 0;
+
+        for (c = 0; c < channels; c++) {
+            double sample = src[c];
+
+            buffer[s->pos + c] = sample;
+            peak = FFMAX(peak, fabs(sample));
+        }
+
+        if (s->auto_release && peak > limit) {
+            s->asc += peak;
+            s->asc_c++;
+        }
+
+        if (peak > limit) {
+            double patt = FFMIN(limit / peak, 1.);
+            double rdelta = get_rdelta(s, release, inlink->sample_rate,
+                                       peak, limit, patt, 0);
+            double delta = (limit / peak - s->att) / buffer_size * channels;
+            int found = 0;
+
+            if (delta < s->delta) {
+                s->delta = delta;
+                nextpos[0] = s->pos;
+                nextpos[1] = -1;
+                nextdelta[0] = rdelta;
+                s->nextlen = 1;
+                s->nextiter= 0;
+            } else {
+                for (i = s->nextiter; i < s->nextiter + s->nextlen; i++) {
+                    int j = i % buffer_size;
+                    double ppeak, pdelta;
+
+                    ppeak = fabs(buffer[nextpos[j]]) > fabs(buffer[nextpos[j] + 1]) ?
+                            fabs(buffer[nextpos[j]]) : fabs(buffer[nextpos[j] + 1]);
+                    pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] + s->pos) % buffer_size) / channels);
+                    if (pdelta < nextdelta[j]) {
+                        nextdelta[j] = pdelta;
+                        found = 1;
+                        break;
+                    }
+                }
+                if (found) {
+                    s->nextlen = i - s->nextiter + 1;
+                    nextpos[(s->nextiter + s->nextlen) % buffer_size] = s->pos;
+                    nextdelta[(s->nextiter + s->nextlen) % buffer_size] = rdelta;
+                    nextpos[(s->nextiter + s->nextlen + 1) % buffer_size] = -1;
+                    s->nextlen++;
+                }
+            }
+        }
+
+        buf = &s->buffer[(s->pos + channels) % buffer_size];
+        peak = 0;
+        for (c = 0; c < channels; c++) {
+            double sample = buf[c];
+
+            peak = FFMAX(peak, fabs(sample));
+        }
+
+        if (s->pos == s->asc_pos && !s->asc_changed)
+            s->asc_pos = -1;
+
+        if (s->auto_release && s->asc_pos == -1 && peak > limit) {
+            s->asc -= peak;
+            s->asc_c--;
+        }
+
+        s->att += s->delta;
+
+        for (c = 0; c < channels; c++)
+            dst[c] = buf[c] * s->att;
+
+        if ((s->pos + channels) % buffer_size == nextpos[s->nextiter]) {
+            if (s->auto_release) {
+                s->delta = get_rdelta(s, release, inlink->sample_rate,
+                                      peak, limit, s->att, 1);
+                if (s->nextlen > 1) {
+                    int pnextpos = nextpos[(s->nextiter + 1) % buffer_size];
+                    double ppeak = fabs(buffer[pnextpos]) > fabs(buffer[pnextpos + 1]) ?
+                                                            fabs(buffer[pnextpos]) :
+                                                            fabs(buffer[pnextpos + 1]);
+                    double pdelta = (limit / ppeak - s->att) /
+                                    (((buffer_size + pnextpos -
+                                    ((s->pos + channels) % buffer_size)) %
+                                    buffer_size) / channels);
+                    if (pdelta < s->delta)
+                        s->delta = pdelta;
+                }
+            } else {
+                s->delta = nextdelta[s->nextiter];
+                s->att = limit / peak;
+            }
+
+            s->nextlen -= 1;
+            nextpos[s->nextiter] = -1;
+            s->nextiter = (s->nextiter + 1) % buffer_size;
+        }
+
+        if (s->att > 1.) {
+            s->att = 1.;
+            s->delta = 0.;
+            s->nextiter = 0;
+            s->nextlen = 0;
+            nextpos[0] = -1;
+        }
+
+        if (s->att <= 0.) {
+            s->att = 0.0000000000001;
+            s->delta = (1.0 - s->att) / (inlink->sample_rate * release);
+        }
+
+        if (s->att != 1. && (1. - s->att) < 0.0000000000001)
+            s->att = 1.;
+
+        if (s->delta != 0. && fabs(s->delta) < 0.00000000000001)
+            s->delta = 0.;
+
+        for (c = 0; c < channels; c++)
+            dst[c] = av_clipd(dst[c], -limit, limit);
+
+        s->pos = (s->pos + channels) % buffer_size;
+        src += channels;
+        dst += channels;
+    }
+
+    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 int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AudioLimiterContext *s = ctx->priv;
+    int obuffer_size;
+
+    obuffer_size = inlink->sample_rate * inlink->channels * 100 / 1000. + inlink->channels;
+    if (obuffer_size < inlink->channels)
+        return AVERROR(EINVAL);
+
+    s->buffer = av_calloc(obuffer_size, sizeof(*s->buffer));
+    s->nextdelta = av_calloc(obuffer_size, sizeof(*s->nextdelta));
+    s->nextpos = av_malloc_array(obuffer_size, sizeof(*s->nextpos));
+    if (!s->buffer || !s->nextdelta || !s->nextpos)
+        return AVERROR(ENOMEM);
+
+    memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
+    s->buffer_size = inlink->sample_rate * s->attack * inlink->channels;
+    s->buffer_size -= s->buffer_size % inlink->channels;
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AudioLimiterContext *s = ctx->priv;
+
+    av_freep(&s->buffer);
+    av_freep(&s->nextdelta);
+    av_freep(&s->nextpos);
+}
+
+static const AVFilterPad alimiter_inputs[] = {
+    {
+        .name         = "main",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+        .config_props = config_input,
+    },
+    { NULL }
+};
+
+static const AVFilterPad alimiter_outputs[] = {
+    {
+        .name = "default",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_alimiter = {
+    .name           = "alimiter",
+    .description    = NULL_IF_CONFIG_SMALL("Lookahead limiter."),
+    .priv_size      = sizeof(AudioLimiterContext),
+    .priv_class     = &alimiter_class,
+    .init           = init,
+    .uninit         = uninit,
+    .query_formats  = query_formats,
+    .inputs         = alimiter_inputs,
+    .outputs        = alimiter_outputs,
+};