[FFmpeg-devel] avfilter: add anlmeans audio filter

Submitted by Paul B Mahol on May 26, 2018, 7:20 p.m.

Details

Message ID 20180526192014.23226-1-onemda@gmail.com
State New
Headers show

Commit Message

Paul B Mahol May 26, 2018, 7:20 p.m.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 libavfilter/Makefile      |   1 +
 libavfilter/af_anlmeans.c | 282 ++++++++++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c  |   1 +
 3 files changed, 284 insertions(+)
 create mode 100644 libavfilter/af_anlmeans.c

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diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index c68ef05fdc..ac7e205b75 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -54,6 +54,7 @@  OBJS-$(CONFIG_AMERGE_FILTER)                 += af_amerge.o
 OBJS-$(CONFIG_AMETADATA_FILTER)              += f_metadata.o
 OBJS-$(CONFIG_AMIX_FILTER)                   += af_amix.o
 OBJS-$(CONFIG_ANEQUALIZER_FILTER)            += af_anequalizer.o
+OBJS-$(CONFIG_ANLMEANS_FILTER)               += af_anlmeans.o
 OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
 OBJS-$(CONFIG_APAD_FILTER)                   += af_apad.o
 OBJS-$(CONFIG_APERMS_FILTER)                 += f_perms.o
diff --git a/libavfilter/af_anlmeans.c b/libavfilter/af_anlmeans.c
new file mode 100644
index 0000000000..01c4ca3239
--- /dev/null
+++ b/libavfilter/af_anlmeans.c
@@ -0,0 +1,282 @@ 
+/*
+ * Copyright (c) 2018 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 <float.h>
+
+#include "libavutil/avassert.h"
+#include "libavutil/audio_fifo.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+#define SQR(x) ((x) * (x))
+
+typedef struct AudioNLMeansContext {
+    const AVClass *class;
+
+    double g;
+    double B;
+    double h;
+    double a;
+    double m;
+    int n;
+    int K;
+    int S;
+
+    int N;
+    int hop_size;
+
+    AVFrame *in;
+
+    double *kernel;
+
+    int64_t pts;
+    uint64_t nb_samples;
+    int samples_left;
+
+    AVAudioFifo *fifo;
+} AudioNLMeansContext;
+
+#define OFFSET(x) offsetof(AudioNLMeansContext, x)
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption anlmeans_options[] = {
+    { "n", "set number of patches", OFFSET(n), AV_OPT_TYPE_INT,    {.i64=1},        1,         4, AF },
+    { "K", "set patch radius",      OFFSET(K), AV_OPT_TYPE_INT,    {.i64=32},       0,      4096, AF },
+    { "S", "set research radius",   OFFSET(S), AV_OPT_TYPE_INT,    {.i64=64},       1,      4096, AF },
+    { "h", "set strength",          OFFSET(h), AV_OPT_TYPE_DOUBLE, {.dbl=1},        1,   FLT_MAX, AF },
+    { "m", "set max allowed diff",  OFFSET(m), AV_OPT_TYPE_DOUBLE, {.dbl=0.1},      0,         2, AF },
+    { "a", "set denoising amount",  OFFSET(a), AV_OPT_TYPE_DOUBLE, {.dbl=1},        1,   FLT_MAX, AF },
+    { "B", "set smooth factor",     OFFSET(B), AV_OPT_TYPE_DOUBLE, {.dbl=1},        1,   FLT_MAX, AF },
+    { "g", "set output gain",       OFFSET(g), AV_OPT_TYPE_DOUBLE, {.dbl=1},        0,        10, AF },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(anlmeans);
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats = NULL;
+    AVFilterChannelLayouts *layouts = NULL;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_FLTP,
+        AV_SAMPLE_FMT_NONE
+    };
+    int 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;
+
+    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_all_samplerates();
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static void compute_kernel(double *kernel, int K, double B2)
+{
+    int k;
+
+    for (k = 0; k <= 2 * K; k++) {
+        double kk = k - K;
+
+        kernel[k] = exp(-(kk * kk) / B2);
+    }
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AudioNLMeansContext *s = ctx->priv;
+
+    s->m = SQR(s->m * 65536.);
+    s->pts = AV_NOPTS_VALUE;
+    s->N = s->n * s->K * 2 + 1 + (s->K + s->S) * 2;
+    s->hop_size = s->n * s->K * 2 + 1;
+
+    av_frame_free(&s->in);
+    s->in = ff_get_audio_buffer(outlink, s->N);
+    if (!s->in)
+        return AVERROR(ENOMEM);
+
+    s->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, s->N);
+    if (!s->fifo)
+        return AVERROR(ENOMEM);
+
+    s->kernel = av_calloc(s->K * 2 + 1, sizeof(*s->kernel));
+    if (!s->kernel)
+        return AVERROR(ENOMEM);
+
+    compute_kernel(s->kernel, s->K, 1. / (s->B * s->B));
+
+    return 0;
+}
+
+typedef struct ThreadData {
+    AVFrame *out;
+} ThreadData;
+
+static double compute_distance(const double *kernel,
+                               const float *f1, const float *f2, int K)
+{
+    double sum = 0.;
+    int k;
+
+    for (k = -K; k <= K; k++) {
+        sum += kernel[k] * SQR(65536. * f1[k] - 65536. * f2[k]);
+    }
+
+    return sum;
+}
+
+static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
+{
+    AudioNLMeansContext *s = ctx->priv;
+    ThreadData *td = arg;
+    AVFrame *out = td->out;
+    const int S = s->S;
+    const int K = s->K;
+    const double m = s->m;
+    const double g = s->g;
+    const double *kernel = s->kernel + K;
+    const float *f = (const float *)(s->in->extended_data[ch]) + K;
+    const double sw = s->a / (s->h * s->h);
+    float *dst = (float *)out->extended_data[ch];
+    int i, j;
+
+    for (i = S; i < s->hop_size + S; i++) {
+        double P = 0., Q = 0.;
+
+        for (j = i - S; j <= i + S; j++) {
+            double w, d2;
+
+            d2 = compute_distance(kernel, f + i, f + j, K);
+
+            if (d2 > m)
+                continue;
+
+            w = exp(-d2 * sw);
+            P += w * f[j];
+            Q += w;
+        }
+
+        dst[i - S] = g * P / Q;
+    }
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AVFilterLink *outlink = ctx->outputs[0];
+    AudioNLMeansContext *s = ctx->priv;
+    AVFrame *out = NULL;
+    int ret = 0;
+
+    if (s->pts == AV_NOPTS_VALUE)
+        s->pts = in->pts;
+
+    ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
+                              in->nb_samples);
+    av_frame_free(&in);
+
+    while (av_audio_fifo_size(s->fifo) >= s->N) {
+        ThreadData td;
+
+        out = ff_get_audio_buffer(outlink, s->hop_size);
+        if (!out)
+            return AVERROR(ENOMEM);
+
+        ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data,
+                                 s->N);
+        if (ret < 0)
+            break;
+
+        td.out = out;
+        ctx->internal->execute(ctx, filter_channel, &td, NULL, inlink->channels);
+
+        av_audio_fifo_drain(s->fifo, s->hop_size);
+
+        if (s->samples_left > 0)
+            out->nb_samples = FFMIN(s->hop_size, s->samples_left);
+
+        out->pts = s->pts;
+        s->pts += s->hop_size;
+
+        ret = ff_filter_frame(outlink, out);
+        if (ret < 0)
+            break;
+    }
+
+    if (ret < 0)
+        av_frame_free(&out);
+    return ret;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AudioNLMeansContext *s = ctx->priv;
+
+    av_audio_fifo_free(s->fifo);
+    av_frame_free(&s->in);
+    av_freep(&s->kernel);
+}
+
+static const AVFilterPad inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+    },
+    { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+    {
+        .name          = "default",
+        .type          = AVMEDIA_TYPE_AUDIO,
+        .config_props  = config_output,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_anlmeans = {
+    .name          = "anlmeans",
+    .description   = NULL_IF_CONFIG_SMALL("Reduce broadband noise from input audio using Non-Local Means."),
+    .query_formats = query_formats,
+    .priv_size     = sizeof(AudioNLMeansContext),
+    .priv_class    = &anlmeans_class,
+    .uninit        = uninit,
+    .inputs        = inputs,
+    .outputs       = outputs,
+    .flags         = AVFILTER_FLAG_SLICE_THREADS,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index b44093d21b..90a033a28f 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -49,6 +49,7 @@  extern AVFilter ff_af_amerge;
 extern AVFilter ff_af_ametadata;
 extern AVFilter ff_af_amix;
 extern AVFilter ff_af_anequalizer;
+extern AVFilter ff_af_anlmeans;
 extern AVFilter ff_af_anull;
 extern AVFilter ff_af_apad;
 extern AVFilter ff_af_aperms;