[FFmpeg-devel] avfilter: add anlmdn audio filter

Submitted by Paul B Mahol on Jan. 7, 2019, 5:11 p.m.

Details

Message ID 20190107171106.15580-1-onemda@gmail.com
State New
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Commit Message

Paul B Mahol Jan. 7, 2019, 5:11 p.m.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 doc/filters.texi         |  27 ++++
 libavfilter/Makefile     |   1 +
 libavfilter/af_anlmdn.c  | 257 +++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c |   1 +
 4 files changed, 286 insertions(+)
 create mode 100644 libavfilter/af_anlmdn.c

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diff --git a/doc/filters.texi b/doc/filters.texi
index 98858c5356..35d68cf85a 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1750,6 +1750,33 @@  Full filter invocation with asendcmd may look like this:
 asendcmd=c='4.0 anequalizer change 0|f=200|w=50|g=1',anequalizer=...
 @end table
 
+@section anlmdn
+
+Reduce broadband noise in audio samples using Non-Local Means algorithm.
+
+Each sample is adjusted by looking for other samples with similar contexts. This
+context similarity is defined by comparing their surrounding patches of size
+@option{p}. Patches are searched in an area of @option{r} around the sample.
+
+The filter accepts the following options.
+
+@table @option
+@item s
+Set denoising strength. Allowed range is from 1 to 9999. Default value is 1.
+
+@item p
+Set patch radius duration. Allowed range is from 1 to 24 milliseconds.
+Default value is 2 milliseconds.
+
+@item r
+Set research radius duration. Allowed range is from 2 to 48 milliseconds.
+Default value is 12 milliseconds.
+
+@item n
+Set number of patches to filter at once. Allowed range is from 1 to 100.
+Default value is 1.
+@end table
+
 @section anull
 
 Pass the audio source unchanged to the output.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 6e2658186d..d31fa59e0d 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -63,6 +63,7 @@  OBJS-$(CONFIG_AMETADATA_FILTER)              += f_metadata.o
 OBJS-$(CONFIG_AMIX_FILTER)                   += af_amix.o
 OBJS-$(CONFIG_AMULTIPLY_FILTER)              += af_amultiply.o
 OBJS-$(CONFIG_ANEQUALIZER_FILTER)            += af_anequalizer.o
+OBJS-$(CONFIG_ANLMDN_FILTER)                 += af_anlmdn.o
 OBJS-$(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_anlmdn.c b/libavfilter/af_anlmdn.c
new file mode 100644
index 0000000000..b7efd8387b
--- /dev/null
+++ b/libavfilter/af_anlmdn.c
@@ -0,0 +1,257 @@ 
+/*
+ * Copyright (c) 2019 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;
+
+    float a;
+    int64_t pd;
+    int64_t rd;
+    int n;
+
+    int K;
+    int S;
+
+    int N;
+    int hop_size;
+
+    AVFrame *in;
+    AVFrame *cache;
+
+    int64_t pts;
+
+    AVAudioFifo *fifo;
+
+    float (*compute_distance)(const float *f1, const float *f2, int K);
+} AudioNLMeansContext;
+
+#define OFFSET(x) offsetof(AudioNLMeansContext, x)
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption anlmdn_options[] = {
+    { "s", "set denoising strength", OFFSET(a),  AV_OPT_TYPE_FLOAT,    {.dbl=1},        1,  9999, AF },
+    { "p", "set patch duration",     OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000},  1000, 24000, AF },
+    { "r", "set research duration",  OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=12000}, 2000, 48000, AF },
+    { "n", "set number of patches to filter at once", OFFSET(n), AV_OPT_TYPE_INT, {.i64=1}, 1, 100, AF },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(anlmdn);
+
+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 float compute_distance_ssd(const float *f1, const float *f2, int K)
+{
+    float distance = 0.;
+
+    for (int k = -K; k <= K; k++) {
+        distance += SQR(f1[k] - f2[k]);
+    }
+
+    return distance;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AudioNLMeansContext *s = ctx->priv;
+
+    s->K = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
+    s->S = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
+
+    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);
+    av_frame_free(&s->cache);
+    s->in = ff_get_audio_buffer(outlink, s->N);
+    if (!s->in)
+        return AVERROR(ENOMEM);
+
+    s->cache = ff_get_audio_buffer(outlink, s->S * 2 + 1);
+    if (!s->cache)
+        return AVERROR(ENOMEM);
+
+    s->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, s->N);
+    if (!s->fifo)
+        return AVERROR(ENOMEM);
+
+    s->compute_distance = compute_distance_ssd;
+
+    return 0;
+}
+
+static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
+{
+    AudioNLMeansContext *s = ctx->priv;
+    AVFrame *out = arg;
+    const int S = s->S;
+    const int K = s->K;
+    const float *f = (const float *)(s->in->extended_data[ch]) + K;
+    float *cache = (float *)s->cache->extended_data[ch];
+    const float sw = 32768.f / s->a;
+    float *dst = (float *)out->extended_data[ch];
+
+    for (int i = S; i < s->hop_size + S; i++) {
+        float P = 0.f, Q = 0.f;
+        int v = 0;
+
+        for (int j = i - S; j <= i + S; j++, v++) {
+            float w, distance;
+
+            if (i == j) {
+                distance = 0.f;
+            } else if (i == S) {
+                cache[v] = distance = s->compute_distance(f + i, f + j, K);
+            } else {
+                distance = cache[v] = cache[v] - SQR(f[i - K - 1] - f[j - K - 1]) + SQR(f[i + K] - f[j + K]);
+            }
+
+            av_assert0(distance >= 0.f);
+            w = expf(-distance * sw);
+            P += w * f[j];
+            Q += w;
+        }
+
+        dst[i - S] = 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) {
+        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;
+
+        ctx->internal->execute(ctx, filter_channel, out, NULL, inlink->channels);
+
+        av_audio_fifo_drain(s->fifo, s->hop_size);
+
+        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_frame_free(&s->cache);
+}
+
+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_anlmdn = {
+    .name          = "anlmdn",
+    .description   = NULL_IF_CONFIG_SMALL("Reduce broadband noise from stream using Non-Local Means."),
+    .query_formats = query_formats,
+    .priv_size     = sizeof(AudioNLMeansContext),
+    .priv_class    = &anlmdn_class,
+    .uninit        = uninit,
+    .inputs        = inputs,
+    .outputs       = outputs,
+    .flags         = AVFILTER_FLAG_SLICE_THREADS,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index a600069500..c8cff6e435 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -55,6 +55,7 @@  extern AVFilter ff_af_ametadata;
 extern AVFilter ff_af_amix;
 extern AVFilter ff_af_amultiply;
 extern AVFilter ff_af_anequalizer;
+extern AVFilter ff_af_anlmdn;
 extern AVFilter ff_af_anull;
 extern AVFilter ff_af_apad;
 extern AVFilter ff_af_aperms;