avfilter: add anlmdn audio filter

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

1 files changed, 271 insertions, 0 deletions

diff --git a/libavfilter/af_anlmdn.c b/libavfilter/af_anlmdn.c
new file mode 100644
index 0000000000..62931e37cc
--- /dev/null
+++ b/libavfilter/af_anlmdn.c
@@ -0,0 +1,271 @@
+/*
+ * 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 K;
+    int S;
+    int N;
+    int H;
+
+    int offset;
+    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, 100000, AF },
+    { "r", "set research duration",  OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, 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->H = s->K * 2 + 1;
+    s->N = s->H + (s->K + s->S) * 2;
+
+    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);
+    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] + s->offset;
+
+    for (int i = S; i < s->H + S; i++) {
+        float P = 0.f, Q = 0.f;
+        int v = 0;
+
+        if (i == S) {
+            for (int j = i - S; j <= i + S; j++) {
+                if (i == j)
+                    continue;
+                cache[v++] = s->compute_distance(f + i, f + j, K);
+            }
+        } else {
+            for (int j = i - S; j < i; j++, v++)
+                cache[v] = cache[v] - SQR(f[i - K - 1] - f[j - K - 1]) + SQR(f[i + K] - f[j + K]);
+
+            for (int j = i + 1; j <= i + S; j++, v++)
+                cache[v] = cache[v] - SQR(f[i - K - 1] - f[j - K - 1]) + SQR(f[i + K] - f[j + K]);
+        }
+
+        for (int j = 0; j < v; j++) {
+            const float distance = cache[j];
+            float w;
+
+            av_assert0(distance >= 0.f);
+            w = expf(-distance * sw);
+            P += w * f[i - S + j + (j >= S)];
+            Q += w;
+        }
+
+        P += f[i];
+        Q += 1;
+
+        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 available, wanted, ret;
+
+    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);
+
+    s->offset = 0;
+    available = av_audio_fifo_size(s->fifo);
+    wanted = (available / s->H) * s->H;
+
+    if (wanted >= s->H && available >= s->N) {
+        out = ff_get_audio_buffer(outlink, wanted);
+        if (!out)
+            return AVERROR(ENOMEM);
+    }
+
+    while (available >= s->N) {
+        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->H);
+
+        s->offset += s->H;
+        available -= s->H;
+    }
+
+    if (out) {
+        out->pts = s->pts;
+        out->nb_samples = s->offset;
+        s->pts += s->offset;
+
+        return ff_filter_frame(outlink, 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,
+};