avfilter: add anlmdn audio filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2018-05-19 22:06:27 +02:00
parent 10931a0661
commit 8a1fc95840
6 changed files with 299 additions and 2 deletions

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@ -13,6 +13,7 @@ version <next>:
- GIF parser
- vividas demuxer
- hymt decoder
- anlmdn filter
version 4.1:

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@ -1750,6 +1750,29 @@ 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 100 milliseconds.
Default value is 2 milliseconds.
@item r
Set research radius duration. Allowed range is from 2 to 300 milliseconds.
Default value is 6 milliseconds.
@end table
@section anull
Pass the audio source unchanged to the output.

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@ -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

271
libavfilter/af_anlmdn.c Normal file
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@ -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,
};

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@ -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;

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@ -30,8 +30,8 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7
#define LIBAVFILTER_VERSION_MINOR 46
#define LIBAVFILTER_VERSION_MICRO 101
#define LIBAVFILTER_VERSION_MINOR 47
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \