mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
avfilter: add anlms filter
This commit is contained in:
Paul B Mahol
parent
a27c0781dd
commit
e37edc70bd
@ -14,6 +14,7 @@ version <next>:
|
||||
- sierpinski video source
|
||||
- scroll video filter
|
||||
- photosensitivity filter
|
||||
- anlms filter
|
||||
|
||||
|
||||
version 4.2:
|
||||
|
||||
@ -1814,6 +1814,58 @@ Change output mode.
|
||||
Syntax for the command is : "i", "o" or "n" string.
|
||||
@end table
|
||||
|
||||
@section anlms
|
||||
Apply Normalized Least-Mean-Squares algorithm to the first audio stream using the second audio stream.
|
||||
|
||||
This adaptive filter is used to mimic a desired filter by finding the filter coefficients that
|
||||
relate to producing the least mean square of the error signal (difference between the desired,
|
||||
2nd input audio stream and the actual signal, the 1st input audio stream).
|
||||
|
||||
A description of the accepted options follows.
|
||||
|
||||
@table @option
|
||||
@item order
|
||||
Set filter order.
|
||||
|
||||
@item mu
|
||||
Set filter mu.
|
||||
|
||||
@item eps
|
||||
Set the filter eps.
|
||||
|
||||
@item leakage
|
||||
Set the filter leakage.
|
||||
|
||||
@item out_mode
|
||||
It accepts the following values:
|
||||
@table @option
|
||||
@item i
|
||||
Pass the 1st input.
|
||||
|
||||
@item d
|
||||
Pass the 2nd input.
|
||||
|
||||
@item o
|
||||
Pass filtered samples.
|
||||
|
||||
@item n
|
||||
Pass difference between desired and filtered samples.
|
||||
|
||||
Default value is @var{o}.
|
||||
@end table
|
||||
@end table
|
||||
|
||||
@subsection Examples
|
||||
|
||||
@itemize
|
||||
@item
|
||||
One of many usages of this filter is noise reduction, input audio is filtered
|
||||
with same samples that are delayed by fixed ammount, one such example for stereo audio is:
|
||||
@example
|
||||
asplit[a][b],[a]adelay=32S|32S[a],[b][a]anlms=order=128:leakage=0.0005:mu=.5:out_mode=o
|
||||
@end example
|
||||
@end itemize
|
||||
|
||||
@section anull
|
||||
|
||||
Pass the audio source unchanged to the output.
|
||||
|
||||
@ -63,6 +63,7 @@ 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-$(CONFIG_ANLMS_FILTER) += af_anlms.o
|
||||
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
|
||||
OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
|
||||
OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
|
||||
|
||||
328
libavfilter/af_anlms.c
Normal file
328
libavfilter/af_anlms.c
Normal file
@ -0,0 +1,328 @@
|
||||
/*
|
||||
* 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 "libavutil/avassert.h"
|
||||
#include "libavutil/channel_layout.h"
|
||||
#include "libavutil/common.h"
|
||||
#include "libavutil/float_dsp.h"
|
||||
#include "libavutil/opt.h"
|
||||
|
||||
#include "audio.h"
|
||||
#include "avfilter.h"
|
||||
#include "formats.h"
|
||||
#include "filters.h"
|
||||
#include "internal.h"
|
||||
|
||||
enum OutModes {
|
||||
IN_MODE,
|
||||
DESIRED_MODE,
|
||||
OUT_MODE,
|
||||
NOISE_MODE,
|
||||
NB_OMODES
|
||||
};
|
||||
|
||||
typedef struct AudioNLMSContext {
|
||||
const AVClass *class;
|
||||
|
||||
int order;
|
||||
float mu;
|
||||
float eps;
|
||||
float leakage;
|
||||
int output_mode;
|
||||
|
||||
int kernel_size;
|
||||
AVFrame *offset;
|
||||
AVFrame *delay;
|
||||
AVFrame *coeffs;
|
||||
AVFrame *tmp;
|
||||
|
||||
AVFrame *frame[2];
|
||||
|
||||
AVFloatDSPContext *fdsp;
|
||||
} AudioNLMSContext;
|
||||
|
||||
#define OFFSET(x) offsetof(AudioNLMSContext, x)
|
||||
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
|
||||
|
||||
static const AVOption anlms_options[] = {
|
||||
{ "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=256}, 1, INT16_MAX, A },
|
||||
{ "mu", "set the filter mu", OFFSET(mu), AV_OPT_TYPE_FLOAT, {.dbl=0.75}, 0, 1, A },
|
||||
{ "eps", "set the filter eps", OFFSET(eps), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, A },
|
||||
{ "leakage", "set the filter leakage", OFFSET(leakage), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, A },
|
||||
{ "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, A, "mode" },
|
||||
{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, A, "mode" },
|
||||
{ "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, A, "mode" },
|
||||
{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, A, "mode" },
|
||||
{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, A, "mode" },
|
||||
{ NULL }
|
||||
};
|
||||
|
||||
AVFILTER_DEFINE_CLASS(anlms);
|
||||
|
||||
static int query_formats(AVFilterContext *ctx)
|
||||
{
|
||||
AVFilterFormats *formats;
|
||||
AVFilterChannelLayouts *layouts;
|
||||
static const enum AVSampleFormat sample_fmts[] = {
|
||||
AV_SAMPLE_FMT_FLTP,
|
||||
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 float fir_sample(AudioNLMSContext *s, float sample, float *delay,
|
||||
float *coeffs, float *tmp, int *offset)
|
||||
{
|
||||
const int order = s->order;
|
||||
float output;
|
||||
|
||||
delay[*offset] = sample;
|
||||
|
||||
memcpy(tmp, coeffs + order - *offset, order * sizeof(float));
|
||||
|
||||
output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
|
||||
|
||||
if (--(*offset) < 0)
|
||||
*offset = order - 1;
|
||||
|
||||
return output;
|
||||
}
|
||||
|
||||
static float process_sample(AudioNLMSContext *s, float input, float desired,
|
||||
float *delay, float *coeffs, float *tmp, int *offsetp)
|
||||
{
|
||||
const int order = s->order;
|
||||
const float leakage = s->leakage;
|
||||
const float mu = s->mu;
|
||||
const float a = 1.f - leakage * mu;
|
||||
float sum, output, e, norm, b;
|
||||
int offset = *offsetp;
|
||||
|
||||
delay[offset + order] = input;
|
||||
|
||||
output = fir_sample(s, input, delay, coeffs, tmp, offsetp);
|
||||
e = desired - output;
|
||||
|
||||
sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size);
|
||||
|
||||
norm = s->eps + sum;
|
||||
b = mu * e / norm;
|
||||
|
||||
memcpy(tmp, delay + offset, order * sizeof(float));
|
||||
|
||||
s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size);
|
||||
|
||||
s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size);
|
||||
|
||||
memcpy(coeffs + order, coeffs, order * sizeof(float));
|
||||
|
||||
switch (s->output_mode) {
|
||||
case IN_MODE: output = input; break;
|
||||
case DESIRED_MODE: output = desired; break;
|
||||
case OUT_MODE: /*output = output;*/ break;
|
||||
case NOISE_MODE: output = desired - output; break;
|
||||
}
|
||||
return output;
|
||||
}
|
||||
|
||||
static int process_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
|
||||
{
|
||||
AudioNLMSContext *s = ctx->priv;
|
||||
AVFrame *out = arg;
|
||||
const int start = (out->channels * jobnr) / nb_jobs;
|
||||
const int end = (out->channels * (jobnr+1)) / nb_jobs;
|
||||
|
||||
for (int c = start; c < end; c++) {
|
||||
const float *input = (const float *)s->frame[0]->extended_data[c];
|
||||
const float *desired = (const float *)s->frame[1]->extended_data[c];
|
||||
float *delay = (float *)s->delay->extended_data[c];
|
||||
float *coeffs = (float *)s->coeffs->extended_data[c];
|
||||
float *tmp = (float *)s->tmp->extended_data[c];
|
||||
int *offset = (int *)s->offset->extended_data[c];
|
||||
float *output = (float *)out->extended_data[c];
|
||||
|
||||
for (int n = 0; n < out->nb_samples; n++)
|
||||
output[n] = process_sample(s, input[n], desired[n], delay, coeffs, tmp, offset);
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int activate(AVFilterContext *ctx)
|
||||
{
|
||||
AudioNLMSContext *s = ctx->priv;
|
||||
int i, ret, status;
|
||||
int nb_samples;
|
||||
int64_t pts;
|
||||
|
||||
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
|
||||
|
||||
nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]),
|
||||
ff_inlink_queued_samples(ctx->inputs[1]));
|
||||
for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) {
|
||||
if (s->frame[i])
|
||||
continue;
|
||||
|
||||
if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) {
|
||||
ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frame[i]);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
}
|
||||
}
|
||||
|
||||
if (s->frame[0] && s->frame[1]) {
|
||||
AVFrame *out;
|
||||
|
||||
out = ff_get_audio_buffer(ctx->outputs[0], s->frame[0]->nb_samples);
|
||||
if (!out) {
|
||||
av_frame_free(&s->frame[0]);
|
||||
av_frame_free(&s->frame[1]);
|
||||
return AVERROR(ENOMEM);
|
||||
}
|
||||
|
||||
ctx->internal->execute(ctx, process_channels, out, NULL, FFMIN(ctx->outputs[0]->channels,
|
||||
ff_filter_get_nb_threads(ctx)));
|
||||
|
||||
out->pts = s->frame[0]->pts;
|
||||
|
||||
av_frame_free(&s->frame[0]);
|
||||
av_frame_free(&s->frame[1]);
|
||||
|
||||
ret = ff_filter_frame(ctx->outputs[0], out);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
}
|
||||
|
||||
if (!nb_samples) {
|
||||
for (i = 0; i < 2; i++) {
|
||||
if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
|
||||
ff_outlink_set_status(ctx->outputs[0], status, pts);
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
if (ff_outlink_frame_wanted(ctx->outputs[0])) {
|
||||
for (i = 0; i < 2; i++) {
|
||||
if (ff_inlink_queued_samples(ctx->inputs[i]) > 0)
|
||||
continue;
|
||||
ff_inlink_request_frame(ctx->inputs[i]);
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int config_output(AVFilterLink *outlink)
|
||||
{
|
||||
AVFilterContext *ctx = outlink->src;
|
||||
AudioNLMSContext *s = ctx->priv;
|
||||
|
||||
s->kernel_size = FFALIGN(s->order, 16);
|
||||
|
||||
if (!s->offset)
|
||||
s->offset = ff_get_audio_buffer(outlink, 1);
|
||||
if (!s->delay)
|
||||
s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
|
||||
if (!s->coeffs)
|
||||
s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
|
||||
if (!s->tmp)
|
||||
s->tmp = ff_get_audio_buffer(outlink, s->kernel_size);
|
||||
if (!s->delay || !s->coeffs || !s->offset || !s->tmp)
|
||||
return AVERROR(ENOMEM);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static av_cold int init(AVFilterContext *ctx)
|
||||
{
|
||||
AudioNLMSContext *s = ctx->priv;
|
||||
|
||||
s->fdsp = avpriv_float_dsp_alloc(0);
|
||||
if (!s->fdsp)
|
||||
return AVERROR(ENOMEM);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static av_cold void uninit(AVFilterContext *ctx)
|
||||
{
|
||||
AudioNLMSContext *s = ctx->priv;
|
||||
|
||||
av_freep(&s->fdsp);
|
||||
av_frame_free(&s->delay);
|
||||
av_frame_free(&s->coeffs);
|
||||
av_frame_free(&s->offset);
|
||||
av_frame_free(&s->tmp);
|
||||
}
|
||||
|
||||
static const AVFilterPad inputs[] = {
|
||||
{
|
||||
.name = "input",
|
||||
.type = AVMEDIA_TYPE_AUDIO,
|
||||
},
|
||||
{
|
||||
.name = "desired",
|
||||
.type = AVMEDIA_TYPE_AUDIO,
|
||||
},
|
||||
{ NULL }
|
||||
};
|
||||
|
||||
static const AVFilterPad outputs[] = {
|
||||
{
|
||||
.name = "default",
|
||||
.type = AVMEDIA_TYPE_AUDIO,
|
||||
.config_props = config_output,
|
||||
},
|
||||
{ NULL }
|
||||
};
|
||||
|
||||
AVFilter ff_af_anlms = {
|
||||
.name = "anlms",
|
||||
.description = NULL_IF_CONFIG_SMALL("Apply Normalized Least-Mean-Squares algorithm to first audio stream."),
|
||||
.priv_size = sizeof(AudioNLMSContext),
|
||||
.priv_class = &anlms_class,
|
||||
.init = init,
|
||||
.uninit = uninit,
|
||||
.activate = activate,
|
||||
.query_formats = query_formats,
|
||||
.inputs = inputs,
|
||||
.outputs = outputs,
|
||||
.flags = AVFILTER_FLAG_SLICE_THREADS,
|
||||
};
|
||||
@ -56,6 +56,7 @@ extern AVFilter ff_af_amix;
|
||||
extern AVFilter ff_af_amultiply;
|
||||
extern AVFilter ff_af_anequalizer;
|
||||
extern AVFilter ff_af_anlmdn;
|
||||
extern AVFilter ff_af_anlms;
|
||||
extern AVFilter ff_af_anull;
|
||||
extern AVFilter ff_af_apad;
|
||||
extern AVFilter ff_af_aperms;
|
||||
|
||||
@ -30,7 +30,7 @@
|
||||
#include "libavutil/version.h"
|
||||
|
||||
#define LIBAVFILTER_VERSION_MAJOR 7
|
||||
#define LIBAVFILTER_VERSION_MINOR 61
|
||||
#define LIBAVFILTER_VERSION_MINOR 62
|
||||
#define LIBAVFILTER_VERSION_MICRO 100
|
||||
|
||||
|
||||
|
||||
Loading…
Reference in New Issue
Block a user