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FFmpeg/libavfilter/af_anlms.c
2022-04-27 21:30:43 +02:00

340 lines
10 KiB
C

/*
* 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/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];
int anlmf;
AVFloatDSPContext *fdsp;
} AudioNLMSContext;
#define OFFSET(x) offsetof(AudioNLMSContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define AT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_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, 2, AT },
{ "eps", "set the filter eps", OFFSET(eps), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AT },
{ "leakage", "set the filter leakage", OFFSET(leakage), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, AT },
{ "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, AT, "mode" },
{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AT, "mode" },
{ "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, AT, "mode" },
{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, "mode" },
{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, "mode" },
{ NULL }
};
AVFILTER_DEFINE_CLASS_EXT(anlms, "anlm(f|s)", anlms_options);
static int query_formats(AVFilterContext *ctx)
{
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE
};
int ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
ret = ff_set_common_formats_from_list(ctx, sample_fmts);
if (ret < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
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;
if (s->anlmf)
b *= 4.f * e * e;
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->ch_layout.nb_channels * jobnr) / nb_jobs;
const int end = (out->ch_layout.nb_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);
if (ctx->is_disabled)
output[n] = input[n];
}
}
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);
}
ff_filter_execute(ctx, process_channels, out, NULL,
FFMIN(ctx->outputs[0]->ch_layout.nb_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->anlmf = !strcmp(ctx->filter->name, "anlmf");
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,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const 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,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
.process_command = ff_filter_process_command,
};
const AVFilter ff_af_anlmf = {
.name = "anlmf",
.description = NULL_IF_CONFIG_SMALL("Apply Normalized Least-Mean-Fourth algorithm to first audio stream."),
.priv_size = sizeof(AudioNLMSContext),
.priv_class = &anlms_class,
.init = init,
.uninit = uninit,
.activate = activate,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
.process_command = ff_filter_process_command,
};