mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-21 10:55:51 +02:00
340 lines
10 KiB
C
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,
|
|
};
|