mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-21 10:55:51 +02:00
777 lines
23 KiB
C
777 lines
23 KiB
C
/*
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* Copyright (c) 2018 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/audio_fifo.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "filters.h"
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#include "formats.h"
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#include "internal.h"
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typedef struct DeclickChannel {
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double *auxiliary;
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double *detection;
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double *acoefficients;
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double *acorrelation;
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double *tmp;
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double *interpolated;
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double *matrix;
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int matrix_size;
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double *vector;
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int vector_size;
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double *y;
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int y_size;
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uint8_t *click;
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int *index;
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unsigned *histogram;
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int histogram_size;
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} DeclickChannel;
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typedef struct AudioDeclickContext {
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const AVClass *class;
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double w;
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double overlap;
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double threshold;
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double ar;
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double burst;
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int method;
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int nb_hbins;
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int is_declip;
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int ar_order;
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int nb_burst_samples;
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int window_size;
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int hop_size;
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int overlap_skip;
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AVFrame *in;
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AVFrame *out;
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AVFrame *buffer;
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AVFrame *is;
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DeclickChannel *chan;
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int64_t pts;
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int nb_channels;
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uint64_t nb_samples;
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uint64_t detected_errors;
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int samples_left;
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int eof;
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AVAudioFifo *fifo;
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double *window_func_lut;
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int (*detector)(struct AudioDeclickContext *s, DeclickChannel *c,
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double sigmae, double *detection,
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double *acoefficients, uint8_t *click, int *index,
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const double *src, double *dst);
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} AudioDeclickContext;
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#define OFFSET(x) offsetof(AudioDeclickContext, x)
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#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption adeclick_options[] = {
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{ "w", "set window size", OFFSET(w), AV_OPT_TYPE_DOUBLE, {.dbl=55}, 10, 100, AF },
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{ "o", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=75}, 50, 95, AF },
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{ "a", "set autoregression order", OFFSET(ar), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 25, AF },
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{ "t", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 100, AF },
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{ "b", "set burst fusion", OFFSET(burst), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, AF },
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{ "m", "set overlap method", OFFSET(method), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "m" },
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{ "a", "overlap-add", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
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{ "s", "overlap-save", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(adeclick);
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layouts = NULL;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats(ctx, formats);
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if (ret < 0)
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return ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_all_samplerates();
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return ff_set_common_samplerates(ctx, formats);
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}
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioDeclickContext *s = ctx->priv;
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int i;
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s->pts = AV_NOPTS_VALUE;
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s->window_size = inlink->sample_rate * s->w / 1000.;
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if (s->window_size < 100)
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return AVERROR(EINVAL);
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s->ar_order = FFMAX(s->window_size * s->ar / 100., 1);
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s->nb_burst_samples = s->window_size * s->burst / 1000.;
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s->hop_size = s->window_size * (1. - (s->overlap / 100.));
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if (s->hop_size < 1)
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return AVERROR(EINVAL);
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s->window_func_lut = av_calloc(s->window_size, sizeof(*s->window_func_lut));
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if (!s->window_func_lut)
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return AVERROR(ENOMEM);
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for (i = 0; i < s->window_size; i++)
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s->window_func_lut[i] = sin(M_PI * i / s->window_size) *
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(1. - (s->overlap / 100.)) * M_PI_2;
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av_frame_free(&s->in);
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av_frame_free(&s->out);
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av_frame_free(&s->buffer);
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av_frame_free(&s->is);
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s->in = ff_get_audio_buffer(inlink, s->window_size);
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s->out = ff_get_audio_buffer(inlink, s->window_size);
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s->buffer = ff_get_audio_buffer(inlink, s->window_size * 2);
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s->is = ff_get_audio_buffer(inlink, s->window_size);
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if (!s->in || !s->out || !s->buffer || !s->is)
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return AVERROR(ENOMEM);
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s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->window_size);
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if (!s->fifo)
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return AVERROR(ENOMEM);
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s->overlap_skip = s->method ? (s->window_size - s->hop_size) / 2 : 0;
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if (s->overlap_skip > 0) {
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av_audio_fifo_write(s->fifo, (void **)s->in->extended_data,
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s->overlap_skip);
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}
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s->nb_channels = inlink->channels;
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s->chan = av_calloc(inlink->channels, sizeof(*s->chan));
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if (!s->chan)
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return AVERROR(ENOMEM);
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for (i = 0; i < inlink->channels; i++) {
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DeclickChannel *c = &s->chan[i];
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c->detection = av_calloc(s->window_size, sizeof(*c->detection));
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c->auxiliary = av_calloc(s->ar_order + 1, sizeof(*c->auxiliary));
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c->acoefficients = av_calloc(s->ar_order + 1, sizeof(*c->acoefficients));
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c->acorrelation = av_calloc(s->ar_order + 1, sizeof(*c->acorrelation));
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c->tmp = av_calloc(s->ar_order, sizeof(*c->tmp));
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c->click = av_calloc(s->window_size, sizeof(*c->click));
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c->index = av_calloc(s->window_size, sizeof(*c->index));
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c->interpolated = av_calloc(s->window_size, sizeof(*c->interpolated));
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if (!c->auxiliary || !c->acoefficients || !c->detection || !c->click ||
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!c->index || !c->interpolated || !c->acorrelation || !c->tmp)
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return AVERROR(ENOMEM);
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}
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return 0;
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}
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static void autocorrelation(const double *input, int order, int size,
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double *output, double scale)
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{
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int i, j;
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for (i = 0; i <= order; i++) {
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double value = 0.;
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for (j = i; j < size; j++)
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value += input[j] * input[j - i];
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output[i] = value * scale;
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}
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}
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static double autoregression(const double *samples, int ar_order,
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int nb_samples, double *k, double *r, double *a)
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{
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double alpha;
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int i, j;
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memset(a, 0, ar_order * sizeof(*a));
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autocorrelation(samples, ar_order, nb_samples, r, 1. / nb_samples);
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/* Levinson-Durbin algorithm */
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k[0] = a[0] = -r[1] / r[0];
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alpha = r[0] * (1. - k[0] * k[0]);
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for (i = 1; i < ar_order; i++) {
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double epsilon = 0.;
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for (j = 0; j < i; j++)
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epsilon += a[j] * r[i - j];
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epsilon += r[i + 1];
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k[i] = -epsilon / alpha;
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alpha *= (1. - k[i] * k[i]);
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for (j = i - 1; j >= 0; j--)
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k[j] = a[j] + k[i] * a[i - j - 1];
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for (j = 0; j <= i; j++)
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a[j] = k[j];
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}
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k[0] = 1.;
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for (i = 1; i <= ar_order; i++)
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k[i] = a[i - 1];
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return sqrt(alpha);
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}
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static int isfinite_array(double *samples, int nb_samples)
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{
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int i;
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for (i = 0; i < nb_samples; i++)
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if (!isfinite(samples[i]))
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return 0;
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return 1;
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}
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static int find_index(int *index, int value, int size)
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{
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int i, start, end;
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if ((value < index[0]) || (value > index[size - 1]))
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return 1;
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i = start = 0;
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end = size - 1;
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while (start <= end) {
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i = (end + start) / 2;
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if (index[i] == value)
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return 0;
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if (value < index[i])
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end = i - 1;
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if (value > index[i])
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start = i + 1;
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}
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return 1;
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}
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static int factorization(double *matrix, int n)
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{
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int i, j, k;
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for (i = 0; i < n; i++) {
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const int in = i * n;
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double value;
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value = matrix[in + i];
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for (j = 0; j < i; j++)
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value -= matrix[j * n + j] * matrix[in + j] * matrix[in + j];
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if (value == 0.) {
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return -1;
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}
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matrix[in + i] = value;
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for (j = i + 1; j < n; j++) {
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const int jn = j * n;
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double x;
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x = matrix[jn + i];
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for (k = 0; k < i; k++)
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x -= matrix[k * n + k] * matrix[in + k] * matrix[jn + k];
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matrix[jn + i] = x / matrix[in + i];
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}
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}
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return 0;
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}
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static int do_interpolation(DeclickChannel *c, double *matrix,
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double *vector, int n, double *out)
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{
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int i, j, ret;
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double *y;
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ret = factorization(matrix, n);
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if (ret < 0)
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return ret;
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av_fast_malloc(&c->y, &c->y_size, n * sizeof(*c->y));
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y = c->y;
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if (!y)
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return AVERROR(ENOMEM);
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for (i = 0; i < n; i++) {
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const int in = i * n;
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double value;
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value = vector[i];
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for (j = 0; j < i; j++)
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value -= matrix[in + j] * y[j];
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y[i] = value;
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}
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for (i = n - 1; i >= 0; i--) {
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out[i] = y[i] / matrix[i * n + i];
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for (j = i + 1; j < n; j++)
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out[i] -= matrix[j * n + i] * out[j];
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}
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return 0;
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}
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static int interpolation(DeclickChannel *c, const double *src, int ar_order,
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double *acoefficients, int *index, int nb_errors,
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double *auxiliary, double *interpolated)
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{
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double *vector, *matrix;
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int i, j;
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av_fast_malloc(&c->matrix, &c->matrix_size, nb_errors * nb_errors * sizeof(*c->matrix));
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matrix = c->matrix;
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if (!matrix)
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return AVERROR(ENOMEM);
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av_fast_malloc(&c->vector, &c->vector_size, nb_errors * sizeof(*c->vector));
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vector = c->vector;
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if (!vector)
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return AVERROR(ENOMEM);
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autocorrelation(acoefficients, ar_order, ar_order + 1, auxiliary, 1.);
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for (i = 0; i < nb_errors; i++) {
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const int im = i * nb_errors;
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for (j = i; j < nb_errors; j++) {
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if (abs(index[j] - index[i]) <= ar_order) {
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matrix[j * nb_errors + i] = matrix[im + j] = auxiliary[abs(index[j] - index[i])];
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} else {
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matrix[j * nb_errors + i] = matrix[im + j] = 0;
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}
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}
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}
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for (i = 0; i < nb_errors; i++) {
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double value = 0.;
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for (j = -ar_order; j <= ar_order; j++)
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if (find_index(index, index[i] - j, nb_errors))
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value -= src[index[i] - j] * auxiliary[abs(j)];
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vector[i] = value;
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}
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return do_interpolation(c, matrix, vector, nb_errors, interpolated);
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}
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static int detect_clips(AudioDeclickContext *s, DeclickChannel *c,
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double unused0,
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double *unused1, double *unused2,
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uint8_t *clip, int *index,
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const double *src, double *dst)
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{
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const double threshold = s->threshold;
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double max_amplitude = 0;
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unsigned *histogram;
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int i, nb_clips = 0;
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av_fast_malloc(&c->histogram, &c->histogram_size, s->nb_hbins * sizeof(*c->histogram));
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if (!c->histogram)
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return AVERROR(ENOMEM);
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histogram = c->histogram;
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memset(histogram, 0, sizeof(*histogram) * s->nb_hbins);
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for (i = 0; i < s->window_size; i++) {
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const unsigned index = fmin(fabs(src[i]), 1) * (s->nb_hbins - 1);
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histogram[index]++;
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dst[i] = src[i];
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clip[i] = 0;
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}
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for (i = s->nb_hbins - 1; i > 1; i--) {
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if (histogram[i]) {
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if (histogram[i] / (double)FFMAX(histogram[i - 1], 1) > threshold) {
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max_amplitude = i / (double)s->nb_hbins;
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}
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break;
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}
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}
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if (max_amplitude > 0.) {
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for (i = 0; i < s->window_size; i++) {
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clip[i] = fabs(src[i]) >= max_amplitude;
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}
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}
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memset(clip, 0, s->ar_order * sizeof(*clip));
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memset(clip + (s->window_size - s->ar_order), 0, s->ar_order * sizeof(*clip));
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for (i = s->ar_order; i < s->window_size - s->ar_order; i++)
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if (clip[i])
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index[nb_clips++] = i;
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return nb_clips;
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}
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static int detect_clicks(AudioDeclickContext *s, DeclickChannel *c,
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double sigmae,
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double *detection, double *acoefficients,
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uint8_t *click, int *index,
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const double *src, double *dst)
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{
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const double threshold = s->threshold;
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int i, j, nb_clicks = 0, prev = -1;
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memset(detection, 0, s->window_size * sizeof(*detection));
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for (i = s->ar_order; i < s->window_size; i++) {
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for (j = 0; j <= s->ar_order; j++) {
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detection[i] += acoefficients[j] * src[i - j];
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}
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}
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for (i = 0; i < s->window_size; i++) {
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click[i] = fabs(detection[i]) > sigmae * threshold;
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dst[i] = src[i];
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}
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for (i = 0; i < s->window_size; i++) {
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if (!click[i])
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continue;
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if (prev >= 0 && (i > prev + 1) && (i <= s->nb_burst_samples + prev))
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for (j = prev + 1; j < i; j++)
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click[j] = 1;
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prev = i;
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}
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memset(click, 0, s->ar_order * sizeof(*click));
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memset(click + (s->window_size - s->ar_order), 0, s->ar_order * sizeof(*click));
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for (i = s->ar_order; i < s->window_size - s->ar_order; i++)
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if (click[i])
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index[nb_clicks++] = i;
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return nb_clicks;
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}
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typedef struct ThreadData {
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AVFrame *out;
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} ThreadData;
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static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
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{
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AudioDeclickContext *s = ctx->priv;
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ThreadData *td = arg;
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AVFrame *out = td->out;
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const double *src = (const double *)s->in->extended_data[ch];
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double *is = (double *)s->is->extended_data[ch];
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double *dst = (double *)s->out->extended_data[ch];
|
|
double *ptr = (double *)out->extended_data[ch];
|
|
double *buf = (double *)s->buffer->extended_data[ch];
|
|
const double *w = s->window_func_lut;
|
|
DeclickChannel *c = &s->chan[ch];
|
|
double sigmae;
|
|
int j, ret;
|
|
|
|
sigmae = autoregression(src, s->ar_order, s->window_size, c->acoefficients, c->acorrelation, c->tmp);
|
|
|
|
if (isfinite_array(c->acoefficients, s->ar_order + 1)) {
|
|
double *interpolated = c->interpolated;
|
|
int *index = c->index;
|
|
int nb_errors;
|
|
|
|
nb_errors = s->detector(s, c, sigmae, c->detection, c->acoefficients,
|
|
c->click, index, src, dst);
|
|
if (nb_errors > 0) {
|
|
ret = interpolation(c, src, s->ar_order, c->acoefficients, index,
|
|
nb_errors, c->auxiliary, interpolated);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
for (j = 0; j < nb_errors; j++) {
|
|
dst[index[j]] = interpolated[j];
|
|
is[index[j]] = 1;
|
|
}
|
|
}
|
|
} else {
|
|
memcpy(dst, src, s->window_size * sizeof(*dst));
|
|
}
|
|
|
|
if (s->method == 0) {
|
|
for (j = 0; j < s->window_size; j++)
|
|
buf[j] += dst[j] * w[j];
|
|
} else {
|
|
const int skip = s->overlap_skip;
|
|
|
|
for (j = 0; j < s->hop_size; j++)
|
|
buf[j] = dst[skip + j];
|
|
}
|
|
for (j = 0; j < s->hop_size; j++)
|
|
ptr[j] = buf[j];
|
|
|
|
memmove(buf, buf + s->hop_size, (s->window_size * 2 - s->hop_size) * sizeof(*buf));
|
|
memmove(is, is + s->hop_size, (s->window_size - s->hop_size) * sizeof(*is));
|
|
memset(buf + s->window_size * 2 - s->hop_size, 0, s->hop_size * sizeof(*buf));
|
|
memset(is + s->window_size - s->hop_size, 0, s->hop_size * sizeof(*is));
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int filter_frame(AVFilterLink *inlink)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
|
|
AVFilterLink *outlink = ctx->outputs[0];
|
|
AudioDeclickContext *s = ctx->priv;
|
|
AVFrame *out = NULL;
|
|
int ret = 0, j, ch, detected_errors = 0;
|
|
ThreadData td;
|
|
|
|
out = ff_get_audio_buffer(outlink, s->hop_size);
|
|
if (!out)
|
|
return AVERROR(ENOMEM);
|
|
|
|
ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data,
|
|
s->window_size);
|
|
if (ret < 0)
|
|
goto fail;
|
|
|
|
td.out = out;
|
|
ret = ctx->internal->execute(ctx, filter_channel, &td, NULL, inlink->channels);
|
|
if (ret < 0)
|
|
goto fail;
|
|
|
|
for (ch = 0; ch < s->in->channels; ch++) {
|
|
double *is = (double *)s->is->extended_data[ch];
|
|
|
|
for (j = 0; j < s->hop_size; j++) {
|
|
if (is[j])
|
|
detected_errors++;
|
|
}
|
|
}
|
|
|
|
av_audio_fifo_drain(s->fifo, s->hop_size);
|
|
|
|
if (s->samples_left > 0)
|
|
out->nb_samples = FFMIN(s->hop_size, s->samples_left);
|
|
|
|
out->pts = s->pts;
|
|
s->pts += s->hop_size;
|
|
|
|
s->detected_errors += detected_errors;
|
|
s->nb_samples += out->nb_samples * inlink->channels;
|
|
|
|
ret = ff_filter_frame(outlink, out);
|
|
if (ret < 0)
|
|
goto fail;
|
|
|
|
if (s->samples_left > 0) {
|
|
s->samples_left -= s->hop_size;
|
|
if (s->samples_left <= 0)
|
|
av_audio_fifo_drain(s->fifo, av_audio_fifo_size(s->fifo));
|
|
}
|
|
|
|
fail:
|
|
if (ret < 0)
|
|
av_frame_free(&out);
|
|
return ret;
|
|
}
|
|
|
|
static int activate(AVFilterContext *ctx)
|
|
{
|
|
AVFilterLink *inlink = ctx->inputs[0];
|
|
AVFilterLink *outlink = ctx->outputs[0];
|
|
AudioDeclickContext *s = ctx->priv;
|
|
AVFrame *in;
|
|
int ret, status;
|
|
int64_t pts;
|
|
|
|
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
|
|
|
|
ret = ff_inlink_consume_samples(inlink, s->window_size, s->window_size, &in);
|
|
if (ret < 0)
|
|
return ret;
|
|
if (ret > 0) {
|
|
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);
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
|
|
if (av_audio_fifo_size(s->fifo) >= s->window_size ||
|
|
s->samples_left > 0)
|
|
return filter_frame(inlink);
|
|
|
|
if (av_audio_fifo_size(s->fifo) >= s->window_size) {
|
|
ff_filter_set_ready(ctx, 100);
|
|
return 0;
|
|
}
|
|
|
|
if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
|
|
if (status == AVERROR_EOF) {
|
|
s->eof = 1;
|
|
s->samples_left = av_audio_fifo_size(s->fifo) - s->overlap_skip;
|
|
ff_filter_set_ready(ctx, 100);
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
if (s->eof && s->samples_left <= 0) {
|
|
ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
|
|
return 0;
|
|
}
|
|
|
|
if (!s->eof)
|
|
FF_FILTER_FORWARD_WANTED(outlink, inlink);
|
|
|
|
return FFERROR_NOT_READY;
|
|
}
|
|
|
|
static av_cold int init(AVFilterContext *ctx)
|
|
{
|
|
AudioDeclickContext *s = ctx->priv;
|
|
|
|
s->is_declip = !strcmp(ctx->filter->name, "adeclip");
|
|
if (s->is_declip) {
|
|
s->detector = detect_clips;
|
|
} else {
|
|
s->detector = detect_clicks;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold void uninit(AVFilterContext *ctx)
|
|
{
|
|
AudioDeclickContext *s = ctx->priv;
|
|
int i;
|
|
|
|
av_log(ctx, AV_LOG_INFO, "Detected %s in %"PRId64" of %"PRId64" samples (%g%%).\n",
|
|
s->is_declip ? "clips" : "clicks", s->detected_errors,
|
|
s->nb_samples, 100. * s->detected_errors / s->nb_samples);
|
|
|
|
av_audio_fifo_free(s->fifo);
|
|
av_freep(&s->window_func_lut);
|
|
av_frame_free(&s->in);
|
|
av_frame_free(&s->out);
|
|
av_frame_free(&s->buffer);
|
|
av_frame_free(&s->is);
|
|
|
|
if (s->chan) {
|
|
for (i = 0; i < s->nb_channels; i++) {
|
|
DeclickChannel *c = &s->chan[i];
|
|
|
|
av_freep(&c->detection);
|
|
av_freep(&c->auxiliary);
|
|
av_freep(&c->acoefficients);
|
|
av_freep(&c->acorrelation);
|
|
av_freep(&c->tmp);
|
|
av_freep(&c->click);
|
|
av_freep(&c->index);
|
|
av_freep(&c->interpolated);
|
|
av_freep(&c->matrix);
|
|
c->matrix_size = 0;
|
|
av_freep(&c->histogram);
|
|
c->histogram_size = 0;
|
|
av_freep(&c->vector);
|
|
c->vector_size = 0;
|
|
av_freep(&c->y);
|
|
c->y_size = 0;
|
|
}
|
|
}
|
|
av_freep(&s->chan);
|
|
s->nb_channels = 0;
|
|
}
|
|
|
|
static const AVFilterPad inputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.config_props = config_input,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
static const AVFilterPad outputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
AVFilter ff_af_adeclick = {
|
|
.name = "adeclick",
|
|
.description = NULL_IF_CONFIG_SMALL("Remove impulsive noise from input audio."),
|
|
.query_formats = query_formats,
|
|
.priv_size = sizeof(AudioDeclickContext),
|
|
.priv_class = &adeclick_class,
|
|
.init = init,
|
|
.activate = activate,
|
|
.uninit = uninit,
|
|
.inputs = inputs,
|
|
.outputs = outputs,
|
|
.flags = AVFILTER_FLAG_SLICE_THREADS,
|
|
};
|
|
|
|
static const AVOption adeclip_options[] = {
|
|
{ "w", "set window size", OFFSET(w), AV_OPT_TYPE_DOUBLE, {.dbl=55}, 10, 100, AF },
|
|
{ "o", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=75}, 50, 95, AF },
|
|
{ "a", "set autoregression order", OFFSET(ar), AV_OPT_TYPE_DOUBLE, {.dbl=8}, 0, 25, AF },
|
|
{ "t", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=10}, 1, 100, AF },
|
|
{ "n", "set histogram size", OFFSET(nb_hbins), AV_OPT_TYPE_INT, {.i64=1000}, 100, 9999, AF },
|
|
{ "m", "set overlap method", OFFSET(method), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "m" },
|
|
{ "a", "overlap-add", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
|
|
{ "s", "overlap-save", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" },
|
|
{ NULL }
|
|
};
|
|
|
|
AVFILTER_DEFINE_CLASS(adeclip);
|
|
|
|
AVFilter ff_af_adeclip = {
|
|
.name = "adeclip",
|
|
.description = NULL_IF_CONFIG_SMALL("Remove clipping from input audio."),
|
|
.query_formats = query_formats,
|
|
.priv_size = sizeof(AudioDeclickContext),
|
|
.priv_class = &adeclip_class,
|
|
.init = init,
|
|
.activate = activate,
|
|
.uninit = uninit,
|
|
.inputs = inputs,
|
|
.outputs = outputs,
|
|
.flags = AVFILTER_FLAG_SLICE_THREADS,
|
|
};
|