mirror of https://git.videolan.org/git/ffmpeg.git
245 lines
7.0 KiB
C
245 lines
7.0 KiB
C
/*
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* Copyright (c) 2013-2020 Michael Barbour <barbour.michael.0@gmail.com>
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* Copyright (c) 2021 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/channel_layout.h"
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#include "libavutil/ffmath.h"
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#include "libavutil/lfg.h"
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#include "libavutil/mem.h"
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#include "libavutil/random_seed.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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#define MAX_STAGES 16
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#define FILTER_FC 1100.0
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#define RT60_LF 0.1
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#define RT60_HF 0.008
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typedef struct APContext {
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int len, p;
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double *mx, *my;
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double b0, b1, a0, a1;
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} APContext;
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typedef struct ADecorrelateContext {
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const AVClass *class;
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int stages;
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int64_t seed;
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int nb_channels;
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APContext (*ap)[MAX_STAGES];
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AVLFG c;
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void (*filter_channel)(AVFilterContext *ctx,
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int channel,
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AVFrame *in, AVFrame *out);
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} ADecorrelateContext;
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static int ap_init(APContext *ap, int fs, double delay)
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{
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const int delay_samples = lrint(round(delay * fs));
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const double gain_lf = -60.0 / (RT60_LF * fs) * delay_samples;
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const double gain_hf = -60.0 / (RT60_HF * fs) * delay_samples;
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const double w0 = 2.0 * M_PI * FILTER_FC / fs;
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const double t = tan(w0 / 2.0);
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const double g_hf = ff_exp10(gain_hf / 20.0);
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const double gd = ff_exp10((gain_lf-gain_hf) / 20.0);
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const double sgd = sqrt(gd);
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ap->len = delay_samples + 1;
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ap->p = 0;
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ap->mx = av_calloc(ap->len, sizeof(*ap->mx));
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ap->my = av_calloc(ap->len, sizeof(*ap->my));
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if (!ap->mx || !ap->my)
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return AVERROR(ENOMEM);
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ap->a0 = t + sgd;
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ap->a1 = (t - sgd) / ap->a0;
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ap->b0 = (gd*t - sgd) / ap->a0 * g_hf;
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ap->b1 = (gd*t + sgd) / ap->a0 * g_hf;
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ap->a0 = 1.0;
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return 0;
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}
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static void ap_free(APContext *ap)
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{
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av_freep(&ap->mx);
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av_freep(&ap->my);
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}
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static double ap_run(APContext *ap, double x)
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{
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const int i0 = ((ap->p < 1) ? ap->len : ap->p)-1, i_n1 = ap->p, i_n2 = (ap->p+1 >= ap->len) ? 0 : ap->p+1;
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const double r = ap->b1*x + ap->b0*ap->mx[i0] + ap->a1*ap->mx[i_n2] + ap->a0*ap->mx[i_n1] -
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ap->a1*ap->my[i0] - ap->b0*ap->my[i_n2] - ap->b1*ap->my[i_n1];
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ap->mx[ap->p] = x;
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ap->my[ap->p] = r;
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ap->p = (ap->p+1 >= ap->len) ? 0 : ap->p+1;
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return r;
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}
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static void filter_channel_dbl(AVFilterContext *ctx, int ch,
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AVFrame *in, AVFrame *out)
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{
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ADecorrelateContext *s = ctx->priv;
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const double *src = (const double *)in->extended_data[ch];
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double *dst = (double *)out->extended_data[ch];
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const int nb_samples = in->nb_samples;
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const int stages = s->stages;
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APContext *ap0 = &s->ap[ch][0];
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for (int n = 0; n < nb_samples; n++) {
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dst[n] = ap_run(ap0, src[n]);
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for (int i = 1; i < stages; i++) {
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APContext *ap = &s->ap[ch][i];
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dst[n] = ap_run(ap, dst[n]);
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}
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}
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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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ADecorrelateContext *s = ctx->priv;
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int ret;
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if (s->seed == -1)
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s->seed = av_get_random_seed();
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av_lfg_init(&s->c, s->seed);
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s->nb_channels = inlink->ch_layout.nb_channels;
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s->ap = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->ap));
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if (!s->ap)
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return AVERROR(ENOMEM);
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for (int i = 0; i < inlink->ch_layout.nb_channels; i++) {
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for (int j = 0; j < s->stages; j++) {
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ret = ap_init(&s->ap[i][j], inlink->sample_rate,
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(double)av_lfg_get(&s->c) / 0xffffffff * 2.2917e-3 + 0.83333e-3);
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if (ret < 0)
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return ret;
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}
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}
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s->filter_channel = filter_channel_dbl;
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return 0;
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}
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typedef struct ThreadData {
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AVFrame *in, *out;
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} ThreadData;
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static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
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{
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ADecorrelateContext *s = ctx->priv;
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ThreadData *td = arg;
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AVFrame *out = td->out;
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AVFrame *in = td->in;
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const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
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const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
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for (int ch = start; ch < end; ch++)
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s->filter_channel(ctx, ch, in, out);
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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AVFrame *out;
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ThreadData td;
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if (av_frame_is_writable(in)) {
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out = in;
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} else {
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out = ff_get_audio_buffer(outlink, in->nb_samples);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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}
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td.in = in; td.out = out;
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ff_filter_execute(ctx, filter_channels, &td, NULL,
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FFMIN(inlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
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if (out != in)
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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ADecorrelateContext *s = ctx->priv;
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if (s->ap) {
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for (int ch = 0; ch < s->nb_channels; ch++) {
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for (int stage = 0; stage < s->stages; stage++)
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ap_free(&s->ap[ch][stage]);
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}
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}
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av_freep(&s->ap);
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}
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#define OFFSET(x) offsetof(ADecorrelateContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption adecorrelate_options[] = {
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{ "stages", "set filtering stages", OFFSET(stages), AV_OPT_TYPE_INT, {.i64=6}, 1, MAX_STAGES, FLAGS },
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{ "seed", "set random seed", OFFSET(seed), AV_OPT_TYPE_INT64, {.i64=-1}, -1, UINT_MAX, FLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(adecorrelate);
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static const AVFilterPad inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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};
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const AVFilter ff_af_adecorrelate = {
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.name = "adecorrelate",
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.description = NULL_IF_CONFIG_SMALL("Apply decorrelation to input audio."),
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.priv_size = sizeof(ADecorrelateContext),
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.priv_class = &adecorrelate_class,
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.uninit = uninit,
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(ff_audio_default_filterpad),
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FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
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AVFILTER_FLAG_SLICE_THREADS,
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};
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