mirror of https://git.ffmpeg.org/ffmpeg.git
300 lines
9.2 KiB
C
300 lines
9.2 KiB
C
/*
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* Copyright (c) 2023 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/common.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "formats.h"
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#include "filters.h"
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#include "internal.h"
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enum OutModes {
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IN_MODE,
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DESIRED_MODE,
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OUT_MODE,
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NOISE_MODE,
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ERROR_MODE,
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NB_OMODES
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};
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typedef struct AudioRLSContext {
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const AVClass *class;
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int order;
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float lambda;
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float delta;
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int output_mode;
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int precision;
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int kernel_size;
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AVFrame *offset;
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AVFrame *delay;
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AVFrame *coeffs;
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AVFrame *p, *dp;
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AVFrame *gains;
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AVFrame *u, *tmp;
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AVFrame *frame[2];
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int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
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AVFloatDSPContext *fdsp;
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} AudioRLSContext;
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#define OFFSET(x) offsetof(AudioRLSContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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#define AT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption arls_options[] = {
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{ "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=16}, 1, INT16_MAX, A },
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{ "lambda", "set the filter lambda", OFFSET(lambda), AV_OPT_TYPE_FLOAT, {.dbl=1.f}, 0, 1, AT },
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{ "delta", "set the filter delta", OFFSET(delta), AV_OPT_TYPE_FLOAT, {.dbl=2.f}, 0, INT16_MAX, A },
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{ "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, AT, .unit = "mode" },
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{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AT, .unit = "mode" },
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{ "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, AT, .unit = "mode" },
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{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, .unit = "mode" },
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{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, .unit = "mode" },
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{ "e", "error", 0, AV_OPT_TYPE_CONST, {.i64=ERROR_MODE}, 0, 0, AT, .unit = "mode" },
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{ "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, .unit = "precision" },
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{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "precision" },
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{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "precision" },
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{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, .unit = "precision" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(arls);
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static int query_formats(AVFilterContext *ctx)
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{
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AudioRLSContext *s = ctx->priv;
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static const enum AVSampleFormat sample_fmts[3][3] = {
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{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
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{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
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{ AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
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};
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int ret;
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if ((ret = ff_set_common_all_channel_counts(ctx)) < 0)
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return ret;
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if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
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return ret;
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return ff_set_common_all_samplerates(ctx);
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}
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static int activate(AVFilterContext *ctx)
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{
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AudioRLSContext *s = ctx->priv;
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int i, ret, status;
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int nb_samples;
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int64_t pts;
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FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
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nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]),
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ff_inlink_queued_samples(ctx->inputs[1]));
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for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) {
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if (s->frame[i])
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continue;
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if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) {
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ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frame[i]);
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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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if (s->frame[0] && s->frame[1]) {
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AVFrame *out;
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out = ff_get_audio_buffer(ctx->outputs[0], s->frame[0]->nb_samples);
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if (!out) {
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av_frame_free(&s->frame[0]);
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av_frame_free(&s->frame[1]);
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return AVERROR(ENOMEM);
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}
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ff_filter_execute(ctx, s->filter_channels, out, NULL,
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FFMIN(ctx->outputs[0]->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
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out->pts = s->frame[0]->pts;
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out->duration = s->frame[0]->duration;
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av_frame_free(&s->frame[0]);
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av_frame_free(&s->frame[1]);
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ret = ff_filter_frame(ctx->outputs[0], out);
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if (ret < 0)
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return ret;
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}
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if (!nb_samples) {
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for (i = 0; i < 2; i++) {
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if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
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ff_outlink_set_status(ctx->outputs[0], status, pts);
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return 0;
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}
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}
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}
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if (ff_outlink_frame_wanted(ctx->outputs[0])) {
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for (i = 0; i < 2; i++) {
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if (s->frame[i] || ff_inlink_queued_samples(ctx->inputs[i]) > 0)
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continue;
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ff_inlink_request_frame(ctx->inputs[i]);
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return 0;
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}
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}
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return 0;
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}
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#define DEPTH 32
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#include "arls_template.c"
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#undef DEPTH
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#define DEPTH 64
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#include "arls_template.c"
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AudioRLSContext *s = ctx->priv;
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s->kernel_size = FFALIGN(s->order, 16);
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if (!s->offset)
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s->offset = ff_get_audio_buffer(outlink, 1);
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if (!s->delay)
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s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
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if (!s->coeffs)
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s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
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if (!s->gains)
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s->gains = ff_get_audio_buffer(outlink, s->kernel_size);
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if (!s->p)
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s->p = ff_get_audio_buffer(outlink, s->kernel_size * s->kernel_size);
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if (!s->dp)
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s->dp = ff_get_audio_buffer(outlink, s->kernel_size * s->kernel_size);
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if (!s->u)
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s->u = ff_get_audio_buffer(outlink, s->kernel_size);
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if (!s->tmp)
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s->tmp = ff_get_audio_buffer(outlink, s->kernel_size);
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if (!s->delay || !s->coeffs || !s->p || !s->dp || !s->gains || !s->offset || !s->u || !s->tmp)
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return AVERROR(ENOMEM);
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for (int ch = 0; ch < s->offset->ch_layout.nb_channels; ch++) {
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int *dst = (int *)s->offset->extended_data[ch];
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for (int i = 0; i < s->kernel_size; i++)
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dst[0] = s->kernel_size - 1;
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}
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switch (outlink->format) {
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case AV_SAMPLE_FMT_DBLP:
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for (int ch = 0; ch < s->p->ch_layout.nb_channels; ch++) {
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double *dst = (double *)s->p->extended_data[ch];
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for (int i = 0; i < s->kernel_size; i++)
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dst[i * s->kernel_size + i] = s->delta;
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}
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s->filter_channels = filter_channels_double;
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break;
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case AV_SAMPLE_FMT_FLTP:
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for (int ch = 0; ch < s->p->ch_layout.nb_channels; ch++) {
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float *dst = (float *)s->p->extended_data[ch];
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for (int i = 0; i < s->kernel_size; i++)
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dst[i * s->kernel_size + i] = s->delta;
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}
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s->filter_channels = filter_channels_float;
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break;
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}
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return 0;
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}
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static av_cold int init(AVFilterContext *ctx)
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{
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AudioRLSContext *s = ctx->priv;
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s->fdsp = avpriv_float_dsp_alloc(0);
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if (!s->fdsp)
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return AVERROR(ENOMEM);
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioRLSContext *s = ctx->priv;
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av_freep(&s->fdsp);
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av_frame_free(&s->delay);
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av_frame_free(&s->coeffs);
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av_frame_free(&s->gains);
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av_frame_free(&s->offset);
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av_frame_free(&s->p);
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av_frame_free(&s->dp);
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av_frame_free(&s->u);
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av_frame_free(&s->tmp);
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}
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static const AVFilterPad inputs[] = {
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{
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.name = "input",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{
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.name = "desired",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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};
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static const AVFilterPad outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output,
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},
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};
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const AVFilter ff_af_arls = {
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.name = "arls",
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.description = NULL_IF_CONFIG_SMALL("Apply Recursive Least Squares algorithm to first audio stream."),
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.priv_size = sizeof(AudioRLSContext),
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.priv_class = &arls_class,
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.init = init,
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.uninit = uninit,
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.activate = activate,
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(outputs),
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FILTER_QUERY_FUNC(query_formats),
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
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AVFILTER_FLAG_SLICE_THREADS,
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.process_command = ff_filter_process_command,
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};
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