| /* |
| * Copyright (c) 2023 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/common.h" |
| #include "libavutil/float_dsp.h" |
| #include "libavutil/mem.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, |
| ERROR_MODE, |
| NB_OMODES |
| }; |
| |
| typedef struct AudioRLSContext { |
| const AVClass *class; |
| |
| int order; |
| float lambda; |
| float delta; |
| int output_mode; |
| int precision; |
| |
| int kernel_size; |
| AVFrame *offset; |
| AVFrame *delay; |
| AVFrame *coeffs; |
| AVFrame *p, *dp; |
| AVFrame *gains; |
| AVFrame *u, *tmp; |
| |
| AVFrame *frame[2]; |
| |
| int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs); |
| |
| AVFloatDSPContext *fdsp; |
| } AudioRLSContext; |
| |
| #define OFFSET(x) offsetof(AudioRLSContext, 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 arls_options[] = { |
| { "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=16}, 1, INT16_MAX, A }, |
| { "lambda", "set the filter lambda", OFFSET(lambda), AV_OPT_TYPE_FLOAT, {.dbl=1.f}, 0, 1, AT }, |
| { "delta", "set the filter delta", OFFSET(delta), AV_OPT_TYPE_FLOAT, {.dbl=2.f}, 0, INT16_MAX, A }, |
| { "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, AT, .unit = "mode" }, |
| { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AT, .unit = "mode" }, |
| { "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, AT, .unit = "mode" }, |
| { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, .unit = "mode" }, |
| { "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, .unit = "mode" }, |
| { "e", "error", 0, AV_OPT_TYPE_CONST, {.i64=ERROR_MODE}, 0, 0, AT, .unit = "mode" }, |
| { "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, .unit = "precision" }, |
| { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "precision" }, |
| { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "precision" }, |
| { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, .unit = "precision" }, |
| { NULL } |
| }; |
| |
| AVFILTER_DEFINE_CLASS(arls); |
| |
| static int query_formats(AVFilterContext *ctx) |
| { |
| AudioRLSContext *s = ctx->priv; |
| static const enum AVSampleFormat sample_fmts[3][3] = { |
| { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE }, |
| { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, |
| { AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE }, |
| }; |
| int ret; |
| |
| if ((ret = ff_set_common_all_channel_counts(ctx)) < 0) |
| return ret; |
| |
| if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0) |
| return ret; |
| |
| return ff_set_common_all_samplerates(ctx); |
| } |
| |
| static int activate(AVFilterContext *ctx) |
| { |
| AudioRLSContext *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, s->filter_channels, out, NULL, |
| FFMIN(ctx->outputs[0]->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx))); |
| |
| out->pts = s->frame[0]->pts; |
| out->duration = s->frame[0]->duration; |
| |
| 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 (s->frame[i] || ff_inlink_queued_samples(ctx->inputs[i]) > 0) |
| continue; |
| ff_inlink_request_frame(ctx->inputs[i]); |
| return 0; |
| } |
| } |
| return 0; |
| } |
| |
| #define DEPTH 32 |
| #include "arls_template.c" |
| |
| #undef DEPTH |
| #define DEPTH 64 |
| #include "arls_template.c" |
| |
| static int config_output(AVFilterLink *outlink) |
| { |
| AVFilterContext *ctx = outlink->src; |
| AudioRLSContext *s = ctx->priv; |
| |
| 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->gains) |
| s->gains = ff_get_audio_buffer(outlink, s->kernel_size); |
| if (!s->p) |
| s->p = ff_get_audio_buffer(outlink, s->kernel_size * s->kernel_size); |
| if (!s->dp) |
| s->dp = ff_get_audio_buffer(outlink, s->kernel_size * s->kernel_size); |
| if (!s->u) |
| s->u = ff_get_audio_buffer(outlink, s->kernel_size); |
| if (!s->tmp) |
| s->tmp = ff_get_audio_buffer(outlink, s->kernel_size); |
| |
| if (!s->delay || !s->coeffs || !s->p || !s->dp || !s->gains || !s->offset || !s->u || !s->tmp) |
| return AVERROR(ENOMEM); |
| |
| for (int ch = 0; ch < s->offset->ch_layout.nb_channels; ch++) { |
| int *dst = (int *)s->offset->extended_data[ch]; |
| |
| for (int i = 0; i < s->kernel_size; i++) |
| dst[0] = s->kernel_size - 1; |
| } |
| |
| switch (outlink->format) { |
| case AV_SAMPLE_FMT_DBLP: |
| for (int ch = 0; ch < s->p->ch_layout.nb_channels; ch++) { |
| double *dst = (double *)s->p->extended_data[ch]; |
| |
| for (int i = 0; i < s->kernel_size; i++) |
| dst[i * s->kernel_size + i] = s->delta; |
| } |
| |
| s->filter_channels = filter_channels_double; |
| break; |
| case AV_SAMPLE_FMT_FLTP: |
| for (int ch = 0; ch < s->p->ch_layout.nb_channels; ch++) { |
| float *dst = (float *)s->p->extended_data[ch]; |
| |
| for (int i = 0; i < s->kernel_size; i++) |
| dst[i * s->kernel_size + i] = s->delta; |
| } |
| |
| s->filter_channels = filter_channels_float; |
| break; |
| } |
| |
| return 0; |
| } |
| |
| static av_cold int init(AVFilterContext *ctx) |
| { |
| AudioRLSContext *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) |
| { |
| AudioRLSContext *s = ctx->priv; |
| |
| av_freep(&s->fdsp); |
| av_frame_free(&s->delay); |
| av_frame_free(&s->coeffs); |
| av_frame_free(&s->gains); |
| av_frame_free(&s->offset); |
| av_frame_free(&s->p); |
| av_frame_free(&s->dp); |
| av_frame_free(&s->u); |
| 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_arls = { |
| .name = "arls", |
| .description = NULL_IF_CONFIG_SMALL("Apply Recursive Least Squares algorithm to first audio stream."), |
| .priv_size = sizeof(AudioRLSContext), |
| .priv_class = &arls_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, |
| }; |