diff options
author | Paul B Mahol <onemda@gmail.com> | 2023-04-30 17:06:00 +0200 |
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committer | Paul B Mahol <onemda@gmail.com> | 2023-11-28 15:40:34 +0100 |
commit | f66536cc5816a4a3e9af67de26d41ea581505e30 (patch) | |
tree | 4c6a286fbb4a5410296dc57d66747926a48d5603 /libavfilter/af_aap.c | |
parent | cc86343b960793a822d6c51b58a1a7e3319cb217 (diff) |
avfilter: add Affine Projection adaptive audio filter
Diffstat (limited to 'libavfilter/af_aap.c')
-rw-r--r-- | libavfilter/af_aap.c | 332 |
1 files changed, 332 insertions, 0 deletions
diff --git a/libavfilter/af_aap.c b/libavfilter/af_aap.c new file mode 100644 index 0000000000..96c8d27af4 --- /dev/null +++ b/libavfilter/af_aap.c @@ -0,0 +1,332 @@ +/* + * 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/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, + ERROR_MODE, + NB_OMODES +}; + +typedef struct AudioAPContext { + const AVClass *class; + + int order; + int projection; + float mu; + float delta; + int output_mode; + int precision; + + int kernel_size; + AVFrame *offset; + AVFrame *delay; + AVFrame *coeffs; + AVFrame *e; + AVFrame *p; + AVFrame *x; + AVFrame *w; + AVFrame *dcoeffs; + AVFrame *tmp; + AVFrame *tmpm; + AVFrame *itmpm; + + void **tmpmp; + void **itmpmp; + + AVFrame *frame[2]; + + int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs); + + AVFloatDSPContext *fdsp; +} AudioAPContext; + +#define OFFSET(x) offsetof(AudioAPContext, 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 aap_options[] = { + { "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=16}, 1, INT16_MAX, A }, + { "projection", "set the filter projection", OFFSET(projection), AV_OPT_TYPE_INT, {.i64=2}, 1, 256, A }, + { "mu", "set the filter mu", OFFSET(mu), AV_OPT_TYPE_FLOAT, {.dbl=0.0001},0,1, AT }, + { "delta", "set the filter delta", OFFSET(delta), AV_OPT_TYPE_FLOAT, {.dbl=0.001},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" }, + { "e", "error", 0, AV_OPT_TYPE_CONST, {.i64=ERROR_MODE}, 0, 0, AT, "mode" }, + { "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "precision" }, + { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "precision" }, + { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "precision" }, + { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "precision" }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(aap); + +static int query_formats(AVFilterContext *ctx) +{ + AudioAPContext *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) +{ + AudioAPContext *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 "aap_template.c" + +#undef DEPTH +#define DEPTH 64 +#include "aap_template.c" + +static int config_output(AVFilterLink *outlink) +{ + const int channels = outlink->ch_layout.nb_channels; + AVFilterContext *ctx = outlink->src; + AudioAPContext *s = ctx->priv; + + s->kernel_size = FFALIGN(s->order, 16); + + if (!s->offset) + s->offset = ff_get_audio_buffer(outlink, 3); + if (!s->delay) + s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size); + if (!s->dcoeffs) + s->dcoeffs = ff_get_audio_buffer(outlink, s->kernel_size); + if (!s->coeffs) + s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size); + if (!s->e) + s->e = ff_get_audio_buffer(outlink, 2 * s->projection); + if (!s->p) + s->p = ff_get_audio_buffer(outlink, s->projection + 1); + if (!s->x) + s->x = ff_get_audio_buffer(outlink, 2 * (s->projection + s->order)); + if (!s->w) + s->w = ff_get_audio_buffer(outlink, s->projection); + if (!s->tmp) + s->tmp = ff_get_audio_buffer(outlink, s->kernel_size); + if (!s->tmpm) + s->tmpm = ff_get_audio_buffer(outlink, s->projection * s->projection); + if (!s->itmpm) + s->itmpm = ff_get_audio_buffer(outlink, s->projection * s->projection); + + if (!s->tmpmp) + s->tmpmp = av_calloc(s->projection * channels, sizeof(*s->tmpmp)); + if (!s->itmpmp) + s->itmpmp = av_calloc(s->projection * channels, sizeof(*s->itmpmp)); + + if (!s->offset || !s->delay || !s->dcoeffs || !s->coeffs || !s->tmpmp || !s->itmpmp || + !s->e || !s->p || !s->x || !s->w || !s->tmp || !s->tmpm || !s->itmpm) + return AVERROR(ENOMEM); + + switch (outlink->format) { + case AV_SAMPLE_FMT_DBLP: + for (int ch = 0; ch < channels; ch++) { + double *itmpm = (double *)s->itmpm->extended_data[ch]; + double *tmpm = (double *)s->tmpm->extended_data[ch]; + double **itmpmp = (double **)&s->itmpmp[s->projection * ch]; + double **tmpmp = (double **)&s->tmpmp[s->projection * ch]; + + for (int i = 0; i < s->projection; i++) { + itmpmp[i] = &itmpm[i * s->projection]; + tmpmp[i] = &tmpm[i * s->projection]; + } + } + + s->filter_channels = filter_channels_double; + break; + case AV_SAMPLE_FMT_FLTP: + for (int ch = 0; ch < channels; ch++) { + float *itmpm = (float *)s->itmpm->extended_data[ch]; + float *tmpm = (float *)s->tmpm->extended_data[ch]; + float **itmpmp = (float **)&s->itmpmp[s->projection * ch]; + float **tmpmp = (float **)&s->tmpmp[s->projection * ch]; + + for (int i = 0; i < s->projection; i++) { + itmpmp[i] = &itmpm[i * s->projection]; + tmpmp[i] = &tmpm[i * s->projection]; + } + } + + s->filter_channels = filter_channels_float; + break; + } + + return 0; +} + +static av_cold int init(AVFilterContext *ctx) +{ + AudioAPContext *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) +{ + AudioAPContext *s = ctx->priv; + + av_freep(&s->fdsp); + + av_frame_free(&s->offset); + av_frame_free(&s->delay); + av_frame_free(&s->dcoeffs); + av_frame_free(&s->coeffs); + av_frame_free(&s->e); + av_frame_free(&s->p); + av_frame_free(&s->w); + av_frame_free(&s->x); + av_frame_free(&s->tmp); + av_frame_free(&s->tmpm); + av_frame_free(&s->itmpm); + + av_freep(&s->tmpmp); + av_freep(&s->itmpmp); +} + +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_aap = { + .name = "aap", + .description = NULL_IF_CONFIG_SMALL("Apply Affine Projection algorithm to first audio stream."), + .priv_size = sizeof(AudioAPContext), + .priv_class = &aap_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, +}; 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