| Line | Branch | Exec | Source |
|---|---|---|---|
| 1 | /* | ||
| 2 | * Copyright (c) 2023 Paul B Mahol | ||
| 3 | * | ||
| 4 | * This file is part of FFmpeg. | ||
| 5 | * | ||
| 6 | * FFmpeg is free software; you can redistribute it and/or | ||
| 7 | * modify it under the terms of the GNU Lesser General Public | ||
| 8 | * License as published by the Free Software Foundation; either | ||
| 9 | * version 2.1 of the License, or (at your option) any later version. | ||
| 10 | * | ||
| 11 | * FFmpeg is distributed in the hope that it will be useful, | ||
| 12 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | ||
| 13 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | ||
| 14 | * Lesser General Public License for more details. | ||
| 15 | * | ||
| 16 | * You should have received a copy of the GNU Lesser General Public | ||
| 17 | * License along with FFmpeg; if not, write to the Free Software | ||
| 18 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | ||
| 19 | */ | ||
| 20 | |||
| 21 | #include "libavutil/channel_layout.h" | ||
| 22 | #include "libavutil/common.h" | ||
| 23 | #include "libavutil/float_dsp.h" | ||
| 24 | #include "libavutil/mem.h" | ||
| 25 | #include "libavutil/opt.h" | ||
| 26 | |||
| 27 | #include "audio.h" | ||
| 28 | #include "avfilter.h" | ||
| 29 | #include "formats.h" | ||
| 30 | #include "filters.h" | ||
| 31 | |||
| 32 | enum OutModes { | ||
| 33 | IN_MODE, | ||
| 34 | DESIRED_MODE, | ||
| 35 | OUT_MODE, | ||
| 36 | NOISE_MODE, | ||
| 37 | ERROR_MODE, | ||
| 38 | NB_OMODES | ||
| 39 | }; | ||
| 40 | |||
| 41 | typedef struct AudioAPContext { | ||
| 42 | const AVClass *class; | ||
| 43 | |||
| 44 | int order; | ||
| 45 | int projection; | ||
| 46 | float mu; | ||
| 47 | float delta; | ||
| 48 | int output_mode; | ||
| 49 | int precision; | ||
| 50 | |||
| 51 | int kernel_size; | ||
| 52 | AVFrame *offset; | ||
| 53 | AVFrame *delay; | ||
| 54 | AVFrame *coeffs; | ||
| 55 | AVFrame *e; | ||
| 56 | AVFrame *p; | ||
| 57 | AVFrame *x; | ||
| 58 | AVFrame *w; | ||
| 59 | AVFrame *dcoeffs; | ||
| 60 | AVFrame *tmp; | ||
| 61 | AVFrame *tmpm; | ||
| 62 | AVFrame *itmpm; | ||
| 63 | |||
| 64 | void **tmpmp; | ||
| 65 | void **itmpmp; | ||
| 66 | |||
| 67 | AVFrame *frame[2]; | ||
| 68 | |||
| 69 | int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs); | ||
| 70 | |||
| 71 | AVFloatDSPContext *fdsp; | ||
| 72 | } AudioAPContext; | ||
| 73 | |||
| 74 | #define OFFSET(x) offsetof(AudioAPContext, x) | ||
| 75 | #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM | ||
| 76 | #define AT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM | ||
| 77 | |||
| 78 | static const AVOption aap_options[] = { | ||
| 79 | { "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=16}, 1, INT16_MAX, A }, | ||
| 80 | { "projection", "set the filter projection", OFFSET(projection), AV_OPT_TYPE_INT, {.i64=2}, 1, 256, A }, | ||
| 81 | { "mu", "set the filter mu", OFFSET(mu), AV_OPT_TYPE_FLOAT, {.dbl=0.0001},0,1, AT }, | ||
| 82 | { "delta", "set the filter delta", OFFSET(delta), AV_OPT_TYPE_FLOAT, {.dbl=0.001},0, 1, AT }, | ||
| 83 | { "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, AT, .unit = "mode" }, | ||
| 84 | { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AT, .unit = "mode" }, | ||
| 85 | { "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, AT, .unit = "mode" }, | ||
| 86 | { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, .unit = "mode" }, | ||
| 87 | { "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, .unit = "mode" }, | ||
| 88 | { "e", "error", 0, AV_OPT_TYPE_CONST, {.i64=ERROR_MODE}, 0, 0, AT, .unit = "mode" }, | ||
| 89 | { "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, .unit = "precision" }, | ||
| 90 | { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "precision" }, | ||
| 91 | { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "precision" }, | ||
| 92 | { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, .unit = "precision" }, | ||
| 93 | { NULL } | ||
| 94 | }; | ||
| 95 | |||
| 96 | AVFILTER_DEFINE_CLASS(aap); | ||
| 97 | |||
| 98 | ✗ | static int query_formats(const AVFilterContext *ctx, | |
| 99 | AVFilterFormatsConfig **cfg_in, | ||
| 100 | AVFilterFormatsConfig **cfg_out) | ||
| 101 | { | ||
| 102 | ✗ | const AudioAPContext *s = ctx->priv; | |
| 103 | static const enum AVSampleFormat sample_fmts[3][3] = { | ||
| 104 | { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE }, | ||
| 105 | { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, | ||
| 106 | { AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE }, | ||
| 107 | }; | ||
| 108 | int ret; | ||
| 109 | |||
| 110 | ✗ | if ((ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, | |
| 111 | ✗ | sample_fmts[s->precision])) < 0) | |
| 112 | ✗ | return ret; | |
| 113 | |||
| 114 | ✗ | return 0; | |
| 115 | } | ||
| 116 | |||
| 117 | ✗ | static int activate(AVFilterContext *ctx) | |
| 118 | { | ||
| 119 | ✗ | AudioAPContext *s = ctx->priv; | |
| 120 | int i, ret, status; | ||
| 121 | int nb_samples; | ||
| 122 | int64_t pts; | ||
| 123 | |||
| 124 | ✗ | FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx); | |
| 125 | |||
| 126 | ✗ | nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]), | |
| 127 | ff_inlink_queued_samples(ctx->inputs[1])); | ||
| 128 | ✗ | for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) { | |
| 129 | ✗ | if (s->frame[i]) | |
| 130 | ✗ | continue; | |
| 131 | |||
| 132 | ✗ | if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) { | |
| 133 | ✗ | ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frame[i]); | |
| 134 | ✗ | if (ret < 0) | |
| 135 | ✗ | return ret; | |
| 136 | } | ||
| 137 | } | ||
| 138 | |||
| 139 | ✗ | if (s->frame[0] && s->frame[1]) { | |
| 140 | AVFrame *out; | ||
| 141 | |||
| 142 | ✗ | out = ff_get_audio_buffer(ctx->outputs[0], s->frame[0]->nb_samples); | |
| 143 | ✗ | if (!out) { | |
| 144 | ✗ | av_frame_free(&s->frame[0]); | |
| 145 | ✗ | av_frame_free(&s->frame[1]); | |
| 146 | ✗ | return AVERROR(ENOMEM); | |
| 147 | } | ||
| 148 | |||
| 149 | ✗ | ff_filter_execute(ctx, s->filter_channels, out, NULL, | |
| 150 | ✗ | FFMIN(ctx->outputs[0]->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx))); | |
| 151 | |||
| 152 | ✗ | out->pts = s->frame[0]->pts; | |
| 153 | ✗ | out->duration = s->frame[0]->duration; | |
| 154 | |||
| 155 | ✗ | av_frame_free(&s->frame[0]); | |
| 156 | ✗ | av_frame_free(&s->frame[1]); | |
| 157 | |||
| 158 | ✗ | ret = ff_filter_frame(ctx->outputs[0], out); | |
| 159 | ✗ | if (ret < 0) | |
| 160 | ✗ | return ret; | |
| 161 | } | ||
| 162 | |||
| 163 | ✗ | if (!nb_samples) { | |
| 164 | ✗ | for (i = 0; i < 2; i++) { | |
| 165 | ✗ | if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) { | |
| 166 | ✗ | ff_outlink_set_status(ctx->outputs[0], status, pts); | |
| 167 | ✗ | return 0; | |
| 168 | } | ||
| 169 | } | ||
| 170 | } | ||
| 171 | |||
| 172 | ✗ | if (ff_outlink_frame_wanted(ctx->outputs[0])) { | |
| 173 | ✗ | for (i = 0; i < 2; i++) { | |
| 174 | ✗ | if (s->frame[i] || ff_inlink_queued_samples(ctx->inputs[i]) > 0) | |
| 175 | ✗ | continue; | |
| 176 | ✗ | ff_inlink_request_frame(ctx->inputs[i]); | |
| 177 | ✗ | return 0; | |
| 178 | } | ||
| 179 | } | ||
| 180 | ✗ | return 0; | |
| 181 | } | ||
| 182 | |||
| 183 | #define DEPTH 32 | ||
| 184 | #include "aap_template.c" | ||
| 185 | |||
| 186 | #undef DEPTH | ||
| 187 | #define DEPTH 64 | ||
| 188 | #include "aap_template.c" | ||
| 189 | |||
| 190 | ✗ | static int config_output(AVFilterLink *outlink) | |
| 191 | { | ||
| 192 | ✗ | const int channels = outlink->ch_layout.nb_channels; | |
| 193 | ✗ | AVFilterContext *ctx = outlink->src; | |
| 194 | ✗ | AudioAPContext *s = ctx->priv; | |
| 195 | |||
| 196 | ✗ | s->kernel_size = FFALIGN(s->order, 16); | |
| 197 | |||
| 198 | ✗ | if (!s->offset) | |
| 199 | ✗ | s->offset = ff_get_audio_buffer(outlink, 3); | |
| 200 | ✗ | if (!s->delay) | |
| 201 | ✗ | s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size); | |
| 202 | ✗ | if (!s->dcoeffs) | |
| 203 | ✗ | s->dcoeffs = ff_get_audio_buffer(outlink, s->kernel_size); | |
| 204 | ✗ | if (!s->coeffs) | |
| 205 | ✗ | s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size); | |
| 206 | ✗ | if (!s->e) | |
| 207 | ✗ | s->e = ff_get_audio_buffer(outlink, 2 * s->projection); | |
| 208 | ✗ | if (!s->p) | |
| 209 | ✗ | s->p = ff_get_audio_buffer(outlink, s->projection + 1); | |
| 210 | ✗ | if (!s->x) | |
| 211 | ✗ | s->x = ff_get_audio_buffer(outlink, 2 * (s->projection + s->order)); | |
| 212 | ✗ | if (!s->w) | |
| 213 | ✗ | s->w = ff_get_audio_buffer(outlink, s->projection); | |
| 214 | ✗ | if (!s->tmp) | |
| 215 | ✗ | s->tmp = ff_get_audio_buffer(outlink, s->kernel_size); | |
| 216 | ✗ | if (!s->tmpm) | |
| 217 | ✗ | s->tmpm = ff_get_audio_buffer(outlink, s->projection * s->projection); | |
| 218 | ✗ | if (!s->itmpm) | |
| 219 | ✗ | s->itmpm = ff_get_audio_buffer(outlink, s->projection * s->projection); | |
| 220 | |||
| 221 | ✗ | if (!s->tmpmp) | |
| 222 | ✗ | s->tmpmp = av_calloc(s->projection * channels, sizeof(*s->tmpmp)); | |
| 223 | ✗ | if (!s->itmpmp) | |
| 224 | ✗ | s->itmpmp = av_calloc(s->projection * channels, sizeof(*s->itmpmp)); | |
| 225 | |||
| 226 | ✗ | if (!s->offset || !s->delay || !s->dcoeffs || !s->coeffs || !s->tmpmp || !s->itmpmp || | |
| 227 | ✗ | !s->e || !s->p || !s->x || !s->w || !s->tmp || !s->tmpm || !s->itmpm) | |
| 228 | ✗ | return AVERROR(ENOMEM); | |
| 229 | |||
| 230 | ✗ | switch (outlink->format) { | |
| 231 | ✗ | case AV_SAMPLE_FMT_DBLP: | |
| 232 | ✗ | for (int ch = 0; ch < channels; ch++) { | |
| 233 | ✗ | double *itmpm = (double *)s->itmpm->extended_data[ch]; | |
| 234 | ✗ | double *tmpm = (double *)s->tmpm->extended_data[ch]; | |
| 235 | ✗ | double **itmpmp = (double **)&s->itmpmp[s->projection * ch]; | |
| 236 | ✗ | double **tmpmp = (double **)&s->tmpmp[s->projection * ch]; | |
| 237 | |||
| 238 | ✗ | for (int i = 0; i < s->projection; i++) { | |
| 239 | ✗ | itmpmp[i] = &itmpm[i * s->projection]; | |
| 240 | ✗ | tmpmp[i] = &tmpm[i * s->projection]; | |
| 241 | } | ||
| 242 | } | ||
| 243 | |||
| 244 | ✗ | s->filter_channels = filter_channels_double; | |
| 245 | ✗ | break; | |
| 246 | ✗ | case AV_SAMPLE_FMT_FLTP: | |
| 247 | ✗ | for (int ch = 0; ch < channels; ch++) { | |
| 248 | ✗ | float *itmpm = (float *)s->itmpm->extended_data[ch]; | |
| 249 | ✗ | float *tmpm = (float *)s->tmpm->extended_data[ch]; | |
| 250 | ✗ | float **itmpmp = (float **)&s->itmpmp[s->projection * ch]; | |
| 251 | ✗ | float **tmpmp = (float **)&s->tmpmp[s->projection * ch]; | |
| 252 | |||
| 253 | ✗ | for (int i = 0; i < s->projection; i++) { | |
| 254 | ✗ | itmpmp[i] = &itmpm[i * s->projection]; | |
| 255 | ✗ | tmpmp[i] = &tmpm[i * s->projection]; | |
| 256 | } | ||
| 257 | } | ||
| 258 | |||
| 259 | ✗ | s->filter_channels = filter_channels_float; | |
| 260 | ✗ | break; | |
| 261 | } | ||
| 262 | |||
| 263 | ✗ | return 0; | |
| 264 | } | ||
| 265 | |||
| 266 | ✗ | static av_cold int init(AVFilterContext *ctx) | |
| 267 | { | ||
| 268 | ✗ | AudioAPContext *s = ctx->priv; | |
| 269 | |||
| 270 | ✗ | s->fdsp = avpriv_float_dsp_alloc(0); | |
| 271 | ✗ | if (!s->fdsp) | |
| 272 | ✗ | return AVERROR(ENOMEM); | |
| 273 | |||
| 274 | ✗ | return 0; | |
| 275 | } | ||
| 276 | |||
| 277 | ✗ | static av_cold void uninit(AVFilterContext *ctx) | |
| 278 | { | ||
| 279 | ✗ | AudioAPContext *s = ctx->priv; | |
| 280 | |||
| 281 | ✗ | av_freep(&s->fdsp); | |
| 282 | |||
| 283 | ✗ | av_frame_free(&s->offset); | |
| 284 | ✗ | av_frame_free(&s->delay); | |
| 285 | ✗ | av_frame_free(&s->dcoeffs); | |
| 286 | ✗ | av_frame_free(&s->coeffs); | |
| 287 | ✗ | av_frame_free(&s->e); | |
| 288 | ✗ | av_frame_free(&s->p); | |
| 289 | ✗ | av_frame_free(&s->w); | |
| 290 | ✗ | av_frame_free(&s->x); | |
| 291 | ✗ | av_frame_free(&s->tmp); | |
| 292 | ✗ | av_frame_free(&s->tmpm); | |
| 293 | ✗ | av_frame_free(&s->itmpm); | |
| 294 | |||
| 295 | ✗ | av_freep(&s->tmpmp); | |
| 296 | ✗ | av_freep(&s->itmpmp); | |
| 297 | ✗ | } | |
| 298 | |||
| 299 | static const AVFilterPad inputs[] = { | ||
| 300 | { | ||
| 301 | .name = "input", | ||
| 302 | .type = AVMEDIA_TYPE_AUDIO, | ||
| 303 | }, | ||
| 304 | { | ||
| 305 | .name = "desired", | ||
| 306 | .type = AVMEDIA_TYPE_AUDIO, | ||
| 307 | }, | ||
| 308 | }; | ||
| 309 | |||
| 310 | static const AVFilterPad outputs[] = { | ||
| 311 | { | ||
| 312 | .name = "default", | ||
| 313 | .type = AVMEDIA_TYPE_AUDIO, | ||
| 314 | .config_props = config_output, | ||
| 315 | }, | ||
| 316 | }; | ||
| 317 | |||
| 318 | const FFFilter ff_af_aap = { | ||
| 319 | .p.name = "aap", | ||
| 320 | .p.description = NULL_IF_CONFIG_SMALL("Apply Affine Projection algorithm to first audio stream."), | ||
| 321 | .p.priv_class = &aap_class, | ||
| 322 | .p.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL | | ||
| 323 | AVFILTER_FLAG_SLICE_THREADS, | ||
| 324 | .priv_size = sizeof(AudioAPContext), | ||
| 325 | .init = init, | ||
| 326 | .uninit = uninit, | ||
| 327 | .activate = activate, | ||
| 328 | FILTER_INPUTS(inputs), | ||
| 329 | FILTER_OUTPUTS(outputs), | ||
| 330 | FILTER_QUERY_FUNC2(query_formats), | ||
| 331 | .process_command = ff_filter_process_command, | ||
| 332 | }; | ||
| 333 |