FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/asrc_afirsrc.c
Date: 2024-02-16 17:37:06
Exec Total Coverage
Lines: 0 238 0.0%
Functions: 0 9 0.0%
Branches: 0 136 0.0%

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1 /*
2 * Copyright (c) 2020 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 License
8 * 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
14 * GNU Lesser General Public License for more details.
15 *
16 * You should have received a copy of the GNU Lesser General Public License
17 * along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
18 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19 */
20
21 #include "libavutil/cpu.h"
22 #include "libavutil/channel_layout.h"
23 #include "libavutil/ffmath.h"
24 #include "libavutil/eval.h"
25 #include "libavutil/opt.h"
26 #include "libavutil/tx.h"
27 #include "audio.h"
28 #include "avfilter.h"
29 #include "filters.h"
30 #include "formats.h"
31 #include "internal.h"
32 #include "window_func.h"
33
34 typedef struct AudioFIRSourceContext {
35 const AVClass *class;
36
37 char *freq_points_str;
38 char *magnitude_str;
39 char *phase_str;
40 int nb_taps;
41 int sample_rate;
42 int nb_samples;
43 int win_func;
44 int preset;
45 int interp;
46 int phaset;
47
48 AVComplexFloat *complexf;
49 float *freq;
50 float *magnitude;
51 float *phase;
52 int freq_size;
53 int magnitude_size;
54 int phase_size;
55 int nb_freq;
56 int nb_magnitude;
57 int nb_phase;
58
59 float *taps;
60 float *win;
61 int64_t pts;
62
63 AVTXContext *tx_ctx, *itx_ctx;
64 av_tx_fn tx_fn, itx_fn;
65 } AudioFIRSourceContext;
66
67 #define OFFSET(x) offsetof(AudioFIRSourceContext, x)
68 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
69
70 static const AVOption afirsrc_options[] = {
71 { "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS },
72 { "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS },
73 { "frequency", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },
74 { "f", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },
75 { "magnitude", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },
76 { "m", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },
77 { "phase", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },
78 { "p", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },
79 { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
80 { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
81 { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
82 { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
83 WIN_FUNC_OPTION("win_func", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),
84 WIN_FUNC_OPTION("w", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),
85 {NULL}
86 };
87
88 AVFILTER_DEFINE_CLASS(afirsrc);
89
90 static av_cold int init(AVFilterContext *ctx)
91 {
92 AudioFIRSourceContext *s = ctx->priv;
93
94 if (!(s->nb_taps & 1)) {
95 av_log(s, AV_LOG_WARNING, "Number of taps %d must be odd length.\n", s->nb_taps);
96 s->nb_taps |= 1;
97 }
98
99 return 0;
100 }
101
102 static av_cold void uninit(AVFilterContext *ctx)
103 {
104 AudioFIRSourceContext *s = ctx->priv;
105
106 av_freep(&s->win);
107 av_freep(&s->taps);
108 av_freep(&s->freq);
109 av_freep(&s->magnitude);
110 av_freep(&s->phase);
111 av_freep(&s->complexf);
112 av_tx_uninit(&s->tx_ctx);
113 av_tx_uninit(&s->itx_ctx);
114 }
115
116 static av_cold int query_formats(AVFilterContext *ctx)
117 {
118 AudioFIRSourceContext *s = ctx->priv;
119 static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } };
120 int sample_rates[] = { s->sample_rate, -1 };
121 static const enum AVSampleFormat sample_fmts[] = {
122 AV_SAMPLE_FMT_FLT,
123 AV_SAMPLE_FMT_NONE
124 };
125 int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
126 if (ret < 0)
127 return ret;
128
129 ret = ff_set_common_channel_layouts_from_list(ctx, chlayouts);
130 if (ret < 0)
131 return ret;
132
133 return ff_set_common_samplerates_from_list(ctx, sample_rates);
134 }
135
136 static int parse_string(char *str, float **items, int *nb_items, int *items_size)
137 {
138 float *new_items;
139 char *tail;
140
141 new_items = av_fast_realloc(NULL, items_size, sizeof(float));
142 if (!new_items)
143 return AVERROR(ENOMEM);
144 *items = new_items;
145
146 tail = str;
147 if (!tail)
148 return AVERROR(EINVAL);
149
150 do {
151 (*items)[(*nb_items)++] = av_strtod(tail, &tail);
152 new_items = av_fast_realloc(*items, items_size, (*nb_items + 2) * sizeof(float));
153 if (!new_items)
154 return AVERROR(ENOMEM);
155 *items = new_items;
156 if (tail && *tail)
157 tail++;
158 } while (tail && *tail);
159
160 return 0;
161 }
162
163 static void lininterp(AVComplexFloat *complexf,
164 const float *freq,
165 const float *magnitude,
166 const float *phase,
167 int m, int minterp)
168 {
169 for (int i = 0; i < minterp; i++) {
170 for (int j = 1; j < m; j++) {
171 const float x = i / (float)minterp;
172
173 if (x <= freq[j]) {
174 const float mg = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (magnitude[j] - magnitude[j-1]) + magnitude[j-1];
175 const float ph = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (phase[j] - phase[j-1]) + phase[j-1];
176
177 complexf[i].re = mg * cosf(ph);
178 complexf[i].im = mg * sinf(ph);
179 break;
180 }
181 }
182 }
183 }
184
185 static av_cold int config_output(AVFilterLink *outlink)
186 {
187 AVFilterContext *ctx = outlink->src;
188 AudioFIRSourceContext *s = ctx->priv;
189 float overlap, scale = 1.f, compensation;
190 int fft_size, middle, ret;
191
192 s->nb_freq = s->nb_magnitude = s->nb_phase = 0;
193
194 ret = parse_string(s->freq_points_str, &s->freq, &s->nb_freq, &s->freq_size);
195 if (ret < 0)
196 return ret;
197
198 ret = parse_string(s->magnitude_str, &s->magnitude, &s->nb_magnitude, &s->magnitude_size);
199 if (ret < 0)
200 return ret;
201
202 ret = parse_string(s->phase_str, &s->phase, &s->nb_phase, &s->phase_size);
203 if (ret < 0)
204 return ret;
205
206 if (s->nb_freq != s->nb_magnitude && s->nb_freq != s->nb_phase && s->nb_freq >= 2) {
207 av_log(ctx, AV_LOG_ERROR, "Number of frequencies, magnitudes and phases must be same and >= 2.\n");
208 return AVERROR(EINVAL);
209 }
210
211 for (int i = 0; i < s->nb_freq; i++) {
212 if (i == 0 && s->freq[i] != 0.f) {
213 av_log(ctx, AV_LOG_ERROR, "First frequency must be 0.\n");
214 return AVERROR(EINVAL);
215 }
216
217 if (i == s->nb_freq - 1 && s->freq[i] != 1.f) {
218 av_log(ctx, AV_LOG_ERROR, "Last frequency must be 1.\n");
219 return AVERROR(EINVAL);
220 }
221
222 if (i && s->freq[i] < s->freq[i-1]) {
223 av_log(ctx, AV_LOG_ERROR, "Frequencies must be in increasing order.\n");
224 return AVERROR(EINVAL);
225 }
226 }
227
228 fft_size = 1 << (av_log2(s->nb_taps) + 1);
229 s->complexf = av_calloc(fft_size * 2, sizeof(*s->complexf));
230 if (!s->complexf)
231 return AVERROR(ENOMEM);
232
233 ret = av_tx_init(&s->tx_ctx, &s->tx_fn, AV_TX_FLOAT_FFT, 1, fft_size, &scale, 0);
234 if (ret < 0)
235 return ret;
236
237 s->taps = av_calloc(s->nb_taps, sizeof(*s->taps));
238 if (!s->taps)
239 return AVERROR(ENOMEM);
240
241 s->win = av_calloc(s->nb_taps, sizeof(*s->win));
242 if (!s->win)
243 return AVERROR(ENOMEM);
244
245 generate_window_func(s->win, s->nb_taps, s->win_func, &overlap);
246
247 lininterp(s->complexf, s->freq, s->magnitude, s->phase, s->nb_freq, fft_size / 2);
248
249 s->tx_fn(s->tx_ctx, s->complexf + fft_size, s->complexf, sizeof(*s->complexf));
250
251 compensation = 2.f / fft_size;
252 middle = s->nb_taps / 2;
253
254 for (int i = 0; i <= middle; i++) {
255 s->taps[ i] = s->complexf[fft_size + middle - i].re * compensation * s->win[i];
256 s->taps[middle + i] = s->complexf[fft_size + i].re * compensation * s->win[middle + i];
257 }
258
259 s->pts = 0;
260
261 return 0;
262 }
263
264 static int activate(AVFilterContext *ctx)
265 {
266 AVFilterLink *outlink = ctx->outputs[0];
267 AudioFIRSourceContext *s = ctx->priv;
268 AVFrame *frame;
269 int nb_samples;
270
271 if (!ff_outlink_frame_wanted(outlink))
272 return FFERROR_NOT_READY;
273
274 nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts);
275 if (nb_samples <= 0) {
276 ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
277 return 0;
278 }
279
280 if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
281 return AVERROR(ENOMEM);
282
283 memcpy(frame->data[0], s->taps + s->pts, nb_samples * sizeof(float));
284
285 frame->pts = s->pts;
286 s->pts += nb_samples;
287 return ff_filter_frame(outlink, frame);
288 }
289
290 static const AVFilterPad afirsrc_outputs[] = {
291 {
292 .name = "default",
293 .type = AVMEDIA_TYPE_AUDIO,
294 .config_props = config_output,
295 },
296 };
297
298 const AVFilter ff_asrc_afirsrc = {
299 .name = "afirsrc",
300 .description = NULL_IF_CONFIG_SMALL("Generate a FIR coefficients audio stream."),
301 .init = init,
302 .uninit = uninit,
303 .activate = activate,
304 .priv_size = sizeof(AudioFIRSourceContext),
305 .inputs = NULL,
306 FILTER_OUTPUTS(afirsrc_outputs),
307 FILTER_QUERY_FUNC(query_formats),
308 .priv_class = &afirsrc_class,
309 };
310
311 #define DEFAULT_BANDS "25 40 63 100 160 250 400 630 1000 1600 2500 4000 6300 10000 16000 24000"
312
313 typedef struct EqPreset {
314 char name[16];
315 float gains[16];
316 } EqPreset;
317
318 static const EqPreset eq_presets[] = {
319 { "flat", { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } },
320 { "acoustic", { 5.0, 4.5, 4.0, 3.5, 1.5, 1.0, 1.5, 1.5, 2.0, 3.0, 3.5, 4.0, 3.7, 3.0, 3.0 } },
321 { "bass", { 10.0, 8.8, 8.5, 6.5, 2.5, 1.5, 0, 0, 0, 0, 0, 0, 0, 0, 0 } },
322 { "beats", { -5.5, -5.0, -4.5, -4.2, -3.5, -3.0, -1.9, 0, 0, 0, 0, 0, 0, 0, 0 } },
323 { "classic", { -0.3, 0.3, -3.5, -9.0, -1.0, 0.0, 1.8, 2.1, 0.0, 0.0, 0.0, 4.4, 9.0, 9.0, 9.0 } },
324 { "clear", { 3.5, 5.5, 6.5, 9.5, 8.0, 6.5, 3.5, 2.5, 1.3, 5.0, 7.0, 9.0, 10.0, 11.0, 9.0 } },
325 { "deep bass", { 12.0, 8.0, 0.0, -6.7, -12.0, -9.0, -3.5, -3.5, -6.1, 0.0, -3.0, -5.0, 0.0, 1.2, 3.0 } },
326 { "dubstep", { 12.0, 10.0, 0.5, -1.0, -3.0, -5.0, -5.0, -4.8, -4.5, -2.5, -1.0, 0.0, -2.5, -2.5, 0.0 } },
327 { "electronic", { 4.0, 4.0, 3.5, 1.0, 0.0, -0.5, -2.0, 0.0, 2.0, 0.0, 0.0, 1.0, 3.0, 4.0, 4.5 } },
328 { "hardstyle", { 6.1, 7.0, 12.0, 6.1, -5.0, -12.0, -2.5, 3.0, 6.5, 0.0, -2.2, -4.5, -6.1, -9.2, -10.0 } },
329 { "hip-hop", { 4.5, 4.3, 4.0, 2.5, 1.5, 3.0, -1.0, -1.5, -1.5, 1.5, 0.0, -1.0, 0.0, 1.5, 3.0 } },
330 { "jazz", { 0.0, 0.0, 0.0, 2.0, 4.0, 5.9, -5.9, -4.5, -2.5, 2.5, 1.0, -0.8, -0.8, -0.8, -0.8 } },
331 { "metal", { 10.5, 10.5, 7.5, 0.0, 2.0, 5.5, 0.0, 0.0, 0.0, 6.1, 0.0, 0.0, 6.1, 10.0, 12.0 } },
332 { "movie", { 3.0, 3.0, 6.1, 8.5, 9.0, 7.0, 6.1, 6.1, 5.0, 8.0, 3.5, 3.5, 8.0, 10.0, 8.0 } },
333 { "pop", { 0.0, 0.0, 0.0, 0.0, 0.0, 1.3, 2.0, 2.5, 5.0, -1.5, -2.0, -3.0, -3.0, -3.0, -3.0 } },
334 { "r&b", { 3.0, 3.0, 7.0, 6.1, 4.5, 1.5, -1.5, -2.0, -1.5, 2.0, 2.5, 3.0, 3.5, 3.8, 4.0 } },
335 { "rock", { 0.0, 0.0, 0.0, 3.0, 3.0, -10.0, -4.0, -1.0, 0.8, 3.0, 3.0, 3.0, 3.0, 3.0, 3.0 } },
336 { "vocal booster", { -1.5, -2.0, -3.0, -3.0, -0.5, 1.5, 3.5, 3.5, 3.5, 3.0, 2.0, 1.5, 0.0, 0.0, -1.5 } },
337 };
338
339 static const AVOption afireqsrc_options[] = {
340 { "preset","set equalizer preset", OFFSET(preset), AV_OPT_TYPE_INT, {.i64=0}, -1, FF_ARRAY_ELEMS(eq_presets)-1, FLAGS, .unit = "preset" },
341 { "p", "set equalizer preset", OFFSET(preset), AV_OPT_TYPE_INT, {.i64=0}, -1, FF_ARRAY_ELEMS(eq_presets)-1, FLAGS, .unit = "preset" },
342 { "custom", NULL, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, .unit = "preset" },
343 { eq_presets[ 0].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 0}, 0, 0, FLAGS, .unit = "preset" },
344 { eq_presets[ 1].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 1}, 0, 0, FLAGS, .unit = "preset" },
345 { eq_presets[ 2].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 2}, 0, 0, FLAGS, .unit = "preset" },
346 { eq_presets[ 3].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 3}, 0, 0, FLAGS, .unit = "preset" },
347 { eq_presets[ 4].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 4}, 0, 0, FLAGS, .unit = "preset" },
348 { eq_presets[ 5].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 5}, 0, 0, FLAGS, .unit = "preset" },
349 { eq_presets[ 6].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 6}, 0, 0, FLAGS, .unit = "preset" },
350 { eq_presets[ 7].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 7}, 0, 0, FLAGS, .unit = "preset" },
351 { eq_presets[ 8].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 8}, 0, 0, FLAGS, .unit = "preset" },
352 { eq_presets[ 9].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 9}, 0, 0, FLAGS, .unit = "preset" },
353 { eq_presets[10].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=10}, 0, 0, FLAGS, .unit = "preset" },
354 { eq_presets[11].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=11}, 0, 0, FLAGS, .unit = "preset" },
355 { eq_presets[12].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=12}, 0, 0, FLAGS, .unit = "preset" },
356 { eq_presets[13].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=13}, 0, 0, FLAGS, .unit = "preset" },
357 { eq_presets[14].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=14}, 0, 0, FLAGS, .unit = "preset" },
358 { eq_presets[15].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=15}, 0, 0, FLAGS, .unit = "preset" },
359 { eq_presets[16].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=16}, 0, 0, FLAGS, .unit = "preset" },
360 { eq_presets[17].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=17}, 0, 0, FLAGS, .unit = "preset" },
361 { "gains", "set gain values per band", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0"}, 0, 0, FLAGS },
362 { "g", "set gain values per band", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0"}, 0, 0, FLAGS },
363 { "bands", "set central frequency values per band", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str=DEFAULT_BANDS}, 0, 0, FLAGS },
364 { "b", "set central frequency values per band", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str=DEFAULT_BANDS}, 0, 0, FLAGS },
365 { "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=4096}, 16, UINT16_MAX, FLAGS },
366 { "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=4096}, 16, UINT16_MAX, FLAGS },
367 { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
368 { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
369 { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
370 { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
371 { "interp","set the interpolation", OFFSET(interp), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, .unit = "interp" },
372 { "i", "set the interpolation", OFFSET(interp), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, .unit = "interp" },
373 { "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "interp" },
374 { "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "interp" },
375 { "phase","set the phase", OFFSET(phaset), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS, .unit = "phase" },
376 { "h", "set the phase", OFFSET(phaset), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS, .unit = "phase" },
377 { "linear", "linear phase", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "phase" },
378 { "min", "minimum phase", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "phase" },
379 {NULL}
380 };
381
382 AVFILTER_DEFINE_CLASS(afireqsrc);
383
384 static void eq_interp(AVComplexFloat *complexf,
385 const float *freq,
386 const float *magnitude,
387 int m, int interp, int minterp,
388 const float factor)
389 {
390 for (int i = 0; i < minterp; i++) {
391 for (int j = 0; j < m; j++) {
392 const float x = factor * i;
393
394 if (x <= freq[j+1]) {
395 float g;
396
397 if (interp == 0) {
398 const float d = freq[j+1] - freq[j];
399 const float d0 = x - freq[j];
400 const float d1 = freq[j+1] - x;
401 const float g0 = magnitude[j];
402 const float g1 = magnitude[j+1];
403
404 if (d0 && d1) {
405 g = (d0 * g1 + d1 * g0) / d;
406 } else if (d0) {
407 g = g1;
408 } else {
409 g = g0;
410 }
411 } else {
412 if (x <= freq[j]) {
413 g = magnitude[j];
414 } else {
415 float x1, x2, x3;
416 float a, b, c, d;
417 float m0, m1, m2, msum;
418 const float unit = freq[j+1] - freq[j];
419
420 m0 = j != 0 ? unit * (magnitude[j] - magnitude[j-1]) / (freq[j] - freq[j-1]) : 0;
421 m1 = magnitude[j+1] - magnitude[j];
422 m2 = j != minterp - 1 ? unit * (magnitude[j+2] - magnitude[j+1]) / (freq[j+2] - freq[j+1]) : 0;
423
424 msum = fabsf(m0) + fabsf(m1);
425 m0 = msum > 0.f ? (fabsf(m0) * m1 + fabsf(m1) * m0) / msum : 0.f;
426 msum = fabsf(m1) + fabsf(m2);
427 m1 = msum > 0.f ? (fabsf(m1) * m2 + fabsf(m2) * m1) / msum : 0.f;
428
429 d = magnitude[j];
430 c = m0;
431 b = 3.f * magnitude[j+1] - m1 - 2.f * c - 3.f * d;
432 a = magnitude[j+1] - b - c - d;
433
434 x1 = (x - freq[j]) / unit;
435 x2 = x1 * x1;
436 x3 = x2 * x1;
437
438 g = a * x3 + b * x2 + c * x1 + d;
439 }
440 }
441
442 complexf[i].re = g;
443 complexf[i].im = 0;
444 complexf[minterp * 2 - i - 1].re = g;
445 complexf[minterp * 2 - i - 1].im = 0;
446
447 break;
448 }
449 }
450 }
451 }
452
453 static av_cold int config_eq_output(AVFilterLink *outlink)
454 {
455 AVFilterContext *ctx = outlink->src;
456 AudioFIRSourceContext *s = ctx->priv;
457 int fft_size, middle, asize, ret;
458 float scale, factor;
459
460 s->nb_freq = s->nb_magnitude = 0;
461 if (s->preset < 0) {
462 ret = parse_string(s->freq_points_str, &s->freq, &s->nb_freq, &s->freq_size);
463 if (ret < 0)
464 return ret;
465
466 ret = parse_string(s->magnitude_str, &s->magnitude, &s->nb_magnitude, &s->magnitude_size);
467 if (ret < 0)
468 return ret;
469 } else {
470 char *freq_str;
471
472 s->nb_magnitude = FF_ARRAY_ELEMS(eq_presets[s->preset].gains);
473
474 freq_str = av_strdup(DEFAULT_BANDS);
475 if (!freq_str)
476 return AVERROR(ENOMEM);
477
478 ret = parse_string(freq_str, &s->freq, &s->nb_freq, &s->freq_size);
479 av_free(freq_str);
480 if (ret < 0)
481 return ret;
482
483 s->magnitude = av_calloc(s->nb_magnitude + 1, sizeof(*s->magnitude));
484 if (!s->magnitude)
485 return AVERROR(ENOMEM);
486 memcpy(s->magnitude, eq_presets[s->preset].gains, sizeof(*s->magnitude) * s->nb_magnitude);
487 }
488
489 if (s->nb_freq != s->nb_magnitude || s->nb_freq < 2) {
490 av_log(ctx, AV_LOG_ERROR, "Number of bands and gains must be same and >= 2.\n");
491 return AVERROR(EINVAL);
492 }
493
494 s->freq[s->nb_freq] = outlink->sample_rate * 0.5f;
495 s->magnitude[s->nb_freq] = s->magnitude[s->nb_freq-1];
496
497 fft_size = s->nb_taps * 2;
498 factor = FFMIN(outlink->sample_rate * 0.5f, s->freq[s->nb_freq - 1]) / (float)fft_size;
499 asize = FFALIGN(fft_size, av_cpu_max_align());
500 s->complexf = av_calloc(asize * 2, sizeof(*s->complexf));
501 if (!s->complexf)
502 return AVERROR(ENOMEM);
503
504 scale = 1.f;
505 ret = av_tx_init(&s->itx_ctx, &s->itx_fn, AV_TX_FLOAT_FFT, 1, fft_size, &scale, 0);
506 if (ret < 0)
507 return ret;
508
509 s->taps = av_calloc(s->nb_taps, sizeof(*s->taps));
510 if (!s->taps)
511 return AVERROR(ENOMEM);
512
513 eq_interp(s->complexf, s->freq, s->magnitude, s->nb_freq, s->interp, s->nb_taps, factor);
514
515 for (int i = 0; i < fft_size; i++)
516 s->complexf[i].re = ff_exp10f(s->complexf[i].re / 20.f);
517
518 if (s->phaset) {
519 const float threshold = powf(10.f, -100.f / 20.f);
520 const float logt = logf(threshold);
521
522 scale = 1.f;
523 ret = av_tx_init(&s->tx_ctx, &s->tx_fn, AV_TX_FLOAT_FFT, 0, fft_size, &scale, 0);
524 if (ret < 0)
525 return ret;
526
527 for (int i = 0; i < fft_size; i++)
528 s->complexf[i].re = s->complexf[i].re < threshold ? logt : logf(s->complexf[i].re);
529
530 s->itx_fn(s->itx_ctx, s->complexf + asize, s->complexf, sizeof(float));
531 for (int i = 0; i < fft_size; i++) {
532 s->complexf[i + asize].re /= fft_size;
533 s->complexf[i + asize].im /= fft_size;
534 }
535
536 for (int i = 1; i < s->nb_taps; i++) {
537 s->complexf[asize + i].re += s->complexf[asize + fft_size - i].re;
538 s->complexf[asize + i].im -= s->complexf[asize + fft_size - i].im;
539 s->complexf[asize + fft_size - i].re = 0.f;
540 s->complexf[asize + fft_size - i].im = 0.f;
541 }
542 s->complexf[asize + s->nb_taps - 1].im *= -1.f;
543
544 s->tx_fn(s->tx_ctx, s->complexf, s->complexf + asize, sizeof(float));
545
546 for (int i = 0; i < fft_size; i++) {
547 float eR = expf(s->complexf[i].re);
548
549 s->complexf[i].re = eR * cosf(s->complexf[i].im);
550 s->complexf[i].im = eR * sinf(s->complexf[i].im);
551 }
552
553 s->itx_fn(s->itx_ctx, s->complexf + asize, s->complexf, sizeof(float));
554
555 for (int i = 0; i < s->nb_taps; i++)
556 s->taps[i] = s->complexf[i + asize].re / fft_size;
557 } else {
558 s->itx_fn(s->itx_ctx, s->complexf + asize, s->complexf, sizeof(float));
559
560 middle = s->nb_taps / 2;
561 for (int i = 0; i < middle; i++) {
562 s->taps[middle - i] = s->complexf[i + asize].re / fft_size;
563 s->taps[middle + i] = s->complexf[i + asize].re / fft_size;
564 }
565 }
566
567 s->pts = 0;
568
569 return 0;
570 }
571
572 static const AVFilterPad afireqsrc_outputs[] = {
573 {
574 .name = "default",
575 .type = AVMEDIA_TYPE_AUDIO,
576 .config_props = config_eq_output,
577 },
578 };
579
580 const AVFilter ff_asrc_afireqsrc = {
581 .name = "afireqsrc",
582 .description = NULL_IF_CONFIG_SMALL("Generate a FIR equalizer coefficients audio stream."),
583 .uninit = uninit,
584 .activate = activate,
585 .priv_size = sizeof(AudioFIRSourceContext),
586 .inputs = NULL,
587 FILTER_OUTPUTS(afireqsrc_outputs),
588 FILTER_QUERY_FUNC(query_formats),
589 .priv_class = &afireqsrc_class,
590 };
591