FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/asrc_afirsrc.c
Date: 2024-07-24 19:24:46
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/mem.h"
26 #include "libavutil/opt.h"
27 #include "libavutil/tx.h"
28 #include "audio.h"
29 #include "avfilter.h"
30 #include "filters.h"
31 #include "formats.h"
32 #include "internal.h"
33 #include "window_func.h"
34
35 typedef struct AudioFIRSourceContext {
36 const AVClass *class;
37
38 char *freq_points_str;
39 char *magnitude_str;
40 char *phase_str;
41 int nb_taps;
42 int sample_rate;
43 int nb_samples;
44 int win_func;
45 int preset;
46 int interp;
47 int phaset;
48
49 AVComplexFloat *complexf;
50 float *freq;
51 float *magnitude;
52 float *phase;
53 int freq_size;
54 int magnitude_size;
55 int phase_size;
56 int nb_freq;
57 int nb_magnitude;
58 int nb_phase;
59
60 float *taps;
61 float *win;
62 int64_t pts;
63
64 AVTXContext *tx_ctx, *itx_ctx;
65 av_tx_fn tx_fn, itx_fn;
66 } AudioFIRSourceContext;
67
68 #define OFFSET(x) offsetof(AudioFIRSourceContext, x)
69 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
70
71 static const AVOption afirsrc_options[] = {
72 { "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS },
73 { "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS },
74 { "frequency", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },
75 { "f", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },
76 { "magnitude", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },
77 { "m", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },
78 { "phase", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },
79 { "p", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },
80 { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
81 { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
82 { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
83 { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
84 WIN_FUNC_OPTION("win_func", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),
85 WIN_FUNC_OPTION("w", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN),
86 {NULL}
87 };
88
89 AVFILTER_DEFINE_CLASS(afirsrc);
90
91 static av_cold int init(AVFilterContext *ctx)
92 {
93 AudioFIRSourceContext *s = ctx->priv;
94
95 if (!(s->nb_taps & 1)) {
96 av_log(s, AV_LOG_WARNING, "Number of taps %d must be odd length.\n", s->nb_taps);
97 s->nb_taps |= 1;
98 }
99
100 return 0;
101 }
102
103 static av_cold void uninit(AVFilterContext *ctx)
104 {
105 AudioFIRSourceContext *s = ctx->priv;
106
107 av_freep(&s->win);
108 av_freep(&s->taps);
109 av_freep(&s->freq);
110 av_freep(&s->magnitude);
111 av_freep(&s->phase);
112 av_freep(&s->complexf);
113 av_tx_uninit(&s->tx_ctx);
114 av_tx_uninit(&s->itx_ctx);
115 }
116
117 static av_cold int query_formats(AVFilterContext *ctx)
118 {
119 AudioFIRSourceContext *s = ctx->priv;
120 static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } };
121 int sample_rates[] = { s->sample_rate, -1 };
122 static const enum AVSampleFormat sample_fmts[] = {
123 AV_SAMPLE_FMT_FLT,
124 AV_SAMPLE_FMT_NONE
125 };
126 int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
127 if (ret < 0)
128 return ret;
129
130 ret = ff_set_common_channel_layouts_from_list(ctx, chlayouts);
131 if (ret < 0)
132 return ret;
133
134 return ff_set_common_samplerates_from_list(ctx, sample_rates);
135 }
136
137 static int parse_string(char *str, float **items, int *nb_items, int *items_size)
138 {
139 float *new_items;
140 char *tail;
141
142 new_items = av_fast_realloc(NULL, items_size, sizeof(float));
143 if (!new_items)
144 return AVERROR(ENOMEM);
145 *items = new_items;
146
147 tail = str;
148 if (!tail)
149 return AVERROR(EINVAL);
150
151 do {
152 (*items)[(*nb_items)++] = av_strtod(tail, &tail);
153 new_items = av_fast_realloc(*items, items_size, (*nb_items + 2) * sizeof(float));
154 if (!new_items)
155 return AVERROR(ENOMEM);
156 *items = new_items;
157 if (tail && *tail)
158 tail++;
159 } while (tail && *tail);
160
161 return 0;
162 }
163
164 static void lininterp(AVComplexFloat *complexf,
165 const float *freq,
166 const float *magnitude,
167 const float *phase,
168 int m, int minterp)
169 {
170 for (int i = 0; i < minterp; i++) {
171 for (int j = 1; j < m; j++) {
172 const float x = i / (float)minterp;
173
174 if (x <= freq[j]) {
175 const float mg = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (magnitude[j] - magnitude[j-1]) + magnitude[j-1];
176 const float ph = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (phase[j] - phase[j-1]) + phase[j-1];
177
178 complexf[i].re = mg * cosf(ph);
179 complexf[i].im = mg * sinf(ph);
180 break;
181 }
182 }
183 }
184 }
185
186 static av_cold int config_output(AVFilterLink *outlink)
187 {
188 AVFilterContext *ctx = outlink->src;
189 AudioFIRSourceContext *s = ctx->priv;
190 float overlap, scale = 1.f, compensation;
191 int fft_size, middle, ret;
192
193 s->nb_freq = s->nb_magnitude = s->nb_phase = 0;
194
195 ret = parse_string(s->freq_points_str, &s->freq, &s->nb_freq, &s->freq_size);
196 if (ret < 0)
197 return ret;
198
199 ret = parse_string(s->magnitude_str, &s->magnitude, &s->nb_magnitude, &s->magnitude_size);
200 if (ret < 0)
201 return ret;
202
203 ret = parse_string(s->phase_str, &s->phase, &s->nb_phase, &s->phase_size);
204 if (ret < 0)
205 return ret;
206
207 if (s->nb_freq != s->nb_magnitude && s->nb_freq != s->nb_phase && s->nb_freq >= 2) {
208 av_log(ctx, AV_LOG_ERROR, "Number of frequencies, magnitudes and phases must be same and >= 2.\n");
209 return AVERROR(EINVAL);
210 }
211
212 for (int i = 0; i < s->nb_freq; i++) {
213 if (i == 0 && s->freq[i] != 0.f) {
214 av_log(ctx, AV_LOG_ERROR, "First frequency must be 0.\n");
215 return AVERROR(EINVAL);
216 }
217
218 if (i == s->nb_freq - 1 && s->freq[i] != 1.f) {
219 av_log(ctx, AV_LOG_ERROR, "Last frequency must be 1.\n");
220 return AVERROR(EINVAL);
221 }
222
223 if (i && s->freq[i] < s->freq[i-1]) {
224 av_log(ctx, AV_LOG_ERROR, "Frequencies must be in increasing order.\n");
225 return AVERROR(EINVAL);
226 }
227 }
228
229 fft_size = 1 << (av_log2(s->nb_taps) + 1);
230 s->complexf = av_calloc(fft_size * 2, sizeof(*s->complexf));
231 if (!s->complexf)
232 return AVERROR(ENOMEM);
233
234 ret = av_tx_init(&s->tx_ctx, &s->tx_fn, AV_TX_FLOAT_FFT, 1, fft_size, &scale, 0);
235 if (ret < 0)
236 return ret;
237
238 s->taps = av_calloc(s->nb_taps, sizeof(*s->taps));
239 if (!s->taps)
240 return AVERROR(ENOMEM);
241
242 s->win = av_calloc(s->nb_taps, sizeof(*s->win));
243 if (!s->win)
244 return AVERROR(ENOMEM);
245
246 generate_window_func(s->win, s->nb_taps, s->win_func, &overlap);
247
248 lininterp(s->complexf, s->freq, s->magnitude, s->phase, s->nb_freq, fft_size / 2);
249
250 s->tx_fn(s->tx_ctx, s->complexf + fft_size, s->complexf, sizeof(*s->complexf));
251
252 compensation = 2.f / fft_size;
253 middle = s->nb_taps / 2;
254
255 for (int i = 0; i <= middle; i++) {
256 s->taps[ i] = s->complexf[fft_size + middle - i].re * compensation * s->win[i];
257 s->taps[middle + i] = s->complexf[fft_size + i].re * compensation * s->win[middle + i];
258 }
259
260 s->pts = 0;
261
262 return 0;
263 }
264
265 static int activate(AVFilterContext *ctx)
266 {
267 AVFilterLink *outlink = ctx->outputs[0];
268 AudioFIRSourceContext *s = ctx->priv;
269 AVFrame *frame;
270 int nb_samples;
271
272 if (!ff_outlink_frame_wanted(outlink))
273 return FFERROR_NOT_READY;
274
275 nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts);
276 if (nb_samples <= 0) {
277 ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
278 return 0;
279 }
280
281 if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
282 return AVERROR(ENOMEM);
283
284 memcpy(frame->data[0], s->taps + s->pts, nb_samples * sizeof(float));
285
286 frame->pts = s->pts;
287 s->pts += nb_samples;
288 return ff_filter_frame(outlink, frame);
289 }
290
291 static const AVFilterPad afirsrc_outputs[] = {
292 {
293 .name = "default",
294 .type = AVMEDIA_TYPE_AUDIO,
295 .config_props = config_output,
296 },
297 };
298
299 const AVFilter ff_asrc_afirsrc = {
300 .name = "afirsrc",
301 .description = NULL_IF_CONFIG_SMALL("Generate a FIR coefficients audio stream."),
302 .init = init,
303 .uninit = uninit,
304 .activate = activate,
305 .priv_size = sizeof(AudioFIRSourceContext),
306 .inputs = NULL,
307 FILTER_OUTPUTS(afirsrc_outputs),
308 FILTER_QUERY_FUNC(query_formats),
309 .priv_class = &afirsrc_class,
310 };
311
312 #define DEFAULT_BANDS "25 40 63 100 160 250 400 630 1000 1600 2500 4000 6300 10000 16000 24000"
313
314 typedef struct EqPreset {
315 char name[16];
316 float gains[16];
317 } EqPreset;
318
319 static const EqPreset eq_presets[] = {
320 { "flat", { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } },
321 { "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 } },
322 { "bass", { 10.0, 8.8, 8.5, 6.5, 2.5, 1.5, 0, 0, 0, 0, 0, 0, 0, 0, 0 } },
323 { "beats", { -5.5, -5.0, -4.5, -4.2, -3.5, -3.0, -1.9, 0, 0, 0, 0, 0, 0, 0, 0 } },
324 { "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 } },
325 { "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 } },
326 { "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 } },
327 { "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 } },
328 { "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 } },
329 { "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 } },
330 { "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 } },
331 { "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 } },
332 { "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 } },
333 { "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 } },
334 { "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 } },
335 { "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 } },
336 { "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 } },
337 { "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 } },
338 };
339
340 static const AVOption afireqsrc_options[] = {
341 { "preset","set equalizer preset", OFFSET(preset), AV_OPT_TYPE_INT, {.i64=0}, -1, FF_ARRAY_ELEMS(eq_presets)-1, FLAGS, .unit = "preset" },
342 { "p", "set equalizer preset", OFFSET(preset), AV_OPT_TYPE_INT, {.i64=0}, -1, FF_ARRAY_ELEMS(eq_presets)-1, FLAGS, .unit = "preset" },
343 { "custom", NULL, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, .unit = "preset" },
344 { eq_presets[ 0].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 0}, 0, 0, FLAGS, .unit = "preset" },
345 { eq_presets[ 1].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 1}, 0, 0, FLAGS, .unit = "preset" },
346 { eq_presets[ 2].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 2}, 0, 0, FLAGS, .unit = "preset" },
347 { eq_presets[ 3].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 3}, 0, 0, FLAGS, .unit = "preset" },
348 { eq_presets[ 4].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 4}, 0, 0, FLAGS, .unit = "preset" },
349 { eq_presets[ 5].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 5}, 0, 0, FLAGS, .unit = "preset" },
350 { eq_presets[ 6].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 6}, 0, 0, FLAGS, .unit = "preset" },
351 { eq_presets[ 7].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 7}, 0, 0, FLAGS, .unit = "preset" },
352 { eq_presets[ 8].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 8}, 0, 0, FLAGS, .unit = "preset" },
353 { eq_presets[ 9].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 9}, 0, 0, FLAGS, .unit = "preset" },
354 { eq_presets[10].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=10}, 0, 0, FLAGS, .unit = "preset" },
355 { eq_presets[11].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=11}, 0, 0, FLAGS, .unit = "preset" },
356 { eq_presets[12].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=12}, 0, 0, FLAGS, .unit = "preset" },
357 { eq_presets[13].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=13}, 0, 0, FLAGS, .unit = "preset" },
358 { eq_presets[14].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=14}, 0, 0, FLAGS, .unit = "preset" },
359 { eq_presets[15].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=15}, 0, 0, FLAGS, .unit = "preset" },
360 { eq_presets[16].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=16}, 0, 0, FLAGS, .unit = "preset" },
361 { eq_presets[17].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=17}, 0, 0, FLAGS, .unit = "preset" },
362 { "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 },
363 { "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 },
364 { "bands", "set central frequency values per band", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str=DEFAULT_BANDS}, 0, 0, FLAGS },
365 { "b", "set central frequency values per band", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str=DEFAULT_BANDS}, 0, 0, FLAGS },
366 { "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=4096}, 16, UINT16_MAX, FLAGS },
367 { "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=4096}, 16, UINT16_MAX, FLAGS },
368 { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
369 { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
370 { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
371 { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
372 { "interp","set the interpolation", OFFSET(interp), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, .unit = "interp" },
373 { "i", "set the interpolation", OFFSET(interp), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, .unit = "interp" },
374 { "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "interp" },
375 { "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "interp" },
376 { "phase","set the phase", OFFSET(phaset), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS, .unit = "phase" },
377 { "h", "set the phase", OFFSET(phaset), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS, .unit = "phase" },
378 { "linear", "linear phase", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "phase" },
379 { "min", "minimum phase", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "phase" },
380 {NULL}
381 };
382
383 AVFILTER_DEFINE_CLASS(afireqsrc);
384
385 static void eq_interp(AVComplexFloat *complexf,
386 const float *freq,
387 const float *magnitude,
388 int m, int interp, int minterp,
389 const float factor)
390 {
391 for (int i = 0; i < minterp; i++) {
392 for (int j = 0; j < m; j++) {
393 const float x = factor * i;
394
395 if (x <= freq[j+1]) {
396 float g;
397
398 if (interp == 0) {
399 const float d = freq[j+1] - freq[j];
400 const float d0 = x - freq[j];
401 const float d1 = freq[j+1] - x;
402 const float g0 = magnitude[j];
403 const float g1 = magnitude[j+1];
404
405 if (d0 && d1) {
406 g = (d0 * g1 + d1 * g0) / d;
407 } else if (d0) {
408 g = g1;
409 } else {
410 g = g0;
411 }
412 } else {
413 if (x <= freq[j]) {
414 g = magnitude[j];
415 } else {
416 float x1, x2, x3;
417 float a, b, c, d;
418 float m0, m1, m2, msum;
419 const float unit = freq[j+1] - freq[j];
420
421 m0 = j != 0 ? unit * (magnitude[j] - magnitude[j-1]) / (freq[j] - freq[j-1]) : 0;
422 m1 = magnitude[j+1] - magnitude[j];
423 m2 = j != minterp - 1 ? unit * (magnitude[j+2] - magnitude[j+1]) / (freq[j+2] - freq[j+1]) : 0;
424
425 msum = fabsf(m0) + fabsf(m1);
426 m0 = msum > 0.f ? (fabsf(m0) * m1 + fabsf(m1) * m0) / msum : 0.f;
427 msum = fabsf(m1) + fabsf(m2);
428 m1 = msum > 0.f ? (fabsf(m1) * m2 + fabsf(m2) * m1) / msum : 0.f;
429
430 d = magnitude[j];
431 c = m0;
432 b = 3.f * magnitude[j+1] - m1 - 2.f * c - 3.f * d;
433 a = magnitude[j+1] - b - c - d;
434
435 x1 = (x - freq[j]) / unit;
436 x2 = x1 * x1;
437 x3 = x2 * x1;
438
439 g = a * x3 + b * x2 + c * x1 + d;
440 }
441 }
442
443 complexf[i].re = g;
444 complexf[i].im = 0;
445 complexf[minterp * 2 - i - 1].re = g;
446 complexf[minterp * 2 - i - 1].im = 0;
447
448 break;
449 }
450 }
451 }
452 }
453
454 static av_cold int config_eq_output(AVFilterLink *outlink)
455 {
456 AVFilterContext *ctx = outlink->src;
457 AudioFIRSourceContext *s = ctx->priv;
458 int fft_size, middle, asize, ret;
459 float scale, factor;
460
461 s->nb_freq = s->nb_magnitude = 0;
462 if (s->preset < 0) {
463 ret = parse_string(s->freq_points_str, &s->freq, &s->nb_freq, &s->freq_size);
464 if (ret < 0)
465 return ret;
466
467 ret = parse_string(s->magnitude_str, &s->magnitude, &s->nb_magnitude, &s->magnitude_size);
468 if (ret < 0)
469 return ret;
470 } else {
471 char *freq_str;
472
473 s->nb_magnitude = FF_ARRAY_ELEMS(eq_presets[s->preset].gains);
474
475 freq_str = av_strdup(DEFAULT_BANDS);
476 if (!freq_str)
477 return AVERROR(ENOMEM);
478
479 ret = parse_string(freq_str, &s->freq, &s->nb_freq, &s->freq_size);
480 av_free(freq_str);
481 if (ret < 0)
482 return ret;
483
484 s->magnitude = av_calloc(s->nb_magnitude + 1, sizeof(*s->magnitude));
485 if (!s->magnitude)
486 return AVERROR(ENOMEM);
487 memcpy(s->magnitude, eq_presets[s->preset].gains, sizeof(*s->magnitude) * s->nb_magnitude);
488 }
489
490 if (s->nb_freq != s->nb_magnitude || s->nb_freq < 2) {
491 av_log(ctx, AV_LOG_ERROR, "Number of bands and gains must be same and >= 2.\n");
492 return AVERROR(EINVAL);
493 }
494
495 s->freq[s->nb_freq] = outlink->sample_rate * 0.5f;
496 s->magnitude[s->nb_freq] = s->magnitude[s->nb_freq-1];
497
498 fft_size = s->nb_taps * 2;
499 factor = FFMIN(outlink->sample_rate * 0.5f, s->freq[s->nb_freq - 1]) / (float)fft_size;
500 asize = FFALIGN(fft_size, av_cpu_max_align());
501 s->complexf = av_calloc(asize * 2, sizeof(*s->complexf));
502 if (!s->complexf)
503 return AVERROR(ENOMEM);
504
505 scale = 1.f;
506 ret = av_tx_init(&s->itx_ctx, &s->itx_fn, AV_TX_FLOAT_FFT, 1, fft_size, &scale, 0);
507 if (ret < 0)
508 return ret;
509
510 s->taps = av_calloc(s->nb_taps, sizeof(*s->taps));
511 if (!s->taps)
512 return AVERROR(ENOMEM);
513
514 eq_interp(s->complexf, s->freq, s->magnitude, s->nb_freq, s->interp, s->nb_taps, factor);
515
516 for (int i = 0; i < fft_size; i++)
517 s->complexf[i].re = ff_exp10f(s->complexf[i].re / 20.f);
518
519 if (s->phaset) {
520 const float threshold = powf(10.f, -100.f / 20.f);
521 const float logt = logf(threshold);
522
523 scale = 1.f;
524 ret = av_tx_init(&s->tx_ctx, &s->tx_fn, AV_TX_FLOAT_FFT, 0, fft_size, &scale, 0);
525 if (ret < 0)
526 return ret;
527
528 for (int i = 0; i < fft_size; i++)
529 s->complexf[i].re = s->complexf[i].re < threshold ? logt : logf(s->complexf[i].re);
530
531 s->itx_fn(s->itx_ctx, s->complexf + asize, s->complexf, sizeof(float));
532 for (int i = 0; i < fft_size; i++) {
533 s->complexf[i + asize].re /= fft_size;
534 s->complexf[i + asize].im /= fft_size;
535 }
536
537 for (int i = 1; i < s->nb_taps; i++) {
538 s->complexf[asize + i].re += s->complexf[asize + fft_size - i].re;
539 s->complexf[asize + i].im -= s->complexf[asize + fft_size - i].im;
540 s->complexf[asize + fft_size - i].re = 0.f;
541 s->complexf[asize + fft_size - i].im = 0.f;
542 }
543 s->complexf[asize + s->nb_taps - 1].im *= -1.f;
544
545 s->tx_fn(s->tx_ctx, s->complexf, s->complexf + asize, sizeof(float));
546
547 for (int i = 0; i < fft_size; i++) {
548 float eR = expf(s->complexf[i].re);
549
550 s->complexf[i].re = eR * cosf(s->complexf[i].im);
551 s->complexf[i].im = eR * sinf(s->complexf[i].im);
552 }
553
554 s->itx_fn(s->itx_ctx, s->complexf + asize, s->complexf, sizeof(float));
555
556 for (int i = 0; i < s->nb_taps; i++)
557 s->taps[i] = s->complexf[i + asize].re / fft_size;
558 } else {
559 s->itx_fn(s->itx_ctx, s->complexf + asize, s->complexf, sizeof(float));
560
561 middle = s->nb_taps / 2;
562 for (int i = 0; i < middle; i++) {
563 s->taps[middle - i] = s->complexf[i + asize].re / fft_size;
564 s->taps[middle + i] = s->complexf[i + asize].re / fft_size;
565 }
566 }
567
568 s->pts = 0;
569
570 return 0;
571 }
572
573 static const AVFilterPad afireqsrc_outputs[] = {
574 {
575 .name = "default",
576 .type = AVMEDIA_TYPE_AUDIO,
577 .config_props = config_eq_output,
578 },
579 };
580
581 const AVFilter ff_asrc_afireqsrc = {
582 .name = "afireqsrc",
583 .description = NULL_IF_CONFIG_SMALL("Generate a FIR equalizer coefficients audio stream."),
584 .uninit = uninit,
585 .activate = activate,
586 .priv_size = sizeof(AudioFIRSourceContext),
587 .inputs = NULL,
588 FILTER_OUTPUTS(afireqsrc_outputs),
589 FILTER_QUERY_FUNC(query_formats),
590 .priv_class = &afireqsrc_class,
591 };
592