FFmpeg coverage

Line	Branch	Exec	Source
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