2 * Copyright (c) 2018 Paul B Mahol
4 * This file is part of FFmpeg.
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.
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.
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
23 #include "libavutil/avassert.h"
24 #include "libavutil/avstring.h"
25 #include "libavutil/intreadwrite.h"
26 #include "libavutil/opt.h"
27 #include "libavutil/xga_font_data.h"
32 typedef struct ThreadData {
40 typedef struct BiquadContext {
47 typedef struct IIRChannel {
52 BiquadContext *biquads;
56 typedef struct AudioIIRContext {
58 char *a_str, *b_str, *g_str;
59 double dry_gain, wet_gain;
74 enum AVSampleFormat sample_format;
76 int (*iir_channel)(AVFilterContext *ctx, void *arg, int ch, int nb_jobs);
79 static int query_formats(AVFilterContext *ctx)
81 AudioIIRContext *s = ctx->priv;
82 AVFilterFormats *formats;
83 AVFilterChannelLayouts *layouts;
84 enum AVSampleFormat sample_fmts[] = {
88 static const enum AVPixelFormat pix_fmts[] = {
95 AVFilterLink *videolink = ctx->outputs[1];
97 formats = ff_make_format_list(pix_fmts);
98 if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
102 layouts = ff_all_channel_counts();
104 return AVERROR(ENOMEM);
105 ret = ff_set_common_channel_layouts(ctx, layouts);
109 sample_fmts[0] = s->sample_format;
110 formats = ff_make_format_list(sample_fmts);
112 return AVERROR(ENOMEM);
113 ret = ff_set_common_formats(ctx, formats);
117 formats = ff_all_samplerates();
119 return AVERROR(ENOMEM);
120 return ff_set_common_samplerates(ctx, formats);
123 #define IIR_CH(name, type, min, max, need_clipping) \
124 static int iir_ch_## name(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) \
126 AudioIIRContext *s = ctx->priv; \
127 const double ig = s->dry_gain; \
128 const double og = s->wet_gain; \
129 const double mix = s->mix; \
130 ThreadData *td = arg; \
131 AVFrame *in = td->in, *out = td->out; \
132 const type *src = (const type *)in->extended_data[ch]; \
133 double *oc = (double *)s->iir[ch].cache[0]; \
134 double *ic = (double *)s->iir[ch].cache[1]; \
135 const int nb_a = s->iir[ch].nb_ab[0]; \
136 const int nb_b = s->iir[ch].nb_ab[1]; \
137 const double *a = s->iir[ch].ab[0]; \
138 const double *b = s->iir[ch].ab[1]; \
139 const double g = s->iir[ch].g; \
140 int *clippings = &s->iir[ch].clippings; \
141 type *dst = (type *)out->extended_data[ch]; \
144 for (n = 0; n < in->nb_samples; n++) { \
145 double sample = 0.; \
148 memmove(&ic[1], &ic[0], (nb_b - 1) * sizeof(*ic)); \
149 memmove(&oc[1], &oc[0], (nb_a - 1) * sizeof(*oc)); \
150 ic[0] = src[n] * ig; \
151 for (x = 0; x < nb_b; x++) \
152 sample += b[x] * ic[x]; \
154 for (x = 1; x < nb_a; x++) \
155 sample -= a[x] * oc[x]; \
159 sample = sample * mix + ic[0] * (1. - mix); \
160 if (need_clipping && sample < min) { \
163 } else if (need_clipping && sample > max) { \
174 IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
175 IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
176 IIR_CH(fltp, float, -1., 1., 0)
177 IIR_CH(dblp, double, -1., 1., 0)
179 #define SERIAL_IIR_CH(name, type, min, max, need_clipping) \
180 static int iir_ch_serial_## name(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) \
182 AudioIIRContext *s = ctx->priv; \
183 const double ig = s->dry_gain; \
184 const double og = s->wet_gain; \
185 const double mix = s->mix; \
186 ThreadData *td = arg; \
187 AVFrame *in = td->in, *out = td->out; \
188 const type *src = (const type *)in->extended_data[ch]; \
189 type *dst = (type *)out->extended_data[ch]; \
190 IIRChannel *iir = &s->iir[ch]; \
191 const double g = iir->g; \
192 int *clippings = &iir->clippings; \
193 int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; \
196 for (i = 0; i < nb_biquads; i++) { \
197 const double a1 = -iir->biquads[i].a[1]; \
198 const double a2 = -iir->biquads[i].a[2]; \
199 const double b0 = iir->biquads[i].b[0]; \
200 const double b1 = iir->biquads[i].b[1]; \
201 const double b2 = iir->biquads[i].b[2]; \
202 double i1 = iir->biquads[i].i1; \
203 double i2 = iir->biquads[i].i2; \
204 double o1 = iir->biquads[i].o1; \
205 double o2 = iir->biquads[i].o2; \
207 for (n = 0; n < in->nb_samples; n++) { \
208 double sample = ig * (i ? dst[n] : src[n]); \
209 double o0 = sample * b0 + i1 * b1 + i2 * b2 + o1 * a1 + o2 * a2; \
217 o0 = o0 * mix + (1. - mix) * sample; \
218 if (need_clipping && o0 < min) { \
221 } else if (need_clipping && o0 > max) { \
228 iir->biquads[i].i1 = i1; \
229 iir->biquads[i].i2 = i2; \
230 iir->biquads[i].o1 = o1; \
231 iir->biquads[i].o2 = o2; \
237 SERIAL_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
238 SERIAL_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
239 SERIAL_IIR_CH(fltp, float, -1., 1., 0)
240 SERIAL_IIR_CH(dblp, double, -1., 1., 0)
242 static void count_coefficients(char *item_str, int *nb_items)
250 for (p = item_str; *p && *p != '|'; p++) {
256 static int read_gains(AVFilterContext *ctx, char *item_str, int nb_items)
258 AudioIIRContext *s = ctx->priv;
259 char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
262 p = old_str = av_strdup(item_str);
264 return AVERROR(ENOMEM);
265 for (i = 0; i < nb_items; i++) {
266 if (!(arg = av_strtok(p, "|", &saveptr)))
271 return AVERROR(EINVAL);
275 if (sscanf(arg, "%lf", &s->iir[i].g) != 1) {
276 av_log(ctx, AV_LOG_ERROR, "Invalid gains supplied: %s\n", arg);
278 return AVERROR(EINVAL);
289 static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
291 char *p, *arg, *old_str, *saveptr = NULL;
294 p = old_str = av_strdup(item_str);
296 return AVERROR(ENOMEM);
297 for (i = 0; i < nb_items; i++) {
298 if (!(arg = av_strtok(p, " ", &saveptr)))
302 if (sscanf(arg, "%lf", &dst[i]) != 1) {
303 av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
305 return AVERROR(EINVAL);
314 static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst, const char *format)
316 char *p, *arg, *old_str, *saveptr = NULL;
319 p = old_str = av_strdup(item_str);
321 return AVERROR(ENOMEM);
322 for (i = 0; i < nb_items; i++) {
323 if (!(arg = av_strtok(p, " ", &saveptr)))
327 if (sscanf(arg, format, &dst[i*2], &dst[i*2+1]) != 2) {
328 av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
330 return AVERROR(EINVAL);
339 static const char *format[] = { "%lf", "%lf %lfi", "%lf %lfr", "%lf %lfd", "%lf %lfi" };
341 static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int ab)
343 AudioIIRContext *s = ctx->priv;
344 char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
347 p = old_str = av_strdup(item_str);
349 return AVERROR(ENOMEM);
350 for (i = 0; i < channels; i++) {
351 IIRChannel *iir = &s->iir[i];
353 if (!(arg = av_strtok(p, "|", &saveptr)))
358 return AVERROR(EINVAL);
361 count_coefficients(arg, &iir->nb_ab[ab]);
364 iir->cache[ab] = av_calloc(iir->nb_ab[ab] + 1, sizeof(double));
365 iir->ab[ab] = av_calloc(iir->nb_ab[ab] * (!!s->format + 1), sizeof(double));
366 if (!iir->ab[ab] || !iir->cache[ab]) {
368 return AVERROR(ENOMEM);
372 ret = read_zp_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab], format[s->format]);
374 ret = read_tf_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab]);
388 static void cmul(double re, double im, double re2, double im2, double *RE, double *IM)
390 *RE = re * re2 - im * im2;
391 *IM = re * im2 + re2 * im;
394 static int expand(AVFilterContext *ctx, double *pz, int n, double *coefs)
398 for (int i = 1; i <= n; i++) {
399 for (int j = n - i; j < n; j++) {
402 cmul(coefs[2 * (j + 1)], coefs[2 * (j + 1) + 1],
403 pz[2 * (i - 1)], pz[2 * (i - 1) + 1], &re, &im);
406 coefs[2 * j + 1] -= im;
410 for (int i = 0; i < n + 1; i++) {
411 if (fabs(coefs[2 * i + 1]) > FLT_EPSILON) {
412 av_log(ctx, AV_LOG_ERROR, "coefs: %f of z^%d is not real; poles/zeros are not complex conjugates.\n",
413 coefs[2 * i + 1], i);
414 return AVERROR(EINVAL);
421 static void normalize_coeffs(AVFilterContext *ctx, int ch)
423 AudioIIRContext *s = ctx->priv;
424 IIRChannel *iir = &s->iir[ch];
430 for (int i = 0; i < iir->nb_ab[1]; i++) {
431 sum_den += iir->ab[1][i];
434 if (sum_den > 1e-6) {
435 double factor, sum_num = 0.;
437 for (int i = 0; i < iir->nb_ab[0]; i++) {
438 sum_num += iir->ab[0][i];
441 factor = sum_num / sum_den;
443 for (int i = 0; i < iir->nb_ab[1]; i++) {
444 iir->ab[1][i] *= factor;
449 static int convert_zp2tf(AVFilterContext *ctx, int channels)
451 AudioIIRContext *s = ctx->priv;
452 int ch, i, j, ret = 0;
454 for (ch = 0; ch < channels; ch++) {
455 IIRChannel *iir = &s->iir[ch];
458 topc = av_calloc((iir->nb_ab[1] + 1) * 2, sizeof(*topc));
459 botc = av_calloc((iir->nb_ab[0] + 1) * 2, sizeof(*botc));
460 if (!topc || !botc) {
461 ret = AVERROR(ENOMEM);
465 ret = expand(ctx, iir->ab[0], iir->nb_ab[0], botc);
470 ret = expand(ctx, iir->ab[1], iir->nb_ab[1], topc);
475 for (j = 0, i = iir->nb_ab[1]; i >= 0; j++, i--) {
476 iir->ab[1][j] = topc[2 * i];
480 for (j = 0, i = iir->nb_ab[0]; i >= 0; j++, i--) {
481 iir->ab[0][j] = botc[2 * i];
485 normalize_coeffs(ctx, ch);
497 static int decompose_zp2biquads(AVFilterContext *ctx, int channels)
499 AudioIIRContext *s = ctx->priv;
502 for (ch = 0; ch < channels; ch++) {
503 IIRChannel *iir = &s->iir[ch];
504 int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2;
505 int current_biquad = 0;
507 iir->biquads = av_calloc(nb_biquads, sizeof(BiquadContext));
509 return AVERROR(ENOMEM);
511 while (nb_biquads--) {
512 Pair outmost_pole = { -1, -1 };
513 Pair nearest_zero = { -1, -1 };
514 double zeros[4] = { 0 };
515 double poles[4] = { 0 };
518 double min_distance = DBL_MAX;
523 for (i = 0; i < iir->nb_ab[0]; i++) {
526 if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1]))
528 mag = hypot(iir->ab[0][2 * i], iir->ab[0][2 * i + 1]);
536 for (i = 0; i < iir->nb_ab[0]; i++) {
537 if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1]))
540 if (iir->ab[0][2 * i ] == iir->ab[0][2 * outmost_pole.a ] &&
541 iir->ab[0][2 * i + 1] == -iir->ab[0][2 * outmost_pole.a + 1]) {
547 av_log(ctx, AV_LOG_VERBOSE, "outmost_pole is %d.%d\n", outmost_pole.a, outmost_pole.b);
549 if (outmost_pole.a < 0 || outmost_pole.b < 0)
550 return AVERROR(EINVAL);
552 for (i = 0; i < iir->nb_ab[1]; i++) {
555 if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1]))
557 distance = hypot(iir->ab[0][2 * outmost_pole.a ] - iir->ab[1][2 * i ],
558 iir->ab[0][2 * outmost_pole.a + 1] - iir->ab[1][2 * i + 1]);
560 if (distance < min_distance) {
561 min_distance = distance;
566 for (i = 0; i < iir->nb_ab[1]; i++) {
567 if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1]))
570 if (iir->ab[1][2 * i ] == iir->ab[1][2 * nearest_zero.a ] &&
571 iir->ab[1][2 * i + 1] == -iir->ab[1][2 * nearest_zero.a + 1]) {
577 av_log(ctx, AV_LOG_VERBOSE, "nearest_zero is %d.%d\n", nearest_zero.a, nearest_zero.b);
579 if (nearest_zero.a < 0 || nearest_zero.b < 0)
580 return AVERROR(EINVAL);
582 poles[0] = iir->ab[0][2 * outmost_pole.a ];
583 poles[1] = iir->ab[0][2 * outmost_pole.a + 1];
585 zeros[0] = iir->ab[1][2 * nearest_zero.a ];
586 zeros[1] = iir->ab[1][2 * nearest_zero.a + 1];
588 if (nearest_zero.a == nearest_zero.b && outmost_pole.a == outmost_pole.b) {
595 poles[2] = iir->ab[0][2 * outmost_pole.b ];
596 poles[3] = iir->ab[0][2 * outmost_pole.b + 1];
598 zeros[2] = iir->ab[1][2 * nearest_zero.b ];
599 zeros[3] = iir->ab[1][2 * nearest_zero.b + 1];
602 ret = expand(ctx, zeros, 2, b);
606 ret = expand(ctx, poles, 2, a);
610 iir->ab[0][2 * outmost_pole.a] = iir->ab[0][2 * outmost_pole.a + 1] = NAN;
611 iir->ab[0][2 * outmost_pole.b] = iir->ab[0][2 * outmost_pole.b + 1] = NAN;
612 iir->ab[1][2 * nearest_zero.a] = iir->ab[1][2 * nearest_zero.a + 1] = NAN;
613 iir->ab[1][2 * nearest_zero.b] = iir->ab[1][2 * nearest_zero.b + 1] = NAN;
615 iir->biquads[current_biquad].a[0] = 1.;
616 iir->biquads[current_biquad].a[1] = a[2] / a[4];
617 iir->biquads[current_biquad].a[2] = a[0] / a[4];
618 iir->biquads[current_biquad].b[0] = b[4] / a[4];
619 iir->biquads[current_biquad].b[1] = b[2] / a[4];
620 iir->biquads[current_biquad].b[2] = b[0] / a[4];
623 fabs(iir->biquads[current_biquad].b[0] +
624 iir->biquads[current_biquad].b[1] +
625 iir->biquads[current_biquad].b[2]) > 1e-6) {
626 factor = (iir->biquads[current_biquad].a[0] +
627 iir->biquads[current_biquad].a[1] +
628 iir->biquads[current_biquad].a[2]) /
629 (iir->biquads[current_biquad].b[0] +
630 iir->biquads[current_biquad].b[1] +
631 iir->biquads[current_biquad].b[2]);
633 av_log(ctx, AV_LOG_VERBOSE, "factor=%f\n", factor);
635 iir->biquads[current_biquad].b[0] *= factor;
636 iir->biquads[current_biquad].b[1] *= factor;
637 iir->biquads[current_biquad].b[2] *= factor;
640 iir->biquads[current_biquad].b[0] *= (current_biquad ? 1.0 : iir->g);
641 iir->biquads[current_biquad].b[1] *= (current_biquad ? 1.0 : iir->g);
642 iir->biquads[current_biquad].b[2] *= (current_biquad ? 1.0 : iir->g);
644 av_log(ctx, AV_LOG_VERBOSE, "a=%f %f %f:b=%f %f %f\n",
645 iir->biquads[current_biquad].a[0],
646 iir->biquads[current_biquad].a[1],
647 iir->biquads[current_biquad].a[2],
648 iir->biquads[current_biquad].b[0],
649 iir->biquads[current_biquad].b[1],
650 iir->biquads[current_biquad].b[2]);
659 static void convert_pr2zp(AVFilterContext *ctx, int channels)
661 AudioIIRContext *s = ctx->priv;
664 for (ch = 0; ch < channels; ch++) {
665 IIRChannel *iir = &s->iir[ch];
668 for (n = 0; n < iir->nb_ab[0]; n++) {
669 double r = iir->ab[0][2*n];
670 double angle = iir->ab[0][2*n+1];
672 iir->ab[0][2*n] = r * cos(angle);
673 iir->ab[0][2*n+1] = r * sin(angle);
676 for (n = 0; n < iir->nb_ab[1]; n++) {
677 double r = iir->ab[1][2*n];
678 double angle = iir->ab[1][2*n+1];
680 iir->ab[1][2*n] = r * cos(angle);
681 iir->ab[1][2*n+1] = r * sin(angle);
686 static void convert_sp2zp(AVFilterContext *ctx, int channels)
688 AudioIIRContext *s = ctx->priv;
691 for (ch = 0; ch < channels; ch++) {
692 IIRChannel *iir = &s->iir[ch];
695 for (n = 0; n < iir->nb_ab[0]; n++) {
696 double sr = iir->ab[0][2*n];
697 double si = iir->ab[0][2*n+1];
698 double snr = 1. + sr;
699 double sdr = 1. - sr;
700 double div = sdr * sdr + si * si;
702 iir->ab[0][2*n] = (snr * sdr - si * si) / div;
703 iir->ab[0][2*n+1] = (sdr * si + snr * si) / div;
706 for (n = 0; n < iir->nb_ab[1]; n++) {
707 double sr = iir->ab[1][2*n];
708 double si = iir->ab[1][2*n+1];
709 double snr = 1. + sr;
710 double sdr = 1. - sr;
711 double div = sdr * sdr + si * si;
713 iir->ab[1][2*n] = (snr * sdr - si * si) / div;
714 iir->ab[1][2*n+1] = (sdr * si + snr * si) / div;
719 static void convert_pd2zp(AVFilterContext *ctx, int channels)
721 AudioIIRContext *s = ctx->priv;
724 for (ch = 0; ch < channels; ch++) {
725 IIRChannel *iir = &s->iir[ch];
728 for (n = 0; n < iir->nb_ab[0]; n++) {
729 double r = iir->ab[0][2*n];
730 double angle = M_PI*iir->ab[0][2*n+1]/180.;
732 iir->ab[0][2*n] = r * cos(angle);
733 iir->ab[0][2*n+1] = r * sin(angle);
736 for (n = 0; n < iir->nb_ab[1]; n++) {
737 double r = iir->ab[1][2*n];
738 double angle = M_PI*iir->ab[1][2*n+1]/180.;
740 iir->ab[1][2*n] = r * cos(angle);
741 iir->ab[1][2*n+1] = r * sin(angle);
746 static void check_stability(AVFilterContext *ctx, int channels)
748 AudioIIRContext *s = ctx->priv;
751 for (ch = 0; ch < channels; ch++) {
752 IIRChannel *iir = &s->iir[ch];
754 for (int n = 0; n < iir->nb_ab[0]; n++) {
755 double pr = hypot(iir->ab[0][2*n], iir->ab[0][2*n+1]);
758 av_log(ctx, AV_LOG_WARNING, "pole %d at channel %d is unstable\n", n, ch);
765 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
771 font = avpriv_cga_font, font_height = 8;
773 for (i = 0; txt[i]; i++) {
776 uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
777 for (char_y = 0; char_y < font_height; char_y++) {
778 for (mask = 0x80; mask; mask >>= 1) {
779 if (font[txt[i] * font_height + char_y] & mask)
783 p += pic->linesize[0] - 8 * 4;
788 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
790 int dx = FFABS(x1-x0);
791 int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
792 int err = (dx>dy ? dx : -dy) / 2, e2;
795 AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
797 if (x0 == x1 && y0 == y1)
814 static double distance(double x0, double x1, double y0, double y1)
816 return hypot(x0 - x1, y0 - y1);
819 static void get_response(int channel, int format, double w,
820 const double *b, const double *a,
821 int nb_b, int nb_a, double *magnitude, double *phase)
829 realz = 0., realp = 0.;
830 imagz = 0., imagp = 0.;
831 for (int x = 0; x < nb_a; x++) {
832 realz += cos(-x * w) * a[x];
833 imagz += sin(-x * w) * a[x];
836 for (int x = 0; x < nb_b; x++) {
837 realp += cos(-x * w) * b[x];
838 imagp += sin(-x * w) * b[x];
841 div = realp * realp + imagp * imagp;
842 real = (realz * realp + imagz * imagp) / div;
843 imag = (imagz * realp - imagp * realz) / div;
845 *magnitude = hypot(real, imag);
846 *phase = atan2(imag, real);
848 double p = 1., z = 1.;
851 for (int x = 0; x < nb_a; x++) {
852 z *= distance(cos(w), a[2 * x], sin(w), a[2 * x + 1]);
853 acc += atan2(sin(w) - a[2 * x + 1], cos(w) - a[2 * x]);
856 for (int x = 0; x < nb_b; x++) {
857 p *= distance(cos(w), b[2 * x], sin(w), b[2 * x + 1]);
858 acc -= atan2(sin(w) - b[2 * x + 1], cos(w) - b[2 * x]);
866 static void draw_response(AVFilterContext *ctx, AVFrame *out, int sample_rate)
868 AudioIIRContext *s = ctx->priv;
869 double *mag, *phase, *temp, *delay, min = DBL_MAX, max = -DBL_MAX;
870 double min_delay = DBL_MAX, max_delay = -DBL_MAX, min_phase, max_phase;
871 int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
875 memset(out->data[0], 0, s->h * out->linesize[0]);
877 phase = av_malloc_array(s->w, sizeof(*phase));
878 temp = av_malloc_array(s->w, sizeof(*temp));
879 mag = av_malloc_array(s->w, sizeof(*mag));
880 delay = av_malloc_array(s->w, sizeof(*delay));
881 if (!mag || !phase || !delay || !temp)
884 ch = av_clip(s->ir_channel, 0, s->channels - 1);
885 for (i = 0; i < s->w; i++) {
886 const double *b = s->iir[ch].ab[0];
887 const double *a = s->iir[ch].ab[1];
888 const int nb_b = s->iir[ch].nb_ab[0];
889 const int nb_a = s->iir[ch].nb_ab[1];
890 double w = i * M_PI / (s->w - 1);
893 get_response(ch, s->format, w, b, a, nb_b, nb_a, &m, &p);
895 mag[i] = s->iir[ch].g * m;
897 min = fmin(min, mag[i]);
898 max = fmax(max, mag[i]);
902 for (i = 0; i < s->w - 1; i++) {
903 double d = phase[i] - phase[i + 1];
904 temp[i + 1] = ceil(fabs(d) / (2. * M_PI)) * 2. * M_PI * ((d > M_PI) - (d < -M_PI));
907 min_phase = phase[0];
908 max_phase = phase[0];
909 for (i = 1; i < s->w; i++) {
910 temp[i] += temp[i - 1];
912 min_phase = fmin(min_phase, phase[i]);
913 max_phase = fmax(max_phase, phase[i]);
916 for (i = 0; i < s->w - 1; i++) {
917 double div = s->w / (double)sample_rate;
919 delay[i + 1] = -(phase[i] - phase[i + 1]) / div;
920 min_delay = fmin(min_delay, delay[i + 1]);
921 max_delay = fmax(max_delay, delay[i + 1]);
925 for (i = 0; i < s->w; i++) {
926 int ymag = mag[i] / max * (s->h - 1);
927 int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
928 int yphase = (phase[i] - min_phase) / (max_phase - min_phase) * (s->h - 1);
930 ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
931 yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
932 ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
937 prev_yphase = yphase;
939 prev_ydelay = ydelay;
941 draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
942 draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
943 draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
946 prev_yphase = yphase;
947 prev_ydelay = ydelay;
950 if (s->w > 400 && s->h > 100) {
951 drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
952 snprintf(text, sizeof(text), "%.2f", max);
953 drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
955 drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
956 snprintf(text, sizeof(text), "%.2f", min);
957 drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
959 drawtext(out, 2, 22, "Max Phase:", 0xDDDDDDDD);
960 snprintf(text, sizeof(text), "%.2f", max_phase);
961 drawtext(out, 15 * 8 + 2, 22, text, 0xDDDDDDDD);
963 drawtext(out, 2, 32, "Min Phase:", 0xDDDDDDDD);
964 snprintf(text, sizeof(text), "%.2f", min_phase);
965 drawtext(out, 15 * 8 + 2, 32, text, 0xDDDDDDDD);
967 drawtext(out, 2, 42, "Max Delay:", 0xDDDDDDDD);
968 snprintf(text, sizeof(text), "%.2f", max_delay);
969 drawtext(out, 11 * 8 + 2, 42, text, 0xDDDDDDDD);
971 drawtext(out, 2, 52, "Min Delay:", 0xDDDDDDDD);
972 snprintf(text, sizeof(text), "%.2f", min_delay);
973 drawtext(out, 11 * 8 + 2, 52, text, 0xDDDDDDDD);
983 static int config_output(AVFilterLink *outlink)
985 AVFilterContext *ctx = outlink->src;
986 AudioIIRContext *s = ctx->priv;
987 AVFilterLink *inlink = ctx->inputs[0];
990 s->channels = inlink->channels;
991 s->iir = av_calloc(s->channels, sizeof(*s->iir));
993 return AVERROR(ENOMEM);
995 ret = read_gains(ctx, s->g_str, inlink->channels);
999 ret = read_channels(ctx, inlink->channels, s->a_str, 0);
1003 ret = read_channels(ctx, inlink->channels, s->b_str, 1);
1007 if (s->format == 2) {
1008 convert_pr2zp(ctx, inlink->channels);
1009 } else if (s->format == 3) {
1010 convert_pd2zp(ctx, inlink->channels);
1011 } else if (s->format == 4) {
1012 convert_sp2zp(ctx, inlink->channels);
1014 if (s->format > 0) {
1015 check_stability(ctx, inlink->channels);
1018 av_frame_free(&s->video);
1020 s->video = ff_get_video_buffer(ctx->outputs[1], s->w, s->h);
1022 return AVERROR(ENOMEM);
1024 draw_response(ctx, s->video, inlink->sample_rate);
1028 av_log(ctx, AV_LOG_WARNING, "tf coefficients format is not recommended for too high number of zeros/poles.\n");
1030 if (s->format > 0 && s->process == 0) {
1031 av_log(ctx, AV_LOG_WARNING, "Direct processsing is not recommended for zp coefficients format.\n");
1033 ret = convert_zp2tf(ctx, inlink->channels);
1036 } else if (s->format == 0 && s->process == 1) {
1037 av_log(ctx, AV_LOG_ERROR, "Serial cascading is not implemented for transfer function.\n");
1038 return AVERROR_PATCHWELCOME;
1039 } else if (s->format > 0 && s->process == 1) {
1040 if (inlink->format == AV_SAMPLE_FMT_S16P)
1041 av_log(ctx, AV_LOG_WARNING, "Serial cascading is not recommended for i16 precision.\n");
1043 ret = decompose_zp2biquads(ctx, inlink->channels);
1048 for (ch = 0; s->format == 0 && ch < inlink->channels; ch++) {
1049 IIRChannel *iir = &s->iir[ch];
1051 for (i = 1; i < iir->nb_ab[0]; i++) {
1052 iir->ab[0][i] /= iir->ab[0][0];
1055 iir->ab[0][0] = 1.0;
1056 for (i = 0; i < iir->nb_ab[1]; i++) {
1057 iir->ab[1][i] *= iir->g;
1060 normalize_coeffs(ctx, ch);
1063 switch (inlink->format) {
1064 case AV_SAMPLE_FMT_DBLP: s->iir_channel = s->process == 1 ? iir_ch_serial_dblp : iir_ch_dblp; break;
1065 case AV_SAMPLE_FMT_FLTP: s->iir_channel = s->process == 1 ? iir_ch_serial_fltp : iir_ch_fltp; break;
1066 case AV_SAMPLE_FMT_S32P: s->iir_channel = s->process == 1 ? iir_ch_serial_s32p : iir_ch_s32p; break;
1067 case AV_SAMPLE_FMT_S16P: s->iir_channel = s->process == 1 ? iir_ch_serial_s16p : iir_ch_s16p; break;
1073 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
1075 AVFilterContext *ctx = inlink->dst;
1076 AudioIIRContext *s = ctx->priv;
1077 AVFilterLink *outlink = ctx->outputs[0];
1082 if (av_frame_is_writable(in)) {
1085 out = ff_get_audio_buffer(outlink, in->nb_samples);
1088 return AVERROR(ENOMEM);
1090 av_frame_copy_props(out, in);
1095 ctx->internal->execute(ctx, s->iir_channel, &td, NULL, outlink->channels);
1097 for (ch = 0; ch < outlink->channels; ch++) {
1098 if (s->iir[ch].clippings > 0)
1099 av_log(ctx, AV_LOG_WARNING, "Channel %d clipping %d times. Please reduce gain.\n",
1100 ch, s->iir[ch].clippings);
1101 s->iir[ch].clippings = 0;
1108 AVFilterLink *outlink = ctx->outputs[1];
1109 int64_t old_pts = s->video->pts;
1110 int64_t new_pts = av_rescale_q(out->pts, ctx->inputs[0]->time_base, outlink->time_base);
1112 if (new_pts > old_pts) {
1115 s->video->pts = new_pts;
1116 clone = av_frame_clone(s->video);
1118 return AVERROR(ENOMEM);
1119 ret = ff_filter_frame(outlink, clone);
1125 return ff_filter_frame(outlink, out);
1128 static int config_video(AVFilterLink *outlink)
1130 AVFilterContext *ctx = outlink->src;
1131 AudioIIRContext *s = ctx->priv;
1133 outlink->sample_aspect_ratio = (AVRational){1,1};
1136 outlink->frame_rate = s->rate;
1137 outlink->time_base = av_inv_q(outlink->frame_rate);
1142 static av_cold int init(AVFilterContext *ctx)
1144 AudioIIRContext *s = ctx->priv;
1145 AVFilterPad pad, vpad;
1148 if (!s->a_str || !s->b_str || !s->g_str) {
1149 av_log(ctx, AV_LOG_ERROR, "Valid coefficients are mandatory.\n");
1150 return AVERROR(EINVAL);
1153 switch (s->precision) {
1154 case 0: s->sample_format = AV_SAMPLE_FMT_DBLP; break;
1155 case 1: s->sample_format = AV_SAMPLE_FMT_FLTP; break;
1156 case 2: s->sample_format = AV_SAMPLE_FMT_S32P; break;
1157 case 3: s->sample_format = AV_SAMPLE_FMT_S16P; break;
1158 default: return AVERROR_BUG;
1161 pad = (AVFilterPad){
1162 .name = av_strdup("default"),
1163 .type = AVMEDIA_TYPE_AUDIO,
1164 .config_props = config_output,
1168 return AVERROR(ENOMEM);
1170 ret = ff_insert_outpad(ctx, 0, &pad);
1175 vpad = (AVFilterPad){
1176 .name = av_strdup("filter_response"),
1177 .type = AVMEDIA_TYPE_VIDEO,
1178 .config_props = config_video,
1181 return AVERROR(ENOMEM);
1183 ret = ff_insert_outpad(ctx, 1, &vpad);
1191 static av_cold void uninit(AVFilterContext *ctx)
1193 AudioIIRContext *s = ctx->priv;
1197 for (ch = 0; ch < s->channels; ch++) {
1198 IIRChannel *iir = &s->iir[ch];
1199 av_freep(&iir->ab[0]);
1200 av_freep(&iir->ab[1]);
1201 av_freep(&iir->cache[0]);
1202 av_freep(&iir->cache[1]);
1203 av_freep(&iir->biquads);
1208 av_freep(&ctx->output_pads[0].name);
1210 av_freep(&ctx->output_pads[1].name);
1211 av_frame_free(&s->video);
1214 static const AVFilterPad inputs[] = {
1217 .type = AVMEDIA_TYPE_AUDIO,
1218 .filter_frame = filter_frame,
1223 #define OFFSET(x) offsetof(AudioIIRContext, x)
1224 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
1225 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
1227 static const AVOption aiir_options[] = {
1228 { "zeros", "set B/numerator/zeros coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1229 { "z", "set B/numerator/zeros coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1230 { "poles", "set A/denominator/poles coefficients", OFFSET(a_str),AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1231 { "p", "set A/denominator/poles coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF },
1232 { "gains", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF },
1233 { "k", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF },
1234 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
1235 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
1236 { "format", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, 0, 4, AF, "format" },
1237 { "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, 0, 4, AF, "format" },
1238 { "tf", "digital transfer function", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "format" },
1239 { "zp", "Z-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "format" },
1240 { "pr", "Z-plane zeros/poles (polar radians)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "format" },
1241 { "pd", "Z-plane zeros/poles (polar degrees)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "format" },
1242 { "sp", "S-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "format" },
1243 { "process", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "process" },
1244 { "r", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "process" },
1245 { "d", "direct", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "process" },
1246 { "s", "serial cascading", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "process" },
1247 { "precision", "set filtering precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, "precision" },
1248 { "e", "set precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, "precision" },
1249 { "dbl", "double-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "precision" },
1250 { "flt", "single-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision" },
1251 { "i32", "32-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision" },
1252 { "i16", "16-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision" },
1253 { "normalize", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
1254 { "n", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
1255 { "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
1256 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
1257 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
1258 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
1259 { "rate", "set video rate", OFFSET(rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
1263 AVFILTER_DEFINE_CLASS(aiir);
1265 AVFilter ff_af_aiir = {
1267 .description = NULL_IF_CONFIG_SMALL("Apply Infinite Impulse Response filter with supplied coefficients."),
1268 .priv_size = sizeof(AudioIIRContext),
1269 .priv_class = &aiir_class,
1272 .query_formats = query_formats,
1274 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
1275 AVFILTER_FLAG_SLICE_THREADS,