2 * Copyright (c) 2017 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 * An arbitrary audio FIR filter
28 #include "libavutil/common.h"
29 #include "libavutil/float_dsp.h"
30 #include "libavutil/intreadwrite.h"
31 #include "libavutil/opt.h"
32 #include "libavutil/xga_font_data.h"
33 #include "libavcodec/avfft.h"
42 static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
46 for (n = 0; n < len; n++) {
47 const float cre = c[2 * n ];
48 const float cim = c[2 * n + 1];
49 const float tre = t[2 * n ];
50 const float tim = t[2 * n + 1];
52 sum[2 * n ] += tre * cre - tim * cim;
53 sum[2 * n + 1] += tre * cim + tim * cre;
56 sum[2 * n] += t[2 * n] * c[2 * n];
59 static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
61 AudioFIRContext *s = ctx->priv;
62 AudioFIRSegment *seg = &s->seg;
63 const float *src = (const float *)s->in[0]->extended_data[ch];
64 float *sum = (float *)seg->sum->extended_data[ch];
66 float *block, *dst, *ptr;
69 memset(sum, 0, sizeof(*sum) * seg->fft_length);
70 block = (float *)seg->block->extended_data[ch] + seg->part_index * seg->block_size;
71 memset(block, 0, sizeof(*block) * seg->fft_length);
73 s->fdsp->vector_fmul_scalar(block, src, s->dry_gain, FFALIGN(out->nb_samples, 4));
76 av_rdft_calc(seg->rdft[ch], block);
77 block[2 * seg->part_size] = block[1];
82 for (i = 0; i < seg->nb_partitions; i++) {
83 const int coffset = i * seg->coeff_size;
84 const float *block = (const float *)seg->block->extended_data[ch] + j * seg->block_size;
85 const FFTComplex *coeff = seg->coeff[ch * !s->one2many] + coffset;
87 s->fcmul_add(sum, block, (const float *)coeff, seg->part_size);
90 j = seg->nb_partitions;
94 sum[1] = sum[2 * seg->part_size];
95 av_rdft_calc(seg->irdft[ch], sum);
97 dst = (float *)seg->buffer->extended_data[ch];
98 for (n = 0; n < seg->part_size; n++) {
102 ptr = (float *)out->extended_data[ch];
103 s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4));
106 dst = (float *)seg->buffer->extended_data[ch];
107 memcpy(dst, sum + seg->part_size, seg->part_size * sizeof(*dst));
112 static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
114 AVFilterContext *ctx = outlink->src;
117 out = ff_get_audio_buffer(outlink, in->nb_samples);
120 return AVERROR(ENOMEM);
123 if (s->pts == AV_NOPTS_VALUE)
126 ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
128 s->seg.part_index = (s->seg.part_index + 1) % s->seg.nb_partitions;
131 if (s->pts != AV_NOPTS_VALUE)
132 s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
137 return ff_filter_frame(outlink, out);
140 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
146 font = avpriv_cga_font, font_height = 8;
148 for (i = 0; txt[i]; i++) {
151 uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
152 for (char_y = 0; char_y < font_height; char_y++) {
153 for (mask = 0x80; mask; mask >>= 1) {
154 if (font[txt[i] * font_height + char_y] & mask)
158 p += pic->linesize[0] - 8 * 4;
163 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
165 int dx = FFABS(x1-x0);
166 int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
167 int err = (dx>dy ? dx : -dy) / 2, e2;
170 AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
172 if (x0 == x1 && y0 == y1)
189 static void draw_response(AVFilterContext *ctx, AVFrame *out)
191 AudioFIRContext *s = ctx->priv;
192 float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
193 float min_delay = FLT_MAX, max_delay = FLT_MIN;
194 int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
198 memset(out->data[0], 0, s->h * out->linesize[0]);
200 phase = av_malloc_array(s->w, sizeof(*phase));
201 mag = av_malloc_array(s->w, sizeof(*mag));
202 delay = av_malloc_array(s->w, sizeof(*delay));
203 if (!mag || !phase || !delay)
206 channel = av_clip(s->ir_channel, 0, s->in[1]->channels - 1);
207 for (i = 0; i < s->w; i++) {
208 const float *src = (const float *)s->in[1]->extended_data[channel];
209 double w = i * M_PI / (s->w - 1);
210 double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
212 for (x = 0; x < s->nb_taps; x++) {
213 real += cos(-x * w) * src[x];
214 imag += sin(-x * w) * src[x];
215 real_num += cos(-x * w) * src[x] * x;
216 imag_num += sin(-x * w) * src[x] * x;
219 mag[i] = hypot(real, imag);
220 phase[i] = atan2(imag, real);
221 div = real * real + imag * imag;
222 delay[i] = (real_num * real + imag_num * imag) / div;
223 min = fminf(min, mag[i]);
224 max = fmaxf(max, mag[i]);
225 min_delay = fminf(min_delay, delay[i]);
226 max_delay = fmaxf(max_delay, delay[i]);
229 for (i = 0; i < s->w; i++) {
230 int ymag = mag[i] / max * (s->h - 1);
231 int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
232 int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
234 ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
235 yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
236 ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
241 prev_yphase = yphase;
243 prev_ydelay = ydelay;
245 draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
246 draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
247 draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
250 prev_yphase = yphase;
251 prev_ydelay = ydelay;
254 if (s->w > 400 && s->h > 100) {
255 drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
256 snprintf(text, sizeof(text), "%.2f", max);
257 drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
259 drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
260 snprintf(text, sizeof(text), "%.2f", min);
261 drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
263 drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
264 snprintf(text, sizeof(text), "%.2f", max_delay);
265 drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
267 drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
268 snprintf(text, sizeof(text), "%.2f", min_delay);
269 drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
278 static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_partitions, int part_size)
280 seg->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*seg->coeff));
281 seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
282 seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
283 if (!seg->coeff || !seg->rdft || !seg->irdft)
284 return AVERROR(ENOMEM);
286 seg->fft_length = part_size * 4 + 1;
287 seg->part_size = part_size;
288 seg->block_size = FFALIGN(seg->fft_length, 32);
289 seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
290 seg->nb_partitions = nb_partitions;
292 for (int ch = 0; ch < ctx->inputs[1]->channels; ch++) {
293 seg->coeff[ch] = av_calloc(seg->nb_partitions * seg->coeff_size, sizeof(**seg->coeff));
295 return AVERROR(ENOMEM);
298 for (int ch = 0; ch < ctx->inputs[0]->channels; ch++) {
299 seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
300 seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);
301 if (!seg->rdft[ch] || !seg->irdft[ch])
302 return AVERROR(ENOMEM);
305 seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
306 seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
307 seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
308 if (!seg->buffer || !seg->sum || !seg->block)
309 return AVERROR(ENOMEM);
314 static int convert_coeffs(AVFilterContext *ctx)
316 AudioFIRContext *s = ctx->priv;
317 int ret, i, ch, n, N;
320 s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1]);
322 return AVERROR(EINVAL);
324 for (n = av_log2(s->minp); (1 << n) < s->nb_taps; n++);
325 N = FFMIN(n, av_log2(s->maxp));
327 ret = init_segment(ctx, &s->seg, (s->nb_taps + (1 << N) - 1) / (1 << N), 1 << N);
331 ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]);
338 draw_response(ctx, s->video);
347 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
348 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
350 for (i = 0; i < s->nb_taps; i++)
351 power += FFABS(time[i]);
353 s->gain = ctx->inputs[1]->channels / power;
356 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
357 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
359 for (i = 0; i < s->nb_taps; i++)
362 s->gain = ctx->inputs[1]->channels / power;
365 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
366 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
368 for (i = 0; i < s->nb_taps; i++)
369 power += time[i] * time[i];
371 s->gain = sqrtf(ch / power);
377 s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
378 av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
379 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
380 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
382 s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
385 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
386 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
387 float *block = (float *)s->seg.block->extended_data[ch];
388 FFTComplex *coeff = s->seg.coeff[ch];
390 for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
393 for (i = 0; i < s->seg.nb_partitions; i++) {
394 const float scale = 1.f / s->seg.part_size;
395 const int toffset = i * s->seg.part_size;
396 const int coffset = i * s->seg.coeff_size;
397 const int remaining = s->nb_taps - (i * s->seg.part_size);
398 const int size = remaining >= s->seg.part_size ? s->seg.part_size : remaining;
400 memset(block, 0, sizeof(*block) * s->seg.fft_length);
401 memcpy(block, time + toffset, size * sizeof(*block));
403 av_rdft_calc(s->seg.rdft[0], block);
405 coeff[coffset].re = block[0] * scale;
406 coeff[coffset].im = 0;
407 for (n = 1; n < s->seg.part_size; n++) {
408 coeff[coffset + n].re = block[2 * n] * scale;
409 coeff[coffset + n].im = block[2 * n + 1] * scale;
411 coeff[coffset + s->seg.part_size].re = block[1] * scale;
412 coeff[coffset + s->seg.part_size].im = 0;
416 av_frame_free(&s->in[1]);
417 av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
418 av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->seg.nb_partitions);
419 av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->seg.part_size);
420 av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->seg.fft_length);
427 static int check_ir(AVFilterLink *link, AVFrame *frame)
429 AVFilterContext *ctx = link->dst;
430 AudioFIRContext *s = ctx->priv;
431 int nb_taps, max_nb_taps;
433 nb_taps = ff_inlink_queued_samples(link);
434 max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
435 if (nb_taps > max_nb_taps) {
436 av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
437 return AVERROR(EINVAL);
443 static int activate(AVFilterContext *ctx)
445 AudioFIRContext *s = ctx->priv;
446 AVFilterLink *outlink = ctx->outputs[0];
451 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
453 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
454 if (!s->eof_coeffs) {
457 ret = check_ir(ctx->inputs[1], ir);
461 if (ff_outlink_get_status(ctx->inputs[1]) == AVERROR_EOF)
464 if (!s->eof_coeffs) {
465 if (ff_outlink_frame_wanted(ctx->outputs[0]))
466 ff_inlink_request_frame(ctx->inputs[1]);
467 else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
468 ff_inlink_request_frame(ctx->inputs[1]);
473 if (!s->have_coeffs && s->eof_coeffs) {
474 ret = convert_coeffs(ctx);
479 ret = ff_inlink_consume_samples(ctx->inputs[0], s->seg.part_size, s->seg.part_size, &in);
481 ret = fir_frame(s, in, outlink);
486 if (s->response && s->have_coeffs) {
487 int64_t old_pts = s->video->pts;
488 int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
490 if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
491 s->video->pts = new_pts;
492 return ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
496 if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->seg.part_size) {
497 ff_filter_set_ready(ctx, 10);
501 if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
502 if (status == AVERROR_EOF) {
503 ff_outlink_set_status(ctx->outputs[0], status, pts);
505 ff_outlink_set_status(ctx->outputs[1], status, pts);
510 if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
511 !ff_outlink_get_status(ctx->inputs[0])) {
512 ff_inlink_request_frame(ctx->inputs[0]);
517 ff_outlink_frame_wanted(ctx->outputs[1]) &&
518 !ff_outlink_get_status(ctx->inputs[0])) {
519 ff_inlink_request_frame(ctx->inputs[0]);
523 return FFERROR_NOT_READY;
526 static int query_formats(AVFilterContext *ctx)
528 AudioFIRContext *s = ctx->priv;
529 AVFilterFormats *formats;
530 AVFilterChannelLayouts *layouts;
531 static const enum AVSampleFormat sample_fmts[] = {
535 static const enum AVPixelFormat pix_fmts[] = {
542 AVFilterLink *videolink = ctx->outputs[1];
543 formats = ff_make_format_list(pix_fmts);
544 if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
548 layouts = ff_all_channel_counts();
550 return AVERROR(ENOMEM);
553 ret = ff_set_common_channel_layouts(ctx, layouts);
557 AVFilterChannelLayouts *mono = NULL;
559 ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
563 if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts)) < 0)
565 if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
567 if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[1]->out_channel_layouts)) < 0)
571 formats = ff_make_format_list(sample_fmts);
572 if ((ret = ff_set_common_formats(ctx, formats)) < 0)
575 formats = ff_all_samplerates();
576 return ff_set_common_samplerates(ctx, formats);
579 static int config_output(AVFilterLink *outlink)
581 AVFilterContext *ctx = outlink->src;
582 AudioFIRContext *s = ctx->priv;
584 s->one2many = ctx->inputs[1]->channels == 1;
585 outlink->sample_rate = ctx->inputs[0]->sample_rate;
586 outlink->time_base = ctx->inputs[0]->time_base;
587 outlink->channel_layout = ctx->inputs[0]->channel_layout;
588 outlink->channels = ctx->inputs[0]->channels;
590 s->nb_channels = outlink->channels;
591 s->nb_coef_channels = ctx->inputs[1]->channels;
592 s->pts = AV_NOPTS_VALUE;
597 static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
599 AudioFIRContext *s = ctx->priv;
602 for (int ch = 0; ch < s->nb_coef_channels; ch++) {
603 av_freep(&seg->coeff[ch]);
606 av_freep(&seg->coeff);
609 for (int ch = 0; ch < s->nb_channels; ch++) {
610 av_rdft_end(seg->rdft[ch]);
613 av_freep(&seg->rdft);
616 for (int ch = 0; ch < s->nb_channels; ch++) {
617 av_rdft_end(seg->irdft[ch]);
620 av_freep(&seg->irdft);
622 av_frame_free(&seg->block);
623 av_frame_free(&seg->sum);
624 av_frame_free(&seg->buffer);
627 static av_cold void uninit(AVFilterContext *ctx)
629 AudioFIRContext *s = ctx->priv;
631 uninit_segment(ctx, &s->seg);
634 av_frame_free(&s->in[1]);
636 for (int i = 0; i < ctx->nb_outputs; i++)
637 av_freep(&ctx->output_pads[i].name);
638 av_frame_free(&s->video);
641 static int config_video(AVFilterLink *outlink)
643 AVFilterContext *ctx = outlink->src;
644 AudioFIRContext *s = ctx->priv;
646 outlink->sample_aspect_ratio = (AVRational){1,1};
649 outlink->frame_rate = s->frame_rate;
650 outlink->time_base = av_inv_q(outlink->frame_rate);
652 av_frame_free(&s->video);
653 s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
655 return AVERROR(ENOMEM);
660 static av_cold int init(AVFilterContext *ctx)
662 AudioFIRContext *s = ctx->priv;
663 AVFilterPad pad, vpad;
667 .name = av_strdup("default"),
668 .type = AVMEDIA_TYPE_AUDIO,
669 .config_props = config_output,
673 return AVERROR(ENOMEM);
676 vpad = (AVFilterPad){
677 .name = av_strdup("filter_response"),
678 .type = AVMEDIA_TYPE_VIDEO,
679 .config_props = config_video,
682 return AVERROR(ENOMEM);
685 ret = ff_insert_outpad(ctx, 0, &pad);
692 ret = ff_insert_outpad(ctx, 1, &vpad);
694 av_freep(&vpad.name);
699 s->fcmul_add = fcmul_add_c;
701 s->fdsp = avpriv_float_dsp_alloc(0);
703 return AVERROR(ENOMEM);
711 static const AVFilterPad afir_inputs[] = {
714 .type = AVMEDIA_TYPE_AUDIO,
717 .type = AVMEDIA_TYPE_AUDIO,
722 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
723 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
724 #define OFFSET(x) offsetof(AudioFIRContext, x)
726 static const AVOption afir_options[] = {
727 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
728 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
729 { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
730 { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
731 { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
732 { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
733 { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
734 { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
735 { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
736 { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
737 { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
738 { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
739 { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
740 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
741 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
742 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
743 { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
744 { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=16}, 16, 32768, AF },
745 { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 16, 32768, AF },
749 AVFILTER_DEFINE_CLASS(afir);
751 AVFilter ff_af_afir = {
753 .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
754 .priv_size = sizeof(AudioFIRContext),
755 .priv_class = &afir_class,
756 .query_formats = query_formats,
758 .activate = activate,
760 .inputs = afir_inputs,
761 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
762 AVFILTER_FLAG_SLICE_THREADS,