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[0];
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 = (const FFTComplex *)seg->coeff->extended_data[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 for (int segment = 0; segment < s->nb_segments; segment++) {
129 AudioFIRSegment *seg = &s->seg[segment];
131 seg->part_index = (seg->part_index + 1) % seg->nb_partitions;
135 if (s->pts != AV_NOPTS_VALUE)
136 s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
141 return ff_filter_frame(outlink, out);
144 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
150 font = avpriv_cga_font, font_height = 8;
152 for (i = 0; txt[i]; i++) {
155 uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
156 for (char_y = 0; char_y < font_height; char_y++) {
157 for (mask = 0x80; mask; mask >>= 1) {
158 if (font[txt[i] * font_height + char_y] & mask)
162 p += pic->linesize[0] - 8 * 4;
167 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
169 int dx = FFABS(x1-x0);
170 int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
171 int err = (dx>dy ? dx : -dy) / 2, e2;
174 AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
176 if (x0 == x1 && y0 == y1)
193 static void draw_response(AVFilterContext *ctx, AVFrame *out)
195 AudioFIRContext *s = ctx->priv;
196 float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
197 float min_delay = FLT_MAX, max_delay = FLT_MIN;
198 int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
202 memset(out->data[0], 0, s->h * out->linesize[0]);
204 phase = av_malloc_array(s->w, sizeof(*phase));
205 mag = av_malloc_array(s->w, sizeof(*mag));
206 delay = av_malloc_array(s->w, sizeof(*delay));
207 if (!mag || !phase || !delay)
210 channel = av_clip(s->ir_channel, 0, s->in[1]->channels - 1);
211 for (i = 0; i < s->w; i++) {
212 const float *src = (const float *)s->in[1]->extended_data[channel];
213 double w = i * M_PI / (s->w - 1);
214 double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
216 for (x = 0; x < s->nb_taps; x++) {
217 real += cos(-x * w) * src[x];
218 imag += sin(-x * w) * src[x];
219 real_num += cos(-x * w) * src[x] * x;
220 imag_num += sin(-x * w) * src[x] * x;
223 mag[i] = hypot(real, imag);
224 phase[i] = atan2(imag, real);
225 div = real * real + imag * imag;
226 delay[i] = (real_num * real + imag_num * imag) / div;
227 min = fminf(min, mag[i]);
228 max = fmaxf(max, mag[i]);
229 min_delay = fminf(min_delay, delay[i]);
230 max_delay = fmaxf(max_delay, delay[i]);
233 for (i = 0; i < s->w; i++) {
234 int ymag = mag[i] / max * (s->h - 1);
235 int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
236 int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
238 ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
239 yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
240 ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
245 prev_yphase = yphase;
247 prev_ydelay = ydelay;
249 draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
250 draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
251 draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
254 prev_yphase = yphase;
255 prev_ydelay = ydelay;
258 if (s->w > 400 && s->h > 100) {
259 drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
260 snprintf(text, sizeof(text), "%.2f", max);
261 drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
263 drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
264 snprintf(text, sizeof(text), "%.2f", min);
265 drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
267 drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
268 snprintf(text, sizeof(text), "%.2f", max_delay);
269 drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
271 drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
272 snprintf(text, sizeof(text), "%.2f", min_delay);
273 drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
282 static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_partitions, int part_size)
284 seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
285 seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
286 if (!seg->rdft || !seg->irdft)
287 return AVERROR(ENOMEM);
289 seg->fft_length = part_size * 4 + 1;
290 seg->part_size = part_size;
291 seg->block_size = FFALIGN(seg->fft_length, 32);
292 seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
293 seg->nb_partitions = nb_partitions;
294 seg->segment_size = part_size * nb_partitions;
296 for (int ch = 0; ch < ctx->inputs[0]->channels; ch++) {
297 seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
298 seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);
299 if (!seg->rdft[ch] || !seg->irdft[ch])
300 return AVERROR(ENOMEM);
303 seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
304 seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
305 seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
306 seg->coeff = ff_get_audio_buffer(ctx->inputs[1], seg->nb_partitions * seg->coeff_size * 2);
307 if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff)
308 return AVERROR(ENOMEM);
313 static int convert_coeffs(AVFilterContext *ctx)
315 AudioFIRContext *s = ctx->priv;
316 int ret, i, ch, n, N;
319 s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1]);
321 return AVERROR(EINVAL);
323 for (n = av_log2(s->minp); (1 << n) < s->nb_taps; n++);
324 N = FFMIN(n, av_log2(s->maxp));
327 ret = init_segment(ctx, &s->seg[0], (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 av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
386 av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
388 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
389 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
391 for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
394 av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
396 for (int segment = 0; segment < s->nb_segments; segment++) {
397 AudioFIRSegment *seg = &s->seg[segment];
398 float *block = (float *)seg->block->extended_data[ch];
399 FFTComplex *coeff = (FFTComplex *)seg->coeff->extended_data[ch];
401 av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
403 for (i = 0; i < seg->nb_partitions; i++) {
404 const float scale = 1.f / seg->part_size;
405 const int toffset = i * seg->part_size;
406 const int coffset = i * seg->coeff_size;
407 const int remaining = s->nb_taps - (i * seg->part_size);
408 const int size = remaining >= seg->part_size ? seg->part_size : remaining;
410 memset(block, 0, sizeof(*block) * seg->fft_length);
411 memcpy(block, time + toffset, size * sizeof(*block));
413 av_rdft_calc(seg->rdft[0], block);
415 coeff[coffset].re = block[0] * scale;
416 coeff[coffset].im = 0;
417 for (n = 1; n < seg->part_size; n++) {
418 coeff[coffset + n].re = block[2 * n] * scale;
419 coeff[coffset + n].im = block[2 * n + 1] * scale;
421 coeff[coffset + seg->part_size].re = block[1] * scale;
422 coeff[coffset + seg->part_size].im = 0;
425 av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
426 av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
427 av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
431 av_frame_free(&s->in[1]);
437 static int check_ir(AVFilterLink *link, AVFrame *frame)
439 AVFilterContext *ctx = link->dst;
440 AudioFIRContext *s = ctx->priv;
441 int nb_taps, max_nb_taps;
443 nb_taps = ff_inlink_queued_samples(link);
444 max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
445 if (nb_taps > max_nb_taps) {
446 av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
447 return AVERROR(EINVAL);
453 static int activate(AVFilterContext *ctx)
455 AudioFIRContext *s = ctx->priv;
456 AVFilterLink *outlink = ctx->outputs[0];
461 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
463 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
464 if (!s->eof_coeffs) {
467 ret = check_ir(ctx->inputs[1], ir);
471 if (ff_outlink_get_status(ctx->inputs[1]) == AVERROR_EOF)
474 if (!s->eof_coeffs) {
475 if (ff_outlink_frame_wanted(ctx->outputs[0]))
476 ff_inlink_request_frame(ctx->inputs[1]);
477 else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
478 ff_inlink_request_frame(ctx->inputs[1]);
483 if (!s->have_coeffs && s->eof_coeffs) {
484 ret = convert_coeffs(ctx);
489 ret = ff_inlink_consume_samples(ctx->inputs[0], s->seg[0].part_size, s->seg[0].part_size, &in);
491 ret = fir_frame(s, in, outlink);
496 if (s->response && s->have_coeffs) {
497 int64_t old_pts = s->video->pts;
498 int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
500 if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
501 s->video->pts = new_pts;
502 return ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
506 if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->seg[0].part_size) {
507 ff_filter_set_ready(ctx, 10);
511 if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
512 if (status == AVERROR_EOF) {
513 ff_outlink_set_status(ctx->outputs[0], status, pts);
515 ff_outlink_set_status(ctx->outputs[1], status, pts);
520 if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
521 !ff_outlink_get_status(ctx->inputs[0])) {
522 ff_inlink_request_frame(ctx->inputs[0]);
527 ff_outlink_frame_wanted(ctx->outputs[1]) &&
528 !ff_outlink_get_status(ctx->inputs[0])) {
529 ff_inlink_request_frame(ctx->inputs[0]);
533 return FFERROR_NOT_READY;
536 static int query_formats(AVFilterContext *ctx)
538 AudioFIRContext *s = ctx->priv;
539 AVFilterFormats *formats;
540 AVFilterChannelLayouts *layouts;
541 static const enum AVSampleFormat sample_fmts[] = {
545 static const enum AVPixelFormat pix_fmts[] = {
552 AVFilterLink *videolink = ctx->outputs[1];
553 formats = ff_make_format_list(pix_fmts);
554 if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
558 layouts = ff_all_channel_counts();
560 return AVERROR(ENOMEM);
563 ret = ff_set_common_channel_layouts(ctx, layouts);
567 AVFilterChannelLayouts *mono = NULL;
569 ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
573 if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts)) < 0)
575 if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
577 if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[1]->out_channel_layouts)) < 0)
581 formats = ff_make_format_list(sample_fmts);
582 if ((ret = ff_set_common_formats(ctx, formats)) < 0)
585 formats = ff_all_samplerates();
586 return ff_set_common_samplerates(ctx, formats);
589 static int config_output(AVFilterLink *outlink)
591 AVFilterContext *ctx = outlink->src;
592 AudioFIRContext *s = ctx->priv;
594 s->one2many = ctx->inputs[1]->channels == 1;
595 outlink->sample_rate = ctx->inputs[0]->sample_rate;
596 outlink->time_base = ctx->inputs[0]->time_base;
597 outlink->channel_layout = ctx->inputs[0]->channel_layout;
598 outlink->channels = ctx->inputs[0]->channels;
600 s->nb_channels = outlink->channels;
601 s->nb_coef_channels = ctx->inputs[1]->channels;
602 s->pts = AV_NOPTS_VALUE;
607 static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
609 AudioFIRContext *s = ctx->priv;
612 for (int ch = 0; ch < s->nb_channels; ch++) {
613 av_rdft_end(seg->rdft[ch]);
616 av_freep(&seg->rdft);
619 for (int ch = 0; ch < s->nb_channels; ch++) {
620 av_rdft_end(seg->irdft[ch]);
623 av_freep(&seg->irdft);
625 av_frame_free(&seg->block);
626 av_frame_free(&seg->sum);
627 av_frame_free(&seg->buffer);
628 av_frame_free(&seg->coeff);
631 static av_cold void uninit(AVFilterContext *ctx)
633 AudioFIRContext *s = ctx->priv;
635 for (int i = 0; i < s->nb_segments; i++) {
636 uninit_segment(ctx, &s->seg[i]);
640 av_frame_free(&s->in[1]);
642 for (int i = 0; i < ctx->nb_outputs; i++)
643 av_freep(&ctx->output_pads[i].name);
644 av_frame_free(&s->video);
647 static int config_video(AVFilterLink *outlink)
649 AVFilterContext *ctx = outlink->src;
650 AudioFIRContext *s = ctx->priv;
652 outlink->sample_aspect_ratio = (AVRational){1,1};
655 outlink->frame_rate = s->frame_rate;
656 outlink->time_base = av_inv_q(outlink->frame_rate);
658 av_frame_free(&s->video);
659 s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
661 return AVERROR(ENOMEM);
666 static av_cold int init(AVFilterContext *ctx)
668 AudioFIRContext *s = ctx->priv;
669 AVFilterPad pad, vpad;
673 .name = av_strdup("default"),
674 .type = AVMEDIA_TYPE_AUDIO,
675 .config_props = config_output,
679 return AVERROR(ENOMEM);
682 vpad = (AVFilterPad){
683 .name = av_strdup("filter_response"),
684 .type = AVMEDIA_TYPE_VIDEO,
685 .config_props = config_video,
688 return AVERROR(ENOMEM);
691 ret = ff_insert_outpad(ctx, 0, &pad);
698 ret = ff_insert_outpad(ctx, 1, &vpad);
700 av_freep(&vpad.name);
705 s->fcmul_add = fcmul_add_c;
707 s->fdsp = avpriv_float_dsp_alloc(0);
709 return AVERROR(ENOMEM);
717 static const AVFilterPad afir_inputs[] = {
720 .type = AVMEDIA_TYPE_AUDIO,
723 .type = AVMEDIA_TYPE_AUDIO,
728 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
729 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
730 #define OFFSET(x) offsetof(AudioFIRContext, x)
732 static const AVOption afir_options[] = {
733 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
734 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
735 { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
736 { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
737 { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
738 { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
739 { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
740 { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
741 { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
742 { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
743 { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
744 { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
745 { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
746 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
747 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
748 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
749 { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
750 { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=16}, 16, 32768, AF },
751 { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 16, 32768, AF },
755 AVFILTER_DEFINE_CLASS(afir);
757 AVFilter ff_af_afir = {
759 .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
760 .priv_size = sizeof(AudioFIRContext),
761 .priv_class = &afir_class,
762 .query_formats = query_formats,
764 .activate = activate,
766 .inputs = afir_inputs,
767 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
768 AVFILTER_FLAG_SLICE_THREADS,