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 const float *src = (const float *)s->in[0]->extended_data[ch];
63 float *sum = (float *)s->seg.sum->extended_data[ch];
65 float *block, *dst, *ptr;
68 memset(sum, 0, sizeof(*sum) * s->seg.fft_length);
69 block = (float *)s->seg.block->extended_data[ch] + s->seg.part_index * s->seg.block_size;
70 memset(block, 0, sizeof(*block) * s->seg.fft_length);
72 s->fdsp->vector_fmul_scalar(block, src, s->dry_gain, FFALIGN(out->nb_samples, 4));
75 av_rdft_calc(s->seg.rdft[ch], block);
76 block[2 * s->seg.part_size] = block[1];
79 j = s->seg.part_index;
81 for (i = 0; i < s->seg.nb_partitions; i++) {
82 const int coffset = i * s->seg.coeff_size;
83 const float *block = (const float *)s->seg.block->extended_data[ch] + j * s->seg.block_size;
84 const FFTComplex *coeff = s->seg.coeff[ch * !s->one2many] + coffset;
86 s->fcmul_add(sum, block, (const float *)coeff, s->seg.part_size);
89 j = s->seg.nb_partitions;
93 sum[1] = sum[2 * s->seg.part_size];
94 av_rdft_calc(s->seg.irdft[ch], sum);
96 dst = (float *)s->seg.buffer->extended_data[ch];
97 for (n = 0; n < s->seg.part_size; n++) {
101 ptr = (float *)out->extended_data[ch];
102 s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4));
105 dst = (float *)s->seg.buffer->extended_data[ch];
106 memcpy(dst, sum + s->seg.part_size, s->seg.part_size * sizeof(*dst));
111 static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
113 AVFilterContext *ctx = outlink->src;
116 out = ff_get_audio_buffer(outlink, in->nb_samples);
119 return AVERROR(ENOMEM);
122 if (s->pts == AV_NOPTS_VALUE)
125 ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
127 s->seg.part_index = (s->seg.part_index + 1) % s->seg.nb_partitions;
130 if (s->pts != AV_NOPTS_VALUE)
131 s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
136 return ff_filter_frame(outlink, out);
139 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
145 font = avpriv_cga_font, font_height = 8;
147 for (i = 0; txt[i]; i++) {
150 uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
151 for (char_y = 0; char_y < font_height; char_y++) {
152 for (mask = 0x80; mask; mask >>= 1) {
153 if (font[txt[i] * font_height + char_y] & mask)
157 p += pic->linesize[0] - 8 * 4;
162 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
164 int dx = FFABS(x1-x0);
165 int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
166 int err = (dx>dy ? dx : -dy) / 2, e2;
169 AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
171 if (x0 == x1 && y0 == y1)
188 static void draw_response(AVFilterContext *ctx, AVFrame *out)
190 AudioFIRContext *s = ctx->priv;
191 float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
192 float min_delay = FLT_MAX, max_delay = FLT_MIN;
193 int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
197 memset(out->data[0], 0, s->h * out->linesize[0]);
199 phase = av_malloc_array(s->w, sizeof(*phase));
200 mag = av_malloc_array(s->w, sizeof(*mag));
201 delay = av_malloc_array(s->w, sizeof(*delay));
202 if (!mag || !phase || !delay)
205 channel = av_clip(s->ir_channel, 0, s->in[1]->channels - 1);
206 for (i = 0; i < s->w; i++) {
207 const float *src = (const float *)s->in[1]->extended_data[channel];
208 double w = i * M_PI / (s->w - 1);
209 double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
211 for (x = 0; x < s->nb_taps; x++) {
212 real += cos(-x * w) * src[x];
213 imag += sin(-x * w) * src[x];
214 real_num += cos(-x * w) * src[x] * x;
215 imag_num += sin(-x * w) * src[x] * x;
218 mag[i] = hypot(real, imag);
219 phase[i] = atan2(imag, real);
220 div = real * real + imag * imag;
221 delay[i] = (real_num * real + imag_num * imag) / div;
222 min = fminf(min, mag[i]);
223 max = fmaxf(max, mag[i]);
224 min_delay = fminf(min_delay, delay[i]);
225 max_delay = fmaxf(max_delay, delay[i]);
228 for (i = 0; i < s->w; i++) {
229 int ymag = mag[i] / max * (s->h - 1);
230 int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
231 int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
233 ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
234 yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
235 ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
240 prev_yphase = yphase;
242 prev_ydelay = ydelay;
244 draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
245 draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
246 draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
249 prev_yphase = yphase;
250 prev_ydelay = ydelay;
253 if (s->w > 400 && s->h > 100) {
254 drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
255 snprintf(text, sizeof(text), "%.2f", max);
256 drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
258 drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
259 snprintf(text, sizeof(text), "%.2f", min);
260 drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
262 drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
263 snprintf(text, sizeof(text), "%.2f", max_delay);
264 drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
266 drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
267 snprintf(text, sizeof(text), "%.2f", min_delay);
268 drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
277 static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_partitions, int part_size)
279 seg->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*seg->coeff));
280 seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
281 seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
282 if (!seg->coeff || !seg->rdft || !seg->irdft)
283 return AVERROR(ENOMEM);
285 seg->fft_length = part_size * 4 + 1;
286 seg->part_size = part_size;
287 seg->block_size = FFALIGN(seg->fft_length, 32);
288 seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
289 seg->nb_partitions = nb_partitions;
291 for (int ch = 0; ch < ctx->inputs[1]->channels; ch++) {
292 seg->coeff[ch] = av_calloc(seg->nb_partitions * seg->coeff_size, sizeof(**seg->coeff));
294 return AVERROR(ENOMEM);
297 for (int ch = 0; ch < ctx->inputs[0]->channels; ch++) {
298 seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
299 seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);
300 if (!seg->rdft[ch] || !seg->irdft[ch])
301 return AVERROR(ENOMEM);
304 seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
305 seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
306 seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
307 if (!seg->buffer || !seg->sum || !seg->block)
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));
326 ret = init_segment(ctx, &s->seg, (s->nb_taps + (1 << N) - 1) / (1 << N), 1 << N);
330 ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]);
337 draw_response(ctx, s->video);
346 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
347 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
349 for (i = 0; i < s->nb_taps; i++)
350 power += FFABS(time[i]);
352 s->gain = ctx->inputs[1]->channels / power;
355 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
356 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
358 for (i = 0; i < s->nb_taps; i++)
361 s->gain = ctx->inputs[1]->channels / power;
364 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
365 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
367 for (i = 0; i < s->nb_taps; i++)
368 power += time[i] * time[i];
370 s->gain = sqrtf(ch / power);
376 s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
377 av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
378 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
379 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
381 s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
384 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
385 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
386 float *block = (float *)s->seg.block->extended_data[ch];
387 FFTComplex *coeff = s->seg.coeff[ch];
389 for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
392 for (i = 0; i < s->seg.nb_partitions; i++) {
393 const float scale = 1.f / s->seg.part_size;
394 const int toffset = i * s->seg.part_size;
395 const int coffset = i * s->seg.coeff_size;
396 const int remaining = s->nb_taps - (i * s->seg.part_size);
397 const int size = remaining >= s->seg.part_size ? s->seg.part_size : remaining;
399 memset(block, 0, sizeof(*block) * s->seg.fft_length);
400 memcpy(block, time + toffset, size * sizeof(*block));
402 av_rdft_calc(s->seg.rdft[0], block);
404 coeff[coffset].re = block[0] * scale;
405 coeff[coffset].im = 0;
406 for (n = 1; n < s->seg.part_size; n++) {
407 coeff[coffset + n].re = block[2 * n] * scale;
408 coeff[coffset + n].im = block[2 * n + 1] * scale;
410 coeff[coffset + s->seg.part_size].re = block[1] * scale;
411 coeff[coffset + s->seg.part_size].im = 0;
415 av_frame_free(&s->in[1]);
416 av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
417 av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->seg.nb_partitions);
418 av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->seg.part_size);
419 av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->seg.fft_length);
426 static int check_ir(AVFilterLink *link, AVFrame *frame)
428 AVFilterContext *ctx = link->dst;
429 AudioFIRContext *s = ctx->priv;
430 int nb_taps, max_nb_taps;
432 nb_taps = ff_inlink_queued_samples(link);
433 max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
434 if (nb_taps > max_nb_taps) {
435 av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
436 return AVERROR(EINVAL);
442 static int activate(AVFilterContext *ctx)
444 AudioFIRContext *s = ctx->priv;
445 AVFilterLink *outlink = ctx->outputs[0];
450 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
452 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
453 if (!s->eof_coeffs) {
456 ret = check_ir(ctx->inputs[1], ir);
460 if (ff_outlink_get_status(ctx->inputs[1]) == AVERROR_EOF)
463 if (!s->eof_coeffs) {
464 if (ff_outlink_frame_wanted(ctx->outputs[0]))
465 ff_inlink_request_frame(ctx->inputs[1]);
466 else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
467 ff_inlink_request_frame(ctx->inputs[1]);
472 if (!s->have_coeffs && s->eof_coeffs) {
473 ret = convert_coeffs(ctx);
478 ret = ff_inlink_consume_samples(ctx->inputs[0], s->seg.part_size, s->seg.part_size, &in);
480 ret = fir_frame(s, in, outlink);
485 if (s->response && s->have_coeffs) {
486 int64_t old_pts = s->video->pts;
487 int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
489 if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
490 s->video->pts = new_pts;
491 return ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
495 if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->seg.part_size) {
496 ff_filter_set_ready(ctx, 10);
500 if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
501 if (status == AVERROR_EOF) {
502 ff_outlink_set_status(ctx->outputs[0], status, pts);
504 ff_outlink_set_status(ctx->outputs[1], status, pts);
509 if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
510 !ff_outlink_get_status(ctx->inputs[0])) {
511 ff_inlink_request_frame(ctx->inputs[0]);
516 ff_outlink_frame_wanted(ctx->outputs[1]) &&
517 !ff_outlink_get_status(ctx->inputs[0])) {
518 ff_inlink_request_frame(ctx->inputs[0]);
522 return FFERROR_NOT_READY;
525 static int query_formats(AVFilterContext *ctx)
527 AudioFIRContext *s = ctx->priv;
528 AVFilterFormats *formats;
529 AVFilterChannelLayouts *layouts;
530 static const enum AVSampleFormat sample_fmts[] = {
534 static const enum AVPixelFormat pix_fmts[] = {
541 AVFilterLink *videolink = ctx->outputs[1];
542 formats = ff_make_format_list(pix_fmts);
543 if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
547 layouts = ff_all_channel_counts();
549 return AVERROR(ENOMEM);
552 ret = ff_set_common_channel_layouts(ctx, layouts);
556 AVFilterChannelLayouts *mono = NULL;
558 ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
562 if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts)) < 0)
564 if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
566 if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[1]->out_channel_layouts)) < 0)
570 formats = ff_make_format_list(sample_fmts);
571 if ((ret = ff_set_common_formats(ctx, formats)) < 0)
574 formats = ff_all_samplerates();
575 return ff_set_common_samplerates(ctx, formats);
578 static int config_output(AVFilterLink *outlink)
580 AVFilterContext *ctx = outlink->src;
581 AudioFIRContext *s = ctx->priv;
583 s->one2many = ctx->inputs[1]->channels == 1;
584 outlink->sample_rate = ctx->inputs[0]->sample_rate;
585 outlink->time_base = ctx->inputs[0]->time_base;
586 outlink->channel_layout = ctx->inputs[0]->channel_layout;
587 outlink->channels = ctx->inputs[0]->channels;
589 s->nb_channels = outlink->channels;
590 s->nb_coef_channels = ctx->inputs[1]->channels;
591 s->pts = AV_NOPTS_VALUE;
596 static av_cold void uninit(AVFilterContext *ctx)
598 AudioFIRContext *s = ctx->priv;
602 for (ch = 0; ch < s->nb_coef_channels; ch++) {
603 av_freep(&s->seg.coeff[ch]);
606 av_freep(&s->seg.coeff);
609 for (ch = 0; ch < s->nb_channels; ch++) {
610 av_rdft_end(s->seg.rdft[ch]);
613 av_freep(&s->seg.rdft);
616 for (ch = 0; ch < s->nb_channels; ch++) {
617 av_rdft_end(s->seg.irdft[ch]);
620 av_freep(&s->seg.irdft);
622 av_frame_free(&s->in[1]);
624 av_frame_free(&s->seg.block);
625 av_frame_free(&s->seg.sum);
626 av_frame_free(&s->seg.buffer);
630 for (int i = 0; i < ctx->nb_outputs; i++)
631 av_freep(&ctx->output_pads[i].name);
632 av_frame_free(&s->video);
635 static int config_video(AVFilterLink *outlink)
637 AVFilterContext *ctx = outlink->src;
638 AudioFIRContext *s = ctx->priv;
640 outlink->sample_aspect_ratio = (AVRational){1,1};
643 outlink->frame_rate = s->frame_rate;
644 outlink->time_base = av_inv_q(outlink->frame_rate);
646 av_frame_free(&s->video);
647 s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
649 return AVERROR(ENOMEM);
654 static av_cold int init(AVFilterContext *ctx)
656 AudioFIRContext *s = ctx->priv;
657 AVFilterPad pad, vpad;
661 .name = av_strdup("default"),
662 .type = AVMEDIA_TYPE_AUDIO,
663 .config_props = config_output,
667 return AVERROR(ENOMEM);
670 vpad = (AVFilterPad){
671 .name = av_strdup("filter_response"),
672 .type = AVMEDIA_TYPE_VIDEO,
673 .config_props = config_video,
676 return AVERROR(ENOMEM);
679 ret = ff_insert_outpad(ctx, 0, &pad);
686 ret = ff_insert_outpad(ctx, 1, &vpad);
688 av_freep(&vpad.name);
693 s->fcmul_add = fcmul_add_c;
695 s->fdsp = avpriv_float_dsp_alloc(0);
697 return AVERROR(ENOMEM);
705 static const AVFilterPad afir_inputs[] = {
708 .type = AVMEDIA_TYPE_AUDIO,
711 .type = AVMEDIA_TYPE_AUDIO,
716 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
717 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
718 #define OFFSET(x) offsetof(AudioFIRContext, x)
720 static const AVOption afir_options[] = {
721 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
722 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
723 { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
724 { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
725 { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
726 { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
727 { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
728 { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
729 { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
730 { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
731 { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
732 { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
733 { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
734 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
735 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
736 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
737 { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
738 { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=16}, 16, 32768, AF },
739 { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 16, 32768, AF },
743 AVFILTER_DEFINE_CLASS(afir);
745 AVFilter ff_af_afir = {
747 .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
748 .priv_size = sizeof(AudioFIRContext),
749 .priv_class = &afir_class,
750 .query_formats = query_formats,
752 .activate = activate,
754 .inputs = afir_inputs,
755 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
756 AVFILTER_FLAG_SLICE_THREADS,