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 convert_coeffs(AVFilterContext *ctx)
279 AudioFIRContext *s = ctx->priv;
280 int ret, i, ch, n, N;
283 s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1]);
285 return AVERROR(EINVAL);
287 for (n = av_log2(s->minp); (1 << n) < s->nb_taps; n++);
288 N = FFMIN(n, av_log2(s->maxp));
290 s->seg.coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->seg.coeff));
291 s->seg.rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*s->seg.rdft));
292 s->seg.irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*s->seg.irdft));
293 if (!s->seg.coeff || !s->seg.rdft || !s->seg.irdft)
294 return AVERROR(ENOMEM);
296 s->seg.fft_length = (1 << (N + 1)) + 1;
297 s->seg.part_size = 1 << (N - 1);
298 s->seg.block_size = FFALIGN(s->seg.fft_length, 32);
299 s->seg.coeff_size = FFALIGN(s->seg.part_size + 1, 32);
300 s->seg.nb_partitions = (s->nb_taps + s->seg.part_size - 1) / s->seg.part_size;
302 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
303 s->seg.coeff[ch] = av_calloc(s->seg.nb_partitions * s->seg.coeff_size, sizeof(**s->seg.coeff));
304 if (!s->seg.coeff[ch])
305 return AVERROR(ENOMEM);
308 for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
309 s->seg.rdft[ch] = av_rdft_init(N, DFT_R2C);
310 s->seg.irdft[ch] = av_rdft_init(N, IDFT_C2R);
311 if (!s->seg.rdft[ch] || !s->seg.irdft[ch])
312 return AVERROR(ENOMEM);
315 s->seg.sum = ff_get_audio_buffer(ctx->inputs[0], s->seg.fft_length);
316 s->seg.block = ff_get_audio_buffer(ctx->inputs[0], s->seg.nb_partitions * s->seg.block_size);
317 s->seg.buffer = ff_get_audio_buffer(ctx->inputs[0], s->seg.part_size);
318 if (!s->seg.buffer || !s->seg.sum || !s->seg.block)
319 return AVERROR(ENOMEM);
321 ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]);
328 draw_response(ctx, s->video);
337 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
338 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
340 for (i = 0; i < s->nb_taps; i++)
341 power += FFABS(time[i]);
343 s->gain = ctx->inputs[1]->channels / power;
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++)
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++)
359 power += time[i] * time[i];
361 s->gain = sqrtf(ch / power);
367 s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
368 av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
369 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
370 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
372 s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
375 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
376 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
377 float *block = (float *)s->seg.block->extended_data[ch];
378 FFTComplex *coeff = s->seg.coeff[ch];
380 for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
383 for (i = 0; i < s->seg.nb_partitions; i++) {
384 const float scale = 1.f / s->seg.part_size;
385 const int toffset = i * s->seg.part_size;
386 const int coffset = i * s->seg.coeff_size;
387 const int remaining = s->nb_taps - (i * s->seg.part_size);
388 const int size = remaining >= s->seg.part_size ? s->seg.part_size : remaining;
390 memset(block, 0, sizeof(*block) * s->seg.fft_length);
391 memcpy(block, time + toffset, size * sizeof(*block));
393 av_rdft_calc(s->seg.rdft[0], block);
395 coeff[coffset].re = block[0] * scale;
396 coeff[coffset].im = 0;
397 for (n = 1; n < s->seg.part_size; n++) {
398 coeff[coffset + n].re = block[2 * n] * scale;
399 coeff[coffset + n].im = block[2 * n + 1] * scale;
401 coeff[coffset + s->seg.part_size].re = block[1] * scale;
402 coeff[coffset + s->seg.part_size].im = 0;
406 av_frame_free(&s->in[1]);
407 av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
408 av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->seg.nb_partitions);
409 av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->seg.part_size);
410 av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->seg.fft_length);
417 static int check_ir(AVFilterLink *link, AVFrame *frame)
419 AVFilterContext *ctx = link->dst;
420 AudioFIRContext *s = ctx->priv;
421 int nb_taps, max_nb_taps;
423 nb_taps = ff_inlink_queued_samples(link);
424 max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
425 if (nb_taps > max_nb_taps) {
426 av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
427 return AVERROR(EINVAL);
433 static int activate(AVFilterContext *ctx)
435 AudioFIRContext *s = ctx->priv;
436 AVFilterLink *outlink = ctx->outputs[0];
441 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
443 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
444 if (!s->eof_coeffs) {
447 ret = check_ir(ctx->inputs[1], ir);
451 if (ff_outlink_get_status(ctx->inputs[1]) == AVERROR_EOF)
454 if (!s->eof_coeffs) {
455 if (ff_outlink_frame_wanted(ctx->outputs[0]))
456 ff_inlink_request_frame(ctx->inputs[1]);
457 else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
458 ff_inlink_request_frame(ctx->inputs[1]);
463 if (!s->have_coeffs && s->eof_coeffs) {
464 ret = convert_coeffs(ctx);
469 ret = ff_inlink_consume_samples(ctx->inputs[0], s->seg.part_size, s->seg.part_size, &in);
471 ret = fir_frame(s, in, outlink);
476 if (s->response && s->have_coeffs) {
477 int64_t old_pts = s->video->pts;
478 int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
480 if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
481 s->video->pts = new_pts;
482 return ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
486 if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->seg.part_size) {
487 ff_filter_set_ready(ctx, 10);
491 if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
492 if (status == AVERROR_EOF) {
493 ff_outlink_set_status(ctx->outputs[0], status, pts);
495 ff_outlink_set_status(ctx->outputs[1], status, pts);
500 if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
501 !ff_outlink_get_status(ctx->inputs[0])) {
502 ff_inlink_request_frame(ctx->inputs[0]);
507 ff_outlink_frame_wanted(ctx->outputs[1]) &&
508 !ff_outlink_get_status(ctx->inputs[0])) {
509 ff_inlink_request_frame(ctx->inputs[0]);
513 return FFERROR_NOT_READY;
516 static int query_formats(AVFilterContext *ctx)
518 AudioFIRContext *s = ctx->priv;
519 AVFilterFormats *formats;
520 AVFilterChannelLayouts *layouts;
521 static const enum AVSampleFormat sample_fmts[] = {
525 static const enum AVPixelFormat pix_fmts[] = {
532 AVFilterLink *videolink = ctx->outputs[1];
533 formats = ff_make_format_list(pix_fmts);
534 if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
538 layouts = ff_all_channel_counts();
540 return AVERROR(ENOMEM);
543 ret = ff_set_common_channel_layouts(ctx, layouts);
547 AVFilterChannelLayouts *mono = NULL;
549 ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
553 if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts)) < 0)
555 if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
557 if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[1]->out_channel_layouts)) < 0)
561 formats = ff_make_format_list(sample_fmts);
562 if ((ret = ff_set_common_formats(ctx, formats)) < 0)
565 formats = ff_all_samplerates();
566 return ff_set_common_samplerates(ctx, formats);
569 static int config_output(AVFilterLink *outlink)
571 AVFilterContext *ctx = outlink->src;
572 AudioFIRContext *s = ctx->priv;
574 s->one2many = ctx->inputs[1]->channels == 1;
575 outlink->sample_rate = ctx->inputs[0]->sample_rate;
576 outlink->time_base = ctx->inputs[0]->time_base;
577 outlink->channel_layout = ctx->inputs[0]->channel_layout;
578 outlink->channels = ctx->inputs[0]->channels;
580 s->nb_channels = outlink->channels;
581 s->nb_coef_channels = ctx->inputs[1]->channels;
582 s->pts = AV_NOPTS_VALUE;
587 static av_cold void uninit(AVFilterContext *ctx)
589 AudioFIRContext *s = ctx->priv;
593 for (ch = 0; ch < s->nb_coef_channels; ch++) {
594 av_freep(&s->seg.coeff[ch]);
597 av_freep(&s->seg.coeff);
600 for (ch = 0; ch < s->nb_channels; ch++) {
601 av_rdft_end(s->seg.rdft[ch]);
604 av_freep(&s->seg.rdft);
607 for (ch = 0; ch < s->nb_channels; ch++) {
608 av_rdft_end(s->seg.irdft[ch]);
611 av_freep(&s->seg.irdft);
613 av_frame_free(&s->in[1]);
615 av_frame_free(&s->seg.block);
616 av_frame_free(&s->seg.sum);
617 av_frame_free(&s->seg.buffer);
621 for (int i = 0; i < ctx->nb_outputs; i++)
622 av_freep(&ctx->output_pads[i].name);
623 av_frame_free(&s->video);
626 static int config_video(AVFilterLink *outlink)
628 AVFilterContext *ctx = outlink->src;
629 AudioFIRContext *s = ctx->priv;
631 outlink->sample_aspect_ratio = (AVRational){1,1};
634 outlink->frame_rate = s->frame_rate;
635 outlink->time_base = av_inv_q(outlink->frame_rate);
637 av_frame_free(&s->video);
638 s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
640 return AVERROR(ENOMEM);
645 static av_cold int init(AVFilterContext *ctx)
647 AudioFIRContext *s = ctx->priv;
648 AVFilterPad pad, vpad;
652 .name = av_strdup("default"),
653 .type = AVMEDIA_TYPE_AUDIO,
654 .config_props = config_output,
658 return AVERROR(ENOMEM);
661 vpad = (AVFilterPad){
662 .name = av_strdup("filter_response"),
663 .type = AVMEDIA_TYPE_VIDEO,
664 .config_props = config_video,
667 return AVERROR(ENOMEM);
670 ret = ff_insert_outpad(ctx, 0, &pad);
677 ret = ff_insert_outpad(ctx, 1, &vpad);
679 av_freep(&vpad.name);
684 s->fcmul_add = fcmul_add_c;
686 s->fdsp = avpriv_float_dsp_alloc(0);
688 return AVERROR(ENOMEM);
696 static const AVFilterPad afir_inputs[] = {
699 .type = AVMEDIA_TYPE_AUDIO,
702 .type = AVMEDIA_TYPE_AUDIO,
707 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
708 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
709 #define OFFSET(x) offsetof(AudioFIRContext, x)
711 static const AVOption afir_options[] = {
712 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
713 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
714 { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
715 { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
716 { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
717 { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
718 { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
719 { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
720 { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
721 { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
722 { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
723 { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
724 { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
725 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
726 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
727 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
728 { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
729 { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=16}, 16, 65536, AF },
730 { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=65536}, 16, 65536, AF },
734 AVFILTER_DEFINE_CLASS(afir);
736 AVFilter ff_af_afir = {
738 .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
739 .priv_size = sizeof(AudioFIRContext),
740 .priv_class = &afir_class,
741 .query_formats = query_formats,
743 .activate = activate,
745 .inputs = afir_inputs,
746 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
747 AVFILTER_FLAG_SLICE_THREADS,