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)
61 AudioFIRContext *s = ctx->priv;
62 const float *in = (const float *)s->in[0]->extended_data[ch];
64 float *block, *buf, *ptr = (float *)out->extended_data[ch];
67 for (int segment = 0; segment < s->nb_segments; segment++) {
68 AudioFIRSegment *seg = &s->seg[segment];
69 float *src = (float *)seg->input->extended_data[ch];
70 float *dst = (float *)seg->output->extended_data[ch];
71 float *sum = (float *)seg->sum->extended_data[ch];
73 s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(out->nb_samples, 4));
76 seg->output_offset[ch] += s->min_part_size;
77 if (seg->output_offset[ch] == seg->part_size) {
78 seg->output_offset[ch] = 0;
80 memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
82 dst += seg->output_offset[ch];
83 for (n = 0; n < out->nb_samples; n++) {
89 memset(sum, 0, sizeof(*sum) * seg->fft_length);
90 block = (float *)seg->block->extended_data[ch] + seg->part_index[ch] * seg->block_size;
91 memset(block + seg->part_size, 0, sizeof(*block) * (seg->fft_length - seg->part_size));
93 memcpy(block, src, sizeof(*src) * seg->part_size);
95 av_rdft_calc(seg->rdft[ch], block);
96 block[2 * seg->part_size] = block[1];
99 j = seg->part_index[ch];
101 for (i = 0; i < seg->nb_partitions; i++) {
102 const int coffset = j * seg->coeff_size;
103 const float *block = (const float *)seg->block->extended_data[ch] + i * seg->block_size;
104 const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
106 s->fcmul_add(sum, block, (const float *)coeff, seg->part_size);
109 j = seg->nb_partitions;
113 sum[1] = sum[2 * seg->part_size];
114 av_rdft_calc(seg->irdft[ch], sum);
116 buf = (float *)seg->buffer->extended_data[ch];
117 for (n = 0; n < seg->part_size; n++) {
121 memcpy(dst, buf, seg->part_size * sizeof(*dst));
123 buf = (float *)seg->buffer->extended_data[ch];
124 memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf));
126 seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
128 memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
130 for (n = 0; n < out->nb_samples; n++) {
135 s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(out->nb_samples, 4));
141 static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
144 const int start = (out->channels * jobnr) / nb_jobs;
145 const int end = (out->channels * (jobnr+1)) / nb_jobs;
147 for (int ch = start; ch < end; ch++) {
148 fir_channel(ctx, out, ch);
154 static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
156 AVFilterContext *ctx = outlink->src;
159 out = ff_get_audio_buffer(outlink, in->nb_samples);
162 return AVERROR(ENOMEM);
165 if (s->pts == AV_NOPTS_VALUE)
168 ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels,
169 ff_filter_get_nb_threads(ctx)));
172 if (s->pts != AV_NOPTS_VALUE)
173 s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
178 return ff_filter_frame(outlink, out);
181 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
187 font = avpriv_cga_font, font_height = 8;
189 for (i = 0; txt[i]; i++) {
192 uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
193 for (char_y = 0; char_y < font_height; char_y++) {
194 for (mask = 0x80; mask; mask >>= 1) {
195 if (font[txt[i] * font_height + char_y] & mask)
199 p += pic->linesize[0] - 8 * 4;
204 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
206 int dx = FFABS(x1-x0);
207 int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
208 int err = (dx>dy ? dx : -dy) / 2, e2;
211 AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
213 if (x0 == x1 && y0 == y1)
230 static void draw_response(AVFilterContext *ctx, AVFrame *out)
232 AudioFIRContext *s = ctx->priv;
233 float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
234 float min_delay = FLT_MAX, max_delay = FLT_MIN;
235 int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
239 memset(out->data[0], 0, s->h * out->linesize[0]);
241 phase = av_malloc_array(s->w, sizeof(*phase));
242 mag = av_malloc_array(s->w, sizeof(*mag));
243 delay = av_malloc_array(s->w, sizeof(*delay));
244 if (!mag || !phase || !delay)
247 channel = av_clip(s->ir_channel, 0, s->in[1]->channels - 1);
248 for (i = 0; i < s->w; i++) {
249 const float *src = (const float *)s->in[1]->extended_data[channel];
250 double w = i * M_PI / (s->w - 1);
251 double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
253 for (x = 0; x < s->nb_taps; x++) {
254 real += cos(-x * w) * src[x];
255 imag += sin(-x * w) * src[x];
256 real_num += cos(-x * w) * src[x] * x;
257 imag_num += sin(-x * w) * src[x] * x;
260 mag[i] = hypot(real, imag);
261 phase[i] = atan2(imag, real);
262 div = real * real + imag * imag;
263 delay[i] = (real_num * real + imag_num * imag) / div;
264 min = fminf(min, mag[i]);
265 max = fmaxf(max, mag[i]);
266 min_delay = fminf(min_delay, delay[i]);
267 max_delay = fmaxf(max_delay, delay[i]);
270 for (i = 0; i < s->w; i++) {
271 int ymag = mag[i] / max * (s->h - 1);
272 int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
273 int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
275 ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
276 yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
277 ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
282 prev_yphase = yphase;
284 prev_ydelay = ydelay;
286 draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
287 draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
288 draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
291 prev_yphase = yphase;
292 prev_ydelay = ydelay;
295 if (s->w > 400 && s->h > 100) {
296 drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
297 snprintf(text, sizeof(text), "%.2f", max);
298 drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
300 drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
301 snprintf(text, sizeof(text), "%.2f", min);
302 drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
304 drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
305 snprintf(text, sizeof(text), "%.2f", max_delay);
306 drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
308 drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
309 snprintf(text, sizeof(text), "%.2f", min_delay);
310 drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
319 static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
320 int offset, int nb_partitions, int part_size)
322 AudioFIRContext *s = ctx->priv;
324 seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
325 seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
326 if (!seg->rdft || !seg->irdft)
327 return AVERROR(ENOMEM);
329 seg->fft_length = part_size * 2 + 1;
330 seg->part_size = part_size;
331 seg->block_size = FFALIGN(seg->fft_length, 32);
332 seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
333 seg->nb_partitions = nb_partitions;
334 seg->input_size = offset + s->min_part_size;
335 seg->input_offset = offset;
337 seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index));
338 seg->output_offset = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->output_offset));
339 if (!seg->part_index || !seg->output_offset)
340 return AVERROR(ENOMEM);
342 for (int ch = 0; ch < ctx->inputs[0]->channels; ch++) {
343 seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
344 seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);
345 if (!seg->rdft[ch] || !seg->irdft[ch])
346 return AVERROR(ENOMEM);
349 seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
350 seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
351 seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
352 seg->coeff = ff_get_audio_buffer(ctx->inputs[1], seg->nb_partitions * seg->coeff_size * 2);
353 seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
354 seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
355 if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff || !seg->input || !seg->output)
356 return AVERROR(ENOMEM);
361 static int convert_coeffs(AVFilterContext *ctx)
363 AudioFIRContext *s = ctx->priv;
364 int left, offset = 0, part_size, max_part_size;
368 s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1]);
370 return AVERROR(EINVAL);
372 if (s->minp > s->maxp) {
377 part_size = 1 << av_log2(s->minp);
378 max_part_size = 1 << av_log2(s->maxp);
380 s->min_part_size = part_size;
382 for (i = 0; left > 0; i++) {
383 int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);
384 int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
386 s->nb_segments = i + 1;
387 ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);
390 offset += nb_partitions * part_size;
391 left -= nb_partitions * part_size;
393 part_size = FFMIN(part_size, max_part_size);
396 ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]);
403 draw_response(ctx, s->video);
412 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
413 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
415 for (i = 0; i < s->nb_taps; i++)
416 power += FFABS(time[i]);
418 s->gain = ctx->inputs[1]->channels / power;
421 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
422 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
424 for (i = 0; i < s->nb_taps; i++)
427 s->gain = ctx->inputs[1]->channels / power;
430 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
431 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
433 for (i = 0; i < s->nb_taps; i++)
434 power += time[i] * time[i];
436 s->gain = sqrtf(ch / power);
442 s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
443 av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
444 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
445 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
447 s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
450 av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
451 av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
453 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
454 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
457 for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
460 av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
462 for (int segment = 0; segment < s->nb_segments; segment++) {
463 AudioFIRSegment *seg = &s->seg[segment];
464 float *block = (float *)seg->block->extended_data[ch];
465 FFTComplex *coeff = (FFTComplex *)seg->coeff->extended_data[ch];
467 av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
469 for (i = 0; i < seg->nb_partitions; i++) {
470 const float scale = 1.f / seg->part_size;
471 const int coffset = i * seg->coeff_size;
472 const int remaining = s->nb_taps - toffset;
473 const int size = remaining >= seg->part_size ? seg->part_size : remaining;
475 memset(block, 0, sizeof(*block) * seg->fft_length);
476 memcpy(block, time + toffset, size * sizeof(*block));
478 av_rdft_calc(seg->rdft[0], block);
480 coeff[coffset].re = block[0] * scale;
481 coeff[coffset].im = 0;
482 for (n = 1; n < seg->part_size; n++) {
483 coeff[coffset + n].re = block[2 * n] * scale;
484 coeff[coffset + n].im = block[2 * n + 1] * scale;
486 coeff[coffset + seg->part_size].re = block[1] * scale;
487 coeff[coffset + seg->part_size].im = 0;
492 av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
493 av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
494 av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
495 av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
496 av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
497 av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
498 av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
502 av_frame_free(&s->in[1]);
508 static int check_ir(AVFilterLink *link, AVFrame *frame)
510 AVFilterContext *ctx = link->dst;
511 AudioFIRContext *s = ctx->priv;
512 int nb_taps, max_nb_taps;
514 nb_taps = ff_inlink_queued_samples(link);
515 max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
516 if (nb_taps > max_nb_taps) {
517 av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
518 return AVERROR(EINVAL);
524 static int activate(AVFilterContext *ctx)
526 AudioFIRContext *s = ctx->priv;
527 AVFilterLink *outlink = ctx->outputs[0];
532 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
534 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
535 if (!s->eof_coeffs) {
538 ret = check_ir(ctx->inputs[1], ir);
542 if (ff_outlink_get_status(ctx->inputs[1]) == AVERROR_EOF)
545 if (!s->eof_coeffs) {
546 if (ff_outlink_frame_wanted(ctx->outputs[0]))
547 ff_inlink_request_frame(ctx->inputs[1]);
548 else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
549 ff_inlink_request_frame(ctx->inputs[1]);
554 if (!s->have_coeffs && s->eof_coeffs) {
555 ret = convert_coeffs(ctx);
560 ret = ff_inlink_consume_samples(ctx->inputs[0], s->min_part_size, s->min_part_size, &in);
562 ret = fir_frame(s, in, outlink);
567 if (s->response && s->have_coeffs) {
568 int64_t old_pts = s->video->pts;
569 int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
571 if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
572 s->video->pts = new_pts;
573 return ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
577 if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
578 ff_filter_set_ready(ctx, 10);
582 if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
583 if (status == AVERROR_EOF) {
584 ff_outlink_set_status(ctx->outputs[0], status, pts);
586 ff_outlink_set_status(ctx->outputs[1], status, pts);
591 if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
592 !ff_outlink_get_status(ctx->inputs[0])) {
593 ff_inlink_request_frame(ctx->inputs[0]);
598 ff_outlink_frame_wanted(ctx->outputs[1]) &&
599 !ff_outlink_get_status(ctx->inputs[0])) {
600 ff_inlink_request_frame(ctx->inputs[0]);
604 return FFERROR_NOT_READY;
607 static int query_formats(AVFilterContext *ctx)
609 AudioFIRContext *s = ctx->priv;
610 AVFilterFormats *formats;
611 AVFilterChannelLayouts *layouts;
612 static const enum AVSampleFormat sample_fmts[] = {
616 static const enum AVPixelFormat pix_fmts[] = {
623 AVFilterLink *videolink = ctx->outputs[1];
624 formats = ff_make_format_list(pix_fmts);
625 if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
629 layouts = ff_all_channel_counts();
631 return AVERROR(ENOMEM);
634 ret = ff_set_common_channel_layouts(ctx, layouts);
638 AVFilterChannelLayouts *mono = NULL;
640 ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
644 if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts)) < 0)
646 if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
648 if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[1]->out_channel_layouts)) < 0)
652 formats = ff_make_format_list(sample_fmts);
653 if ((ret = ff_set_common_formats(ctx, formats)) < 0)
656 formats = ff_all_samplerates();
657 return ff_set_common_samplerates(ctx, formats);
660 static int config_output(AVFilterLink *outlink)
662 AVFilterContext *ctx = outlink->src;
663 AudioFIRContext *s = ctx->priv;
665 s->one2many = ctx->inputs[1]->channels == 1;
666 outlink->sample_rate = ctx->inputs[0]->sample_rate;
667 outlink->time_base = ctx->inputs[0]->time_base;
668 outlink->channel_layout = ctx->inputs[0]->channel_layout;
669 outlink->channels = ctx->inputs[0]->channels;
671 s->nb_channels = outlink->channels;
672 s->nb_coef_channels = ctx->inputs[1]->channels;
673 s->pts = AV_NOPTS_VALUE;
678 static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
680 AudioFIRContext *s = ctx->priv;
683 for (int ch = 0; ch < s->nb_channels; ch++) {
684 av_rdft_end(seg->rdft[ch]);
687 av_freep(&seg->rdft);
690 for (int ch = 0; ch < s->nb_channels; ch++) {
691 av_rdft_end(seg->irdft[ch]);
694 av_freep(&seg->irdft);
696 av_freep(&seg->output_offset);
697 av_freep(&seg->part_index);
699 av_frame_free(&seg->block);
700 av_frame_free(&seg->sum);
701 av_frame_free(&seg->buffer);
702 av_frame_free(&seg->coeff);
703 av_frame_free(&seg->input);
704 av_frame_free(&seg->output);
708 static av_cold void uninit(AVFilterContext *ctx)
710 AudioFIRContext *s = ctx->priv;
712 for (int i = 0; i < s->nb_segments; i++) {
713 uninit_segment(ctx, &s->seg[i]);
717 av_frame_free(&s->in[1]);
719 for (int i = 0; i < ctx->nb_outputs; i++)
720 av_freep(&ctx->output_pads[i].name);
721 av_frame_free(&s->video);
724 static int config_video(AVFilterLink *outlink)
726 AVFilterContext *ctx = outlink->src;
727 AudioFIRContext *s = ctx->priv;
729 outlink->sample_aspect_ratio = (AVRational){1,1};
732 outlink->frame_rate = s->frame_rate;
733 outlink->time_base = av_inv_q(outlink->frame_rate);
735 av_frame_free(&s->video);
736 s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
738 return AVERROR(ENOMEM);
743 static av_cold int init(AVFilterContext *ctx)
745 AudioFIRContext *s = ctx->priv;
746 AVFilterPad pad, vpad;
750 .name = av_strdup("default"),
751 .type = AVMEDIA_TYPE_AUDIO,
752 .config_props = config_output,
756 return AVERROR(ENOMEM);
759 vpad = (AVFilterPad){
760 .name = av_strdup("filter_response"),
761 .type = AVMEDIA_TYPE_VIDEO,
762 .config_props = config_video,
765 return AVERROR(ENOMEM);
768 ret = ff_insert_outpad(ctx, 0, &pad);
775 ret = ff_insert_outpad(ctx, 1, &vpad);
777 av_freep(&vpad.name);
782 s->fcmul_add = fcmul_add_c;
784 s->fdsp = avpriv_float_dsp_alloc(0);
786 return AVERROR(ENOMEM);
794 static const AVFilterPad afir_inputs[] = {
797 .type = AVMEDIA_TYPE_AUDIO,
800 .type = AVMEDIA_TYPE_AUDIO,
805 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
806 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
807 #define OFFSET(x) offsetof(AudioFIRContext, x)
809 static const AVOption afir_options[] = {
810 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
811 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
812 { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
813 { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
814 { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
815 { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
816 { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
817 { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
818 { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
819 { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
820 { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
821 { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
822 { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
823 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
824 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
825 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
826 { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
827 { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 16, 32768, AF },
828 { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 16, 32768, AF },
832 AVFILTER_DEFINE_CLASS(afir);
834 AVFilter ff_af_afir = {
836 .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
837 .priv_size = sizeof(AudioFIRContext),
838 .priv_class = &afir_class,
839 .query_formats = query_formats,
841 .activate = activate,
843 .inputs = afir_inputs,
844 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
845 AVFILTER_FLAG_SLICE_THREADS,