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_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
61 AudioFIRContext *s = ctx->priv;
62 const float *in = (const float *)s->in[0]->extended_data[ch] + offset;
63 float *block, *buf, *ptr = (float *)out->extended_data[ch] + offset;
64 const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);
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(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 < 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->afirdsp.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 < nb_samples; n++) {
135 s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));
141 static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
143 AudioFIRContext *s = ctx->priv;
145 for (int offset = 0; offset < out->nb_samples; offset += s->min_part_size) {
146 fir_quantum(ctx, out, ch, offset);
152 static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
155 const int start = (out->channels * jobnr) / nb_jobs;
156 const int end = (out->channels * (jobnr+1)) / nb_jobs;
158 for (int ch = start; ch < end; ch++) {
159 fir_channel(ctx, out, ch);
165 static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
167 AVFilterContext *ctx = outlink->src;
170 out = ff_get_audio_buffer(outlink, in->nb_samples);
173 return AVERROR(ENOMEM);
176 if (s->pts == AV_NOPTS_VALUE)
179 ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels,
180 ff_filter_get_nb_threads(ctx)));
183 if (s->pts != AV_NOPTS_VALUE)
184 s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
189 return ff_filter_frame(outlink, out);
192 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
198 font = avpriv_cga_font, font_height = 8;
200 for (i = 0; txt[i]; i++) {
203 uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
204 for (char_y = 0; char_y < font_height; char_y++) {
205 for (mask = 0x80; mask; mask >>= 1) {
206 if (font[txt[i] * font_height + char_y] & mask)
210 p += pic->linesize[0] - 8 * 4;
215 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
217 int dx = FFABS(x1-x0);
218 int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
219 int err = (dx>dy ? dx : -dy) / 2, e2;
222 AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
224 if (x0 == x1 && y0 == y1)
241 static void draw_response(AVFilterContext *ctx, AVFrame *out)
243 AudioFIRContext *s = ctx->priv;
244 float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
245 float min_delay = FLT_MAX, max_delay = FLT_MIN;
246 int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
250 memset(out->data[0], 0, s->h * out->linesize[0]);
252 phase = av_malloc_array(s->w, sizeof(*phase));
253 mag = av_malloc_array(s->w, sizeof(*mag));
254 delay = av_malloc_array(s->w, sizeof(*delay));
255 if (!mag || !phase || !delay)
258 channel = av_clip(s->ir_channel, 0, s->in[1]->channels - 1);
259 for (i = 0; i < s->w; i++) {
260 const float *src = (const float *)s->in[1]->extended_data[channel];
261 double w = i * M_PI / (s->w - 1);
262 double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
264 for (x = 0; x < s->nb_taps; x++) {
265 real += cos(-x * w) * src[x];
266 imag += sin(-x * w) * src[x];
267 real_num += cos(-x * w) * src[x] * x;
268 imag_num += sin(-x * w) * src[x] * x;
271 mag[i] = hypot(real, imag);
272 phase[i] = atan2(imag, real);
273 div = real * real + imag * imag;
274 delay[i] = (real_num * real + imag_num * imag) / div;
275 min = fminf(min, mag[i]);
276 max = fmaxf(max, mag[i]);
277 min_delay = fminf(min_delay, delay[i]);
278 max_delay = fmaxf(max_delay, delay[i]);
281 for (i = 0; i < s->w; i++) {
282 int ymag = mag[i] / max * (s->h - 1);
283 int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
284 int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
286 ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
287 yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
288 ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
293 prev_yphase = yphase;
295 prev_ydelay = ydelay;
297 draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
298 draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
299 draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
302 prev_yphase = yphase;
303 prev_ydelay = ydelay;
306 if (s->w > 400 && s->h > 100) {
307 drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
308 snprintf(text, sizeof(text), "%.2f", max);
309 drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
311 drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
312 snprintf(text, sizeof(text), "%.2f", min);
313 drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
315 drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
316 snprintf(text, sizeof(text), "%.2f", max_delay);
317 drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
319 drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
320 snprintf(text, sizeof(text), "%.2f", min_delay);
321 drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
330 static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
331 int offset, int nb_partitions, int part_size)
333 AudioFIRContext *s = ctx->priv;
335 seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
336 seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
337 if (!seg->rdft || !seg->irdft)
338 return AVERROR(ENOMEM);
340 seg->fft_length = part_size * 2 + 1;
341 seg->part_size = part_size;
342 seg->block_size = FFALIGN(seg->fft_length, 32);
343 seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
344 seg->nb_partitions = nb_partitions;
345 seg->input_size = offset + s->min_part_size;
346 seg->input_offset = offset;
348 seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index));
349 seg->output_offset = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->output_offset));
350 if (!seg->part_index || !seg->output_offset)
351 return AVERROR(ENOMEM);
353 for (int ch = 0; ch < ctx->inputs[0]->channels; ch++) {
354 seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
355 seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);
356 if (!seg->rdft[ch] || !seg->irdft[ch])
357 return AVERROR(ENOMEM);
360 seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
361 seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
362 seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
363 seg->coeff = ff_get_audio_buffer(ctx->inputs[1], seg->nb_partitions * seg->coeff_size * 2);
364 seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
365 seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
366 if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff || !seg->input || !seg->output)
367 return AVERROR(ENOMEM);
372 static int convert_coeffs(AVFilterContext *ctx)
374 AudioFIRContext *s = ctx->priv;
375 int left, offset = 0, part_size, max_part_size;
379 s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1]);
381 return AVERROR(EINVAL);
383 if (s->minp > s->maxp) {
388 part_size = 1 << av_log2(s->minp);
389 max_part_size = 1 << av_log2(s->maxp);
391 s->min_part_size = part_size;
393 for (i = 0; left > 0; i++) {
394 int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);
395 int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
397 s->nb_segments = i + 1;
398 ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);
401 offset += nb_partitions * part_size;
402 left -= nb_partitions * part_size;
404 part_size = FFMIN(part_size, max_part_size);
407 ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]);
414 draw_response(ctx, s->video);
423 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
424 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
426 for (i = 0; i < s->nb_taps; i++)
427 power += FFABS(time[i]);
429 s->gain = ctx->inputs[1]->channels / power;
432 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
433 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
435 for (i = 0; i < s->nb_taps; i++)
438 s->gain = ctx->inputs[1]->channels / power;
441 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
442 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
444 for (i = 0; i < s->nb_taps; i++)
445 power += time[i] * time[i];
447 s->gain = sqrtf(ch / power);
453 s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
454 av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
455 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
456 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
458 s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
461 av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
462 av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
464 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
465 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
468 for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
471 av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
473 for (int segment = 0; segment < s->nb_segments; segment++) {
474 AudioFIRSegment *seg = &s->seg[segment];
475 float *block = (float *)seg->block->extended_data[ch];
476 FFTComplex *coeff = (FFTComplex *)seg->coeff->extended_data[ch];
478 av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
480 for (i = 0; i < seg->nb_partitions; i++) {
481 const float scale = 1.f / seg->part_size;
482 const int coffset = i * seg->coeff_size;
483 const int remaining = s->nb_taps - toffset;
484 const int size = remaining >= seg->part_size ? seg->part_size : remaining;
486 memset(block, 0, sizeof(*block) * seg->fft_length);
487 memcpy(block, time + toffset, size * sizeof(*block));
489 av_rdft_calc(seg->rdft[0], block);
491 coeff[coffset].re = block[0] * scale;
492 coeff[coffset].im = 0;
493 for (n = 1; n < seg->part_size; n++) {
494 coeff[coffset + n].re = block[2 * n] * scale;
495 coeff[coffset + n].im = block[2 * n + 1] * scale;
497 coeff[coffset + seg->part_size].re = block[1] * scale;
498 coeff[coffset + seg->part_size].im = 0;
503 av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
504 av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
505 av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
506 av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
507 av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
508 av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
509 av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
513 av_frame_free(&s->in[1]);
519 static int check_ir(AVFilterLink *link, AVFrame *frame)
521 AVFilterContext *ctx = link->dst;
522 AudioFIRContext *s = ctx->priv;
523 int nb_taps, max_nb_taps;
525 nb_taps = ff_inlink_queued_samples(link);
526 max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
527 if (nb_taps > max_nb_taps) {
528 av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
529 return AVERROR(EINVAL);
535 static int activate(AVFilterContext *ctx)
537 AudioFIRContext *s = ctx->priv;
538 AVFilterLink *outlink = ctx->outputs[0];
539 int ret, status, available, wanted;
543 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
545 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
546 if (!s->eof_coeffs) {
549 ret = check_ir(ctx->inputs[1], ir);
553 if (ff_outlink_get_status(ctx->inputs[1]) == AVERROR_EOF)
556 if (!s->eof_coeffs) {
557 if (ff_outlink_frame_wanted(ctx->outputs[0]))
558 ff_inlink_request_frame(ctx->inputs[1]);
559 else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
560 ff_inlink_request_frame(ctx->inputs[1]);
565 if (!s->have_coeffs && s->eof_coeffs) {
566 ret = convert_coeffs(ctx);
571 available = ff_inlink_queued_samples(ctx->inputs[0]);
572 wanted = FFMAX(s->min_part_size, (available / s->min_part_size) * s->min_part_size);
573 ret = ff_inlink_consume_samples(ctx->inputs[0], wanted, wanted, &in);
575 ret = fir_frame(s, in, outlink);
580 if (s->response && s->have_coeffs) {
581 int64_t old_pts = s->video->pts;
582 int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
584 if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
585 s->video->pts = new_pts;
586 return ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
590 if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
591 ff_filter_set_ready(ctx, 10);
595 if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
596 if (status == AVERROR_EOF) {
597 ff_outlink_set_status(ctx->outputs[0], status, pts);
599 ff_outlink_set_status(ctx->outputs[1], status, pts);
604 if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
605 !ff_outlink_get_status(ctx->inputs[0])) {
606 ff_inlink_request_frame(ctx->inputs[0]);
611 ff_outlink_frame_wanted(ctx->outputs[1]) &&
612 !ff_outlink_get_status(ctx->inputs[0])) {
613 ff_inlink_request_frame(ctx->inputs[0]);
617 return FFERROR_NOT_READY;
620 static int query_formats(AVFilterContext *ctx)
622 AudioFIRContext *s = ctx->priv;
623 AVFilterFormats *formats;
624 AVFilterChannelLayouts *layouts;
625 static const enum AVSampleFormat sample_fmts[] = {
629 static const enum AVPixelFormat pix_fmts[] = {
636 AVFilterLink *videolink = ctx->outputs[1];
637 formats = ff_make_format_list(pix_fmts);
638 if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
642 layouts = ff_all_channel_counts();
644 return AVERROR(ENOMEM);
647 ret = ff_set_common_channel_layouts(ctx, layouts);
651 AVFilterChannelLayouts *mono = NULL;
653 ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
657 if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts)) < 0)
659 if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
661 if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[1]->out_channel_layouts)) < 0)
665 formats = ff_make_format_list(sample_fmts);
666 if ((ret = ff_set_common_formats(ctx, formats)) < 0)
669 formats = ff_all_samplerates();
670 return ff_set_common_samplerates(ctx, formats);
673 static int config_output(AVFilterLink *outlink)
675 AVFilterContext *ctx = outlink->src;
676 AudioFIRContext *s = ctx->priv;
678 s->one2many = ctx->inputs[1]->channels == 1;
679 outlink->sample_rate = ctx->inputs[0]->sample_rate;
680 outlink->time_base = ctx->inputs[0]->time_base;
681 outlink->channel_layout = ctx->inputs[0]->channel_layout;
682 outlink->channels = ctx->inputs[0]->channels;
684 s->nb_channels = outlink->channels;
685 s->nb_coef_channels = ctx->inputs[1]->channels;
686 s->pts = AV_NOPTS_VALUE;
691 static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
693 AudioFIRContext *s = ctx->priv;
696 for (int ch = 0; ch < s->nb_channels; ch++) {
697 av_rdft_end(seg->rdft[ch]);
700 av_freep(&seg->rdft);
703 for (int ch = 0; ch < s->nb_channels; ch++) {
704 av_rdft_end(seg->irdft[ch]);
707 av_freep(&seg->irdft);
709 av_freep(&seg->output_offset);
710 av_freep(&seg->part_index);
712 av_frame_free(&seg->block);
713 av_frame_free(&seg->sum);
714 av_frame_free(&seg->buffer);
715 av_frame_free(&seg->coeff);
716 av_frame_free(&seg->input);
717 av_frame_free(&seg->output);
721 static av_cold void uninit(AVFilterContext *ctx)
723 AudioFIRContext *s = ctx->priv;
725 for (int i = 0; i < s->nb_segments; i++) {
726 uninit_segment(ctx, &s->seg[i]);
730 av_frame_free(&s->in[1]);
732 for (int i = 0; i < ctx->nb_outputs; i++)
733 av_freep(&ctx->output_pads[i].name);
734 av_frame_free(&s->video);
737 static int config_video(AVFilterLink *outlink)
739 AVFilterContext *ctx = outlink->src;
740 AudioFIRContext *s = ctx->priv;
742 outlink->sample_aspect_ratio = (AVRational){1,1};
745 outlink->frame_rate = s->frame_rate;
746 outlink->time_base = av_inv_q(outlink->frame_rate);
748 av_frame_free(&s->video);
749 s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
751 return AVERROR(ENOMEM);
756 void ff_afir_init(AudioFIRDSPContext *dsp)
758 dsp->fcmul_add = fcmul_add_c;
761 ff_afir_init_x86(dsp);
764 static av_cold int init(AVFilterContext *ctx)
766 AudioFIRContext *s = ctx->priv;
767 AVFilterPad pad, vpad;
771 .name = av_strdup("default"),
772 .type = AVMEDIA_TYPE_AUDIO,
773 .config_props = config_output,
777 return AVERROR(ENOMEM);
780 vpad = (AVFilterPad){
781 .name = av_strdup("filter_response"),
782 .type = AVMEDIA_TYPE_VIDEO,
783 .config_props = config_video,
786 return AVERROR(ENOMEM);
789 ret = ff_insert_outpad(ctx, 0, &pad);
796 ret = ff_insert_outpad(ctx, 1, &vpad);
798 av_freep(&vpad.name);
803 s->fdsp = avpriv_float_dsp_alloc(0);
805 return AVERROR(ENOMEM);
807 ff_afir_init(&s->afirdsp);
812 static const AVFilterPad afir_inputs[] = {
815 .type = AVMEDIA_TYPE_AUDIO,
818 .type = AVMEDIA_TYPE_AUDIO,
823 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
824 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
825 #define OFFSET(x) offsetof(AudioFIRContext, x)
827 static const AVOption afir_options[] = {
828 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
829 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
830 { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
831 { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
832 { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
833 { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
834 { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
835 { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
836 { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
837 { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
838 { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
839 { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
840 { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
841 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
842 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
843 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
844 { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
845 { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 32768, AF },
846 { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 32768, AF },
850 AVFILTER_DEFINE_CLASS(afir);
852 AVFilter ff_af_afir = {
854 .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
855 .priv_size = sizeof(AudioFIRContext),
856 .priv_class = &afir_class,
857 .query_formats = query_formats,
859 .activate = activate,
861 .inputs = afir_inputs,
862 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
863 AVFILTER_FLAG_SLICE_THREADS,