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/avstring.h"
29 #include "libavutil/common.h"
30 #include "libavutil/float_dsp.h"
31 #include "libavutil/intreadwrite.h"
32 #include "libavutil/opt.h"
33 #include "libavutil/xga_font_data.h"
34 #include "libavcodec/avfft.h"
43 static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
47 for (n = 0; n < len; n++) {
48 const float cre = c[2 * n ];
49 const float cim = c[2 * n + 1];
50 const float tre = t[2 * n ];
51 const float tim = t[2 * n + 1];
53 sum[2 * n ] += tre * cre - tim * cim;
54 sum[2 * n + 1] += tre * cim + tim * cre;
57 sum[2 * n] += t[2 * n] * c[2 * n];
60 static void direct(const float *in, const FFTComplex *ir, int len, float *out)
62 for (int n = 0; n < len; n++)
63 for (int m = 0; m <= n; m++)
64 out[n] += ir[m].re * in[n - m];
67 static void fir_fadd(AudioFIRContext *s, float *dst, const float *src, int nb_samples)
69 if ((nb_samples & 15) == 0 && nb_samples >= 16) {
70 s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples);
72 for (int n = 0; n < nb_samples; n++)
77 static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
79 AudioFIRContext *s = ctx->priv;
80 const float *in = (const float *)s->in->extended_data[ch] + offset;
81 float *block, *buf, *ptr = (float *)out->extended_data[ch] + offset;
82 const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);
85 for (int segment = 0; segment < s->nb_segments; segment++) {
86 AudioFIRSegment *seg = &s->seg[segment];
87 float *src = (float *)seg->input->extended_data[ch];
88 float *dst = (float *)seg->output->extended_data[ch];
89 float *sum = (float *)seg->sum->extended_data[ch];
91 if (s->min_part_size >= 8) {
92 s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4));
95 for (n = 0; n < nb_samples; n++)
96 src[seg->input_offset + n] = in[n] * s->dry_gain;
99 seg->output_offset[ch] += s->min_part_size;
100 if (seg->output_offset[ch] == seg->part_size) {
101 seg->output_offset[ch] = 0;
103 memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
105 dst += seg->output_offset[ch];
106 fir_fadd(s, ptr, dst, nb_samples);
110 if (seg->part_size < 8) {
111 memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);
113 j = seg->part_index[ch];
115 for (i = 0; i < seg->nb_partitions; i++) {
116 const int coffset = j * seg->coeff_size;
117 const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
119 direct(src, coeff, nb_samples, dst);
122 j = seg->nb_partitions;
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++) {
136 memset(sum, 0, sizeof(*sum) * seg->fft_length);
137 block = (float *)seg->block->extended_data[ch] + seg->part_index[ch] * seg->block_size;
138 memset(block + seg->part_size, 0, sizeof(*block) * (seg->fft_length - seg->part_size));
140 memcpy(block, src, sizeof(*src) * seg->part_size);
142 av_rdft_calc(seg->rdft[ch], block);
143 block[2 * seg->part_size] = block[1];
146 j = seg->part_index[ch];
148 for (i = 0; i < seg->nb_partitions; i++) {
149 const int coffset = j * seg->coeff_size;
150 const float *block = (const float *)seg->block->extended_data[ch] + i * seg->block_size;
151 const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
153 s->afirdsp.fcmul_add(sum, block, (const float *)coeff, seg->part_size);
156 j = seg->nb_partitions;
160 sum[1] = sum[2 * seg->part_size];
161 av_rdft_calc(seg->irdft[ch], sum);
163 buf = (float *)seg->buffer->extended_data[ch];
164 fir_fadd(s, buf, sum, seg->part_size);
166 memcpy(dst, buf, seg->part_size * sizeof(*dst));
168 buf = (float *)seg->buffer->extended_data[ch];
169 memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf));
171 seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
173 memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
175 fir_fadd(s, ptr, dst, nb_samples);
178 if (s->min_part_size >= 8) {
179 s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));
182 for (n = 0; n < nb_samples; n++)
183 ptr[n] *= s->wet_gain;
189 static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
191 AudioFIRContext *s = ctx->priv;
193 for (int offset = 0; offset < out->nb_samples; offset += s->min_part_size) {
194 fir_quantum(ctx, out, ch, offset);
200 static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
203 const int start = (out->channels * jobnr) / nb_jobs;
204 const int end = (out->channels * (jobnr+1)) / nb_jobs;
206 for (int ch = start; ch < end; ch++) {
207 fir_channel(ctx, out, ch);
213 static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
215 AVFilterContext *ctx = outlink->src;
218 out = ff_get_audio_buffer(outlink, in->nb_samples);
221 return AVERROR(ENOMEM);
224 if (s->pts == AV_NOPTS_VALUE)
227 ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels,
228 ff_filter_get_nb_threads(ctx)));
231 if (s->pts != AV_NOPTS_VALUE)
232 s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
237 return ff_filter_frame(outlink, out);
240 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
246 font = avpriv_cga_font, font_height = 8;
248 for (i = 0; txt[i]; i++) {
251 uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
252 for (char_y = 0; char_y < font_height; char_y++) {
253 for (mask = 0x80; mask; mask >>= 1) {
254 if (font[txt[i] * font_height + char_y] & mask)
258 p += pic->linesize[0] - 8 * 4;
263 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
265 int dx = FFABS(x1-x0);
266 int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
267 int err = (dx>dy ? dx : -dy) / 2, e2;
270 AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
272 if (x0 == x1 && y0 == y1)
289 static void draw_response(AVFilterContext *ctx, AVFrame *out)
291 AudioFIRContext *s = ctx->priv;
292 float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
293 float min_delay = FLT_MAX, max_delay = FLT_MIN;
294 int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
298 memset(out->data[0], 0, s->h * out->linesize[0]);
300 phase = av_malloc_array(s->w, sizeof(*phase));
301 mag = av_malloc_array(s->w, sizeof(*mag));
302 delay = av_malloc_array(s->w, sizeof(*delay));
303 if (!mag || !phase || !delay)
306 channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->channels - 1);
307 for (i = 0; i < s->w; i++) {
308 const float *src = (const float *)s->ir[s->selir]->extended_data[channel];
309 double w = i * M_PI / (s->w - 1);
310 double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
312 for (x = 0; x < s->nb_taps; x++) {
313 real += cos(-x * w) * src[x];
314 imag += sin(-x * w) * src[x];
315 real_num += cos(-x * w) * src[x] * x;
316 imag_num += sin(-x * w) * src[x] * x;
319 mag[i] = hypot(real, imag);
320 phase[i] = atan2(imag, real);
321 div = real * real + imag * imag;
322 delay[i] = (real_num * real + imag_num * imag) / div;
323 min = fminf(min, mag[i]);
324 max = fmaxf(max, mag[i]);
325 min_delay = fminf(min_delay, delay[i]);
326 max_delay = fmaxf(max_delay, delay[i]);
329 for (i = 0; i < s->w; i++) {
330 int ymag = mag[i] / max * (s->h - 1);
331 int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
332 int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
334 ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
335 yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
336 ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
341 prev_yphase = yphase;
343 prev_ydelay = ydelay;
345 draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
346 draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
347 draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
350 prev_yphase = yphase;
351 prev_ydelay = ydelay;
354 if (s->w > 400 && s->h > 100) {
355 drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
356 snprintf(text, sizeof(text), "%.2f", max);
357 drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
359 drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
360 snprintf(text, sizeof(text), "%.2f", min);
361 drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
363 drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
364 snprintf(text, sizeof(text), "%.2f", max_delay);
365 drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
367 drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
368 snprintf(text, sizeof(text), "%.2f", min_delay);
369 drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
378 static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
379 int offset, int nb_partitions, int part_size)
381 AudioFIRContext *s = ctx->priv;
383 seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
384 seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
385 if (!seg->rdft || !seg->irdft)
386 return AVERROR(ENOMEM);
388 seg->fft_length = part_size * 2 + 1;
389 seg->part_size = part_size;
390 seg->block_size = FFALIGN(seg->fft_length, 32);
391 seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
392 seg->nb_partitions = nb_partitions;
393 seg->input_size = offset + s->min_part_size;
394 seg->input_offset = offset;
396 seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index));
397 seg->output_offset = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->output_offset));
398 if (!seg->part_index || !seg->output_offset)
399 return AVERROR(ENOMEM);
401 for (int ch = 0; ch < ctx->inputs[0]->channels && part_size >= 8; ch++) {
402 seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
403 seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);
404 if (!seg->rdft[ch] || !seg->irdft[ch])
405 return AVERROR(ENOMEM);
408 seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
409 seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
410 seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
411 seg->coeff = ff_get_audio_buffer(ctx->inputs[1 + s->selir], seg->nb_partitions * seg->coeff_size * 2);
412 seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
413 seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
414 if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff || !seg->input || !seg->output)
415 return AVERROR(ENOMEM);
420 static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
422 AudioFIRContext *s = ctx->priv;
425 for (int ch = 0; ch < s->nb_channels; ch++) {
426 av_rdft_end(seg->rdft[ch]);
429 av_freep(&seg->rdft);
432 for (int ch = 0; ch < s->nb_channels; ch++) {
433 av_rdft_end(seg->irdft[ch]);
436 av_freep(&seg->irdft);
438 av_freep(&seg->output_offset);
439 av_freep(&seg->part_index);
441 av_frame_free(&seg->block);
442 av_frame_free(&seg->sum);
443 av_frame_free(&seg->buffer);
444 av_frame_free(&seg->coeff);
445 av_frame_free(&seg->input);
446 av_frame_free(&seg->output);
450 static int convert_coeffs(AVFilterContext *ctx)
452 AudioFIRContext *s = ctx->priv;
453 int ret, i, ch, n, cur_nb_taps;
457 int part_size, max_part_size;
458 int left, offset = 0;
460 s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1 + s->selir]);
462 return AVERROR(EINVAL);
464 if (s->minp > s->maxp) {
469 part_size = 1 << av_log2(s->minp);
470 max_part_size = 1 << av_log2(s->maxp);
472 s->min_part_size = part_size;
474 for (i = 0; left > 0; i++) {
475 int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);
476 int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
478 s->nb_segments = i + 1;
479 ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);
482 offset += nb_partitions * part_size;
483 left -= nb_partitions * part_size;
485 part_size = FFMIN(part_size, max_part_size);
489 if (!s->ir[s->selir]) {
490 ret = ff_inlink_consume_samples(ctx->inputs[1 + s->selir], s->nb_taps, s->nb_taps, &s->ir[s->selir]);
498 draw_response(ctx, s->video);
501 cur_nb_taps = s->ir[s->selir]->nb_samples;
508 for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
509 float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
511 for (i = 0; i < cur_nb_taps; i++)
512 power += FFABS(time[i]);
514 s->gain = ctx->inputs[1 + s->selir]->channels / power;
517 for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
518 float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
520 for (i = 0; i < cur_nb_taps; i++)
523 s->gain = ctx->inputs[1 + s->selir]->channels / power;
526 for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
527 float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
529 for (i = 0; i < cur_nb_taps; i++)
530 power += time[i] * time[i];
532 s->gain = sqrtf(ch / power);
538 s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
539 av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
540 for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
541 float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
543 s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4));
546 av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", cur_nb_taps);
547 av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
549 for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
550 float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
553 for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
556 av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
558 for (int segment = 0; segment < s->nb_segments; segment++) {
559 AudioFIRSegment *seg = &s->seg[segment];
560 float *block = (float *)seg->block->extended_data[ch];
561 FFTComplex *coeff = (FFTComplex *)seg->coeff->extended_data[ch];
563 av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
565 for (i = 0; i < seg->nb_partitions; i++) {
566 const float scale = 1.f / seg->part_size;
567 const int coffset = i * seg->coeff_size;
568 const int remaining = s->nb_taps - toffset;
569 const int size = remaining >= seg->part_size ? seg->part_size : remaining;
572 for (n = 0; n < size; n++)
573 coeff[coffset + n].re = time[toffset + n];
579 memset(block, 0, sizeof(*block) * seg->fft_length);
580 memcpy(block, time + toffset, size * sizeof(*block));
582 av_rdft_calc(seg->rdft[0], block);
584 coeff[coffset].re = block[0] * scale;
585 coeff[coffset].im = 0;
586 for (n = 1; n < seg->part_size; n++) {
587 coeff[coffset + n].re = block[2 * n] * scale;
588 coeff[coffset + n].im = block[2 * n + 1] * scale;
590 coeff[coffset + seg->part_size].re = block[1] * scale;
591 coeff[coffset + seg->part_size].im = 0;
596 av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
597 av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
598 av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
599 av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
600 av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
601 av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
602 av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
611 static int check_ir(AVFilterLink *link)
613 AVFilterContext *ctx = link->dst;
614 AudioFIRContext *s = ctx->priv;
615 int nb_taps, max_nb_taps;
617 nb_taps = ff_inlink_queued_samples(link);
618 max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
619 if (nb_taps > max_nb_taps) {
620 av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
621 return AVERROR(EINVAL);
627 static int activate(AVFilterContext *ctx)
629 AudioFIRContext *s = ctx->priv;
630 AVFilterLink *outlink = ctx->outputs[0];
631 int ret, status, available, wanted;
635 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
637 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
638 if (!s->eof_coeffs[s->selir]) {
639 ret = check_ir(ctx->inputs[1 + s->selir]);
643 if (ff_outlink_get_status(ctx->inputs[1 + s->selir]) == AVERROR_EOF)
644 s->eof_coeffs[s->selir] = 1;
646 if (!s->eof_coeffs[s->selir]) {
647 if (ff_outlink_frame_wanted(ctx->outputs[0]))
648 ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
649 else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
650 ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
655 if (!s->have_coeffs && s->eof_coeffs[s->selir]) {
656 ret = convert_coeffs(ctx);
661 available = ff_inlink_queued_samples(ctx->inputs[0]);
662 wanted = FFMAX(s->min_part_size, (available / s->min_part_size) * s->min_part_size);
663 ret = ff_inlink_consume_samples(ctx->inputs[0], wanted, wanted, &in);
665 ret = fir_frame(s, in, outlink);
670 if (s->response && s->have_coeffs) {
671 int64_t old_pts = s->video->pts;
672 int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
674 if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
676 s->video->pts = new_pts;
677 clone = av_frame_clone(s->video);
679 return AVERROR(ENOMEM);
680 return ff_filter_frame(ctx->outputs[1], clone);
684 if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
685 ff_filter_set_ready(ctx, 10);
689 if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
690 if (status == AVERROR_EOF) {
691 ff_outlink_set_status(ctx->outputs[0], status, pts);
693 ff_outlink_set_status(ctx->outputs[1], status, pts);
698 if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
699 !ff_outlink_get_status(ctx->inputs[0])) {
700 ff_inlink_request_frame(ctx->inputs[0]);
705 ff_outlink_frame_wanted(ctx->outputs[1]) &&
706 !ff_outlink_get_status(ctx->inputs[0])) {
707 ff_inlink_request_frame(ctx->inputs[0]);
711 return FFERROR_NOT_READY;
714 static int query_formats(AVFilterContext *ctx)
716 AudioFIRContext *s = ctx->priv;
717 AVFilterFormats *formats;
718 AVFilterChannelLayouts *layouts;
719 static const enum AVSampleFormat sample_fmts[] = {
723 static const enum AVPixelFormat pix_fmts[] = {
730 AVFilterLink *videolink = ctx->outputs[1];
731 formats = ff_make_format_list(pix_fmts);
732 if ((ret = ff_formats_ref(formats, &videolink->incfg.formats)) < 0)
736 layouts = ff_all_channel_counts();
738 return AVERROR(ENOMEM);
741 ret = ff_set_common_channel_layouts(ctx, layouts);
745 AVFilterChannelLayouts *mono = NULL;
747 if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts)) < 0)
749 if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts)) < 0)
752 ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
755 for (int i = 1; i < ctx->nb_inputs; i++) {
756 if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[i]->outcfg.channel_layouts)) < 0)
761 formats = ff_make_format_list(sample_fmts);
762 if ((ret = ff_set_common_formats(ctx, formats)) < 0)
765 formats = ff_all_samplerates();
766 return ff_set_common_samplerates(ctx, formats);
769 static int config_output(AVFilterLink *outlink)
771 AVFilterContext *ctx = outlink->src;
772 AudioFIRContext *s = ctx->priv;
774 s->one2many = ctx->inputs[1 + s->selir]->channels == 1;
775 outlink->sample_rate = ctx->inputs[0]->sample_rate;
776 outlink->time_base = ctx->inputs[0]->time_base;
777 outlink->channel_layout = ctx->inputs[0]->channel_layout;
778 outlink->channels = ctx->inputs[0]->channels;
780 s->nb_channels = outlink->channels;
781 s->nb_coef_channels = ctx->inputs[1 + s->selir]->channels;
782 s->pts = AV_NOPTS_VALUE;
787 static av_cold void uninit(AVFilterContext *ctx)
789 AudioFIRContext *s = ctx->priv;
791 for (int i = 0; i < s->nb_segments; i++) {
792 uninit_segment(ctx, &s->seg[i]);
797 for (int i = 0; i < s->nb_irs; i++) {
798 av_frame_free(&s->ir[i]);
801 for (unsigned i = 1; i < ctx->nb_inputs; i++)
802 av_freep(&ctx->input_pads[i].name);
804 av_frame_free(&s->video);
807 static int config_video(AVFilterLink *outlink)
809 AVFilterContext *ctx = outlink->src;
810 AudioFIRContext *s = ctx->priv;
812 outlink->sample_aspect_ratio = (AVRational){1,1};
815 outlink->frame_rate = s->frame_rate;
816 outlink->time_base = av_inv_q(outlink->frame_rate);
818 av_frame_free(&s->video);
819 s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
821 return AVERROR(ENOMEM);
826 void ff_afir_init(AudioFIRDSPContext *dsp)
828 dsp->fcmul_add = fcmul_add_c;
831 ff_afir_init_x86(dsp);
834 static av_cold int init(AVFilterContext *ctx)
836 AudioFIRContext *s = ctx->priv;
837 AVFilterPad pad, vpad;
840 pad = (AVFilterPad) {
842 .type = AVMEDIA_TYPE_AUDIO,
845 ret = ff_insert_inpad(ctx, 0, &pad);
849 for (int n = 0; n < s->nb_irs; n++) {
850 pad = (AVFilterPad) {
851 .name = av_asprintf("ir%d", n),
852 .type = AVMEDIA_TYPE_AUDIO,
856 return AVERROR(ENOMEM);
858 ret = ff_insert_inpad(ctx, n + 1, &pad);
865 pad = (AVFilterPad) {
867 .type = AVMEDIA_TYPE_AUDIO,
868 .config_props = config_output,
871 ret = ff_insert_outpad(ctx, 0, &pad);
876 vpad = (AVFilterPad){
877 .name = "filter_response",
878 .type = AVMEDIA_TYPE_VIDEO,
879 .config_props = config_video,
882 ret = ff_insert_outpad(ctx, 1, &vpad);
887 s->fdsp = avpriv_float_dsp_alloc(0);
889 return AVERROR(ENOMEM);
891 ff_afir_init(&s->afirdsp);
896 static int process_command(AVFilterContext *ctx,
903 AudioFIRContext *s = ctx->priv;
904 int prev_ir = s->selir;
905 int ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
910 s->selir = FFMIN(s->nb_irs - 1, s->selir);
912 if (prev_ir != s->selir) {
919 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
920 #define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
921 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
922 #define OFFSET(x) offsetof(AudioFIRContext, x)
924 static const AVOption afir_options[] = {
925 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
926 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
927 { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
928 { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
929 { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
930 { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
931 { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
932 { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
933 { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
934 { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
935 { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
936 { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
937 { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
938 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
939 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
940 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
941 { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
942 { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 1, 32768, AF },
943 { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 32768, AF },
944 { "nbirs", "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT, {.i64=1}, 1, 32, AF },
945 { "ir", "select IR", OFFSET(selir), AV_OPT_TYPE_INT, {.i64=0}, 0, 31, AFR },
949 AVFILTER_DEFINE_CLASS(afir);
951 const AVFilter ff_af_afir = {
953 .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in additional stream(s)."),
954 .priv_size = sizeof(AudioFIRContext),
955 .priv_class = &afir_class,
956 .query_formats = query_formats,
958 .activate = activate,
960 .process_command = process_command,
961 .flags = AVFILTER_FLAG_DYNAMIC_INPUTS |
962 AVFILTER_FLAG_DYNAMIC_OUTPUTS |
963 AVFILTER_FLAG_SLICE_THREADS,