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 void direct(const float *in, const FFTComplex *ir, int len, float *out)
61 for (int n = 0; n < len; n++)
62 for (int m = 0; m <= n; m++)
63 out[n] += ir[m].re * in[n - m];
66 static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
68 AudioFIRContext *s = ctx->priv;
69 const float *in = (const float *)s->in->extended_data[ch] + offset;
70 float *block, *buf, *ptr = (float *)out->extended_data[ch] + offset;
71 const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);
74 for (int segment = 0; segment < s->nb_segments; segment++) {
75 AudioFIRSegment *seg = &s->seg[segment];
76 float *src = (float *)seg->input->extended_data[ch];
77 float *dst = (float *)seg->output->extended_data[ch];
78 float *sum = (float *)seg->sum->extended_data[ch];
80 if (s->min_part_size >= 8) {
81 s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4));
84 for (n = 0; n < nb_samples; n++)
85 src[seg->input_offset + n] = in[n] * s->dry_gain;
88 seg->output_offset[ch] += s->min_part_size;
89 if (seg->output_offset[ch] == seg->part_size) {
90 seg->output_offset[ch] = 0;
92 memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
94 dst += seg->output_offset[ch];
95 for (n = 0; n < nb_samples; n++) {
101 if (seg->part_size < 8) {
102 memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);
104 j = seg->part_index[ch];
106 for (i = 0; i < seg->nb_partitions; i++) {
107 const int coffset = j * seg->coeff_size;
108 const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
110 direct(src, coeff, nb_samples, dst);
113 j = seg->nb_partitions;
117 seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
119 memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
121 for (n = 0; n < nb_samples; n++) {
127 memset(sum, 0, sizeof(*sum) * seg->fft_length);
128 block = (float *)seg->block->extended_data[ch] + seg->part_index[ch] * seg->block_size;
129 memset(block + seg->part_size, 0, sizeof(*block) * (seg->fft_length - seg->part_size));
131 memcpy(block, src, sizeof(*src) * seg->part_size);
133 av_rdft_calc(seg->rdft[ch], block);
134 block[2 * seg->part_size] = block[1];
137 j = seg->part_index[ch];
139 for (i = 0; i < seg->nb_partitions; i++) {
140 const int coffset = j * seg->coeff_size;
141 const float *block = (const float *)seg->block->extended_data[ch] + i * seg->block_size;
142 const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
144 s->afirdsp.fcmul_add(sum, block, (const float *)coeff, seg->part_size);
147 j = seg->nb_partitions;
151 sum[1] = sum[2 * seg->part_size];
152 av_rdft_calc(seg->irdft[ch], sum);
154 buf = (float *)seg->buffer->extended_data[ch];
155 for (n = 0; n < seg->part_size; n++) {
159 memcpy(dst, buf, seg->part_size * sizeof(*dst));
161 buf = (float *)seg->buffer->extended_data[ch];
162 memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf));
164 seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
166 memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
168 for (n = 0; n < nb_samples; n++) {
173 if (s->min_part_size >= 8) {
174 s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));
177 for (n = 0; n < nb_samples; n++)
178 ptr[n] *= s->wet_gain;
184 static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
186 AudioFIRContext *s = ctx->priv;
188 for (int offset = 0; offset < out->nb_samples; offset += s->min_part_size) {
189 fir_quantum(ctx, out, ch, offset);
195 static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
198 const int start = (out->channels * jobnr) / nb_jobs;
199 const int end = (out->channels * (jobnr+1)) / nb_jobs;
201 for (int ch = start; ch < end; ch++) {
202 fir_channel(ctx, out, ch);
208 static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
210 AVFilterContext *ctx = outlink->src;
213 out = ff_get_audio_buffer(outlink, in->nb_samples);
216 return AVERROR(ENOMEM);
219 if (s->pts == AV_NOPTS_VALUE)
222 ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels,
223 ff_filter_get_nb_threads(ctx)));
226 if (s->pts != AV_NOPTS_VALUE)
227 s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
232 return ff_filter_frame(outlink, out);
235 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
241 font = avpriv_cga_font, font_height = 8;
243 for (i = 0; txt[i]; i++) {
246 uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
247 for (char_y = 0; char_y < font_height; char_y++) {
248 for (mask = 0x80; mask; mask >>= 1) {
249 if (font[txt[i] * font_height + char_y] & mask)
253 p += pic->linesize[0] - 8 * 4;
258 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
260 int dx = FFABS(x1-x0);
261 int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
262 int err = (dx>dy ? dx : -dy) / 2, e2;
265 AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
267 if (x0 == x1 && y0 == y1)
284 static void draw_response(AVFilterContext *ctx, AVFrame *out)
286 AudioFIRContext *s = ctx->priv;
287 float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
288 float min_delay = FLT_MAX, max_delay = FLT_MIN;
289 int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
293 memset(out->data[0], 0, s->h * out->linesize[0]);
295 phase = av_malloc_array(s->w, sizeof(*phase));
296 mag = av_malloc_array(s->w, sizeof(*mag));
297 delay = av_malloc_array(s->w, sizeof(*delay));
298 if (!mag || !phase || !delay)
301 channel = av_clip(s->ir_channel, 0, s->ir[0]->channels - 1);
302 for (i = 0; i < s->w; i++) {
303 const float *src = (const float *)s->ir[0]->extended_data[channel];
304 double w = i * M_PI / (s->w - 1);
305 double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
307 for (x = 0; x < s->nb_taps; x++) {
308 real += cos(-x * w) * src[x];
309 imag += sin(-x * w) * src[x];
310 real_num += cos(-x * w) * src[x] * x;
311 imag_num += sin(-x * w) * src[x] * x;
314 mag[i] = hypot(real, imag);
315 phase[i] = atan2(imag, real);
316 div = real * real + imag * imag;
317 delay[i] = (real_num * real + imag_num * imag) / div;
318 min = fminf(min, mag[i]);
319 max = fmaxf(max, mag[i]);
320 min_delay = fminf(min_delay, delay[i]);
321 max_delay = fmaxf(max_delay, delay[i]);
324 for (i = 0; i < s->w; i++) {
325 int ymag = mag[i] / max * (s->h - 1);
326 int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
327 int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
329 ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
330 yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
331 ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
336 prev_yphase = yphase;
338 prev_ydelay = ydelay;
340 draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
341 draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
342 draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
345 prev_yphase = yphase;
346 prev_ydelay = ydelay;
349 if (s->w > 400 && s->h > 100) {
350 drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
351 snprintf(text, sizeof(text), "%.2f", max);
352 drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
354 drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
355 snprintf(text, sizeof(text), "%.2f", min);
356 drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
358 drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
359 snprintf(text, sizeof(text), "%.2f", max_delay);
360 drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
362 drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
363 snprintf(text, sizeof(text), "%.2f", min_delay);
364 drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
373 static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
374 int offset, int nb_partitions, int part_size)
376 AudioFIRContext *s = ctx->priv;
378 seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
379 seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
380 if (!seg->rdft || !seg->irdft)
381 return AVERROR(ENOMEM);
383 seg->fft_length = part_size * 2 + 1;
384 seg->part_size = part_size;
385 seg->block_size = FFALIGN(seg->fft_length, 32);
386 seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
387 seg->nb_partitions = nb_partitions;
388 seg->input_size = offset + s->min_part_size;
389 seg->input_offset = offset;
391 seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index));
392 seg->output_offset = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->output_offset));
393 if (!seg->part_index || !seg->output_offset)
394 return AVERROR(ENOMEM);
396 for (int ch = 0; ch < ctx->inputs[0]->channels && part_size >= 8; ch++) {
397 seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
398 seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);
399 if (!seg->rdft[ch] || !seg->irdft[ch])
400 return AVERROR(ENOMEM);
403 seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
404 seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
405 seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
406 seg->coeff = ff_get_audio_buffer(ctx->inputs[1], seg->nb_partitions * seg->coeff_size * 2);
407 seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
408 seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
409 if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff || !seg->input || !seg->output)
410 return AVERROR(ENOMEM);
415 static int convert_coeffs(AVFilterContext *ctx)
417 AudioFIRContext *s = ctx->priv;
418 int left, offset = 0, part_size, max_part_size;
422 s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1]);
424 return AVERROR(EINVAL);
426 if (s->minp > s->maxp) {
431 part_size = 1 << av_log2(s->minp);
432 max_part_size = 1 << av_log2(s->maxp);
434 s->min_part_size = part_size;
436 for (i = 0; left > 0; i++) {
437 int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);
438 int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
440 s->nb_segments = i + 1;
441 ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);
444 offset += nb_partitions * part_size;
445 left -= nb_partitions * part_size;
447 part_size = FFMIN(part_size, max_part_size);
450 ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->ir[0]);
457 draw_response(ctx, s->video);
466 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
467 float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
469 for (i = 0; i < s->nb_taps; i++)
470 power += FFABS(time[i]);
472 s->gain = ctx->inputs[1]->channels / power;
475 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
476 float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
478 for (i = 0; i < s->nb_taps; i++)
481 s->gain = ctx->inputs[1]->channels / power;
484 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
485 float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
487 for (i = 0; i < s->nb_taps; i++)
488 power += time[i] * time[i];
490 s->gain = sqrtf(ch / power);
496 s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
497 av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
498 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
499 float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
501 s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
504 av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
505 av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
507 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
508 float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
511 for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
514 av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
516 for (int segment = 0; segment < s->nb_segments; segment++) {
517 AudioFIRSegment *seg = &s->seg[segment];
518 float *block = (float *)seg->block->extended_data[ch];
519 FFTComplex *coeff = (FFTComplex *)seg->coeff->extended_data[ch];
521 av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
523 for (i = 0; i < seg->nb_partitions; i++) {
524 const float scale = 1.f / seg->part_size;
525 const int coffset = i * seg->coeff_size;
526 const int remaining = s->nb_taps - toffset;
527 const int size = remaining >= seg->part_size ? seg->part_size : remaining;
530 for (n = 0; n < size; n++)
531 coeff[coffset + n].re = time[toffset + n];
537 memset(block, 0, sizeof(*block) * seg->fft_length);
538 memcpy(block, time + toffset, size * sizeof(*block));
540 av_rdft_calc(seg->rdft[0], block);
542 coeff[coffset].re = block[0] * scale;
543 coeff[coffset].im = 0;
544 for (n = 1; n < seg->part_size; n++) {
545 coeff[coffset + n].re = block[2 * n] * scale;
546 coeff[coffset + n].im = block[2 * n + 1] * scale;
548 coeff[coffset + seg->part_size].re = block[1] * scale;
549 coeff[coffset + seg->part_size].im = 0;
554 av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
555 av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
556 av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
557 av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
558 av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
559 av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
560 av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
564 av_frame_free(&s->ir[0]);
570 static int check_ir(AVFilterLink *link, AVFrame *frame)
572 AVFilterContext *ctx = link->dst;
573 AudioFIRContext *s = ctx->priv;
574 int nb_taps, max_nb_taps;
576 nb_taps = ff_inlink_queued_samples(link);
577 max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
578 if (nb_taps > max_nb_taps) {
579 av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
580 return AVERROR(EINVAL);
586 static int activate(AVFilterContext *ctx)
588 AudioFIRContext *s = ctx->priv;
589 AVFilterLink *outlink = ctx->outputs[0];
590 int ret, status, available, wanted;
594 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
596 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
597 if (!s->eof_coeffs) {
600 ret = check_ir(ctx->inputs[1], ir);
604 if (ff_outlink_get_status(ctx->inputs[1]) == AVERROR_EOF)
607 if (!s->eof_coeffs) {
608 if (ff_outlink_frame_wanted(ctx->outputs[0]))
609 ff_inlink_request_frame(ctx->inputs[1]);
610 else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
611 ff_inlink_request_frame(ctx->inputs[1]);
616 if (!s->have_coeffs && s->eof_coeffs) {
617 ret = convert_coeffs(ctx);
622 available = ff_inlink_queued_samples(ctx->inputs[0]);
623 wanted = FFMAX(s->min_part_size, (available / s->min_part_size) * s->min_part_size);
624 ret = ff_inlink_consume_samples(ctx->inputs[0], wanted, wanted, &in);
626 ret = fir_frame(s, in, outlink);
631 if (s->response && s->have_coeffs) {
632 int64_t old_pts = s->video->pts;
633 int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
635 if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
636 s->video->pts = new_pts;
637 return ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
641 if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
642 ff_filter_set_ready(ctx, 10);
646 if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
647 if (status == AVERROR_EOF) {
648 ff_outlink_set_status(ctx->outputs[0], status, pts);
650 ff_outlink_set_status(ctx->outputs[1], status, pts);
655 if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
656 !ff_outlink_get_status(ctx->inputs[0])) {
657 ff_inlink_request_frame(ctx->inputs[0]);
662 ff_outlink_frame_wanted(ctx->outputs[1]) &&
663 !ff_outlink_get_status(ctx->inputs[0])) {
664 ff_inlink_request_frame(ctx->inputs[0]);
668 return FFERROR_NOT_READY;
671 static int query_formats(AVFilterContext *ctx)
673 AudioFIRContext *s = ctx->priv;
674 AVFilterFormats *formats;
675 AVFilterChannelLayouts *layouts;
676 static const enum AVSampleFormat sample_fmts[] = {
680 static const enum AVPixelFormat pix_fmts[] = {
687 AVFilterLink *videolink = ctx->outputs[1];
688 formats = ff_make_format_list(pix_fmts);
689 if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
693 layouts = ff_all_channel_counts();
695 return AVERROR(ENOMEM);
698 ret = ff_set_common_channel_layouts(ctx, layouts);
702 AVFilterChannelLayouts *mono = NULL;
704 ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
708 if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts)) < 0)
710 if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
712 if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[1]->out_channel_layouts)) < 0)
716 formats = ff_make_format_list(sample_fmts);
717 if ((ret = ff_set_common_formats(ctx, formats)) < 0)
720 formats = ff_all_samplerates();
721 return ff_set_common_samplerates(ctx, formats);
724 static int config_output(AVFilterLink *outlink)
726 AVFilterContext *ctx = outlink->src;
727 AudioFIRContext *s = ctx->priv;
729 s->one2many = ctx->inputs[1]->channels == 1;
730 outlink->sample_rate = ctx->inputs[0]->sample_rate;
731 outlink->time_base = ctx->inputs[0]->time_base;
732 outlink->channel_layout = ctx->inputs[0]->channel_layout;
733 outlink->channels = ctx->inputs[0]->channels;
735 s->nb_channels = outlink->channels;
736 s->nb_coef_channels = ctx->inputs[1]->channels;
737 s->pts = AV_NOPTS_VALUE;
742 static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
744 AudioFIRContext *s = ctx->priv;
747 for (int ch = 0; ch < s->nb_channels; ch++) {
748 av_rdft_end(seg->rdft[ch]);
751 av_freep(&seg->rdft);
754 for (int ch = 0; ch < s->nb_channels; ch++) {
755 av_rdft_end(seg->irdft[ch]);
758 av_freep(&seg->irdft);
760 av_freep(&seg->output_offset);
761 av_freep(&seg->part_index);
763 av_frame_free(&seg->block);
764 av_frame_free(&seg->sum);
765 av_frame_free(&seg->buffer);
766 av_frame_free(&seg->coeff);
767 av_frame_free(&seg->input);
768 av_frame_free(&seg->output);
772 static av_cold void uninit(AVFilterContext *ctx)
774 AudioFIRContext *s = ctx->priv;
776 for (int i = 0; i < s->nb_segments; i++) {
777 uninit_segment(ctx, &s->seg[i]);
781 av_frame_free(&s->ir[0]);
783 for (int i = 0; i < ctx->nb_outputs; i++)
784 av_freep(&ctx->output_pads[i].name);
785 av_frame_free(&s->video);
788 static int config_video(AVFilterLink *outlink)
790 AVFilterContext *ctx = outlink->src;
791 AudioFIRContext *s = ctx->priv;
793 outlink->sample_aspect_ratio = (AVRational){1,1};
796 outlink->frame_rate = s->frame_rate;
797 outlink->time_base = av_inv_q(outlink->frame_rate);
799 av_frame_free(&s->video);
800 s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
802 return AVERROR(ENOMEM);
807 void ff_afir_init(AudioFIRDSPContext *dsp)
809 dsp->fcmul_add = fcmul_add_c;
812 ff_afir_init_x86(dsp);
815 static av_cold int init(AVFilterContext *ctx)
817 AudioFIRContext *s = ctx->priv;
818 AVFilterPad pad, vpad;
822 .name = av_strdup("default"),
823 .type = AVMEDIA_TYPE_AUDIO,
824 .config_props = config_output,
828 return AVERROR(ENOMEM);
831 vpad = (AVFilterPad){
832 .name = av_strdup("filter_response"),
833 .type = AVMEDIA_TYPE_VIDEO,
834 .config_props = config_video,
837 return AVERROR(ENOMEM);
840 ret = ff_insert_outpad(ctx, 0, &pad);
847 ret = ff_insert_outpad(ctx, 1, &vpad);
849 av_freep(&vpad.name);
854 s->fdsp = avpriv_float_dsp_alloc(0);
856 return AVERROR(ENOMEM);
858 ff_afir_init(&s->afirdsp);
863 static const AVFilterPad afir_inputs[] = {
866 .type = AVMEDIA_TYPE_AUDIO,
869 .type = AVMEDIA_TYPE_AUDIO,
874 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
875 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
876 #define OFFSET(x) offsetof(AudioFIRContext, x)
878 static const AVOption afir_options[] = {
879 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
880 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
881 { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
882 { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
883 { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
884 { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
885 { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
886 { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
887 { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
888 { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
889 { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
890 { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
891 { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
892 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
893 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
894 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
895 { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
896 { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 1, 32768, AF },
897 { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 32768, AF },
901 AVFILTER_DEFINE_CLASS(afir);
903 AVFilter ff_af_afir = {
905 .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
906 .priv_size = sizeof(AudioFIRContext),
907 .priv_class = &afir_class,
908 .query_formats = query_formats,
910 .activate = activate,
912 .inputs = afir_inputs,
913 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
914 AVFILTER_FLAG_SLICE_THREADS,