2 * Copyright (c) 2017 Paul B Mahol
4 * This file is part of FFmpeg.
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23 * An arbitrary audio FIR filter
28 #include "libavutil/common.h"
29 #include "libavutil/float_dsp.h"
30 #include "libavutil/intreadwrite.h"
31 #include "libavutil/opt.h"
32 #include "libavutil/xga_font_data.h"
33 #include "libavcodec/avfft.h"
42 static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
46 for (n = 0; n < len; n++) {
47 const float cre = c[2 * n ];
48 const float cim = c[2 * n + 1];
49 const float tre = t[2 * n ];
50 const float tim = t[2 * n + 1];
52 sum[2 * n ] += tre * cre - tim * cim;
53 sum[2 * n + 1] += tre * cim + tim * cre;
56 sum[2 * n] += t[2 * n] * c[2 * n];
59 static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
61 AudioFIRContext *s = ctx->priv;
62 const float *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_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
143 AVFilterContext *ctx = outlink->src;
146 out = ff_get_audio_buffer(outlink, in->nb_samples);
149 return AVERROR(ENOMEM);
152 if (s->pts == AV_NOPTS_VALUE)
155 ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
158 if (s->pts != AV_NOPTS_VALUE)
159 s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
164 return ff_filter_frame(outlink, out);
167 static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
173 font = avpriv_cga_font, font_height = 8;
175 for (i = 0; txt[i]; i++) {
178 uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4;
179 for (char_y = 0; char_y < font_height; char_y++) {
180 for (mask = 0x80; mask; mask >>= 1) {
181 if (font[txt[i] * font_height + char_y] & mask)
185 p += pic->linesize[0] - 8 * 4;
190 static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
192 int dx = FFABS(x1-x0);
193 int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
194 int err = (dx>dy ? dx : -dy) / 2, e2;
197 AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
199 if (x0 == x1 && y0 == y1)
216 static void draw_response(AVFilterContext *ctx, AVFrame *out)
218 AudioFIRContext *s = ctx->priv;
219 float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
220 float min_delay = FLT_MAX, max_delay = FLT_MIN;
221 int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
225 memset(out->data[0], 0, s->h * out->linesize[0]);
227 phase = av_malloc_array(s->w, sizeof(*phase));
228 mag = av_malloc_array(s->w, sizeof(*mag));
229 delay = av_malloc_array(s->w, sizeof(*delay));
230 if (!mag || !phase || !delay)
233 channel = av_clip(s->ir_channel, 0, s->in[1]->channels - 1);
234 for (i = 0; i < s->w; i++) {
235 const float *src = (const float *)s->in[1]->extended_data[channel];
236 double w = i * M_PI / (s->w - 1);
237 double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
239 for (x = 0; x < s->nb_taps; x++) {
240 real += cos(-x * w) * src[x];
241 imag += sin(-x * w) * src[x];
242 real_num += cos(-x * w) * src[x] * x;
243 imag_num += sin(-x * w) * src[x] * x;
246 mag[i] = hypot(real, imag);
247 phase[i] = atan2(imag, real);
248 div = real * real + imag * imag;
249 delay[i] = (real_num * real + imag_num * imag) / div;
250 min = fminf(min, mag[i]);
251 max = fmaxf(max, mag[i]);
252 min_delay = fminf(min_delay, delay[i]);
253 max_delay = fmaxf(max_delay, delay[i]);
256 for (i = 0; i < s->w; i++) {
257 int ymag = mag[i] / max * (s->h - 1);
258 int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
259 int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
261 ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
262 yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
263 ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
268 prev_yphase = yphase;
270 prev_ydelay = ydelay;
272 draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
273 draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
274 draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
277 prev_yphase = yphase;
278 prev_ydelay = ydelay;
281 if (s->w > 400 && s->h > 100) {
282 drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD);
283 snprintf(text, sizeof(text), "%.2f", max);
284 drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD);
286 drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
287 snprintf(text, sizeof(text), "%.2f", min);
288 drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
290 drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
291 snprintf(text, sizeof(text), "%.2f", max_delay);
292 drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
294 drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
295 snprintf(text, sizeof(text), "%.2f", min_delay);
296 drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
305 static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
306 int offset, int nb_partitions, int part_size)
308 AudioFIRContext *s = ctx->priv;
310 seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
311 seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
312 if (!seg->rdft || !seg->irdft)
313 return AVERROR(ENOMEM);
315 seg->fft_length = part_size * 4 + 1;
316 seg->part_size = part_size;
317 seg->block_size = FFALIGN(seg->fft_length, 32);
318 seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
319 seg->nb_partitions = nb_partitions;
320 seg->input_size = offset + s->min_part_size;
321 seg->input_offset = offset;
323 seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index));
324 seg->output_offset = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->output_offset));
325 if (!seg->part_index || !seg->output_offset)
326 return AVERROR(ENOMEM);
328 for (int ch = 0; ch < ctx->inputs[0]->channels; ch++) {
329 seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
330 seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);
331 if (!seg->rdft[ch] || !seg->irdft[ch])
332 return AVERROR(ENOMEM);
335 seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
336 seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
337 seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
338 seg->coeff = ff_get_audio_buffer(ctx->inputs[1], seg->nb_partitions * seg->coeff_size * 2);
339 seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
340 seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
341 if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff || !seg->input || !seg->output)
342 return AVERROR(ENOMEM);
347 static int convert_coeffs(AVFilterContext *ctx)
349 AudioFIRContext *s = ctx->priv;
350 int left, offset = 0, part_size, max_part_size;
354 s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1]);
356 return AVERROR(EINVAL);
358 if (s->minp > s->maxp) {
363 part_size = 1 << av_log2(s->minp);
364 max_part_size = 1 << av_log2(s->maxp);
366 s->min_part_size = part_size;
368 for (i = 0; left > 0; i++) {
369 int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);
370 int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
372 s->nb_segments = i + 1;
373 ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);
376 offset += nb_partitions * part_size;
377 left -= nb_partitions * part_size;
379 part_size = FFMIN(part_size, max_part_size);
382 ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]);
389 draw_response(ctx, s->video);
398 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
399 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
401 for (i = 0; i < s->nb_taps; i++)
402 power += FFABS(time[i]);
404 s->gain = ctx->inputs[1]->channels / power;
407 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
408 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
410 for (i = 0; i < s->nb_taps; i++)
413 s->gain = ctx->inputs[1]->channels / power;
416 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
417 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
419 for (i = 0; i < s->nb_taps; i++)
420 power += time[i] * time[i];
422 s->gain = sqrtf(ch / power);
428 s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
429 av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
430 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
431 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
433 s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
436 av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
437 av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
439 for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
440 float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
443 for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
446 av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
448 for (int segment = 0; segment < s->nb_segments; segment++) {
449 AudioFIRSegment *seg = &s->seg[segment];
450 float *block = (float *)seg->block->extended_data[ch];
451 FFTComplex *coeff = (FFTComplex *)seg->coeff->extended_data[ch];
453 av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
455 for (i = 0; i < seg->nb_partitions; i++) {
456 const float scale = 1.f / seg->part_size;
457 const int coffset = i * seg->coeff_size;
458 const int remaining = s->nb_taps - toffset;
459 const int size = remaining >= seg->part_size ? seg->part_size : remaining;
461 memset(block, 0, sizeof(*block) * seg->fft_length);
462 memcpy(block, time + toffset, size * sizeof(*block));
464 av_rdft_calc(seg->rdft[0], block);
466 coeff[coffset].re = block[0] * scale;
467 coeff[coffset].im = 0;
468 for (n = 1; n < seg->part_size; n++) {
469 coeff[coffset + n].re = block[2 * n] * scale;
470 coeff[coffset + n].im = block[2 * n + 1] * scale;
472 coeff[coffset + seg->part_size].re = block[1] * scale;
473 coeff[coffset + seg->part_size].im = 0;
478 av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
479 av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
480 av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
481 av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
482 av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
483 av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
484 av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
488 av_frame_free(&s->in[1]);
494 static int check_ir(AVFilterLink *link, AVFrame *frame)
496 AVFilterContext *ctx = link->dst;
497 AudioFIRContext *s = ctx->priv;
498 int nb_taps, max_nb_taps;
500 nb_taps = ff_inlink_queued_samples(link);
501 max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
502 if (nb_taps > max_nb_taps) {
503 av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
504 return AVERROR(EINVAL);
510 static int activate(AVFilterContext *ctx)
512 AudioFIRContext *s = ctx->priv;
513 AVFilterLink *outlink = ctx->outputs[0];
518 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
520 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
521 if (!s->eof_coeffs) {
524 ret = check_ir(ctx->inputs[1], ir);
528 if (ff_outlink_get_status(ctx->inputs[1]) == AVERROR_EOF)
531 if (!s->eof_coeffs) {
532 if (ff_outlink_frame_wanted(ctx->outputs[0]))
533 ff_inlink_request_frame(ctx->inputs[1]);
534 else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
535 ff_inlink_request_frame(ctx->inputs[1]);
540 if (!s->have_coeffs && s->eof_coeffs) {
541 ret = convert_coeffs(ctx);
546 ret = ff_inlink_consume_samples(ctx->inputs[0], s->min_part_size, s->min_part_size, &in);
548 ret = fir_frame(s, in, outlink);
553 if (s->response && s->have_coeffs) {
554 int64_t old_pts = s->video->pts;
555 int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
557 if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
558 s->video->pts = new_pts;
559 return ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
563 if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
564 ff_filter_set_ready(ctx, 10);
568 if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
569 if (status == AVERROR_EOF) {
570 ff_outlink_set_status(ctx->outputs[0], status, pts);
572 ff_outlink_set_status(ctx->outputs[1], status, pts);
577 if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
578 !ff_outlink_get_status(ctx->inputs[0])) {
579 ff_inlink_request_frame(ctx->inputs[0]);
584 ff_outlink_frame_wanted(ctx->outputs[1]) &&
585 !ff_outlink_get_status(ctx->inputs[0])) {
586 ff_inlink_request_frame(ctx->inputs[0]);
590 return FFERROR_NOT_READY;
593 static int query_formats(AVFilterContext *ctx)
595 AudioFIRContext *s = ctx->priv;
596 AVFilterFormats *formats;
597 AVFilterChannelLayouts *layouts;
598 static const enum AVSampleFormat sample_fmts[] = {
602 static const enum AVPixelFormat pix_fmts[] = {
609 AVFilterLink *videolink = ctx->outputs[1];
610 formats = ff_make_format_list(pix_fmts);
611 if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
615 layouts = ff_all_channel_counts();
617 return AVERROR(ENOMEM);
620 ret = ff_set_common_channel_layouts(ctx, layouts);
624 AVFilterChannelLayouts *mono = NULL;
626 ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
630 if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts)) < 0)
632 if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
634 if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[1]->out_channel_layouts)) < 0)
638 formats = ff_make_format_list(sample_fmts);
639 if ((ret = ff_set_common_formats(ctx, formats)) < 0)
642 formats = ff_all_samplerates();
643 return ff_set_common_samplerates(ctx, formats);
646 static int config_output(AVFilterLink *outlink)
648 AVFilterContext *ctx = outlink->src;
649 AudioFIRContext *s = ctx->priv;
651 s->one2many = ctx->inputs[1]->channels == 1;
652 outlink->sample_rate = ctx->inputs[0]->sample_rate;
653 outlink->time_base = ctx->inputs[0]->time_base;
654 outlink->channel_layout = ctx->inputs[0]->channel_layout;
655 outlink->channels = ctx->inputs[0]->channels;
657 s->nb_channels = outlink->channels;
658 s->nb_coef_channels = ctx->inputs[1]->channels;
659 s->pts = AV_NOPTS_VALUE;
664 static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
666 AudioFIRContext *s = ctx->priv;
669 for (int ch = 0; ch < s->nb_channels; ch++) {
670 av_rdft_end(seg->rdft[ch]);
673 av_freep(&seg->rdft);
676 for (int ch = 0; ch < s->nb_channels; ch++) {
677 av_rdft_end(seg->irdft[ch]);
680 av_freep(&seg->irdft);
682 av_freep(&seg->output_offset);
683 av_freep(&seg->part_index);
685 av_frame_free(&seg->block);
686 av_frame_free(&seg->sum);
687 av_frame_free(&seg->buffer);
688 av_frame_free(&seg->coeff);
689 av_frame_free(&seg->input);
690 av_frame_free(&seg->output);
694 static av_cold void uninit(AVFilterContext *ctx)
696 AudioFIRContext *s = ctx->priv;
698 for (int i = 0; i < s->nb_segments; i++) {
699 uninit_segment(ctx, &s->seg[i]);
703 av_frame_free(&s->in[1]);
705 for (int i = 0; i < ctx->nb_outputs; i++)
706 av_freep(&ctx->output_pads[i].name);
707 av_frame_free(&s->video);
710 static int config_video(AVFilterLink *outlink)
712 AVFilterContext *ctx = outlink->src;
713 AudioFIRContext *s = ctx->priv;
715 outlink->sample_aspect_ratio = (AVRational){1,1};
718 outlink->frame_rate = s->frame_rate;
719 outlink->time_base = av_inv_q(outlink->frame_rate);
721 av_frame_free(&s->video);
722 s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
724 return AVERROR(ENOMEM);
729 static av_cold int init(AVFilterContext *ctx)
731 AudioFIRContext *s = ctx->priv;
732 AVFilterPad pad, vpad;
736 .name = av_strdup("default"),
737 .type = AVMEDIA_TYPE_AUDIO,
738 .config_props = config_output,
742 return AVERROR(ENOMEM);
745 vpad = (AVFilterPad){
746 .name = av_strdup("filter_response"),
747 .type = AVMEDIA_TYPE_VIDEO,
748 .config_props = config_video,
751 return AVERROR(ENOMEM);
754 ret = ff_insert_outpad(ctx, 0, &pad);
761 ret = ff_insert_outpad(ctx, 1, &vpad);
763 av_freep(&vpad.name);
768 s->fcmul_add = fcmul_add_c;
770 s->fdsp = avpriv_float_dsp_alloc(0);
772 return AVERROR(ENOMEM);
780 static const AVFilterPad afir_inputs[] = {
783 .type = AVMEDIA_TYPE_AUDIO,
786 .type = AVMEDIA_TYPE_AUDIO,
791 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
792 #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
793 #define OFFSET(x) offsetof(AudioFIRContext, x)
795 static const AVOption afir_options[] = {
796 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
797 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
798 { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
799 { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, AF, "gtype" },
800 { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
801 { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
802 { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
803 { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
804 { "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
805 { "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
806 { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },
807 { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
808 { "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
809 { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
810 { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
811 { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
812 { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
813 { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 16, 32768, AF },
814 { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 16, 32768, AF },
818 AVFILTER_DEFINE_CLASS(afir);
820 AVFilter ff_af_afir = {
822 .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
823 .priv_size = sizeof(AudioFIRContext),
824 .priv_class = &afir_class,
825 .query_formats = query_formats,
827 .activate = activate,
829 .inputs = afir_inputs,
830 .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
831 AVFILTER_FLAG_SLICE_THREADS,