GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavfilter/af_afir.c Lines: 14 540 2.6 %
Date: 2020-10-23 17:01:47 Branches: 2 299 0.7 %

Line Branch Exec Source
1
/*
2
 * Copyright (c) 2017 Paul B Mahol
3
 *
4
 * This file is part of FFmpeg.
5
 *
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.
10
 *
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.
15
 *
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
19
 */
20
21
/**
22
 * @file
23
 * An arbitrary audio FIR filter
24
 */
25
26
#include <float.h>
27
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"
35
36
#include "audio.h"
37
#include "avfilter.h"
38
#include "filters.h"
39
#include "formats.h"
40
#include "internal.h"
41
#include "af_afir.h"
42
43
3
static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len)
44
{
45
    int n;
46
47
771
    for (n = 0; n < len; n++) {
48
768
        const float cre = c[2 * n    ];
49
768
        const float cim = c[2 * n + 1];
50
768
        const float tre = t[2 * n    ];
51
768
        const float tim = t[2 * n + 1];
52
53
768
        sum[2 * n    ] += tre * cre - tim * cim;
54
768
        sum[2 * n + 1] += tre * cim + tim * cre;
55
    }
56
57
3
    sum[2 * n] += t[2 * n] * c[2 * n];
58
3
}
59
60
static void direct(const float *in, const FFTComplex *ir, int len, float *out)
61
{
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];
65
}
66
67
static void fir_fadd(AudioFIRContext *s, float *dst, const float *src, int nb_samples)
68
{
69
    if ((nb_samples & 15) == 0 && nb_samples >= 16) {
70
        s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples);
71
    } else {
72
        for (int n = 0; n < nb_samples; n++)
73
            dst[n] += src[n];
74
    }
75
}
76
77
static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
78
{
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);
83
    int n, i, j;
84
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];
90
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));
93
            emms_c();
94
        } else {
95
            for (n = 0; n < nb_samples; n++)
96
                src[seg->input_offset + n] = in[n] * s->dry_gain;
97
        }
98
99
        seg->output_offset[ch] += s->min_part_size;
100
        if (seg->output_offset[ch] == seg->part_size) {
101
            seg->output_offset[ch] = 0;
102
        } else {
103
            memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
104
105
            dst += seg->output_offset[ch];
106
            fir_fadd(s, ptr, dst, nb_samples);
107
            continue;
108
        }
109
110
        if (seg->part_size < 8) {
111
            memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);
112
113
            j = seg->part_index[ch];
114
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;
118
119
                direct(src, coeff, nb_samples, dst);
120
121
                if (j == 0)
122
                    j = seg->nb_partitions;
123
                j--;
124
            }
125
126
            seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
127
128
            memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
129
130
            for (n = 0; n < nb_samples; n++) {
131
                ptr[n] += dst[n];
132
            }
133
            continue;
134
        }
135
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));
139
140
        memcpy(block, src, sizeof(*src) * seg->part_size);
141
142
        av_rdft_calc(seg->rdft[ch], block);
143
        block[2 * seg->part_size] = block[1];
144
        block[1] = 0;
145
146
        j = seg->part_index[ch];
147
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;
152
153
            s->afirdsp.fcmul_add(sum, block, (const float *)coeff, seg->part_size);
154
155
            if (j == 0)
156
                j = seg->nb_partitions;
157
            j--;
158
        }
159
160
        sum[1] = sum[2 * seg->part_size];
161
        av_rdft_calc(seg->irdft[ch], sum);
162
163
        buf = (float *)seg->buffer->extended_data[ch];
164
        fir_fadd(s, buf, sum, seg->part_size);
165
166
        memcpy(dst, buf, seg->part_size * sizeof(*dst));
167
168
        buf = (float *)seg->buffer->extended_data[ch];
169
        memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf));
170
171
        seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
172
173
        memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
174
175
        fir_fadd(s, ptr, dst, nb_samples);
176
    }
177
178
    if (s->min_part_size >= 8) {
179
        s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));
180
        emms_c();
181
    } else {
182
        for (n = 0; n < nb_samples; n++)
183
            ptr[n] *= s->wet_gain;
184
    }
185
186
    return 0;
187
}
188
189
static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
190
{
191
    AudioFIRContext *s = ctx->priv;
192
193
    for (int offset = 0; offset < out->nb_samples; offset += s->min_part_size) {
194
        fir_quantum(ctx, out, ch, offset);
195
    }
196
197
    return 0;
198
}
199
200
static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
201
{
202
    AVFrame *out = arg;
203
    const int start = (out->channels * jobnr) / nb_jobs;
204
    const int end = (out->channels * (jobnr+1)) / nb_jobs;
205
206
    for (int ch = start; ch < end; ch++) {
207
        fir_channel(ctx, out, ch);
208
    }
209
210
    return 0;
211
}
212
213
static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
214
{
215
    AVFilterContext *ctx = outlink->src;
216
    AVFrame *out = NULL;
217
218
    out = ff_get_audio_buffer(outlink, in->nb_samples);
219
    if (!out) {
220
        av_frame_free(&in);
221
        return AVERROR(ENOMEM);
222
    }
223
224
    if (s->pts == AV_NOPTS_VALUE)
225
        s->pts = in->pts;
226
    s->in = in;
227
    ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels,
228
                                                               ff_filter_get_nb_threads(ctx)));
229
230
    out->pts = s->pts;
231
    if (s->pts != AV_NOPTS_VALUE)
232
        s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
233
234
    av_frame_free(&in);
235
    s->in = NULL;
236
237
    return ff_filter_frame(outlink, out);
238
}
239
240
static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
241
{
242
    const uint8_t *font;
243
    int font_height;
244
    int i;
245
246
    font = avpriv_cga_font, font_height = 8;
247
248
    for (i = 0; txt[i]; i++) {
249
        int char_y, mask;
250
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)
255
                    AV_WL32(p, color);
256
                p += 4;
257
            }
258
            p += pic->linesize[0] - 8 * 4;
259
        }
260
    }
261
}
262
263
static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color)
264
{
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;
268
269
    for (;;) {
270
        AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color);
271
272
        if (x0 == x1 && y0 == y1)
273
            break;
274
275
        e2 = err;
276
277
        if (e2 >-dx) {
278
            err -= dy;
279
            x0--;
280
        }
281
282
        if (e2 < dy) {
283
            err += dx;
284
            y0 += sy;
285
        }
286
    }
287
}
288
289
static void draw_response(AVFilterContext *ctx, AVFrame *out)
290
{
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;
295
    char text[32];
296
    int channel, i, x;
297
298
    memset(out->data[0], 0, s->h * out->linesize[0]);
299
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)
304
        goto end;
305
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.;
311
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;
317
        }
318
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]);
327
    }
328
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);
333
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);
337
338
        if (prev_ymag < 0)
339
            prev_ymag = ymag;
340
        if (prev_yphase < 0)
341
            prev_yphase = yphase;
342
        if (prev_ydelay < 0)
343
            prev_ydelay = ydelay;
344
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);
348
349
        prev_ymag   = ymag;
350
        prev_yphase = yphase;
351
        prev_ydelay = ydelay;
352
    }
353
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);
358
359
        drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
360
        snprintf(text, sizeof(text), "%.2f", min);
361
        drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
362
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);
366
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);
370
    }
371
372
end:
373
    av_free(delay);
374
    av_free(phase);
375
    av_free(mag);
376
}
377
378
static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
379
                        int offset, int nb_partitions, int part_size)
380
{
381
    AudioFIRContext *s = ctx->priv;
382
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);
387
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;
395
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);
400
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);
406
    }
407
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);
416
417
    return 0;
418
}
419
420
static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
421
{
422
    AudioFIRContext *s = ctx->priv;
423
424
    if (seg->rdft) {
425
        for (int ch = 0; ch < s->nb_channels; ch++) {
426
            av_rdft_end(seg->rdft[ch]);
427
        }
428
    }
429
    av_freep(&seg->rdft);
430
431
    if (seg->irdft) {
432
        for (int ch = 0; ch < s->nb_channels; ch++) {
433
            av_rdft_end(seg->irdft[ch]);
434
        }
435
    }
436
    av_freep(&seg->irdft);
437
438
    av_freep(&seg->output_offset);
439
    av_freep(&seg->part_index);
440
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);
447
    seg->input_size = 0;
448
}
449
450
static int convert_coeffs(AVFilterContext *ctx)
451
{
452
    AudioFIRContext *s = ctx->priv;
453
    int ret, i, ch, n, cur_nb_taps;
454
    float power = 0;
455
456
    if (!s->nb_taps) {
457
        int part_size, max_part_size;
458
        int left, offset = 0;
459
460
        s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1 + s->selir]);
461
        if (s->nb_taps <= 0)
462
            return AVERROR(EINVAL);
463
464
        if (s->minp > s->maxp) {
465
            s->maxp = s->minp;
466
        }
467
468
        left = s->nb_taps;
469
        part_size = 1 << av_log2(s->minp);
470
        max_part_size = 1 << av_log2(s->maxp);
471
472
        s->min_part_size = part_size;
473
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);
477
478
            s->nb_segments = i + 1;
479
            ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);
480
            if (ret < 0)
481
                return ret;
482
            offset += nb_partitions * part_size;
483
            left -= nb_partitions * part_size;
484
            part_size *= 2;
485
            part_size = FFMIN(part_size, max_part_size);
486
        }
487
    }
488
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]);
491
        if (ret < 0)
492
            return ret;
493
        if (ret == 0)
494
            return AVERROR_BUG;
495
    }
496
497
    if (s->response)
498
        draw_response(ctx, s->video);
499
500
    s->gain = 1;
501
    cur_nb_taps = s->ir[s->selir]->nb_samples;
502
503
    switch (s->gtype) {
504
    case -1:
505
        /* nothing to do */
506
        break;
507
    case 0:
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];
510
511
            for (i = 0; i < cur_nb_taps; i++)
512
                power += FFABS(time[i]);
513
        }
514
        s->gain = ctx->inputs[1 + s->selir]->channels / power;
515
        break;
516
    case 1:
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];
519
520
            for (i = 0; i < cur_nb_taps; i++)
521
                power += time[i];
522
        }
523
        s->gain = ctx->inputs[1 + s->selir]->channels / power;
524
        break;
525
    case 2:
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];
528
529
            for (i = 0; i < cur_nb_taps; i++)
530
                power += time[i] * time[i];
531
        }
532
        s->gain = sqrtf(ch / power);
533
        break;
534
    default:
535
        return AVERROR_BUG;
536
    }
537
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];
542
543
        s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4));
544
    }
545
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);
548
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];
551
        int toffset = 0;
552
553
        for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
554
            time[i] = 0;
555
556
        av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
557
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];
562
563
            av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
564
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;
570
571
                if (size < 8) {
572
                    for (n = 0; n < size; n++)
573
                        coeff[coffset + n].re = time[toffset + n];
574
575
                    toffset += size;
576
                    continue;
577
                }
578
579
                memset(block, 0, sizeof(*block) * seg->fft_length);
580
                memcpy(block, time + toffset, size * sizeof(*block));
581
582
                av_rdft_calc(seg->rdft[0], block);
583
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;
589
                }
590
                coeff[coffset + seg->part_size].re = block[1] * scale;
591
                coeff[coffset + seg->part_size].im = 0;
592
593
                toffset += size;
594
            }
595
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);
603
        }
604
    }
605
606
    s->have_coeffs = 1;
607
608
    return 0;
609
}
610
611
static int check_ir(AVFilterLink *link)
612
{
613
    AVFilterContext *ctx = link->dst;
614
    AudioFIRContext *s = ctx->priv;
615
    int nb_taps, max_nb_taps;
616
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);
622
    }
623
624
    return 0;
625
}
626
627
static int activate(AVFilterContext *ctx)
628
{
629
    AudioFIRContext *s = ctx->priv;
630
    AVFilterLink *outlink = ctx->outputs[0];
631
    int ret, status, available, wanted;
632
    AVFrame *in = NULL;
633
    int64_t pts;
634
635
    FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
636
    if (s->response)
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]);
640
        if (ret < 0)
641
            return ret;
642
643
        if (ff_outlink_get_status(ctx->inputs[1 + s->selir]) == AVERROR_EOF)
644
            s->eof_coeffs[s->selir] = 1;
645
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]);
651
            return 0;
652
        }
653
    }
654
655
    if (!s->have_coeffs && s->eof_coeffs[s->selir]) {
656
        ret = convert_coeffs(ctx);
657
        if (ret < 0)
658
            return ret;
659
    }
660
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);
664
    if (ret > 0)
665
        ret = fir_frame(s, in, outlink);
666
667
    if (ret < 0)
668
        return ret;
669
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);
673
674
        if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
675
            AVFrame *clone;
676
            s->video->pts = new_pts;
677
            clone = av_frame_clone(s->video);
678
            if (!clone)
679
                return AVERROR(ENOMEM);
680
            return ff_filter_frame(ctx->outputs[1], clone);
681
        }
682
    }
683
684
    if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
685
        ff_filter_set_ready(ctx, 10);
686
        return 0;
687
    }
688
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);
692
            if (s->response)
693
                ff_outlink_set_status(ctx->outputs[1], status, pts);
694
            return 0;
695
        }
696
    }
697
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]);
701
        return 0;
702
    }
703
704
    if (s->response &&
705
        ff_outlink_frame_wanted(ctx->outputs[1]) &&
706
        !ff_outlink_get_status(ctx->inputs[0])) {
707
        ff_inlink_request_frame(ctx->inputs[0]);
708
        return 0;
709
    }
710
711
    return FFERROR_NOT_READY;
712
}
713
714
static int query_formats(AVFilterContext *ctx)
715
{
716
    AudioFIRContext *s = ctx->priv;
717
    AVFilterFormats *formats;
718
    AVFilterChannelLayouts *layouts;
719
    static const enum AVSampleFormat sample_fmts[] = {
720
        AV_SAMPLE_FMT_FLTP,
721
        AV_SAMPLE_FMT_NONE
722
    };
723
    static const enum AVPixelFormat pix_fmts[] = {
724
        AV_PIX_FMT_RGB0,
725
        AV_PIX_FMT_NONE
726
    };
727
    int ret;
728
729
    if (s->response) {
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)
733
            return ret;
734
    }
735
736
    layouts = ff_all_channel_counts();
737
    if (!layouts)
738
        return AVERROR(ENOMEM);
739
740
    if (s->ir_format) {
741
        ret = ff_set_common_channel_layouts(ctx, layouts);
742
        if (ret < 0)
743
            return ret;
744
    } else {
745
        AVFilterChannelLayouts *mono = NULL;
746
747
        if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts)) < 0)
748
            return ret;
749
        if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts)) < 0)
750
            return ret;
751
752
        ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
753
        if (ret)
754
            return ret;
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)
757
                return ret;
758
        }
759
    }
760
761
    formats = ff_make_format_list(sample_fmts);
762
    if ((ret = ff_set_common_formats(ctx, formats)) < 0)
763
        return ret;
764
765
    formats = ff_all_samplerates();
766
    return ff_set_common_samplerates(ctx, formats);
767
}
768
769
static int config_output(AVFilterLink *outlink)
770
{
771
    AVFilterContext *ctx = outlink->src;
772
    AudioFIRContext *s = ctx->priv;
773
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;
779
780
    s->nb_channels = outlink->channels;
781
    s->nb_coef_channels = ctx->inputs[1 + s->selir]->channels;
782
    s->pts = AV_NOPTS_VALUE;
783
784
    return 0;
785
}
786
787
static av_cold void uninit(AVFilterContext *ctx)
788
{
789
    AudioFIRContext *s = ctx->priv;
790
791
    for (int i = 0; i < s->nb_segments; i++) {
792
        uninit_segment(ctx, &s->seg[i]);
793
    }
794
795
    av_freep(&s->fdsp);
796
797
    for (int i = 0; i < s->nb_irs; i++) {
798
        av_frame_free(&s->ir[i]);
799
    }
800
801
    for (unsigned i = 1; i < ctx->nb_inputs; i++)
802
        av_freep(&ctx->input_pads[i].name);
803
804
    av_frame_free(&s->video);
805
}
806
807
static int config_video(AVFilterLink *outlink)
808
{
809
    AVFilterContext *ctx = outlink->src;
810
    AudioFIRContext *s = ctx->priv;
811
812
    outlink->sample_aspect_ratio = (AVRational){1,1};
813
    outlink->w = s->w;
814
    outlink->h = s->h;
815
    outlink->frame_rate = s->frame_rate;
816
    outlink->time_base = av_inv_q(outlink->frame_rate);
817
818
    av_frame_free(&s->video);
819
    s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
820
    if (!s->video)
821
        return AVERROR(ENOMEM);
822
823
    return 0;
824
}
825
826
13
void ff_afir_init(AudioFIRDSPContext *dsp)
827
{
828
13
    dsp->fcmul_add = fcmul_add_c;
829
830
    if (ARCH_X86)
831
13
        ff_afir_init_x86(dsp);
832
13
}
833
834
static av_cold int init(AVFilterContext *ctx)
835
{
836
    AudioFIRContext *s = ctx->priv;
837
    AVFilterPad pad, vpad;
838
    int ret;
839
840
    pad = (AVFilterPad) {
841
        .name = "main",
842
        .type = AVMEDIA_TYPE_AUDIO,
843
    };
844
845
    ret = ff_insert_inpad(ctx, 0, &pad);
846
    if (ret < 0)
847
        return ret;
848
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,
853
        };
854
855
        if (!pad.name)
856
            return AVERROR(ENOMEM);
857
858
        ret = ff_insert_inpad(ctx, n + 1, &pad);
859
        if (ret < 0) {
860
            av_freep(&pad.name);
861
            return ret;
862
        }
863
    }
864
865
    pad = (AVFilterPad) {
866
        .name          = "default",
867
        .type          = AVMEDIA_TYPE_AUDIO,
868
        .config_props  = config_output,
869
    };
870
871
    ret = ff_insert_outpad(ctx, 0, &pad);
872
    if (ret < 0)
873
        return ret;
874
875
    if (s->response) {
876
        vpad = (AVFilterPad){
877
            .name         = "filter_response",
878
            .type         = AVMEDIA_TYPE_VIDEO,
879
            .config_props = config_video,
880
        };
881
882
        ret = ff_insert_outpad(ctx, 1, &vpad);
883
        if (ret < 0)
884
            return ret;
885
    }
886
887
    s->fdsp = avpriv_float_dsp_alloc(0);
888
    if (!s->fdsp)
889
        return AVERROR(ENOMEM);
890
891
    ff_afir_init(&s->afirdsp);
892
893
    return 0;
894
}
895
896
static int process_command(AVFilterContext *ctx,
897
                           const char *cmd,
898
                           const char *arg,
899
                           char *res,
900
                           int res_len,
901
                           int flags)
902
{
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);
906
907
    if (ret < 0)
908
        return ret;
909
910
    s->selir = FFMIN(s->nb_irs - 1, s->selir);
911
912
    if (prev_ir != s->selir) {
913
        s->have_coeffs = 0;
914
    }
915
916
    return 0;
917
}
918
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)
923
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 },
946
    { NULL }
947
};
948
949
AVFILTER_DEFINE_CLASS(afir);
950
951
AVFilter ff_af_afir = {
952
    .name          = "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,
957
    .init          = init,
958
    .activate      = activate,
959
    .uninit        = uninit,
960
    .process_command = process_command,
961
    .flags         = AVFILTER_FLAG_DYNAMIC_INPUTS  |
962
                     AVFILTER_FLAG_DYNAMIC_OUTPUTS |
963
                     AVFILTER_FLAG_SLICE_THREADS,
964
};