aboutsummaryrefslogtreecommitdiffstats
path: root/libavfilter/af_sofalizer.c
blob: f9c5fa2f2dc50f96cc2b7c92b4fa2b238c5839fe (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
1001
1002
1003
1004
1005
1006
1007
1008
1009
1010
1011
1012
1013
1014
1015
1016
1017
1018
1019
1020
1021
1022
1023
1024
1025
1026
1027
1028
1029
1030
1031
1032
1033
1034
1035
1036
1037
1038
1039
1040
1041
1042
1043
1044
1045
1046
1047
1048
1049
1050
1051
1052
1053
1054
1055
1056
1057
1058
1059
1060
1061
1062
1063
1064
1065
1066
1067
1068
1069
1070
1071
1072
1073
1074
1075
1076
1077
1078
1079
1080
1081
1082
1083
1084
1085
1086
1087
1088
1089
1090
1091
1092
1093
1094
1095
1096
1097
1098
1099
1100
1101
1102
1103
1104
1105
1106
1107
1108
/*****************************************************************************
 * sofalizer.c : SOFAlizer filter for virtual binaural acoustics
 *****************************************************************************
 * Copyright (C) 2013-2015 Andreas Fuchs, Wolfgang Hrauda,
 *                         Acoustics Research Institute (ARI), Vienna, Austria
 *
 * Authors: Andreas Fuchs <andi.fuchs.mail@gmail.com>
 *          Wolfgang Hrauda <wolfgang.hrauda@gmx.at>
 *
 * SOFAlizer project coordinator at ARI, main developer of SOFA:
 *          Piotr Majdak <piotr@majdak.at>
 *
 * This program is free software; you can redistribute it and/or modify it
 * under the terms of the GNU Lesser General Public License as published by
 * the Free Software Foundation; either version 2.1 of the License, or
 * (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
 * GNU Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public License
 * along with this program; if not, write to the Free Software Foundation,
 * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
 *****************************************************************************/

#include <math.h>
#include <mysofa.h>

#include "libavutil/tx.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/intmath.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"
#include "audio.h"

#define TIME_DOMAIN      0
#define FREQUENCY_DOMAIN 1

typedef struct MySofa {  /* contains data of one SOFA file */
    struct MYSOFA_HRTF *hrtf;
    struct MYSOFA_LOOKUP *lookup;
    struct MYSOFA_NEIGHBORHOOD *neighborhood;
    int ir_samples;      /* length of one impulse response (IR) */
    int n_samples;       /* ir_samples to next power of 2 */
    float *lir, *rir;    /* IRs (time-domain) */
    float *fir;
    int max_delay;
} MySofa;

typedef struct VirtualSpeaker {
    uint8_t set;
    float azim;
    float elev;
} VirtualSpeaker;

typedef struct SOFAlizerContext {
    const AVClass *class;

    char *filename;             /* name of SOFA file */
    MySofa sofa;                /* contains data of the SOFA file */

    int sample_rate;            /* sample rate from SOFA file */
    float *speaker_azim;        /* azimuth of the virtual loudspeakers */
    float *speaker_elev;        /* elevation of the virtual loudspeakers */
    char *speakers_pos;         /* custom positions of the virtual loudspeakers */
    float lfe_gain;             /* initial gain for the LFE channel */
    float gain_lfe;             /* gain applied to LFE channel */
    int lfe_channel;            /* LFE channel position in channel layout */

    int n_conv;                 /* number of channels to convolute */

                                /* buffer variables (for convolution) */
    float *ringbuffer[2];       /* buffers input samples, length of one buffer: */
                                /* no. input ch. (incl. LFE) x buffer_length */
    int write[2];               /* current write position to ringbuffer */
    int buffer_length;          /* is: longest IR plus max. delay in all SOFA files */
                                /* then choose next power of 2 */
    int n_fft;                  /* number of samples in one FFT block */
    int nb_samples;

                                /* netCDF variables */
    int *delay[2];              /* broadband delay for each channel/IR to be convolved */

    float *data_ir[2];          /* IRs for all channels to be convolved */
                                /* (this excludes the LFE) */
    float *temp_src[2];
    AVComplexFloat *in_fft[2];   /* Array to hold input FFT values */
    AVComplexFloat *out_fft[2];  /* Array to hold output FFT values */
    AVComplexFloat *temp_afft[2];   /* Array to accumulate FFT values prior to IFFT */

                         /* control variables */
    float gain;          /* filter gain (in dB) */
    float rotation;      /* rotation of virtual loudspeakers (in degrees)  */
    float elevation;     /* elevation of virtual loudspeakers (in deg.) */
    float radius;        /* distance virtual loudspeakers to listener (in metres) */
    int type;            /* processing type */
    int framesize;       /* size of buffer */
    int normalize;       /* should all IRs be normalized upon import ? */
    int interpolate;     /* should wanted IRs be interpolated from neighbors ? */
    int minphase;        /* should all IRs be minphased upon import ? */
    float anglestep;     /* neighbor search angle step, in agles */
    float radstep;       /* neighbor search radius step, in meters */

    VirtualSpeaker vspkrpos[64];

    AVTXContext *fft[2], *ifft[2];
    av_tx_fn tx_fn[2], itx_fn[2];
    AVComplexFloat *data_hrtf[2];

    AVFloatDSPContext *fdsp;
} SOFAlizerContext;

static int close_sofa(struct MySofa *sofa)
{
    if (sofa->neighborhood)
        mysofa_neighborhood_free(sofa->neighborhood);
    sofa->neighborhood = NULL;
    if (sofa->lookup)
        mysofa_lookup_free(sofa->lookup);
    sofa->lookup = NULL;
    if (sofa->hrtf)
        mysofa_free(sofa->hrtf);
    sofa->hrtf = NULL;
    av_freep(&sofa->fir);

    return 0;
}

static int preload_sofa(AVFilterContext *ctx, char *filename, int *samplingrate)
{
    struct SOFAlizerContext *s = ctx->priv;
    struct MYSOFA_HRTF *mysofa;
    char *license;
    int ret;

    mysofa = mysofa_load(filename, &ret);
    s->sofa.hrtf = mysofa;
    if (ret || !mysofa) {
        av_log(ctx, AV_LOG_ERROR, "Can't find SOFA-file '%s'\n", filename);
        return AVERROR(EINVAL);
    }

    ret = mysofa_check(mysofa);
    if (ret != MYSOFA_OK) {
        av_log(ctx, AV_LOG_ERROR, "Selected SOFA file is invalid. Please select valid SOFA file.\n");
        return ret;
    }

    if (s->normalize)
        mysofa_loudness(s->sofa.hrtf);

    if (s->minphase)
        mysofa_minphase(s->sofa.hrtf, 0.01f);

    mysofa_tocartesian(s->sofa.hrtf);

    s->sofa.lookup = mysofa_lookup_init(s->sofa.hrtf);
    if (s->sofa.lookup == NULL)
        return AVERROR(EINVAL);

    if (s->interpolate)
        s->sofa.neighborhood = mysofa_neighborhood_init_withstepdefine(s->sofa.hrtf,
                                                                       s->sofa.lookup,
                                                                       s->anglestep,
                                                                       s->radstep);

    s->sofa.fir = av_calloc(s->sofa.hrtf->N * s->sofa.hrtf->R, sizeof(*s->sofa.fir));
    if (!s->sofa.fir)
        return AVERROR(ENOMEM);

    if (mysofa->DataSamplingRate.elements != 1)
        return AVERROR(EINVAL);
    av_log(ctx, AV_LOG_DEBUG, "Original IR length: %d.\n", mysofa->N);
    *samplingrate = mysofa->DataSamplingRate.values[0];
    license = mysofa_getAttribute(mysofa->attributes, (char *)"License");
    if (license)
        av_log(ctx, AV_LOG_INFO, "SOFA license: %s\n", license);

    return 0;
}

static int parse_channel_name(AVFilterContext *ctx, char **arg, int *rchannel)
{
    int len;
    enum AVChannel channel_id = 0;
    char buf[8] = {0};

    /* try to parse a channel name, e.g. "FL" */
    if (av_sscanf(*arg, "%7[A-Z]%n", buf, &len)) {
        channel_id = av_channel_from_string(buf);
        if (channel_id < 0 || channel_id >= 64) {
            av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", buf);
            return AVERROR(EINVAL);
        }

        *rchannel = channel_id;
        *arg += len;
        return 0;
    } else if (av_sscanf(*arg, "%d%n", &channel_id, &len) == 1) {
        if (channel_id < 0 || channel_id >= 64) {
            av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%d\' as channel number.\n", channel_id);
            return AVERROR(EINVAL);
        }
        *rchannel = channel_id;
        *arg += len;
        return 0;
    }
    return AVERROR(EINVAL);
}

static void parse_speaker_pos(AVFilterContext *ctx)
{
    SOFAlizerContext *s = ctx->priv;
    char *arg, *tokenizer, *p, *args = av_strdup(s->speakers_pos);

    if (!args)
        return;
    p = args;

    while ((arg = av_strtok(p, "|", &tokenizer))) {
        float azim, elev;
        int out_ch_id;

        p = NULL;
        if (parse_channel_name(ctx, &arg, &out_ch_id)) {
            continue;
        }
        if (av_sscanf(arg, "%f %f", &azim, &elev) == 2) {
            s->vspkrpos[out_ch_id].set = 1;
            s->vspkrpos[out_ch_id].azim = azim;
            s->vspkrpos[out_ch_id].elev = elev;
        } else if (av_sscanf(arg, "%f", &azim) == 1) {
            s->vspkrpos[out_ch_id].set = 1;
            s->vspkrpos[out_ch_id].azim = azim;
            s->vspkrpos[out_ch_id].elev = 0;
        }
    }

    av_free(args);
}

static int get_speaker_pos(AVFilterContext *ctx,
                           float *speaker_azim, float *speaker_elev)
{
    struct SOFAlizerContext *s = ctx->priv;
    AVChannelLayout *channel_layout = &ctx->inputs[0]->ch_layout;
    float azim[64] = { 0 };
    float elev[64] = { 0 };
    int ch, n_conv = ctx->inputs[0]->ch_layout.nb_channels; /* get no. input channels */

    if (n_conv < 0 || n_conv > 64)
        return AVERROR(EINVAL);

    s->lfe_channel = -1;

    if (s->speakers_pos)
        parse_speaker_pos(ctx);

    /* set speaker positions according to input channel configuration: */
    for (ch = 0; ch < n_conv; ch++) {
        int chan = av_channel_layout_channel_from_index(channel_layout, ch);

        switch (chan) {
        case AV_CHAN_FRONT_LEFT:          azim[ch] =  30;      break;
        case AV_CHAN_FRONT_RIGHT:         azim[ch] = 330;      break;
        case AV_CHAN_FRONT_CENTER:        azim[ch] =   0;      break;
        case AV_CHAN_LOW_FREQUENCY:
        case AV_CHAN_LOW_FREQUENCY_2:     s->lfe_channel = ch; break;
        case AV_CHAN_BACK_LEFT:           azim[ch] = 150;      break;
        case AV_CHAN_BACK_RIGHT:          azim[ch] = 210;      break;
        case AV_CHAN_BACK_CENTER:         azim[ch] = 180;      break;
        case AV_CHAN_SIDE_LEFT:           azim[ch] =  90;      break;
        case AV_CHAN_SIDE_RIGHT:          azim[ch] = 270;      break;
        case AV_CHAN_FRONT_LEFT_OF_CENTER:  azim[ch] =  15;    break;
        case AV_CHAN_FRONT_RIGHT_OF_CENTER: azim[ch] = 345;    break;
        case AV_CHAN_TOP_CENTER:          azim[ch] =   0;
                                          elev[ch] =  90;      break;
        case AV_CHAN_TOP_FRONT_LEFT:      azim[ch] =  30;
                                          elev[ch] =  45;      break;
        case AV_CHAN_TOP_FRONT_CENTER:    azim[ch] =   0;
                                          elev[ch] =  45;      break;
        case AV_CHAN_TOP_FRONT_RIGHT:     azim[ch] = 330;
                                          elev[ch] =  45;      break;
        case AV_CHAN_TOP_BACK_LEFT:       azim[ch] = 150;
                                          elev[ch] =  45;      break;
        case AV_CHAN_TOP_BACK_RIGHT:      azim[ch] = 210;
                                          elev[ch] =  45;      break;
        case AV_CHAN_TOP_BACK_CENTER:     azim[ch] = 180;
                                          elev[ch] =  45;      break;
        case AV_CHAN_WIDE_LEFT:           azim[ch] =  90;      break;
        case AV_CHAN_WIDE_RIGHT:          azim[ch] = 270;      break;
        case AV_CHAN_SURROUND_DIRECT_LEFT:  azim[ch] =  90;    break;
        case AV_CHAN_SURROUND_DIRECT_RIGHT: azim[ch] = 270;    break;
        case AV_CHAN_STEREO_LEFT:         azim[ch] =  90;      break;
        case AV_CHAN_STEREO_RIGHT:        azim[ch] = 270;      break;
        default:
            return AVERROR(EINVAL);
        }

        if (s->vspkrpos[ch].set) {
            azim[ch] = s->vspkrpos[ch].azim;
            elev[ch] = s->vspkrpos[ch].elev;
        }
    }

    memcpy(speaker_azim, azim, n_conv * sizeof(float));
    memcpy(speaker_elev, elev, n_conv * sizeof(float));

    return 0;

}

typedef struct ThreadData {
    AVFrame *in, *out;
    int *write;
    int **delay;
    float **ir;
    int *n_clippings;
    float **ringbuffer;
    float **temp_src;
    AVComplexFloat **in_fft;
    AVComplexFloat **out_fft;
    AVComplexFloat **temp_afft;
} ThreadData;

static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
    SOFAlizerContext *s = ctx->priv;
    ThreadData *td = arg;
    AVFrame *in = td->in, *out = td->out;
    int offset = jobnr;
    int *write = &td->write[jobnr];
    const int *const delay = td->delay[jobnr];
    const float *const ir = td->ir[jobnr];
    int *n_clippings = &td->n_clippings[jobnr];
    float *ringbuffer = td->ringbuffer[jobnr];
    float *temp_src = td->temp_src[jobnr];
    const int ir_samples = s->sofa.ir_samples; /* length of one IR */
    const int n_samples = s->sofa.n_samples;
    const int planar = in->format == AV_SAMPLE_FMT_FLTP;
    const int mult = 1 + !planar;
    const float *src = (const float *)in->extended_data[0]; /* get pointer to audio input buffer */
    float *dst = (float *)out->extended_data[jobnr * planar]; /* get pointer to audio output buffer */
    const int in_channels = s->n_conv; /* number of input channels */
    /* ring buffer length is: longest IR plus max. delay -> next power of 2 */
    const int buffer_length = s->buffer_length;
    /* -1 for AND instead of MODULO (applied to powers of 2): */
    const uint32_t modulo = (uint32_t)buffer_length - 1;
    float *buffer[64]; /* holds ringbuffer for each input channel */
    int wr = *write;
    int read;
    int i, l;

    if (!planar)
        dst += offset;

    for (l = 0; l < in_channels; l++) {
        /* get starting address of ringbuffer for each input channel */
        buffer[l] = ringbuffer + l * buffer_length;
    }

    for (i = 0; i < in->nb_samples; i++) {
        const float *temp_ir = ir; /* using same set of IRs for each sample */

        dst[0] = 0;
        if (planar) {
            for (l = 0; l < in_channels; l++) {
                const float *srcp = (const float *)in->extended_data[l];

                /* write current input sample to ringbuffer (for each channel) */
                buffer[l][wr] = srcp[i];
            }
        } else {
            for (l = 0; l < in_channels; l++) {
                /* write current input sample to ringbuffer (for each channel) */
                buffer[l][wr] = src[l];
            }
        }

        /* loop goes through all channels to be convolved */
        for (l = 0; l < in_channels; l++) {
            const float *const bptr = buffer[l];

            if (l == s->lfe_channel) {
                /* LFE is an input channel but requires no convolution */
                /* apply gain to LFE signal and add to output buffer */
                dst[0] += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
                temp_ir += n_samples;
                continue;
            }

            /* current read position in ringbuffer: input sample write position
             * - delay for l-th ch. + diff. betw. IR length and buffer length
             * (mod buffer length) */
            read = (wr - delay[l] - (ir_samples - 1) + buffer_length) & modulo;

            if (read + ir_samples < buffer_length) {
                memmove(temp_src, bptr + read, ir_samples * sizeof(*temp_src));
            } else {
                int len = FFMIN(n_samples - (read % ir_samples), buffer_length - read);

                memmove(temp_src, bptr + read, len * sizeof(*temp_src));
                memmove(temp_src + len, bptr, (n_samples - len) * sizeof(*temp_src));
            }

            /* multiply signal and IR, and add up the results */
            dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, FFALIGN(ir_samples, 32));
            temp_ir += n_samples;
        }

        /* clippings counter */
        if (fabsf(dst[0]) > 1)
            n_clippings[0]++;

        /* move output buffer pointer by +2 to get to next sample of processed channel: */
        dst += mult;
        src += in_channels;
        wr   = (wr + 1) & modulo; /* update ringbuffer write position */
    }

    *write = wr; /* remember write position in ringbuffer for next call */

    return 0;
}

static int sofalizer_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
    SOFAlizerContext *s = ctx->priv;
    ThreadData *td = arg;
    AVFrame *in = td->in, *out = td->out;
    int offset = jobnr;
    int *write = &td->write[jobnr];
    AVComplexFloat *hrtf = s->data_hrtf[jobnr]; /* get pointers to current HRTF data */
    int *n_clippings = &td->n_clippings[jobnr];
    float *ringbuffer = td->ringbuffer[jobnr];
    const int ir_samples = s->sofa.ir_samples; /* length of one IR */
    const int planar = in->format == AV_SAMPLE_FMT_FLTP;
    const int mult = 1 + !planar;
    float *dst = (float *)out->extended_data[jobnr * planar]; /* get pointer to audio output buffer */
    const int in_channels = s->n_conv; /* number of input channels */
    /* ring buffer length is: longest IR plus max. delay -> next power of 2 */
    const int buffer_length = s->buffer_length;
    /* -1 for AND instead of MODULO (applied to powers of 2): */
    const uint32_t modulo = (uint32_t)buffer_length - 1;
    AVComplexFloat *fft_in = s->in_fft[jobnr]; /* temporary array for FFT input data */
    AVComplexFloat *fft_out = s->out_fft[jobnr]; /* temporary array for FFT output data */
    AVComplexFloat *fft_acc = s->temp_afft[jobnr];
    AVTXContext *ifft = s->ifft[jobnr];
    av_tx_fn itx_fn = s->itx_fn[jobnr];
    AVTXContext *fft = s->fft[jobnr];
    av_tx_fn tx_fn = s->tx_fn[jobnr];
    const int n_conv = s->n_conv;
    const int n_fft = s->n_fft;
    const float fft_scale = 1.0f / s->n_fft;
    AVComplexFloat *hrtf_offset;
    int wr = *write;
    int n_read;
    int i, j;

    if (!planar)
        dst += offset;

    /* find minimum between number of samples and output buffer length:
     * (important, if one IR is longer than the output buffer) */
    n_read = FFMIN(ir_samples, in->nb_samples);
    for (j = 0; j < n_read; j++) {
        /* initialize output buf with saved signal from overflow buf */
        dst[mult * j]  = ringbuffer[wr];
        ringbuffer[wr] = 0.0f; /* re-set read samples to zero */
        /* update ringbuffer read/write position */
        wr  = (wr + 1) & modulo;
    }

    /* initialize rest of output buffer with 0 */
    for (j = n_read; j < in->nb_samples; j++) {
        dst[mult * j] = 0;
    }

    /* fill FFT accumulation with 0 */
    memset(fft_acc, 0, sizeof(AVComplexFloat) * n_fft);

    for (i = 0; i < n_conv; i++) {
        const float *src = (const float *)in->extended_data[i * planar]; /* get pointer to audio input buffer */

        if (i == s->lfe_channel) { /* LFE */
            if (in->format == AV_SAMPLE_FMT_FLT) {
                for (j = 0; j < in->nb_samples; j++) {
                    /* apply gain to LFE signal and add to output buffer */
                    dst[2 * j] += src[i + j * in_channels] * s->gain_lfe;
                }
            } else {
                for (j = 0; j < in->nb_samples; j++) {
                    /* apply gain to LFE signal and add to output buffer */
                    dst[j] += src[j] * s->gain_lfe;
                }
            }
            continue;
        }

        /* outer loop: go through all input channels to be convolved */
        offset = i * n_fft; /* no. samples already processed */
        hrtf_offset = hrtf + offset;

        /* fill FFT input with 0 (we want to zero-pad) */
        memset(fft_in, 0, sizeof(AVComplexFloat) * n_fft);

        if (in->format == AV_SAMPLE_FMT_FLT) {
            for (j = 0; j < in->nb_samples; j++) {
                /* prepare input for FFT */
                /* write all samples of current input channel to FFT input array */
                fft_in[j].re = src[j * in_channels + i];
            }
        } else {
            for (j = 0; j < in->nb_samples; j++) {
                /* prepare input for FFT */
                /* write all samples of current input channel to FFT input array */
                fft_in[j].re = src[j];
            }
        }

        /* transform input signal of current channel to frequency domain */
        tx_fn(fft, fft_out, fft_in, sizeof(float));

        for (j = 0; j < n_fft; j++) {
            const AVComplexFloat *hcomplex = hrtf_offset + j;
            const float re = fft_out[j].re;
            const float im = fft_out[j].im;

            /* complex multiplication of input signal and HRTFs */
            /* output channel (real): */
            fft_acc[j].re += re * hcomplex->re - im * hcomplex->im;
            /* output channel (imag): */
            fft_acc[j].im += re * hcomplex->im + im * hcomplex->re;
        }
    }

    /* transform output signal of current channel back to time domain */
    itx_fn(ifft, fft_out, fft_acc, sizeof(float));

    for (j = 0; j < in->nb_samples; j++) {
        /* write output signal of current channel to output buffer */
        dst[mult * j] += fft_out[j].re * fft_scale;
    }

    for (j = 0; j < ir_samples - 1; j++) { /* overflow length is IR length - 1 */
        /* write the rest of output signal to overflow buffer */
        int write_pos = (wr + j) & modulo;

        *(ringbuffer + write_pos) += fft_out[in->nb_samples + j].re * fft_scale;
    }

    /* go through all samples of current output buffer: count clippings */
    for (i = 0; i < out->nb_samples; i++) {
        /* clippings counter */
        if (fabsf(dst[i * mult]) > 1) { /* if current output sample > 1 */
            n_clippings[0]++;
        }
    }

    /* remember read/write position in ringbuffer for next call */
    *write = wr;

    return 0;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
    AVFilterContext *ctx = inlink->dst;
    SOFAlizerContext *s = ctx->priv;
    AVFilterLink *outlink = ctx->outputs[0];
    int n_clippings[2] = { 0 };
    ThreadData td;
    AVFrame *out;

    out = ff_get_audio_buffer(outlink, in->nb_samples);
    if (!out) {
        av_frame_free(&in);
        return AVERROR(ENOMEM);
    }
    av_frame_copy_props(out, in);

    td.in = in; td.out = out; td.write = s->write;
    td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings;
    td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src;
    td.in_fft = s->in_fft;
    td.out_fft = s->out_fft;
    td.temp_afft = s->temp_afft;

    if (s->type == TIME_DOMAIN) {
        ff_filter_execute(ctx, sofalizer_convolute, &td, NULL, 2);
    } else if (s->type == FREQUENCY_DOMAIN) {
        ff_filter_execute(ctx, sofalizer_fast_convolute, &td, NULL, 2);
    }
    emms_c();

    /* display error message if clipping occurred */
    if (n_clippings[0] + n_clippings[1] > 0) {
        av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.\n",
               n_clippings[0] + n_clippings[1], out->nb_samples * 2);
    }

    av_frame_free(&in);
    return ff_filter_frame(outlink, out);
}

static int activate(AVFilterContext *ctx)
{
    AVFilterLink *inlink = ctx->inputs[0];
    AVFilterLink *outlink = ctx->outputs[0];
    SOFAlizerContext *s = ctx->priv;
    AVFrame *in;
    int ret;

    FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);

    if (s->nb_samples)
        ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in);
    else
        ret = ff_inlink_consume_frame(inlink, &in);
    if (ret < 0)
        return ret;
    if (ret > 0)
        return filter_frame(inlink, in);

    FF_FILTER_FORWARD_STATUS(inlink, outlink);
    FF_FILTER_FORWARD_WANTED(outlink, inlink);

    return FFERROR_NOT_READY;
}

static int query_formats(AVFilterContext *ctx)
{
    struct SOFAlizerContext *s = ctx->priv;
    AVFilterChannelLayouts *layouts = NULL;
    int ret, sample_rates[] = { 48000, -1 };
    static const enum AVSampleFormat sample_fmts[] = {
        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
        AV_SAMPLE_FMT_NONE
    };

    ret = ff_set_common_formats_from_list(ctx, sample_fmts);
    if (ret)
        return ret;

    layouts = ff_all_channel_layouts();
    if (!layouts)
        return AVERROR(ENOMEM);

    ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts);
    if (ret)
        return ret;

    layouts = NULL;
    ret = ff_add_channel_layout(&layouts, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
    if (ret)
        return ret;

    ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts);
    if (ret)
        return ret;

    sample_rates[0] = s->sample_rate;
    return ff_set_common_samplerates_from_list(ctx, sample_rates);
}

static int getfilter_float(AVFilterContext *ctx, float x, float y, float z,
                           float *left, float *right,
                           float *delay_left, float *delay_right)
{
    struct SOFAlizerContext *s = ctx->priv;
    float c[3], delays[2];
    float *fl, *fr;
    int nearest;
    int *neighbors;
    float *res;

    c[0] = x, c[1] = y, c[2] = z;
    nearest = mysofa_lookup(s->sofa.lookup, c);
    if (nearest < 0)
        return AVERROR(EINVAL);

    if (s->interpolate) {
        neighbors = mysofa_neighborhood(s->sofa.neighborhood, nearest);
        res = mysofa_interpolate(s->sofa.hrtf, c,
                                 nearest, neighbors,
                                 s->sofa.fir, delays);
    } else {
        if (s->sofa.hrtf->DataDelay.elements > s->sofa.hrtf->R) {
            delays[0] = s->sofa.hrtf->DataDelay.values[nearest * s->sofa.hrtf->R];
            delays[1] = s->sofa.hrtf->DataDelay.values[nearest * s->sofa.hrtf->R + 1];
        } else {
            delays[0] = s->sofa.hrtf->DataDelay.values[0];
            delays[1] = s->sofa.hrtf->DataDelay.values[1];
        }
        res = s->sofa.hrtf->DataIR.values + nearest * s->sofa.hrtf->N * s->sofa.hrtf->R;
    }

    *delay_left  = delays[0];
    *delay_right = delays[1];

    fl = res;
    fr = res + s->sofa.hrtf->N;

    memcpy(left, fl, sizeof(float) * s->sofa.hrtf->N);
    memcpy(right, fr, sizeof(float) * s->sofa.hrtf->N);

    return 0;
}

static int load_data(AVFilterContext *ctx, int azim, int elev, float radius, int sample_rate)
{
    struct SOFAlizerContext *s = ctx->priv;
    int n_samples;
    int ir_samples;
    int n_conv = s->n_conv; /* no. channels to convolve */
    int n_fft;
    float delay_l; /* broadband delay for each IR */
    float delay_r;
    int nb_input_channels = ctx->inputs[0]->ch_layout.nb_channels; /* no. input channels */
    float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10); /* gain - 3dB/channel */
    AVComplexFloat *data_hrtf_l = NULL;
    AVComplexFloat *data_hrtf_r = NULL;
    AVComplexFloat *fft_out_l = NULL;
    AVComplexFloat *fft_out_r = NULL;
    AVComplexFloat *fft_in_l = NULL;
    AVComplexFloat *fft_in_r = NULL;
    float *data_ir_l = NULL;
    float *data_ir_r = NULL;
    int offset = 0; /* used for faster pointer arithmetics in for-loop */
    int i, j, azim_orig = azim, elev_orig = elev;
    int ret = 0;
    int n_current;
    int n_max = 0;

    av_log(ctx, AV_LOG_DEBUG, "IR length: %d.\n", s->sofa.hrtf->N);
    s->sofa.ir_samples = s->sofa.hrtf->N;
    s->sofa.n_samples = 1 << (32 - ff_clz(s->sofa.ir_samples));

    n_samples = s->sofa.n_samples;
    ir_samples = s->sofa.ir_samples;

    if (s->type == TIME_DOMAIN) {
        s->data_ir[0] = av_calloc(n_samples, sizeof(float) * s->n_conv);
        s->data_ir[1] = av_calloc(n_samples, sizeof(float) * s->n_conv);

        if (!s->data_ir[0] || !s->data_ir[1]) {
            ret = AVERROR(ENOMEM);
            goto fail;
        }
    }

    s->delay[0] = av_calloc(s->n_conv, sizeof(int));
    s->delay[1] = av_calloc(s->n_conv, sizeof(int));

    if (!s->delay[0] || !s->delay[1]) {
        ret = AVERROR(ENOMEM);
        goto fail;
    }

    /* get temporary IR for L and R channel */
    data_ir_l = av_calloc(n_conv * n_samples, sizeof(*data_ir_l));
    data_ir_r = av_calloc(n_conv * n_samples, sizeof(*data_ir_r));
    if (!data_ir_r || !data_ir_l) {
        ret = AVERROR(ENOMEM);
        goto fail;
    }

    if (s->type == TIME_DOMAIN) {
        s->temp_src[0] = av_calloc(n_samples, sizeof(float));
        s->temp_src[1] = av_calloc(n_samples, sizeof(float));
        if (!s->temp_src[0] || !s->temp_src[1]) {
            ret = AVERROR(ENOMEM);
            goto fail;
        }
    }

    s->speaker_azim = av_calloc(s->n_conv, sizeof(*s->speaker_azim));
    s->speaker_elev = av_calloc(s->n_conv, sizeof(*s->speaker_elev));
    if (!s->speaker_azim || !s->speaker_elev) {
        ret = AVERROR(ENOMEM);
        goto fail;
    }

    /* get speaker positions */
    if ((ret = get_speaker_pos(ctx, s->speaker_azim, s->speaker_elev)) < 0) {
        av_log(ctx, AV_LOG_ERROR, "Couldn't get speaker positions. Input channel configuration not supported.\n");
        goto fail;
    }

    for (i = 0; i < s->n_conv; i++) {
        float coordinates[3];

        /* load and store IRs and corresponding delays */
        azim = (int)(s->speaker_azim[i] + azim_orig) % 360;
        elev = (int)(s->speaker_elev[i] + elev_orig) % 90;

        coordinates[0] = azim;
        coordinates[1] = elev;
        coordinates[2] = radius;

        mysofa_s2c(coordinates);

        /* get id of IR closest to desired position */
        ret = getfilter_float(ctx, coordinates[0], coordinates[1], coordinates[2],
                              data_ir_l + n_samples * i,
                              data_ir_r + n_samples * i,
                              &delay_l, &delay_r);
        if (ret < 0)
            goto fail;

        s->delay[0][i] = delay_l * sample_rate;
        s->delay[1][i] = delay_r * sample_rate;

        s->sofa.max_delay = FFMAX3(s->sofa.max_delay, s->delay[0][i], s->delay[1][i]);
    }

    /* get size of ringbuffer (longest IR plus max. delay) */
    /* then choose next power of 2 for performance optimization */
    n_current = n_samples + s->sofa.max_delay;
    /* length of longest IR plus max. delay */
    n_max = FFMAX(n_max, n_current);

    /* buffer length is longest IR plus max. delay -> next power of 2
       (32 - count leading zeros gives required exponent)  */
    s->buffer_length = 1 << (32 - ff_clz(n_max));
    s->n_fft = n_fft = 1 << (32 - ff_clz(n_max + s->framesize));

    if (s->type == FREQUENCY_DOMAIN) {
        float scale;

        av_tx_uninit(&s->fft[0]);
        av_tx_uninit(&s->fft[1]);
        ret = av_tx_init(&s->fft[0], &s->tx_fn[0], AV_TX_FLOAT_FFT, 0, s->n_fft, &scale, 0);
        if (ret < 0)
            goto fail;
        ret = av_tx_init(&s->fft[1], &s->tx_fn[1], AV_TX_FLOAT_FFT, 0, s->n_fft, &scale, 0);
        if (ret < 0)
            goto fail;
        av_tx_uninit(&s->ifft[0]);
        av_tx_uninit(&s->ifft[1]);
        ret = av_tx_init(&s->ifft[0], &s->itx_fn[0], AV_TX_FLOAT_FFT, 1, s->n_fft, &scale, 0);
        if (ret < 0)
            goto fail;
        ret = av_tx_init(&s->ifft[1], &s->itx_fn[1], AV_TX_FLOAT_FFT, 1, s->n_fft, &scale, 0);
        if (ret < 0)
            goto fail;
    }

    if (s->type == TIME_DOMAIN) {
        s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
        s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
    } else if (s->type == FREQUENCY_DOMAIN) {
        /* get temporary HRTF memory for L and R channel */
        data_hrtf_l = av_malloc_array(n_fft, sizeof(*data_hrtf_l) * n_conv);
        data_hrtf_r = av_malloc_array(n_fft, sizeof(*data_hrtf_r) * n_conv);
        if (!data_hrtf_r || !data_hrtf_l) {
            ret = AVERROR(ENOMEM);
            goto fail;
        }

        s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float));
        s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float));
        s->in_fft[0] = av_malloc_array(s->n_fft, sizeof(AVComplexFloat));
        s->in_fft[1] = av_malloc_array(s->n_fft, sizeof(AVComplexFloat));
        s->out_fft[0] = av_malloc_array(s->n_fft, sizeof(AVComplexFloat));
        s->out_fft[1] = av_malloc_array(s->n_fft, sizeof(AVComplexFloat));
        s->temp_afft[0] = av_malloc_array(s->n_fft, sizeof(AVComplexFloat));
        s->temp_afft[1] = av_malloc_array(s->n_fft, sizeof(AVComplexFloat));
        if (!s->in_fft[0] || !s->in_fft[1] ||
            !s->out_fft[0] || !s->out_fft[1] ||
            !s->temp_afft[0] || !s->temp_afft[1]) {
            ret = AVERROR(ENOMEM);
            goto fail;
        }
    }

    if (!s->ringbuffer[0] || !s->ringbuffer[1]) {
        ret = AVERROR(ENOMEM);
        goto fail;
    }

    if (s->type == FREQUENCY_DOMAIN) {
        fft_out_l = av_calloc(n_fft, sizeof(*fft_out_l));
        fft_out_r = av_calloc(n_fft, sizeof(*fft_out_r));
        fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l));
        fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r));
        if (!fft_in_l || !fft_in_r ||
            !fft_out_l || !fft_out_r) {
            ret = AVERROR(ENOMEM);
            goto fail;
        }
    }

    for (i = 0; i < s->n_conv; i++) {
        float *lir, *rir;

        offset = i * n_samples; /* no. samples already written */

        lir = data_ir_l + offset;
        rir = data_ir_r + offset;

        if (s->type == TIME_DOMAIN) {
            for (j = 0; j < ir_samples; j++) {
                /* load reversed IRs of the specified source position
                 * sample-by-sample for left and right ear; and apply gain */
                s->data_ir[0][offset + j] = lir[ir_samples - 1 - j] * gain_lin;
                s->data_ir[1][offset + j] = rir[ir_samples - 1 - j] * gain_lin;
            }
        } else if (s->type == FREQUENCY_DOMAIN) {
            memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l));
            memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r));

            offset = i * n_fft; /* no. samples already written */
            for (j = 0; j < ir_samples; j++) {
                /* load non-reversed IRs of the specified source position
                 * sample-by-sample and apply gain,
                 * L channel is loaded to real part, R channel to imag part,
                 * IRs are shifted by L and R delay */
                fft_in_l[s->delay[0][i] + j].re = lir[j] * gain_lin;
                fft_in_r[s->delay[1][i] + j].re = rir[j] * gain_lin;
            }

            /* actually transform to frequency domain (IRs -> HRTFs) */
            s->tx_fn[0](s->fft[0], fft_out_l, fft_in_l, sizeof(float));
            memcpy(data_hrtf_l + offset, fft_out_l, n_fft * sizeof(*fft_out_l));
            s->tx_fn[1](s->fft[1], fft_out_r, fft_in_r, sizeof(float));
            memcpy(data_hrtf_r + offset, fft_out_r, n_fft * sizeof(*fft_out_r));
        }
    }

    if (s->type == FREQUENCY_DOMAIN) {
        s->data_hrtf[0] = av_malloc_array(n_fft * s->n_conv, sizeof(AVComplexFloat));
        s->data_hrtf[1] = av_malloc_array(n_fft * s->n_conv, sizeof(AVComplexFloat));
        if (!s->data_hrtf[0] || !s->data_hrtf[1]) {
            ret = AVERROR(ENOMEM);
            goto fail;
        }

        memcpy(s->data_hrtf[0], data_hrtf_l, /* copy HRTF data to */
            sizeof(AVComplexFloat) * n_conv * n_fft); /* filter struct */
        memcpy(s->data_hrtf[1], data_hrtf_r,
            sizeof(AVComplexFloat) * n_conv * n_fft);
    }

fail:
    av_freep(&data_hrtf_l); /* free temporary HRTF memory */
    av_freep(&data_hrtf_r);

    av_freep(&data_ir_l); /* free temprary IR memory */
    av_freep(&data_ir_r);

    av_freep(&fft_out_l); /* free temporary FFT memory */
    av_freep(&fft_out_r);

    av_freep(&fft_in_l); /* free temporary FFT memory */
    av_freep(&fft_in_r);

    return ret;
}

static av_cold int init(AVFilterContext *ctx)
{
    SOFAlizerContext *s = ctx->priv;
    int ret;

    if (!s->filename) {
        av_log(ctx, AV_LOG_ERROR, "Valid SOFA filename must be set.\n");
        return AVERROR(EINVAL);
    }

    /* preload SOFA file, */
    ret = preload_sofa(ctx, s->filename, &s->sample_rate);
    if (ret) {
        /* file loading error */
        av_log(ctx, AV_LOG_ERROR, "Error while loading SOFA file: '%s'\n", s->filename);
    } else { /* no file loading error, resampling not required */
        av_log(ctx, AV_LOG_DEBUG, "File '%s' loaded.\n", s->filename);
    }

    if (ret) {
        av_log(ctx, AV_LOG_ERROR, "No valid SOFA file could be loaded. Please specify valid SOFA file.\n");
        return ret;
    }

    s->fdsp = avpriv_float_dsp_alloc(0);
    if (!s->fdsp)
        return AVERROR(ENOMEM);

    return 0;
}

static int config_input(AVFilterLink *inlink)
{
    AVFilterContext *ctx = inlink->dst;
    SOFAlizerContext *s = ctx->priv;
    int ret;

    if (s->type == FREQUENCY_DOMAIN)
        s->nb_samples = s->framesize;

    /* gain -3 dB per channel */
    s->gain_lfe = expf((s->gain - 3 * inlink->ch_layout.nb_channels + s->lfe_gain) / 20 * M_LN10);

    s->n_conv = inlink->ch_layout.nb_channels;

    /* load IRs to data_ir[0] and data_ir[1] for required directions */
    if ((ret = load_data(ctx, s->rotation, s->elevation, s->radius, inlink->sample_rate)) < 0)
        return ret;

    av_log(ctx, AV_LOG_DEBUG, "Samplerate: %d Channels to convolute: %d, Length of ringbuffer: %d x %d\n",
        inlink->sample_rate, s->n_conv, inlink->ch_layout.nb_channels, s->buffer_length);

    return 0;
}

static av_cold void uninit(AVFilterContext *ctx)
{
    SOFAlizerContext *s = ctx->priv;

    close_sofa(&s->sofa);
    av_tx_uninit(&s->ifft[0]);
    av_tx_uninit(&s->ifft[1]);
    av_tx_uninit(&s->fft[0]);
    av_tx_uninit(&s->fft[1]);
    s->ifft[0] = NULL;
    s->ifft[1] = NULL;
    s->fft[0] = NULL;
    s->fft[1] = NULL;
    av_freep(&s->delay[0]);
    av_freep(&s->delay[1]);
    av_freep(&s->data_ir[0]);
    av_freep(&s->data_ir[1]);
    av_freep(&s->ringbuffer[0]);
    av_freep(&s->ringbuffer[1]);
    av_freep(&s->speaker_azim);
    av_freep(&s->speaker_elev);
    av_freep(&s->temp_src[0]);
    av_freep(&s->temp_src[1]);
    av_freep(&s->temp_afft[0]);
    av_freep(&s->temp_afft[1]);
    av_freep(&s->in_fft[0]);
    av_freep(&s->in_fft[1]);
    av_freep(&s->out_fft[0]);
    av_freep(&s->out_fft[1]);
    av_freep(&s->data_hrtf[0]);
    av_freep(&s->data_hrtf[1]);
    av_freep(&s->fdsp);
}

#define OFFSET(x) offsetof(SOFAlizerContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

static const AVOption sofalizer_options[] = {
    { "sofa",      "sofa filename",  OFFSET(filename),  AV_OPT_TYPE_STRING, {.str=NULL},            .flags = FLAGS },
    { "gain",      "set gain in dB", OFFSET(gain),      AV_OPT_TYPE_FLOAT,  {.dbl=0},     -20,  40, .flags = FLAGS },
    { "rotation",  "set rotation"  , OFFSET(rotation),  AV_OPT_TYPE_FLOAT,  {.dbl=0},    -360, 360, .flags = FLAGS },
    { "elevation", "set elevation",  OFFSET(elevation), AV_OPT_TYPE_FLOAT,  {.dbl=0},     -90,  90, .flags = FLAGS },
    { "radius",    "set radius",     OFFSET(radius),    AV_OPT_TYPE_FLOAT,  {.dbl=1},       0,   5, .flags = FLAGS },
    { "type",      "set processing", OFFSET(type),      AV_OPT_TYPE_INT,    {.i64=1},       0,   1, .flags = FLAGS, "type" },
    { "time",      "time domain",      0,               AV_OPT_TYPE_CONST,  {.i64=0},       0,   0, .flags = FLAGS, "type" },
    { "freq",      "frequency domain", 0,               AV_OPT_TYPE_CONST,  {.i64=1},       0,   0, .flags = FLAGS, "type" },
    { "speakers",  "set speaker custom positions", OFFSET(speakers_pos), AV_OPT_TYPE_STRING,  {.str=0},    0, 0, .flags = FLAGS },
    { "lfegain",   "set lfe gain",                 OFFSET(lfe_gain),     AV_OPT_TYPE_FLOAT,   {.dbl=0},  -20,40, .flags = FLAGS },
    { "framesize", "set frame size", OFFSET(framesize), AV_OPT_TYPE_INT,    {.i64=1024},1024,96000, .flags = FLAGS },
    { "normalize", "normalize IRs",  OFFSET(normalize), AV_OPT_TYPE_BOOL,   {.i64=1},       0,   1, .flags = FLAGS },
    { "interpolate","interpolate IRs from neighbors",   OFFSET(interpolate),AV_OPT_TYPE_BOOL,    {.i64=0},       0,   1, .flags = FLAGS },
    { "minphase",  "minphase IRs",   OFFSET(minphase),  AV_OPT_TYPE_BOOL,   {.i64=0},       0,   1, .flags = FLAGS },
    { "anglestep", "set neighbor search angle step",    OFFSET(anglestep),  AV_OPT_TYPE_FLOAT,   {.dbl=.5},      0.01, 10, .flags = FLAGS },
    { "radstep",   "set neighbor search radius step",   OFFSET(radstep),    AV_OPT_TYPE_FLOAT,   {.dbl=.01},     0.01,  1, .flags = FLAGS },
    { NULL }
};

AVFILTER_DEFINE_CLASS(sofalizer);

static const AVFilterPad inputs[] = {
    {
        .name         = "default",
        .type         = AVMEDIA_TYPE_AUDIO,
        .config_props = config_input,
    },
};

static const AVFilterPad outputs[] = {
    {
        .name = "default",
        .type = AVMEDIA_TYPE_AUDIO,
    },
};

const AVFilter ff_af_sofalizer = {
    .name          = "sofalizer",
    .description   = NULL_IF_CONFIG_SMALL("SOFAlizer (Spatially Oriented Format for Acoustics)."),
    .priv_size     = sizeof(SOFAlizerContext),
    .priv_class    = &sofalizer_class,
    .init          = init,
    .activate      = activate,
    .uninit        = uninit,
    FILTER_INPUTS(inputs),
    FILTER_OUTPUTS(outputs),
    FILTER_QUERY_FUNC(query_formats),
    .flags         = AVFILTER_FLAG_SLICE_THREADS,
};