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FFmpeg/libavfilter/af_acrossover.c

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  1. /*

  2.  * This file is part of FFmpeg.

  3.  *

  4.  * FFmpeg is free software; you can redistribute it and/or

  5.  * modify it under the terms of the GNU Lesser General Public

  6.  * License as published by the Free Software Foundation; either

  7.  * version 2.1 of the License, or (at your option) any later version.

  8.  *

  9.  * FFmpeg is distributed in the hope that it will be useful,

  10.  * but WITHOUT ANY WARRANTY; without even the implied warranty of

  11.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

  12.  * Lesser General Public License for more details.

  13.  *

  14.  * You should have received a copy of the GNU Lesser General Public

  15.  * License along with FFmpeg; if not, write to the Free Software

  16.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

  17.  */

  18. 

  19. /**

  20.  * @file

  21.  * Crossover filter

  22.  *

  23.  * Split an audio stream into several bands.

  24.  */

  25. 

  26. #include "libavutil/attributes.h"

  27. #include "libavutil/avstring.h"

  28. #include "libavutil/channel_layout.h"

  29. #include "libavutil/eval.h"

  30. #include "libavutil/float_dsp.h"

  31. #include "libavutil/internal.h"

  32. #include "libavutil/opt.h"

  33. 

  34. #include "audio.h"

  35. #include "avfilter.h"

  36. #include "formats.h"

  37. #include "internal.h"

  38. 

  39. #define MAX_SPLITS 16

  40. #define MAX_BANDS MAX_SPLITS + 1

  41. 

  42. #define B0 0

  43. #define B1 1

  44. #define B2 2

  45. #define A1 3

  46. #define A2 4

  47. 

  48. typedef struct BiquadCoeffs {

  49.     double cd[5];

  50.     float cf[5];

  51. } BiquadCoeffs;

  52. 

  53. typedef struct AudioCrossoverContext {

  54.     const AVClass *class;

  55. 

  56.     char *splits_str;

  57.     char *gains_str;

  58.     int order_opt;

  59.     float level_in;

  60. 

  61.     int order;

  62.     int filter_count;

  63.     int first_order;

  64.     int ap_filter_count;

  65.     int nb_splits;

  66.     float splits[MAX_SPLITS];

  67. 

  68.     float gains[MAX_BANDS];

  69. 

  70.     BiquadCoeffs lp[MAX_BANDS][20];

  71.     BiquadCoeffs hp[MAX_BANDS][20];

  72.     BiquadCoeffs ap[MAX_BANDS][20];

  73. 

  74.     AVFrame *xover;

  75. 

  76.     AVFrame *input_frame;

  77.     AVFrame *frames[MAX_BANDS];

  78. 

  79.     int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);

  80. 

  81.     AVFloatDSPContext *fdsp;

  82. } AudioCrossoverContext;

  83. 

  84. #define OFFSET(x) offsetof(AudioCrossoverContext, x)

  85. #define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM

  86. 

  87. static const AVOption acrossover_options[] = {

  88.     { "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },

  89.     { "order", "set filter order",      OFFSET(order_opt),  AV_OPT_TYPE_INT,    {.i64=1},     0, 9, AF, "m" },

  90.     { "2nd",   "2nd order (12 dB/8ve)", 0,                  AV_OPT_TYPE_CONST,  {.i64=0},     0, 0, AF, "m" },

  91.     { "4th",   "4th order (24 dB/8ve)", 0,                  AV_OPT_TYPE_CONST,  {.i64=1},     0, 0, AF, "m" },

  92.     { "6th",   "6th order (36 dB/8ve)", 0,                  AV_OPT_TYPE_CONST,  {.i64=2},     0, 0, AF, "m" },

  93.     { "8th",   "8th order (48 dB/8ve)", 0,                  AV_OPT_TYPE_CONST,  {.i64=3},     0, 0, AF, "m" },

  94.     { "10th",  "10th order (60 dB/8ve)",0,                  AV_OPT_TYPE_CONST,  {.i64=4},     0, 0, AF, "m" },

  95.     { "12th",  "12th order (72 dB/8ve)",0,                  AV_OPT_TYPE_CONST,  {.i64=5},     0, 0, AF, "m" },

  96.     { "14th",  "14th order (84 dB/8ve)",0,                  AV_OPT_TYPE_CONST,  {.i64=6},     0, 0, AF, "m" },

  97.     { "16th",  "16th order (96 dB/8ve)",0,                  AV_OPT_TYPE_CONST,  {.i64=7},     0, 0, AF, "m" },

  98.     { "18th",  "18th order (108 dB/8ve)",0,                 AV_OPT_TYPE_CONST,  {.i64=8},     0, 0, AF, "m" },

  99.     { "20th",  "20th order (120 dB/8ve)",0,                 AV_OPT_TYPE_CONST,  {.i64=9},     0, 0, AF, "m" },

  100.     { "level", "set input gain",        OFFSET(level_in),   AV_OPT_TYPE_FLOAT,  {.dbl=1},     0, 1, AF },

  101.     { "gain",  "set output bands gain", OFFSET(gains_str),  AV_OPT_TYPE_STRING, {.str="1.f"}, 0, 0, AF },

  102.     { NULL }

  103. };

  104. 

  105. AVFILTER_DEFINE_CLASS(acrossover);

  106. 

  107. static int parse_gains(AVFilterContext *ctx)

  108. {

  109.     AudioCrossoverContext *s = ctx->priv;

  110.     char *p, *arg, *saveptr = NULL;

  111.     int i, ret = 0;

  112. 

  113.     saveptr = NULL;

  114.     p = s->gains_str;

  115.     for (i = 0; i < MAX_BANDS; i++) {

  116.         float gain;

  117.         char c[3] = { 0 };

  118. 

  119.         if (!(arg = av_strtok(p, " |", &saveptr)))

  120.             break;

  121. 

  122.         p = NULL;

  123. 

  124.         if (av_sscanf(arg, "%f%2s", &gain, c) < 1) {

  125.             av_log(ctx, AV_LOG_ERROR, "Invalid syntax for gain[%d].\n", i);

  126.             ret = AVERROR(EINVAL);

  127.             break;

  128.         }

  129. 

  130.         if (c[0] == 'd' && c[1] == 'B')

  131.             s->gains[i] = expf(gain * M_LN10 / 20.f);

  132.         else

  133.             s->gains[i] = gain;

  134.     }

  135. 

  136.     for (; i < MAX_BANDS; i++)

  137.         s->gains[i] = 1.f;

  138. 

  139.     return ret;

  140. }

  141. 

  142. static av_cold int init(AVFilterContext *ctx)

  143. {

  144.     AudioCrossoverContext *s = ctx->priv;

  145.     char *p, *arg, *saveptr = NULL;

  146.     int i, ret = 0;

  147. 

  148.     s->fdsp = avpriv_float_dsp_alloc(0);

  149.     if (!s->fdsp)

  150.         return AVERROR(ENOMEM);

  151. 

  152.     p = s->splits_str;

  153.     for (i = 0; i < MAX_SPLITS; i++) {

  154.         float freq;

  155. 

  156.         if (!(arg = av_strtok(p, " |", &saveptr)))

  157.             break;

  158. 

  159.         p = NULL;

  160. 

  161.         if (av_sscanf(arg, "%f", &freq) != 1) {

  162.             av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i);

  163.             return AVERROR(EINVAL);

  164.         }

  165.         if (freq <= 0) {

  166.             av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);

  167.             return AVERROR(EINVAL);

  168.         }

  169. 

  170.         if (i > 0 && freq <= s->splits[i-1]) {

  171.             av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);

  172.             return AVERROR(EINVAL);

  173.         }

  174. 

  175.         s->splits[i] = freq;

  176.     }

  177. 

  178.     s->nb_splits = i;

  179. 

  180.     ret = parse_gains(ctx);

  181.     if (ret < 0)

  182.         return ret;

  183. 

  184.     for (i = 0; i <= s->nb_splits; i++) {

  185.         AVFilterPad pad  = { 0 };

  186.         char *name;

  187. 

  188.         pad.type = AVMEDIA_TYPE_AUDIO;

  189.         name = av_asprintf("out%d", ctx->nb_outputs);

  190.         if (!name)

  191.             return AVERROR(ENOMEM);

  192.         pad.name = name;

  193. 

  194.         if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) {

  195.             av_freep(&pad.name);

  196.             return ret;

  197.         }

  198.     }

  199. 

  200.     return ret;

  201. }

  202. 

  203. static void set_lp(BiquadCoeffs *b, double fc, double q, double sr)

  204. {

  205.     double omega = 2. * M_PI * fc / sr;

  206.     double cosine = cos(omega);

  207.     double alpha = sin(omega) / (2. * q);

  208. 

  209.     double b0 = (1. - cosine) / 2.;

  210.     double b1 = 1. - cosine;

  211.     double b2 = (1. - cosine) / 2.;

  212.     double a0 = 1. + alpha;

  213.     double a1 = -2. * cosine;

  214.     double a2 = 1. - alpha;

  215. 

  216.     b->cd[B0] =  b0 / a0;

  217.     b->cd[B1] =  b1 / a0;

  218.     b->cd[B2] =  b2 / a0;

  219.     b->cd[A1] = -a1 / a0;

  220.     b->cd[A2] = -a2 / a0;

  221. 

  222.     b->cf[B0] = b->cd[B0];

  223.     b->cf[B1] = b->cd[B1];

  224.     b->cf[B2] = b->cd[B2];

  225.     b->cf[A1] = b->cd[A1];

  226.     b->cf[A2] = b->cd[A2];

  227. }

  228. 

  229. static void set_hp(BiquadCoeffs *b, double fc, double q, double sr)

  230. {

  231.     double omega = 2. * M_PI * fc / sr;

  232.     double cosine = cos(omega);

  233.     double alpha = sin(omega) / (2. * q);

  234. 

  235.     double b0 = (1. + cosine) / 2.;

  236.     double b1 = -1. - cosine;

  237.     double b2 = (1. + cosine) / 2.;

  238.     double a0 = 1. + alpha;

  239.     double a1 = -2. * cosine;

  240.     double a2 = 1. - alpha;

  241. 

  242.     b->cd[B0] =  b0 / a0;

  243.     b->cd[B1] =  b1 / a0;

  244.     b->cd[B2] =  b2 / a0;

  245.     b->cd[A1] = -a1 / a0;

  246.     b->cd[A2] = -a2 / a0;

  247. 

  248.     b->cf[B0] = b->cd[B0];

  249.     b->cf[B1] = b->cd[B1];

  250.     b->cf[B2] = b->cd[B2];

  251.     b->cf[A1] = b->cd[A1];

  252.     b->cf[A2] = b->cd[A2];

  253. }

  254. 

  255. static void set_ap(BiquadCoeffs *b, double fc, double q, double sr)

  256. {

  257.     double omega = 2. * M_PI * fc / sr;

  258.     double cosine = cos(omega);

  259.     double alpha = sin(omega) / (2. * q);

  260. 

  261.     double a0 = 1. + alpha;

  262.     double a1 = -2. * cosine;

  263.     double a2 = 1. - alpha;

  264.     double b0 = a2;

  265.     double b1 = a1;

  266.     double b2 = a0;

  267. 

  268.     b->cd[B0] =  b0 / a0;

  269.     b->cd[B1] =  b1 / a0;

  270.     b->cd[B2] =  b2 / a0;

  271.     b->cd[A1] = -a1 / a0;

  272.     b->cd[A2] = -a2 / a0;

  273. 

  274.     b->cf[B0] = b->cd[B0];

  275.     b->cf[B1] = b->cd[B1];

  276.     b->cf[B2] = b->cd[B2];

  277.     b->cf[A1] = b->cd[A1];

  278.     b->cf[A2] = b->cd[A2];

  279. }

  280. 

  281. static void set_ap1(BiquadCoeffs *b, double fc, double sr)

  282. {

  283.     double omega = 2. * M_PI * fc / sr;

  284. 

  285.     b->cd[A1] = exp(-omega);

  286.     b->cd[A2] = 0.;

  287.     b->cd[B0] = -b->cd[A1];

  288.     b->cd[B1] = 1.;

  289.     b->cd[B2] = 0.;

  290. 

  291.     b->cf[B0] = b->cd[B0];

  292.     b->cf[B1] = b->cd[B1];

  293.     b->cf[B2] = b->cd[B2];

  294.     b->cf[A1] = b->cd[A1];

  295.     b->cf[A2] = b->cd[A2];

  296. }

  297. 

  298. static void calc_q_factors(int order, double *q)

  299. {

  300.     double n = order / 2.;

  301. 

  302.     for (int i = 0; i < n / 2; i++)

  303.         q[i] = 1. / (-2. * cos(M_PI * (2. * (i + 1) + n - 1.) / (2. * n)));

  304. }

  305. 

  306. static int query_formats(AVFilterContext *ctx)

  307. {

  308.     AVFilterFormats *formats;

  309.     AVFilterChannelLayouts *layouts;

  310.     static const enum AVSampleFormat sample_fmts[] = {

  311.         AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,

  312.         AV_SAMPLE_FMT_NONE

  313.     };

  314.     int ret;

  315. 

  316.     layouts = ff_all_channel_counts();

  317.     if (!layouts)

  318.         return AVERROR(ENOMEM);

  319.     ret = ff_set_common_channel_layouts(ctx, layouts);

  320.     if (ret < 0)

  321.         return ret;

  322. 

  323.     formats = ff_make_format_list(sample_fmts);

  324.     if (!formats)

  325.         return AVERROR(ENOMEM);

  326.     ret = ff_set_common_formats(ctx, formats);

  327.     if (ret < 0)

  328.         return ret;

  329. 

  330.     formats = ff_all_samplerates();

  331.     if (!formats)

  332.         return AVERROR(ENOMEM);

  333.     return ff_set_common_samplerates(ctx, formats);

  334. }

  335. 

  336. #define BIQUAD_PROCESS(name, type)                             \

  337. static void biquad_process_## name(const type *const c,        \

  338.                                    type *b,                    \

  339.                                    type *dst, const type *src, \

  340.                                    int nb_samples)             \

  341. {                                                              \

  342.     const type b0 = c[B0];                                     \

  343.     const type b1 = c[B1];                                     \

  344.     const type b2 = c[B2];                                     \

  345.     const type a1 = c[A1];                                     \

  346.     const type a2 = c[A2];                                     \

  347.     type z1 = b[0];                                            \

  348.     type z2 = b[1];                                            \

  349.                                                                \

  350.     for (int n = 0; n + 1 < nb_samples; n++) {                 \

  351.         type in = src[n];                                      \

  352.         type out;                                              \

  353.                                                                \

  354.         out = in * b0 + z1;                                    \

  355.         z1 = b1 * in + z2 + a1 * out;                          \

  356.         z2 = b2 * in + a2 * out;                               \

  357.         dst[n] = out;                                          \

  358.                                                                \

  359.         n++;                                                   \

  360.         in = src[n];                                           \

  361.         out = in * b0 + z1;                                    \

  362.         z1 = b1 * in + z2 + a1 * out;                          \

  363.         z2 = b2 * in + a2 * out;                               \

  364.         dst[n] = out;                                          \

  365.     }                                                          \

  366.                                                                \

  367.     if (nb_samples & 1) {                                      \

  368.         const int n = nb_samples - 1;                          \

  369.         const type in = src[n];                                \

  370.         type out;                                              \

  371.                                                                \

  372.         out = in * b0 + z1;                                    \

  373.         z1 = b1 * in + z2 + a1 * out;                          \

  374.         z2 = b2 * in + a2 * out;                               \

  375.         dst[n] = out;                                          \

  376.     }                                                          \

  377.                                                                \

  378.     b[0] = z1;                                                 \

  379.     b[1] = z2;                                                 \

  380. }

  381. 

  382. BIQUAD_PROCESS(fltp, float)

  383. BIQUAD_PROCESS(dblp, double)

  384. 

  385. #define XOVER_PROCESS(name, type, one, ff)                                                  \

  386. static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) \

  387. {                                                                                           \

  388.     AudioCrossoverContext *s = ctx->priv;                                                   \

  389.     AVFrame *in = s->input_frame;                                                           \

  390.     AVFrame **frames = s->frames;                                                           \

  391.     const int start = (in->channels * jobnr) / nb_jobs;                                     \

  392.     const int end = (in->channels * (jobnr+1)) / nb_jobs;                                   \

  393.     const int nb_samples = in->nb_samples;                                                  \

  394.     const int nb_outs = ctx->nb_outputs;                                                    \

  395.     const int first_order = s->first_order;                                                 \

  396.                                                                                             \

  397.     for (int ch = start; ch < end; ch++) {                                                  \

  398.         const type *src = (const type *)in->extended_data[ch];                              \

  399.         type *xover = (type *)s->xover->extended_data[ch];                                  \

  400.                                                                                             \

  401.         s->fdsp->vector_## ff ##mul_scalar((type *)frames[0]->extended_data[ch], src,       \

  402.                                     s->level_in, FFALIGN(nb_samples, sizeof(type)));        \

  403.                                                                                             \

  404.         for (int band = 0; band < nb_outs; band++) {                                        \

  405.             for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) {               \

  406.                 const type *prv = (const type *)frames[band]->extended_data[ch];            \

  407.                 type *dst = (type *)frames[band + 1]->extended_data[ch];                    \

  408.                 const type *hsrc = f == 0 ? prv : dst;                                      \

  409.                 type *hp = xover + nb_outs * 20 + band * 20 + f * 2;                        \

  410.                 const type *const hpc = (type *)&s->hp[band][f].c ## ff;                    \

  411.                                                                                             \

  412.                 biquad_process_## name(hpc, hp, dst, hsrc, nb_samples);                     \

  413.             }                                                                               \

  414.                                                                                             \

  415.             for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) {               \

  416.                 type *dst = (type *)frames[band]->extended_data[ch];                        \

  417.                 const type *lsrc = dst;                                                     \

  418.                 type *lp = xover + band * 20 + f * 2;                                       \

  419.                 const type *const lpc = (type *)&s->lp[band][f].c ## ff;                    \

  420.                                                                                             \

  421.                 biquad_process_## name(lpc, lp, dst, lsrc, nb_samples);                     \

  422.             }                                                                               \

  423.                                                                                             \

  424.             for (int aband = band + 1; aband + 1 < nb_outs; aband++) {                      \

  425.                 if (first_order) {                                                          \

  426.                     const type *asrc = (const type *)frames[band]->extended_data[ch];       \

  427.                     type *dst = (type *)frames[band]->extended_data[ch];                    \

  428.                     type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20;        \

  429.                     const type *const apc = (type *)&s->ap[aband][0].c ## ff;               \

  430.                                                                                             \

  431.                     biquad_process_## name(apc, ap, dst, asrc, nb_samples);                 \

  432.                 }                                                                           \

  433.                                                                                             \

  434.                 for (int f = first_order; f < s->ap_filter_count; f++) {                    \

  435.                     const type *asrc = (const type *)frames[band]->extended_data[ch];       \

  436.                     type *dst = (type *)frames[band]->extended_data[ch];                    \

  437.                     type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20 + f * 2;\

  438.                     const type *const apc = (type *)&s->ap[aband][f].c ## ff;               \

  439.                                                                                             \

  440.                     biquad_process_## name(apc, ap, dst, asrc, nb_samples);                 \

  441.                 }                                                                           \

  442.             }                                                                               \

  443.         }                                                                                   \

  444.                                                                                             \

  445.         for (int band = 0; band < nb_outs; band++) {                                        \

  446.             const type gain = s->gains[band] * ((band & 1 && first_order) ? -one : one);    \

  447.             type *dst = (type *)frames[band]->extended_data[ch];                            \

  448.                                                                                             \

  449.             s->fdsp->vector_## ff ##mul_scalar(dst, dst, gain,                              \

  450.                                                FFALIGN(nb_samples, sizeof(type)));          \

  451.         }                                                                                   \

  452.     }                                                                                       \

  453.                                                                                             \

  454.     return 0;                                                                               \

  455. }

  456. 

  457. XOVER_PROCESS(fltp, float, 1.f, f)

  458. XOVER_PROCESS(dblp, double, 1.0, d)

  459. 

  460. static int config_input(AVFilterLink *inlink)

  461. {

  462.     AVFilterContext *ctx = inlink->dst;

  463.     AudioCrossoverContext *s = ctx->priv;

  464.     int sample_rate = inlink->sample_rate;

  465.     double q[16];

  466. 

  467.     s->order = (s->order_opt + 1) * 2;

  468.     s->filter_count = s->order / 2;

  469.     s->first_order = s->filter_count & 1;

  470.     s->ap_filter_count = s->filter_count / 2 + s->first_order;

  471.     calc_q_factors(s->order, q);

  472. 

  473.     for (int band = 0; band <= s->nb_splits; band++) {

  474.         if (s->first_order) {

  475.             set_lp(&s->lp[band][0], s->splits[band], 0.5, sample_rate);

  476.             set_hp(&s->hp[band][0], s->splits[band], 0.5, sample_rate);

  477.         }

  478. 

  479.         for (int n = s->first_order; n < s->filter_count; n++) {

  480.             const int idx = s->filter_count / 2 - ((n + s->first_order) / 2 - s->first_order) - 1;

  481. 

  482.             set_lp(&s->lp[band][n], s->splits[band], q[idx], sample_rate);

  483.             set_hp(&s->hp[band][n], s->splits[band], q[idx], sample_rate);

  484.         }

  485. 

  486.         if (s->first_order)

  487.             set_ap1(&s->ap[band][0], s->splits[band], sample_rate);

  488. 

  489.         for (int n = s->first_order; n < s->ap_filter_count; n++) {

  490.             const int idx = (s->filter_count / 2 - ((n * 2 + s->first_order) / 2 - s->first_order) - 1);

  491. 

  492.             set_ap(&s->ap[band][n], s->splits[band], q[idx], sample_rate);

  493.         }

  494.     }

  495. 

  496.     switch (inlink->format) {

  497.     case AV_SAMPLE_FMT_FLTP: s->filter_channels = filter_channels_fltp; break;

  498.     case AV_SAMPLE_FMT_DBLP: s->filter_channels = filter_channels_dblp; break;

  499.     }

  500. 

  501.     s->xover = ff_get_audio_buffer(inlink, 2 * (ctx->nb_outputs * 10 + ctx->nb_outputs * 10 +

  502.                                                 ctx->nb_outputs * ctx->nb_outputs * 10));

  503.     if (!s->xover)

  504.         return AVERROR(ENOMEM);

  505. 

  506.     return 0;

  507. }

  508. 

  509. static int filter_frame(AVFilterLink *inlink, AVFrame *in)

  510. {

  511.     AVFilterContext *ctx = inlink->dst;

  512.     AudioCrossoverContext *s = ctx->priv;

  513.     AVFrame **frames = s->frames;

  514.     int i, ret = 0;

  515. 

  516.     for (i = 0; i < ctx->nb_outputs; i++) {

  517.         frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);

  518. 

  519.         if (!frames[i]) {

  520.             ret = AVERROR(ENOMEM);

  521.             break;

  522.         }

  523. 

  524.         frames[i]->pts = in->pts;

  525.     }

  526. 

  527.     if (ret < 0)

  528.         goto fail;

  529. 

  530.     s->input_frame = in;

  531.     ctx->internal->execute(ctx, s->filter_channels, NULL, NULL, FFMIN(inlink->channels,

  532.                                                                       ff_filter_get_nb_threads(ctx)));

  533. 

  534.     for (i = 0; i < ctx->nb_outputs; i++) {

  535.         ret = ff_filter_frame(ctx->outputs[i], frames[i]);

  536.         frames[i] = NULL;

  537.         if (ret < 0)

  538.             break;

  539.     }

  540. 

  541. fail:

  542.     for (i = 0; i < ctx->nb_outputs; i++)

  543.         av_frame_free(&frames[i]);

  544.     av_frame_free(&in);

  545.     s->input_frame = NULL;

  546. 

  547.     return ret;

  548. }

  549. 

  550. static av_cold void uninit(AVFilterContext *ctx)

  551. {

  552.     AudioCrossoverContext *s = ctx->priv;

  553.     int i;

  554. 

  555.     av_freep(&s->fdsp);

  556.     av_frame_free(&s->xover);

  557. 

  558.     for (i = 0; i < ctx->nb_outputs; i++)

  559.         av_freep(&ctx->output_pads[i].name);

  560. }

  561. 

  562. static const AVFilterPad inputs[] = {

  563.     {

  564.         .name         = "default",

  565.         .type         = AVMEDIA_TYPE_AUDIO,

  566.         .filter_frame = filter_frame,

  567.         .config_props = config_input,

  568.     },

  569.     { NULL }

  570. };

  571. 

  572. AVFilter ff_af_acrossover = {

  573.     .name           = "acrossover",

  574.     .description    = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),

  575.     .priv_size      = sizeof(AudioCrossoverContext),

  576.     .priv_class     = &acrossover_class,

  577.     .init           = init,

  578.     .uninit         = uninit,

  579.     .query_formats  = query_formats,

  580.     .inputs         = inputs,

  581.     .outputs        = NULL,

  582.     .flags          = AVFILTER_FLAG_DYNAMIC_OUTPUTS |

  583.                       AVFILTER_FLAG_SLICE_THREADS,

  584. };