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FFmpeg/libavfilter/af_anequalizer.c

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  1. /*

  2.  * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others

  3.  * Copyright (c) 2015 Paul B Mahol

  4.  *

  5.  * This file is part of FFmpeg.

  6.  *

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

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

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

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

  11.  *

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

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

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

  15.  * Lesser General Public License for more details.

  16.  *

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

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

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

  20.  */

  21. 

  22. #include <complex.h>

  23. 

  24. #include "libavutil/intreadwrite.h"

  25. #include "libavutil/avstring.h"

  26. #include "libavutil/opt.h"

  27. #include "libavutil/parseutils.h"

  28. #include "avfilter.h"

  29. #include "internal.h"

  30. #include "audio.h"

  31. 

  32. #define FILTER_ORDER 4

  33. 

  34. enum FilterType {

  35.     BUTTERWORTH,

  36.     CHEBYSHEV1,

  37.     CHEBYSHEV2,

  38.     NB_TYPES

  39. };

  40. 

  41. typedef struct FoSection {

  42.     double a0, a1, a2, a3, a4;

  43.     double b0, b1, b2, b3, b4;

  44. 

  45.     double num[4];

  46.     double denum[4];

  47. } FoSection;

  48. 

  49. typedef struct EqualizatorFilter {

  50.     int ignore;

  51.     int channel;

  52.     int type;

  53. 

  54.     double freq;

  55.     double gain;

  56.     double width;

  57. 

  58.     FoSection section[2];

  59. } EqualizatorFilter;

  60. 

  61. typedef struct AudioNEqualizerContext {

  62.     const AVClass *class;

  63.     char *args;

  64.     char *colors;

  65.     int draw_curves;

  66.     int w, h;

  67. 

  68.     double mag;

  69.     int fscale;

  70.     int nb_filters;

  71.     int nb_allocated;

  72.     EqualizatorFilter *filters;

  73.     AVFrame *video;

  74. } AudioNEqualizerContext;

  75. 

  76. #define OFFSET(x) offsetof(AudioNEqualizerContext, x)

  77. #define A AV_OPT_FLAG_AUDIO_PARAM

  78. #define V AV_OPT_FLAG_VIDEO_PARAM

  79. #define F AV_OPT_FLAG_FILTERING_PARAM

  80. 

  81. static const AVOption anequalizer_options[] = {

  82.     { "params", NULL,                             OFFSET(args),        AV_OPT_TYPE_STRING,     {.str=""}, 0, 0, A|F },

  83.     { "curves", "draw frequency response curves", OFFSET(draw_curves), AV_OPT_TYPE_BOOL,       {.i64=0}, 0, 1, V|F },

  84.     { "size",   "set video size",                 OFFSET(w),           AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, V|F },

  85.     { "mgain",  "set max gain",                   OFFSET(mag),         AV_OPT_TYPE_DOUBLE,     {.dbl=60}, -900, 900, V|F },

  86.     { "fscale", "set frequency scale",            OFFSET(fscale),      AV_OPT_TYPE_INT,        {.i64=1}, 0, 1, V|F, "fscale" },

  87.         { "lin",  "linear",                       0,                   AV_OPT_TYPE_CONST,      {.i64=0}, 0, 0, V|F, "fscale" },

  88.         { "log",  "logarithmic",                  0,                   AV_OPT_TYPE_CONST,      {.i64=1}, 0, 0, V|F, "fscale" },

  89.     { "colors", "set channels curves colors",     OFFSET(colors),      AV_OPT_TYPE_STRING,     {.str = "red|green|blue|yellow|orange|lime|pink|magenta|brown" }, 0, 0, V|F },

  90.     { NULL }

  91. };

  92. 

  93. AVFILTER_DEFINE_CLASS(anequalizer);

  94. 

  95. static void draw_curves(AVFilterContext *ctx, AVFilterLink *inlink, AVFrame *out)

  96. {

  97.     AudioNEqualizerContext *s = ctx->priv;

  98.     char *colors, *color, *saveptr = NULL;

  99.     int ch, i, n;

  100. 

  101.     colors = av_strdup(s->colors);

  102.     if (!colors)

  103.         return;

  104. 

  105.     memset(out->data[0], 0, s->h * out->linesize[0]);

  106. 

  107.     for (ch = 0; ch < inlink->channels; ch++) {

  108.         uint8_t fg[4] = { 0xff, 0xff, 0xff, 0xff };

  109.         int prev_v = -1;

  110.         double f;

  111. 

  112.         color = av_strtok(ch == 0 ? colors : NULL, " |", &saveptr);

  113.         if (color)

  114.             av_parse_color(fg, color, -1, ctx);

  115. 

  116.         for (f = 0; f < s->w; f++) {

  117.             double complex z;

  118.             double complex H = 1;

  119.             double w;

  120.             int v, y, x;

  121. 

  122.             w = M_PI * (s->fscale ? pow(s->w - 1, f / s->w) : f) / (s->w - 1);

  123.             z = 1. / cexp(I * w);

  124. 

  125.             for (n = 0; n < s->nb_filters; n++) {

  126.                 if (s->filters[n].channel != ch ||

  127.                     s->filters[n].ignore)

  128.                     continue;

  129. 

  130.                 for (i = 0; i < FILTER_ORDER / 2; i++) {

  131.                     FoSection *S = &s->filters[n].section[i];

  132. 

  133.                     H *= (((((S->b4 * z + S->b3) * z + S->b2) * z + S->b1) * z + S->b0) /

  134.                           ((((S->a4 * z + S->a3) * z + S->a2) * z + S->a1) * z + S->a0));

  135.                 }

  136.             }

  137. 

  138.             v = av_clip((1. + -20 * log10(cabs(H)) / s->mag) * s->h / 2, 0, s->h - 1);

  139.             x = lrint(f);

  140.             if (prev_v == -1)

  141.                 prev_v = v;

  142.             if (v <= prev_v) {

  143.                 for (y = v; y <= prev_v; y++)

  144.                     AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg));

  145.             } else {

  146.                 for (y = prev_v; y <= v; y++)

  147.                     AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg));

  148.             }

  149. 

  150.             prev_v = v;

  151.         }

  152.     }

  153. 

  154.     av_free(colors);

  155. }

  156. 

  157. static int config_video(AVFilterLink *outlink)

  158. {

  159.     AVFilterContext *ctx = outlink->src;

  160.     AudioNEqualizerContext *s = ctx->priv;

  161.     AVFilterLink *inlink = ctx->inputs[0];

  162.     AVFrame *out;

  163. 

  164.     outlink->w = s->w;

  165.     outlink->h = s->h;

  166. 

  167.     av_frame_free(&s->video);

  168.     s->video = out = ff_get_video_buffer(outlink, outlink->w, outlink->h);

  169.     if (!out)

  170.         return AVERROR(ENOMEM);

  171.     outlink->sample_aspect_ratio = (AVRational){1,1};

  172. 

  173.     draw_curves(ctx, inlink, out);

  174. 

  175.     return 0;

  176. }

  177. 

  178. static av_cold int init(AVFilterContext *ctx)

  179. {

  180.     AudioNEqualizerContext *s = ctx->priv;

  181.     AVFilterPad pad, vpad;

  182. 

  183.     pad = (AVFilterPad){

  184.         .name         = av_strdup("out0"),

  185.         .type         = AVMEDIA_TYPE_AUDIO,

  186.     };

  187. 

  188.     if (!pad.name)

  189.         return AVERROR(ENOMEM);

  190. 

  191.     if (s->draw_curves) {

  192.         vpad = (AVFilterPad){

  193.             .name         = av_strdup("out1"),

  194.             .type         = AVMEDIA_TYPE_VIDEO,

  195.             .config_props = config_video,

  196.         };

  197.         if (!vpad.name)

  198.             return AVERROR(ENOMEM);

  199.     }

  200. 

  201.     ff_insert_outpad(ctx, 0, &pad);

  202. 

  203.     if (s->draw_curves)

  204.         ff_insert_outpad(ctx, 1, &vpad);

  205. 

  206.     return 0;

  207. }

  208. 

  209. static int query_formats(AVFilterContext *ctx)

  210. {

  211.     AVFilterLink *inlink = ctx->inputs[0];

  212.     AVFilterLink *outlink = ctx->outputs[0];

  213.     AudioNEqualizerContext *s = ctx->priv;

  214.     AVFilterFormats *formats;

  215.     AVFilterChannelLayouts *layouts;

  216.     static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_RGBA, AV_PIX_FMT_NONE };

  217.     static const enum AVSampleFormat sample_fmts[] = {

  218.         AV_SAMPLE_FMT_DBLP,

  219.         AV_SAMPLE_FMT_NONE

  220.     };

  221.     int ret;

  222. 

  223.     if (s->draw_curves) {

  224.         AVFilterLink *videolink = ctx->outputs[1];

  225.         formats = ff_make_format_list(pix_fmts);

  226.         if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)

  227.             return ret;

  228.     }

  229. 

  230.     formats = ff_make_format_list(sample_fmts);

  231.     if ((ret = ff_formats_ref(formats, &inlink->out_formats)) < 0 ||

  232.         (ret = ff_formats_ref(formats, &outlink->in_formats)) < 0)

  233.         return ret;

  234. 

  235.     layouts = ff_all_channel_counts();

  236.     if ((ret = ff_channel_layouts_ref(layouts, &inlink->out_channel_layouts)) < 0 ||

  237.         (ret = ff_channel_layouts_ref(layouts, &outlink->in_channel_layouts)) < 0)

  238.         return ret;

  239. 

  240.     formats = ff_all_samplerates();

  241.     if ((ret = ff_formats_ref(formats, &inlink->out_samplerates)) < 0 ||

  242.         (ret = ff_formats_ref(formats, &outlink->in_samplerates)) < 0)

  243.         return ret;

  244. 

  245.     return 0;

  246. }

  247. 

  248. static av_cold void uninit(AVFilterContext *ctx)

  249. {

  250.     AudioNEqualizerContext *s = ctx->priv;

  251. 

  252.     av_freep(&ctx->output_pads[0].name);

  253.     if (s->draw_curves)

  254.         av_freep(&ctx->output_pads[1].name);

  255.     av_frame_free(&s->video);

  256.     av_freep(&s->filters);

  257.     s->nb_filters = 0;

  258.     s->nb_allocated = 0;

  259. }

  260. 

  261. static void butterworth_fo_section(FoSection *S, double beta,

  262.                                    double si, double g, double g0,

  263.                                    double D, double c0)

  264. {

  265.     if (c0 == 1 || c0 == -1) {

  266.         S->b0 = (g*g*beta*beta + 2*g*g0*si*beta + g0*g0)/D;

  267.         S->b1 = 2*c0*(g*g*beta*beta - g0*g0)/D;

  268.         S->b2 = (g*g*beta*beta - 2*g0*g*beta*si + g0*g0)/D;

  269.         S->b3 = 0;

  270.         S->b4 = 0;

  271. 

  272.         S->a0 = 1;

  273.         S->a1 = 2*c0*(beta*beta - 1)/D;

  274.         S->a2 = (beta*beta - 2*beta*si + 1)/D;

  275.         S->a3 = 0;

  276.         S->a4 = 0;

  277.     } else {

  278.         S->b0 = (g*g*beta*beta + 2*g*g0*si*beta + g0*g0)/D;

  279.         S->b1 = -4*c0*(g0*g0 + g*g0*si*beta)/D;

  280.         S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - g*g*beta*beta)/D;

  281.         S->b3 = -4*c0*(g0*g0 - g*g0*si*beta)/D;

  282.         S->b4 = (g*g*beta*beta - 2*g*g0*si*beta + g0*g0)/D;

  283. 

  284.         S->a0 = 1;

  285.         S->a1 = -4*c0*(1 + si*beta)/D;

  286.         S->a2 = 2*(1 + 2*c0*c0 - beta*beta)/D;

  287.         S->a3 = -4*c0*(1 - si*beta)/D;

  288.         S->a4 = (beta*beta - 2*si*beta + 1)/D;

  289.     }

  290. }

  291. 

  292. static void butterworth_bp_filter(EqualizatorFilter *f,

  293.                                   int N, double w0, double wb,

  294.                                   double G, double Gb, double G0)

  295. {

  296.     double g, c0, g0, beta;

  297.     double epsilon;

  298.     int r =  N % 2;

  299.     int L = (N - r) / 2;

  300.     int i;

  301. 

  302.     if (G == 0 && G0 == 0) {

  303.         f->section[0].a0 = 1;

  304.         f->section[0].b0 = 1;

  305.         f->section[1].a0 = 1;

  306.         f->section[1].b0 = 1;

  307.         return;

  308.     }

  309. 

  310.     G  = exp10( G/20);

  311.     Gb = exp10(Gb/20);

  312.     G0 = exp10(G0/20);

  313. 

  314.     epsilon = sqrt((G * G - Gb * Gb) / (Gb * Gb - G0 * G0));

  315.     g  = pow(G,  1.0 / N);

  316.     g0 = pow(G0, 1.0 / N);

  317.     beta = pow(epsilon, -1.0 / N) * tan(wb/2);

  318.     c0 = cos(w0);

  319. 

  320.     for (i = 1; i <= L; i++) {

  321.         double ui = (2.0 * i - 1) / N;

  322.         double si = sin(M_PI * ui / 2.0);

  323.         double Di = beta * beta + 2 * si * beta + 1;

  324. 

  325.         butterworth_fo_section(&f->section[i - 1], beta, si, g, g0, Di, c0);

  326.     }

  327. }

  328. 

  329. static void chebyshev1_fo_section(FoSection *S, double a,

  330.                                   double c, double tetta_b,

  331.                                   double g0, double si, double b,

  332.                                   double D, double c0)

  333. {

  334.     if (c0 == 1 || c0 == -1) {

  335.         S->b0 = (tetta_b*tetta_b*(b*b+g0*g0*c*c) + 2*g0*b*si*tetta_b*tetta_b + g0*g0)/D;

  336.         S->b1 = 2*c0*(tetta_b*tetta_b*(b*b+g0*g0*c*c) - g0*g0)/D;

  337.         S->b2 = (tetta_b*tetta_b*(b*b+g0*g0*c*c) - 2*g0*b*si*tetta_b + g0*g0)/D;

  338.         S->b3 = 0;

  339.         S->b4 = 0;

  340. 

  341.         S->a0 = 1;

  342.         S->a1 = 2*c0*(tetta_b*tetta_b*(a*a+c*c) - 1)/D;

  343.         S->a2 = (tetta_b*tetta_b*(a*a+c*c) - 2*a*si*tetta_b + 1)/D;

  344.         S->a3 = 0;

  345.         S->a4 = 0;

  346.     } else {

  347.         S->b0 = ((b*b + g0*g0*c*c)*tetta_b*tetta_b + 2*g0*b*si*tetta_b + g0*g0)/D;

  348.         S->b1 = -4*c0*(g0*g0 + g0*b*si*tetta_b)/D;

  349.         S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - (b*b + g0*g0*c*c)*tetta_b*tetta_b)/D;

  350.         S->b3 = -4*c0*(g0*g0 - g0*b*si*tetta_b)/D;

  351.         S->b4 = ((b*b + g0*g0*c*c)*tetta_b*tetta_b - 2*g0*b*si*tetta_b + g0*g0)/D;

  352. 

  353.         S->a0 = 1;

  354.         S->a1 = -4*c0*(1 + a*si*tetta_b)/D;

  355.         S->a2 = 2*(1 + 2*c0*c0 - (a*a + c*c)*tetta_b*tetta_b)/D;

  356.         S->a3 = -4*c0*(1 - a*si*tetta_b)/D;

  357.         S->a4 = ((a*a + c*c)*tetta_b*tetta_b - 2*a*si*tetta_b + 1)/D;

  358.     }

  359. }

  360. 

  361. static void chebyshev1_bp_filter(EqualizatorFilter *f,

  362.                                  int N, double w0, double wb,

  363.                                  double G, double Gb, double G0)

  364. {

  365.     double a, b, c0, g0, alfa, beta, tetta_b;

  366.     double epsilon;

  367.     int r =  N % 2;

  368.     int L = (N - r) / 2;

  369.     int i;

  370. 

  371.     if (G == 0 && G0 == 0) {

  372.         f->section[0].a0 = 1;

  373.         f->section[0].b0 = 1;

  374.         f->section[1].a0 = 1;

  375.         f->section[1].b0 = 1;

  376.         return;

  377.     }

  378. 

  379.     G  = exp10( G/20);

  380.     Gb = exp10(Gb/20);

  381.     G0 = exp10(G0/20);

  382. 

  383.     epsilon = sqrt((G*G - Gb*Gb) / (Gb*Gb - G0*G0));

  384.     g0 = pow(G0,1.0/N);

  385.     alfa = pow(1.0/epsilon    + sqrt(1 + pow(epsilon,-2.0)), 1.0/N);

  386.     beta = pow(G/epsilon + Gb * sqrt(1 + pow(epsilon,-2.0)), 1.0/N);

  387.     a = 0.5 * (alfa - 1.0/alfa);

  388.     b = 0.5 * (beta - g0*g0*(1/beta));

  389.     tetta_b = tan(wb/2);

  390.     c0 = cos(w0);

  391. 

  392.     for (i = 1; i <= L; i++) {

  393.         double ui = (2.0*i-1.0)/N;

  394.         double ci = cos(M_PI*ui/2.0);

  395.         double si = sin(M_PI*ui/2.0);

  396.         double Di = (a*a + ci*ci)*tetta_b*tetta_b + 2.0*a*si*tetta_b + 1;

  397. 

  398.         chebyshev1_fo_section(&f->section[i - 1], a, ci, tetta_b, g0, si, b, Di, c0);

  399.     }

  400. }

  401. 

  402. static void chebyshev2_fo_section(FoSection *S, double a,

  403.                                   double c, double tetta_b,

  404.                                   double g, double si, double b,

  405.                                   double D, double c0)

  406. {

  407.     if (c0 == 1 || c0 == -1) {

  408.         S->b0 = (g*g*tetta_b*tetta_b + 2*tetta_b*g*b*si + b*b + g*g*c*c)/D;

  409.         S->b1 = 2*c0*(g*g*tetta_b*tetta_b - b*b - g*g*c*c)/D;

  410.         S->b2 = (g*g*tetta_b*tetta_b - 2*tetta_b*g*b*si + b*b + g*g*c*c)/D;

  411.         S->b3 = 0;

  412.         S->b4 = 0;

  413. 

  414.         S->a0 = 1;

  415.         S->a1 = 2*c0*(tetta_b*tetta_b - a*a - c*c)/D;

  416.         S->a2 = (tetta_b*tetta_b - 2*tetta_b*a*si + a*a + c*c)/D;

  417.         S->a3 = 0;

  418.         S->a4 = 0;

  419.     } else {

  420.         S->b0 = (g*g*tetta_b*tetta_b + 2*g*b*si*tetta_b + b*b + g*g*c*c)/D;

  421.         S->b1 = -4*c0*(b*b + g*g*c*c + g*b*si*tetta_b)/D;

  422.         S->b2 = 2*((b*b + g*g*c*c)*(1 + 2*c0*c0) - g*g*tetta_b*tetta_b)/D;

  423.         S->b3 = -4*c0*(b*b + g*g*c*c - g*b*si*tetta_b)/D;

  424.         S->b4 = (g*g*tetta_b*tetta_b - 2*g*b*si*tetta_b + b*b + g*g*c*c)/D;

  425. 

  426.         S->a0 = 1;

  427.         S->a1 = -4*c0*(a*a + c*c + a*si*tetta_b)/D;

  428.         S->a2 = 2*((a*a + c*c)*(1 + 2*c0*c0) - tetta_b*tetta_b)/D;

  429.         S->a3 = -4*c0*(a*a + c*c - a*si*tetta_b)/D;

  430.         S->a4 = (tetta_b*tetta_b - 2*a*si*tetta_b + a*a + c*c)/D;

  431.     }

  432. }

  433. 

  434. static void chebyshev2_bp_filter(EqualizatorFilter *f,

  435.                                  int N, double w0, double wb,

  436.                                  double G, double Gb, double G0)

  437. {

  438.     double a, b, c0, tetta_b;

  439.     double epsilon, g, eu, ew;

  440.     int r =  N % 2;

  441.     int L = (N - r) / 2;

  442.     int i;

  443. 

  444.     if (G == 0 && G0 == 0) {

  445.         f->section[0].a0 = 1;

  446.         f->section[0].b0 = 1;

  447.         f->section[1].a0 = 1;

  448.         f->section[1].b0 = 1;

  449.         return;

  450.     }

  451. 

  452.     G  = exp10( G/20);

  453.     Gb = exp10(Gb/20);

  454.     G0 = exp10(G0/20);

  455. 

  456.     epsilon = sqrt((G*G - Gb*Gb) / (Gb*Gb - G0*G0));

  457.     g  = pow(G, 1.0 / N);

  458.     eu = pow(epsilon + sqrt(1 + epsilon*epsilon), 1.0/N);

  459.     ew = pow(G0*epsilon + Gb*sqrt(1 + epsilon*epsilon), 1.0/N);

  460.     a = (eu - 1.0/eu)/2.0;

  461.     b = (ew - g*g/ew)/2.0;

  462.     tetta_b = tan(wb/2);

  463.     c0 = cos(w0);

  464. 

  465.     for (i = 1; i <= L; i++) {

  466.         double ui = (2.0 * i - 1.0)/N;

  467.         double ci = cos(M_PI * ui / 2.0);

  468.         double si = sin(M_PI * ui / 2.0);

  469.         double Di = tetta_b*tetta_b + 2*a*si*tetta_b + a*a + ci*ci;

  470. 

  471.         chebyshev2_fo_section(&f->section[i - 1], a, ci, tetta_b, g, si, b, Di, c0);

  472.     }

  473. }

  474. 

  475. static double butterworth_compute_bw_gain_db(double gain)

  476. {

  477.     double bw_gain = 0;

  478. 

  479.     if (gain <= -6)

  480.         bw_gain = gain + 3;

  481.     else if(gain > -6 && gain < 6)

  482.         bw_gain = gain * 0.5;

  483.     else if(gain >= 6)

  484.         bw_gain = gain - 3;

  485. 

  486.     return bw_gain;

  487. }

  488. 

  489. static double chebyshev1_compute_bw_gain_db(double gain)

  490. {

  491.     double bw_gain = 0;

  492. 

  493.     if (gain <= -6)

  494.         bw_gain = gain + 1;

  495.     else if(gain > -6 && gain < 6)

  496.         bw_gain = gain * 0.9;

  497.     else if(gain >= 6)

  498.         bw_gain = gain - 1;

  499. 

  500.     return bw_gain;

  501. }

  502. 

  503. static double chebyshev2_compute_bw_gain_db(double gain)

  504. {

  505.     double bw_gain = 0;

  506. 

  507.     if (gain <= -6)

  508.         bw_gain = -3;

  509.     else if(gain > -6 && gain < 6)

  510.         bw_gain = gain * 0.3;

  511.     else if(gain >= 6)

  512.         bw_gain = 3;

  513. 

  514.     return bw_gain;

  515. }

  516. 

  517. static inline double hz_2_rad(double x, double fs)

  518. {

  519.     return 2 * M_PI * x / fs;

  520. }

  521. 

  522. static void equalizer(EqualizatorFilter *f, double sample_rate)

  523. {

  524.     double w0 = hz_2_rad(f->freq,  sample_rate);

  525.     double wb = hz_2_rad(f->width, sample_rate);

  526.     double bw_gain;

  527. 

  528.     switch (f->type) {

  529.     case BUTTERWORTH:

  530.         bw_gain = butterworth_compute_bw_gain_db(f->gain);

  531.         butterworth_bp_filter(f, FILTER_ORDER, w0, wb, f->gain, bw_gain, 0);

  532.         break;

  533.     case CHEBYSHEV1:

  534.         bw_gain = chebyshev1_compute_bw_gain_db(f->gain);

  535.         chebyshev1_bp_filter(f, FILTER_ORDER, w0, wb, f->gain, bw_gain, 0);

  536.         break;

  537.     case CHEBYSHEV2:

  538.         bw_gain = chebyshev2_compute_bw_gain_db(f->gain);

  539.         chebyshev2_bp_filter(f, FILTER_ORDER, w0, wb, f->gain, bw_gain, 0);

  540.         break;

  541.     }

  542. 

  543. }

  544. 

  545. static int add_filter(AudioNEqualizerContext *s, AVFilterLink *inlink)

  546. {

  547.     equalizer(&s->filters[s->nb_filters], inlink->sample_rate);

  548.     if (s->nb_filters >= s->nb_allocated) {

  549.         EqualizatorFilter *filters;

  550. 

  551.         filters = av_calloc(s->nb_allocated, 2 * sizeof(*s->filters));

  552.         if (!filters)

  553.             return AVERROR(ENOMEM);

  554.         memcpy(filters, s->filters, sizeof(*s->filters) * s->nb_allocated);

  555.         av_free(s->filters);

  556.         s->filters = filters;

  557.         s->nb_allocated *= 2;

  558.     }

  559.     s->nb_filters++;

  560. 

  561.     return 0;

  562. }

  563. 

  564. static int config_input(AVFilterLink *inlink)

  565. {

  566.     AVFilterContext *ctx = inlink->dst;

  567.     AudioNEqualizerContext *s = ctx->priv;

  568.     char *args = av_strdup(s->args);

  569.     char *saveptr = NULL;

  570.     int ret = 0;

  571. 

  572.     if (!args)

  573.         return AVERROR(ENOMEM);

  574. 

  575.     s->nb_allocated = 32 * inlink->channels;

  576.     s->filters = av_calloc(inlink->channels, 32 * sizeof(*s->filters));

  577.     if (!s->filters) {

  578.         s->nb_allocated = 0;

  579.         return AVERROR(ENOMEM);

  580.     }

  581. 

  582.     while (1) {

  583.         char *arg = av_strtok(s->nb_filters == 0 ? args : NULL, "|", &saveptr);

  584. 

  585.         if (!arg)

  586.             break;

  587. 

  588.         s->filters[s->nb_filters].type = 0;

  589.         if (sscanf(arg, "c%d f=%lf w=%lf g=%lf t=%d", &s->filters[s->nb_filters].channel,

  590.                                                      &s->filters[s->nb_filters].freq,

  591.                                                      &s->filters[s->nb_filters].width,

  592.                                                      &s->filters[s->nb_filters].gain,

  593.                                                      &s->filters[s->nb_filters].type) != 5 &&

  594.             sscanf(arg, "c%d f=%lf w=%lf g=%lf", &s->filters[s->nb_filters].channel,

  595.                                                 &s->filters[s->nb_filters].freq,

  596.                                                 &s->filters[s->nb_filters].width,

  597.                                                 &s->filters[s->nb_filters].gain) != 4 ) {

  598.             av_free(args);

  599.             return AVERROR(EINVAL);

  600.         }

  601. 

  602.         if (s->filters[s->nb_filters].freq < 0 ||

  603.             s->filters[s->nb_filters].freq > inlink->sample_rate / 2)

  604.             s->filters[s->nb_filters].ignore = 1;

  605. 

  606.         if (s->filters[s->nb_filters].channel < 0 ||

  607.             s->filters[s->nb_filters].channel >= inlink->channels)

  608.             s->filters[s->nb_filters].ignore = 1;

  609. 

  610.         av_clip(s->filters[s->nb_filters].type, 0, NB_TYPES - 1);

  611.         ret = add_filter(s, inlink);

  612.         if (ret < 0)

  613.             break;

  614.     }

  615. 

  616.     av_free(args);

  617. 

  618.     return ret;

  619. }

  620. 

  621. static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,

  622.                            char *res, int res_len, int flags)

  623. {

  624.     AudioNEqualizerContext *s = ctx->priv;

  625.     AVFilterLink *inlink = ctx->inputs[0];

  626.     int ret = AVERROR(ENOSYS);

  627. 

  628.     if (!strcmp(cmd, "change")) {

  629.         double freq, width, gain;

  630.         int filter;

  631. 

  632.         if (sscanf(args, "%d|f=%lf|w=%lf|g=%lf", &filter, &freq, &width, &gain) != 4)

  633.             return AVERROR(EINVAL);

  634. 

  635.         if (filter < 0 || filter >= s->nb_filters)

  636.             return AVERROR(EINVAL);

  637. 

  638.         if (freq < 0 || freq > inlink->sample_rate / 2)

  639.             return AVERROR(EINVAL);

  640. 

  641.         s->filters[filter].freq  = freq;

  642.         s->filters[filter].width = width;

  643.         s->filters[filter].gain  = gain;

  644.         equalizer(&s->filters[filter], inlink->sample_rate);

  645.         if (s->draw_curves)

  646.             draw_curves(ctx, inlink, s->video);

  647. 

  648.         ret = 0;

  649.     }

  650. 

  651.     return ret;

  652. }

  653. 

  654. static inline double section_process(FoSection *S, double in)

  655. {

  656.     double out;

  657. 

  658.     out = S->b0 * in;

  659.     out+= S->b1 * S->num[0] - S->denum[0] * S->a1;

  660.     out+= S->b2 * S->num[1] - S->denum[1] * S->a2;

  661.     out+= S->b3 * S->num[2] - S->denum[2] * S->a3;

  662.     out+= S->b4 * S->num[3] - S->denum[3] * S->a4;

  663. 

  664.     S->num[3] = S->num[2];

  665.     S->num[2] = S->num[1];

  666.     S->num[1] = S->num[0];

  667.     S->num[0] = in;

  668. 

  669.     S->denum[3] = S->denum[2];

  670.     S->denum[2] = S->denum[1];

  671.     S->denum[1] = S->denum[0];

  672.     S->denum[0] = out;

  673. 

  674.     return out;

  675. }

  676. 

  677. static double process_sample(FoSection *s1, double in)

  678. {

  679.     double p0 = in, p1;

  680.     int i;

  681. 

  682.     for (i = 0; i < FILTER_ORDER / 2; i++) {

  683.         p1 = section_process(&s1[i], p0);

  684.         p0 = p1;

  685.     }

  686. 

  687.     return p1;

  688. }

  689. 

  690. static int filter_frame(AVFilterLink *inlink, AVFrame *buf)

  691. {

  692.     AVFilterContext *ctx = inlink->dst;

  693.     AudioNEqualizerContext *s = ctx->priv;

  694.     AVFilterLink *outlink = ctx->outputs[0];

  695.     double *bptr;

  696.     int i, n;

  697. 

  698.     for (i = 0; i < s->nb_filters; i++) {

  699.         EqualizatorFilter *f = &s->filters[i];

  700. 

  701.         if (f->gain == 0. || f->ignore)

  702.             continue;

  703. 

  704.         bptr = (double *)buf->extended_data[f->channel];

  705.         for (n = 0; n < buf->nb_samples; n++) {

  706.             double sample = bptr[n];

  707. 

  708.             sample  = process_sample(f->section, sample);

  709.             bptr[n] = sample;

  710.         }

  711.     }

  712. 

  713.     if (s->draw_curves) {

  714.         const int64_t pts = buf->pts +

  715.             av_rescale_q(buf->nb_samples, (AVRational){ 1, inlink->sample_rate },

  716.                          outlink->time_base);

  717.         int ret;

  718. 

  719.         s->video->pts = pts;

  720.         ret = ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));

  721.         if (ret < 0)

  722.             return ret;

  723.     }

  724. 

  725.     return ff_filter_frame(outlink, buf);

  726. }

  727. 

  728. static const AVFilterPad inputs[] = {

  729.     {

  730.         .name           = "default",

  731.         .type           = AVMEDIA_TYPE_AUDIO,

  732.         .config_props   = config_input,

  733.         .filter_frame   = filter_frame,

  734.         .needs_writable = 1,

  735.     },

  736.     { NULL }

  737. };

  738. 

  739. AVFilter ff_af_anequalizer = {

  740.     .name          = "anequalizer",

  741.     .description   = NULL_IF_CONFIG_SMALL("Apply high-order audio parametric multi band equalizer."),

  742.     .priv_size     = sizeof(AudioNEqualizerContext),

  743.     .priv_class    = &anequalizer_class,

  744.     .init          = init,

  745.     .uninit        = uninit,

  746.     .query_formats = query_formats,

  747.     .inputs        = inputs,

  748.     .outputs       = NULL,

  749.     .flags         = AVFILTER_FLAG_DYNAMIC_OUTPUTS,

  750.     .process_command = process_command,

  751. };