falkTX
/
FFmpeg
mirror of https://github.com/falkTX/FFmpeg.git
1
0
Fork 0
Code Issues 0 Releases 338 Wiki Activity
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
77936 Commits
32 Branches
802MB
Tree: 7e29903526
Branches Tags
${ item.name }
Create branch ${ searchTerm }
from '7e29903526'
${ noResults }
FFmpeg/libavfilter/af_anequalizer.c

761 lines
23KB
Raw Blame History 

  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 "libavutil/intreadwrite.h"

  23. #include "libavutil/avstring.h"

  24. #include "libavutil/opt.h"

  25. #include "libavutil/parseutils.h"

  26. #include "avfilter.h"

  27. #include "internal.h"

  28. #include "audio.h"

  29. 

  30. #define FILTER_ORDER 4

  31. 

  32. enum FilterType {

  33.     BUTTERWORTH,

  34.     CHEBYSHEV1,

  35.     CHEBYSHEV2,

  36.     NB_TYPES

  37. };

  38. 

  39. typedef struct FoSection {

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

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

  42. 

  43.     double num[4];

  44.     double denum[4];

  45. } FoSection;

  46. 

  47. typedef struct EqualizatorFilter {

  48.     int ignore;

  49.     int channel;

  50.     int type;

  51. 

  52.     double freq;

  53.     double gain;

  54.     double width;

  55. 

  56.     FoSection section[2];

  57. } EqualizatorFilter;

  58. 

  59. typedef struct AudioNEqualizerContext {

  60.     const AVClass *class;

  61.     char *args;

  62.     char *colors;

  63.     int draw_curves;

  64.     int w, h;

  65. 

  66.     double mag;

  67.     int fscale;

  68.     int nb_filters;

  69.     int nb_allocated;

  70.     EqualizatorFilter *filters;

  71.     AVFrame *video;

  72. } AudioNEqualizerContext;

  73. 

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

  75. #define A AV_OPT_FLAG_AUDIO_PARAM

  76. #define V AV_OPT_FLAG_VIDEO_PARAM

  77. #define F AV_OPT_FLAG_FILTERING_PARAM

  78. 

  79. static const AVOption anequalizer_options[] = {

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

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

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

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

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

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

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

  87.     { "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 },

  88.     { NULL }

  89. };

  90. 

  91. AVFILTER_DEFINE_CLASS(anequalizer);

  92. 

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

  94. {

  95.     AudioNEqualizerContext *s = ctx->priv;

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

  97.     int ch, i, n;

  98. 

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

  100.     if (!colors)

  101.         return;

  102. 

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

  104. 

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

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

  107.         int prev_v = -1;

  108.         double f;

  109. 

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

  111.         if (color)

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

  113. 

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

  115.             double zr, zi, zr2, zi2;

  116.             double Hr, Hi;

  117.             double Hmag = 1;

  118.             double w;

  119.             int v, y, x;

  120. 

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

  122.             zr = cos(w);

  123.             zr2 = zr * zr;

  124.             zi = -sin(w);

  125.             zi2 = zi * zi;

  126. 

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

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

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

  130.                     continue;

  131. 

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

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

  134. 

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

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

  137. 

  138.                     Hr = S->b4*(1-8*zr2*zi2) + S->b2*(zr2-zi2) + zr*(S->b1+S->b3*(zr2-3*zi2))+ S->b0;

  139.                     Hi = zi*(S->b3*(3*zr2-zi2) + S->b1 + 2*zr*(2*S->b4*(zr2-zi2) + S->b2));

  140.                     Hmag *= hypot(Hr, Hi);

  141.                     Hr = S->a4*(1-8*zr2*zi2) + S->a2*(zr2-zi2) + zr*(S->a1+S->a3*(zr2-3*zi2))+ S->a0;

  142.                     Hi = zi*(S->a3*(3*zr2-zi2) + S->a1 + 2*zr*(2*S->a4*(zr2-zi2) + S->a2));

  143.                     Hmag /= hypot(Hr, Hi);

  144.                 }

  145.             }

  146. 

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

  148.             x = lrint(f);

  149.             if (prev_v == -1)

  150.                 prev_v = v;

  151.             if (v <= prev_v) {

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

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

  154.             } else {

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

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

  157.             }

  158. 

  159.             prev_v = v;

  160.         }

  161.     }

  162. 

  163.     av_free(colors);

  164. }

  165. 

  166. static int config_video(AVFilterLink *outlink)

  167. {

  168.     AVFilterContext *ctx = outlink->src;

  169.     AudioNEqualizerContext *s = ctx->priv;

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

  171.     AVFrame *out;

  172. 

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

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

  175. 

  176.     av_frame_free(&s->video);

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

  178.     if (!out)

  179.         return AVERROR(ENOMEM);

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

  181. 

  182.     draw_curves(ctx, inlink, out);

  183. 

  184.     return 0;

  185. }

  186. 

  187. static av_cold int init(AVFilterContext *ctx)

  188. {

  189.     AudioNEqualizerContext *s = ctx->priv;

  190.     AVFilterPad pad, vpad;

  191. 

  192.     pad = (AVFilterPad){

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

  194.         .type         = AVMEDIA_TYPE_AUDIO,

  195.     };

  196. 

  197.     if (!pad.name)

  198.         return AVERROR(ENOMEM);

  199. 

  200.     if (s->draw_curves) {

  201.         vpad = (AVFilterPad){

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

  203.             .type         = AVMEDIA_TYPE_VIDEO,

  204.             .config_props = config_video,

  205.         };

  206.         if (!vpad.name)

  207.             return AVERROR(ENOMEM);

  208.     }

  209. 

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

  211. 

  212.     if (s->draw_curves)

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

  214. 

  215.     return 0;

  216. }

  217. 

  218. static int query_formats(AVFilterContext *ctx)

  219. {

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

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

  222.     AudioNEqualizerContext *s = ctx->priv;

  223.     AVFilterFormats *formats;

  224.     AVFilterChannelLayouts *layouts;

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

  226.     static const enum AVSampleFormat sample_fmts[] = {

  227.         AV_SAMPLE_FMT_DBLP,

  228.         AV_SAMPLE_FMT_NONE

  229.     };

  230.     int ret;

  231. 

  232.     if (s->draw_curves) {

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

  234.         formats = ff_make_format_list(pix_fmts);

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

  236.             return ret;

  237.     }

  238. 

  239.     formats = ff_make_format_list(sample_fmts);

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

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

  242.         return ret;

  243. 

  244.     layouts = ff_all_channel_counts();

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

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

  247.         return ret;

  248. 

  249.     formats = ff_all_samplerates();

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

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

  252.         return ret;

  253. 

  254.     return 0;

  255. }

  256. 

  257. static av_cold void uninit(AVFilterContext *ctx)

  258. {

  259.     AudioNEqualizerContext *s = ctx->priv;

  260. 

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

  262.     if (s->draw_curves)

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

  264.     av_frame_free(&s->video);

  265.     av_freep(&s->filters);

  266.     s->nb_filters = 0;

  267.     s->nb_allocated = 0;

  268. }

  269. 

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

  271.                                    double si, double g, double g0,

  272.                                    double D, double c0)

  273. {

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

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

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

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

  278.         S->b3 = 0;

  279.         S->b4 = 0;

  280. 

  281.         S->a0 = 1;

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

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

  284.         S->a3 = 0;

  285.         S->a4 = 0;

  286.     } else {

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

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

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

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

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

  292. 

  293.         S->a0 = 1;

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

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

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

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

  298.     }

  299. }

  300. 

  301. static void butterworth_bp_filter(EqualizatorFilter *f,

  302.                                   int N, double w0, double wb,

  303.                                   double G, double Gb, double G0)

  304. {

  305.     double g, c0, g0, beta;

  306.     double epsilon;

  307.     int r =  N % 2;

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

  309.     int i;

  310. 

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

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

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

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

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

  316.         return;

  317.     }

  318. 

  319.     G  = ff_exp10(G/20);

  320.     Gb = ff_exp10(Gb/20);

  321.     G0 = ff_exp10(G0/20);

  322. 

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

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

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

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

  327.     c0 = cos(w0);

  328. 

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

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

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

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

  333. 

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

  335.     }

  336. }

  337. 

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

  339.                                   double c, double tetta_b,

  340.                                   double g0, double si, double b,

  341.                                   double D, double c0)

  342. {

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

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

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

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

  347.         S->b3 = 0;

  348.         S->b4 = 0;

  349. 

  350.         S->a0 = 1;

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

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

  353.         S->a3 = 0;

  354.         S->a4 = 0;

  355.     } else {

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

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

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

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

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

  361. 

  362.         S->a0 = 1;

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

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

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

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

  367.     }

  368. }

  369. 

  370. static void chebyshev1_bp_filter(EqualizatorFilter *f,

  371.                                  int N, double w0, double wb,

  372.                                  double G, double Gb, double G0)

  373. {

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

  375.     double epsilon;

  376.     int r =  N % 2;

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

  378.     int i;

  379. 

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

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

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

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

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

  385.         return;

  386.     }

  387. 

  388.     G  = ff_exp10(G/20);

  389.     Gb = ff_exp10(Gb/20);

  390.     G0 = ff_exp10(G0/20);

  391. 

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

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

  394.     alfa = pow(1.0/epsilon    + sqrt(1 + 1/(epsilon*epsilon)), 1.0/N);

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

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

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

  398.     tetta_b = tan(wb/2);

  399.     c0 = cos(w0);

  400. 

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

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

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

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

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

  406. 

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

  408.     }

  409. }

  410. 

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

  412.                                   double c, double tetta_b,

  413.                                   double g, double si, double b,

  414.                                   double D, double c0)

  415. {

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

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

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

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

  420.         S->b3 = 0;

  421.         S->b4 = 0;

  422. 

  423.         S->a0 = 1;

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

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

  426.         S->a3 = 0;

  427.         S->a4 = 0;

  428.     } else {

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

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

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

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

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

  434. 

  435.         S->a0 = 1;

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

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

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

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

  440.     }

  441. }

  442. 

  443. static void chebyshev2_bp_filter(EqualizatorFilter *f,

  444.                                  int N, double w0, double wb,

  445.                                  double G, double Gb, double G0)

  446. {

  447.     double a, b, c0, tetta_b;

  448.     double epsilon, g, eu, ew;

  449.     int r =  N % 2;

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

  451.     int i;

  452. 

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

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

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

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

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

  458.         return;

  459.     }

  460. 

  461.     G  = ff_exp10(G/20);

  462.     Gb = ff_exp10(Gb/20);

  463.     G0 = ff_exp10(G0/20);

  464. 

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

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

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

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

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

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

  471.     tetta_b = tan(wb/2);

  472.     c0 = cos(w0);

  473. 

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

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

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

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

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

  479. 

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

  481.     }

  482. }

  483. 

  484. static double butterworth_compute_bw_gain_db(double gain)

  485. {

  486.     double bw_gain = 0;

  487. 

  488.     if (gain <= -6)

  489.         bw_gain = gain + 3;

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

  491.         bw_gain = gain * 0.5;

  492.     else if(gain >= 6)

  493.         bw_gain = gain - 3;

  494. 

  495.     return bw_gain;

  496. }

  497. 

  498. static double chebyshev1_compute_bw_gain_db(double gain)

  499. {

  500.     double bw_gain = 0;

  501. 

  502.     if (gain <= -6)

  503.         bw_gain = gain + 1;

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

  505.         bw_gain = gain * 0.9;

  506.     else if(gain >= 6)

  507.         bw_gain = gain - 1;

  508. 

  509.     return bw_gain;

  510. }

  511. 

  512. static double chebyshev2_compute_bw_gain_db(double gain)

  513. {

  514.     double bw_gain = 0;

  515. 

  516.     if (gain <= -6)

  517.         bw_gain = -3;

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

  519.         bw_gain = gain * 0.3;

  520.     else if(gain >= 6)

  521.         bw_gain = 3;

  522. 

  523.     return bw_gain;

  524. }

  525. 

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

  527. {

  528.     return 2 * M_PI * x / fs;

  529. }

  530. 

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

  532. {

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

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

  535.     double bw_gain;

  536. 

  537.     switch (f->type) {

  538.     case BUTTERWORTH:

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

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

  541.         break;

  542.     case CHEBYSHEV1:

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

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

  545.         break;

  546.     case CHEBYSHEV2:

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

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

  549.         break;

  550.     }

  551. 

  552. }

  553. 

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

  555. {

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

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

  558.         EqualizatorFilter *filters;

  559. 

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

  561.         if (!filters)

  562.             return AVERROR(ENOMEM);

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

  564.         av_free(s->filters);

  565.         s->filters = filters;

  566.         s->nb_allocated *= 2;

  567.     }

  568.     s->nb_filters++;

  569. 

  570.     return 0;

  571. }

  572. 

  573. static int config_input(AVFilterLink *inlink)

  574. {

  575.     AVFilterContext *ctx = inlink->dst;

  576.     AudioNEqualizerContext *s = ctx->priv;

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

  578.     char *saveptr = NULL;

  579.     int ret = 0;

  580. 

  581.     if (!args)

  582.         return AVERROR(ENOMEM);

  583. 

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

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

  586.     if (!s->filters) {

  587.         s->nb_allocated = 0;

  588.         return AVERROR(ENOMEM);

  589.     }

  590. 

  591.     while (1) {

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

  593. 

  594.         if (!arg)

  595.             break;

  596. 

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

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

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

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

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

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

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

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

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

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

  607.             av_free(args);

  608.             return AVERROR(EINVAL);

  609.         }

  610. 

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

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

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

  614. 

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

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

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

  618. 

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

  620.         ret = add_filter(s, inlink);

  621.         if (ret < 0)

  622.             break;

  623.     }

  624. 

  625.     av_free(args);

  626. 

  627.     return ret;

  628. }

  629. 

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

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

  632. {

  633.     AudioNEqualizerContext *s = ctx->priv;

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

  635.     int ret = AVERROR(ENOSYS);

  636. 

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

  638.         double freq, width, gain;

  639.         int filter;

  640. 

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

  642.             return AVERROR(EINVAL);

  643. 

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

  645.             return AVERROR(EINVAL);

  646. 

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

  648.             return AVERROR(EINVAL);

  649. 

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

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

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

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

  654.         if (s->draw_curves)

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

  656. 

  657.         ret = 0;

  658.     }

  659. 

  660.     return ret;

  661. }

  662. 

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

  664. {

  665.     double out;

  666. 

  667.     out = S->b0 * in;

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

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

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

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

  672. 

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

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

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

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

  677. 

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

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

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

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

  682. 

  683.     return out;

  684. }

  685. 

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

  687. {

  688.     double p0 = in, p1;

  689.     int i;

  690. 

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

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

  693.         p0 = p1;

  694.     }

  695. 

  696.     return p1;

  697. }

  698. 

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

  700. {

  701.     AVFilterContext *ctx = inlink->dst;

  702.     AudioNEqualizerContext *s = ctx->priv;

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

  704.     double *bptr;

  705.     int i, n;

  706. 

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

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

  709. 

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

  711.             continue;

  712. 

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

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

  715.             double sample = bptr[n];

  716. 

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

  718.             bptr[n] = sample;

  719.         }

  720.     }

  721. 

  722.     if (s->draw_curves) {

  723.         const int64_t pts = buf->pts +

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

  725.                          outlink->time_base);

  726.         int ret;

  727. 

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

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

  730.         if (ret < 0)

  731.             return ret;

  732.     }

  733. 

  734.     return ff_filter_frame(outlink, buf);

  735. }

  736. 

  737. static const AVFilterPad inputs[] = {

  738.     {

  739.         .name           = "default",

  740.         .type           = AVMEDIA_TYPE_AUDIO,

  741.         .config_props   = config_input,

  742.         .filter_frame   = filter_frame,

  743.         .needs_writable = 1,

  744.     },

  745.     { NULL }

  746. };

  747. 

  748. AVFilter ff_af_anequalizer = {

  749.     .name          = "anequalizer",

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

  751.     .priv_size     = sizeof(AudioNEqualizerContext),

  752.     .priv_class    = &anequalizer_class,

  753.     .init          = init,

  754.     .uninit        = uninit,

  755.     .query_formats = query_formats,

  756.     .inputs        = inputs,

  757.     .outputs       = NULL,

  758.     .flags         = AVFILTER_FLAG_DYNAMIC_OUTPUTS,

  759.     .process_command = process_command,

  760. };