libavfilter/af_aemphasis.c

   1 /*
   2  * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit and others
   3  *
   4  * This file is part of FFmpeg.
   5  *
   6  * FFmpeg is free software; you can redistribute it and/or
   7  * modify it under the terms of the GNU Lesser General Public
   8  * License as published by the Free Software Foundation; either
   9  * version 2.1 of the License, or (at your option) any later version.
  10  *
  11  * FFmpeg is distributed in the hope that it will be useful,
  12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  14  * Lesser General Public License for more details.
  15  *
  16  * You should have received a copy of the GNU Lesser General Public
  17  * License along with FFmpeg; if not, write to the Free Software
  18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  19  */
  20
  21 #include "libavutil/opt.h"
  22 #include "avfilter.h"
  23 #include "filters.h"
  24 #include "audio.h"
  25
  26 typedef struct BiquadCoeffs {
  27     double a0, a1, a2, b1, b2;
  28 } BiquadCoeffs;
  29
  30 typedef struct RIAACurve {
  31     BiquadCoeffs r1;
  32     BiquadCoeffs brickw;
  33     int use_brickw;
  34 } RIAACurve;
  35
  36 typedef struct AudioEmphasisContext {
  37     const AVClass *class;
  38     int mode, type;
  39     double level_in, level_out;
  40
  41     RIAACurve rc;
  42
  43     AVFrame *w;
  44 } AudioEmphasisContext;
  45
  46 #define OFFSET(x) offsetof(AudioEmphasisContext, x)
  47 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
  48
  49 static const AVOption aemphasis_options[] = {
  50     { "level_in",      "set input gain", OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
  51     { "level_out",    "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
  52     { "mode",         "set filter mode", OFFSET(mode), AV_OPT_TYPE_INT,   {.i64=0}, 0, 1, FLAGS, .unit = "mode" },
  53     { "reproduction",              NULL,            0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "mode" },
  54     { "production",                NULL,            0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "mode" },
  55     { "type",         "set filter type", OFFSET(type), AV_OPT_TYPE_INT,   {.i64=4}, 0, 8, FLAGS, .unit = "type" },
  56     { "col",                 "Columbia",            0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "type" },
  57     { "emi",                      "EMI",            0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "type" },
  58     { "bsi",              "BSI (78RPM)",            0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, .unit = "type" },
  59     { "riaa",                    "RIAA",            0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, FLAGS, .unit = "type" },
  60     { "cd",         "Compact Disc (CD)",            0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, FLAGS, .unit = "type" },
  61     { "50fm",               "50µs (FM)",            0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, FLAGS, .unit = "type" },
  62     { "75fm",               "75µs (FM)",            0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, FLAGS, .unit = "type" },
  63     { "50kf",            "50µs (FM-KF)",            0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, FLAGS, .unit = "type" },
  64     { "75kf",            "75µs (FM-KF)",            0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, FLAGS, .unit = "type" },
  65     { NULL }
  66 };
  67
  68 AVFILTER_DEFINE_CLASS(aemphasis);
  69
  70 static inline void biquad_process(BiquadCoeffs *bq, double *dst, const double *src, int nb_samples,
  71                                   double *w, double level_in, double level_out)
  72 {
  73     const double a0 = bq->a0;
  74     const double a1 = bq->a1;
  75     const double a2 = bq->a2;
  76     const double b1 = bq->b1;
  77     const double b2 = bq->b2;
  78     double w1 = w[0];
  79     double w2 = w[1];
  80
  81     for (int i = 0; i < nb_samples; i++) {
  82         double n = src[i] * level_in;
  83         double tmp = n - w1 * b1 - w2 * b2;
  84         double out = tmp * a0 + w1 * a1 + w2 * a2;
  85
  86         w2 = w1;
  87         w1 = tmp;
  88
  89         dst[i] = out * level_out;
  90     }
  91
  92     w[0] = w1;
  93     w[1] = w2;
  94 }
  95
  96 typedef struct ThreadData {
  97     AVFrame *in, *out;
  98 } ThreadData;
  99
 100 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
 101 {
 102     AudioEmphasisContext *s = ctx->priv;
 103     const double level_out = s->level_out;
 104     const double level_in = s->level_in;
 105     ThreadData *td = arg;
 106     AVFrame *out = td->out;
 107     AVFrame *in = td->in;
 108     const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
 109     const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
 110
 111     for (int ch = start; ch < end; ch++) {
 112         const double *src = (const double *)in->extended_data[ch];
 113         double *w = (double *)s->w->extended_data[ch];
 114         double *dst = (double *)out->extended_data[ch];
 115
 116         if (s->rc.use_brickw) {
 117             biquad_process(&s->rc.brickw, dst, src, in->nb_samples, w + 2, level_in, 1.);
 118             biquad_process(&s->rc.r1, dst, dst, in->nb_samples, w, 1., level_out);
 119         } else {
 120             biquad_process(&s->rc.r1, dst, src, in->nb_samples, w, level_in, level_out);
 121         }
 122     }
 123
 124     return 0;
 125 }
 126
 127 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 128 {
 129     AVFilterContext *ctx = inlink->dst;
 130     AVFilterLink *outlink = ctx->outputs[0];
 131     ThreadData td;
 132     AVFrame *out;
 133
 134     if (av_frame_is_writable(in)) {
 135         out = in;
 136     } else {
 137         out = ff_get_audio_buffer(outlink, in->nb_samples);
 138         if (!out) {
 139             av_frame_free(&in);
 140             return AVERROR(ENOMEM);
 141         }
 142         av_frame_copy_props(out, in);
 143     }
 144
 145     td.in = in; td.out = out;
 146     ff_filter_execute(ctx, filter_channels, &td, NULL,
 147                       FFMIN(inlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
 148
 149     if (in != out)
 150         av_frame_free(&in);
 151     return ff_filter_frame(outlink, out);
 152 }
 153
 154 static inline void set_highshelf_rbj(BiquadCoeffs *bq, double freq, double q, double peak, double sr)
 155 {
 156     double A = sqrt(peak);
 157     double w0 = freq * 2 * M_PI / sr;
 158     double alpha = sin(w0) / (2 * q);
 159     double cw0 = cos(w0);
 160     double tmp = 2 * sqrt(A) * alpha;
 161     double b0 = 0, ib0 = 0;
 162
 163     bq->a0 =    A*( (A+1) + (A-1)*cw0 + tmp);
 164     bq->a1 = -2*A*( (A-1) + (A+1)*cw0);
 165     bq->a2 =    A*( (A+1) + (A-1)*cw0 - tmp);
 166         b0 =        (A+1) - (A-1)*cw0 + tmp;
 167     bq->b1 =    2*( (A-1) - (A+1)*cw0);
 168     bq->b2 =        (A+1) - (A-1)*cw0 - tmp;
 169
 170     ib0     = 1 / b0;
 171     bq->b1 *= ib0;
 172     bq->b2 *= ib0;
 173     bq->a0 *= ib0;
 174     bq->a1 *= ib0;
 175     bq->a2 *= ib0;
 176 }
 177
 178 static inline void set_lp_rbj(BiquadCoeffs *bq, double fc, double q, double sr, double gain)
 179 {
 180     double omega = 2.0 * M_PI * fc / sr;
 181     double sn = sin(omega);
 182     double cs = cos(omega);
 183     double alpha = sn/(2 * q);
 184     double inv = 1.0/(1.0 + alpha);
 185
 186     bq->a2 = bq->a0 = gain * inv * (1.0 - cs) * 0.5;
 187     bq->a1 = bq->a0 + bq->a0;
 188     bq->b1 = (-2.0 * cs * inv);
 189     bq->b2 = ((1.0 - alpha) * inv);
 190 }
 191
 192 static double freq_gain(BiquadCoeffs *c, double freq, double sr)
 193 {
 194     double zr, zi;
 195
 196     freq *= 2.0 * M_PI / sr;
 197     zr = cos(freq);
 198     zi = -sin(freq);
 199
 200     /* |(a0 + a1*z + a2*z^2)/(1 + b1*z + b2*z^2)| */
 201     return hypot(c->a0 + c->a1*zr + c->a2*(zr*zr-zi*zi), c->a1*zi + 2*c->a2*zr*zi) /
 202            hypot(1 + c->b1*zr + c->b2*(zr*zr-zi*zi), c->b1*zi + 2*c->b2*zr*zi);
 203 }
 204
 205 static int config_input(AVFilterLink *inlink)
 206 {
 207     double i, j, k, g, t, a0, a1, a2, b1, b2, tau1, tau2, tau3;
 208     double cutfreq, gain1kHz, gc, sr = inlink->sample_rate;
 209     AVFilterContext *ctx = inlink->dst;
 210     AudioEmphasisContext *s = ctx->priv;
 211     BiquadCoeffs coeffs;
 212
 213     if (!s->w)
 214         s->w = ff_get_audio_buffer(inlink, 4);
 215     if (!s->w)
 216         return AVERROR(ENOMEM);
 217
 218     switch (s->type) {
 219     case 0: //"Columbia"
 220         i = 100.;
 221         j = 500.;
 222         k = 1590.;
 223         break;
 224     case 1: //"EMI"
 225         i = 70.;
 226         j = 500.;
 227         k = 2500.;
 228         break;
 229     case 2: //"BSI(78rpm)"
 230         i = 50.;
 231         j = 353.;
 232         k = 3180.;
 233         break;
 234     case 3: //"RIAA"
 235     default:
 236         tau1 = 0.003180;
 237         tau2 = 0.000318;
 238         tau3 = 0.000075;
 239         i = 1. / (2. * M_PI * tau1);
 240         j = 1. / (2. * M_PI * tau2);
 241         k = 1. / (2. * M_PI * tau3);
 242         break;
 243     case 4: //"CD Mastering"
 244         tau1 = 0.000050;
 245         tau2 = 0.000015;
 246         tau3 = 0.0000001;// 1.6MHz out of audible range for null impact
 247         i = 1. / (2. * M_PI * tau1);
 248         j = 1. / (2. * M_PI * tau2);
 249         k = 1. / (2. * M_PI * tau3);
 250         break;
 251     case 5: //"50µs FM (Europe)"
 252         tau1 = 0.000050;
 253         tau2 = tau1 / 20;// not used
 254         tau3 = tau1 / 50;//
 255         i = 1. / (2. * M_PI * tau1);
 256         j = 1. / (2. * M_PI * tau2);
 257         k = 1. / (2. * M_PI * tau3);
 258         break;
 259     case 6: //"75µs FM (US)"
 260         tau1 = 0.000075;
 261         tau2 = tau1 / 20;// not used
 262         tau3 = tau1 / 50;//
 263         i = 1. / (2. * M_PI * tau1);
 264         j = 1. / (2. * M_PI * tau2);
 265         k = 1. / (2. * M_PI * tau3);
 266         break;
 267     }
 268
 269     i *= 2 * M_PI;
 270     j *= 2 * M_PI;
 271     k *= 2 * M_PI;
 272
 273     t = 1. / sr;
 274
 275     //swap a1 b1, a2 b2
 276     if (s->type == 7 || s->type == 8) {
 277         double tau = (s->type == 7 ? 0.000050 : 0.000075);
 278         double f = 1.0 / (2 * M_PI * tau);
 279         double nyq = sr * 0.5;
 280         double gain = sqrt(1.0 + nyq * nyq / (f * f)); // gain at Nyquist
 281         double cfreq = sqrt((gain - 1.0) * f * f); // frequency
 282         double q = 1.0;
 283
 284         if (s->type == 8)
 285             q = pow((sr / 3269.0) + 19.5, -0.25); // somewhat poor curve-fit
 286         if (s->type == 7)
 287             q = pow((sr / 4750.0) + 19.5, -0.25);
 288         if (s->mode == 0)
 289             set_highshelf_rbj(&s->rc.r1, cfreq, q, 1. / gain, sr);
 290         else
 291             set_highshelf_rbj(&s->rc.r1, cfreq, q, gain, sr);
 292         s->rc.use_brickw = 0;
 293     } else {
 294         s->rc.use_brickw = 1;
 295         if (s->mode == 0) { // Reproduction
 296             g  = 1. / (4.+2.*i*t+2.*k*t+i*k*t*t);
 297             a0 = (2.*t+j*t*t)*g;
 298             a1 = (2.*j*t*t)*g;
 299             a2 = (-2.*t+j*t*t)*g;
 300             b1 = (-8.+2.*i*k*t*t)*g;
 301             b2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
 302         } else {  // Production
 303             g  = 1. / (2.*t+j*t*t);
 304             a0 = (4.+2.*i*t+2.*k*t+i*k*t*t)*g;
 305             a1 = (-8.+2.*i*k*t*t)*g;
 306             a2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
 307             b1 = (2.*j*t*t)*g;
 308             b2 = (-2.*t+j*t*t)*g;
 309         }
 310
 311         coeffs.a0 = a0;
 312         coeffs.a1 = a1;
 313         coeffs.a2 = a2;
 314         coeffs.b1 = b1;
 315         coeffs.b2 = b2;
 316
 317         // the coeffs above give non-normalized value, so it should be normalized to produce 0dB at 1 kHz
 318         // find actual gain
 319         // Note: for FM emphasis, use 100 Hz for normalization instead
 320         gain1kHz = freq_gain(&coeffs, 1000.0, sr);
 321         // divide one filter's x[n-m] coefficients by that value
 322         gc = 1.0 / gain1kHz;
 323         s->rc.r1.a0 = coeffs.a0 * gc;
 324         s->rc.r1.a1 = coeffs.a1 * gc;
 325         s->rc.r1.a2 = coeffs.a2 * gc;
 326         s->rc.r1.b1 = coeffs.b1;
 327         s->rc.r1.b2 = coeffs.b2;
 328     }
 329
 330     cutfreq = FFMIN(0.45 * sr, 21000.);
 331     set_lp_rbj(&s->rc.brickw, cutfreq, 0.707, sr, 1.);
 332
 333     return 0;
 334 }
 335
 336 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
 337                            char *res, int res_len, int flags)
 338 {
 339     int ret;
 340
 341     ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
 342     if (ret < 0)
 343         return ret;
 344
 345     return config_input(ctx->inputs[0]);
 346 }
 347
 348 static av_cold void uninit(AVFilterContext *ctx)
 349 {
 350     AudioEmphasisContext *s = ctx->priv;
 351
 352     av_frame_free(&s->w);
 353 }
 354
 355 static const AVFilterPad avfilter_af_aemphasis_inputs[] = {
 356     {
 357         .name         = "default",
 358         .type         = AVMEDIA_TYPE_AUDIO,
 359         .config_props = config_input,
 360         .filter_frame = filter_frame,
 361     },
 362 };
 363
 364 const FFFilter ff_af_aemphasis = {
 365     .p.name        = "aemphasis",
 366     .p.description = NULL_IF_CONFIG_SMALL("Audio emphasis."),
 367     .p.priv_class  = &aemphasis_class,
 368     .p.flags       = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
 369                      AVFILTER_FLAG_SLICE_THREADS,
 370     .priv_size     = sizeof(AudioEmphasisContext),
 371     .uninit        = uninit,
 372     FILTER_INPUTS(avfilter_af_aemphasis_inputs),
 373     FILTER_OUTPUTS(ff_audio_default_filterpad),
 374     FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
 375     .process_command = process_command,
 376 };