mirror of https://github.com/FFmpeg/FFmpeg.git
Paul B Mahol
3 years ago
parent
466441a0d2
commit
996b13fac4
6 changed files with 395 additions and 1 deletions
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/*
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#include <float.h> |
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#include "libavutil/opt.h" |
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#include "avfilter.h" |
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#include "audio.h" |
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#include "formats.h" |
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#include "hermite.h" |
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typedef struct AudioDynamicEqualizerContext { |
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const AVClass *class; |
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double threshold; |
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double dfrequency; |
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double dqfactor; |
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double tfrequency; |
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double tqfactor; |
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double ratio; |
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double range; |
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double makeup; |
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double knee; |
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double slew; |
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double attack; |
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double release; |
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double attack_coef; |
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double release_coef; |
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int mode; |
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AVFrame *state; |
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} AudioDynamicEqualizerContext; |
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static int config_input(AVFilterLink *inlink) |
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{ |
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AVFilterContext *ctx = inlink->dst; |
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AudioDynamicEqualizerContext *s = ctx->priv; |
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s->state = ff_get_audio_buffer(inlink, 8); |
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if (!s->state) |
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return AVERROR(ENOMEM); |
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return 0; |
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} |
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static double get_svf(double in, double *m, double *a, double *b) |
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{ |
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const double v0 = in; |
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const double v3 = v0 - b[1]; |
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const double v1 = a[0] * b[0] + a[1] * v3; |
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const double v2 = b[1] + a[1] * b[0] + a[2] * v3; |
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b[0] = 2. * v1 - b[0]; |
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b[1] = 2. * v2 - b[1]; |
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return m[0] * v0 + m[1] * v1 + m[2] * v2; |
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} |
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static inline double from_dB(double x) |
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{ |
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return exp(0.05 * x * M_LN10); |
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} |
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static inline double to_dB(double x) |
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{ |
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return 20. * log10(x); |
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} |
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static inline double sqr(double x) |
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{ |
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return x * x; |
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} |
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static double get_gain(double in, double srate, double makeup, |
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double aattack, double iratio, double knee, double range, |
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double thresdb, double slewfactor, double *state, |
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double attack_coeff, double release_coeff, double nc) |
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{ |
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double width = (6. * knee) + 0.01; |
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double cdb = 0.; |
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double Lgain = 1.; |
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double Lxg, Lxl, Lyg, Lyl, Ly1; |
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double checkwidth = 0.; |
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double slewwidth = 1.8; |
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int attslew = 0; |
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Lyg = 0.; |
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Lxg = to_dB(fabs(in) + DBL_EPSILON); |
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Lyg = Lxg + (iratio - 1.) * sqr(Lxg - thresdb + width * .5) / (2. * width); |
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checkwidth = 2. * fabs(Lxg - thresdb); |
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if (2. * (Lxg - thresdb) < -width) { |
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Lyg = Lxg; |
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} else if (checkwidth <= width) { |
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Lyg = thresdb + (Lxg - thresdb) * iratio; |
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if (checkwidth <= slewwidth) { |
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if (Lyg >= state[2]) |
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attslew = 1; |
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} |
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} else if (2. * (Lxg-thresdb) > width) { |
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Lyg = thresdb + (Lxg - thresdb) * iratio; |
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} |
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attack_coeff = attslew ? aattack : attack_coeff; |
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Lxl = Lxg - Lyg; |
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Ly1 = fmaxf(Lxl, release_coeff * state[1] +(1. - release_coeff) * Lxl); |
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Lyl = attack_coeff * state[0] + (1. - attack_coeff) * Ly1; |
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cdb = -Lyl; |
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Lgain = from_dB(nc * fmin(cdb - makeup, range)); |
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state[0] = Lyl; |
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state[1] = Ly1; |
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state[2] = Lyg; |
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return Lgain; |
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} |
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typedef struct ThreadData { |
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AVFrame *in, *out; |
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} ThreadData; |
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static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) |
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{ |
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AudioDynamicEqualizerContext *s = ctx->priv; |
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ThreadData *td = arg; |
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AVFrame *in = td->in; |
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AVFrame *out = td->out; |
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const double sample_rate = in->sample_rate; |
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const double makeup = s->makeup; |
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const double iratio = 1. / s->ratio; |
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const double range = s->range; |
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const double dfrequency = fmin(s->dfrequency, sample_rate * 0.5); |
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const double tfrequency = fmin(s->tfrequency, sample_rate * 0.5); |
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const double threshold = log(s->threshold + DBL_EPSILON); |
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const double release = s->release_coef; |
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const double attack = s->attack_coef; |
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const double dqfactor = s->dqfactor; |
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const double tqfactor = s->tqfactor; |
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const double fg = tan(M_PI * tfrequency / sample_rate); |
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const double dg = tan(M_PI * dfrequency / sample_rate); |
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const int start = (in->channels * jobnr) / nb_jobs; |
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const int end = (in->channels * (jobnr+1)) / nb_jobs; |
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const int mode = s->mode; |
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const double knee = s->knee; |
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const double slew = s->slew; |
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const double aattack = exp(-1000. / ((s->attack + 2.0 * (slew - 1.)) * sample_rate)); |
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const double nc = mode == 0 ? 1. : -1.; |
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double da[3], dm[3]; |
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{ |
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double k = 1. / dqfactor; |
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da[0] = 1. / (1. + dg * (dg + k)); |
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da[1] = dg * da[0]; |
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da[2] = dg * da[1]; |
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dm[0] = 0.; |
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dm[1] = 1.; |
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dm[2] = 0.; |
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} |
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for (int ch = start; ch < end; ch++) { |
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const double *src = (const double *)in->extended_data[ch]; |
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double *dst = (double *)out->extended_data[ch]; |
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double *state = (double *)s->state->extended_data[ch]; |
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for (int n = 0; n < out->nb_samples; n++) { |
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double detect, gain, v, listen; |
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double fa[3], fm[3]; |
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detect = listen = get_svf(src[n], dm, da, state); |
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detect = fabs(detect); |
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gain = get_gain(detect, sample_rate, makeup, |
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aattack, iratio, knee, range, threshold, slew, |
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&state[4], attack, release, nc); |
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{ |
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double k = 1. / (tqfactor * gain); |
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fa[0] = 1. / (1. + fg * (fg + k)); |
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fa[1] = fg * fa[0]; |
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fa[2] = fg * fa[1]; |
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fm[0] = 1.; |
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fm[1] = k * (gain * gain - 1.); |
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fm[2] = 0.; |
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} |
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v = get_svf(src[n], fm, fa, &state[2]); |
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v = mode == -1 ? listen : v; |
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dst[n] = ctx->is_disabled ? src[n] : v; |
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} |
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} |
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return 0; |
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} |
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static double get_coef(double x, double sr) |
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{ |
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return exp(-1000. / (x * sr)); |
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} |
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static int filter_frame(AVFilterLink *inlink, AVFrame *in) |
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{ |
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AVFilterContext *ctx = inlink->dst; |
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AVFilterLink *outlink = ctx->outputs[0]; |
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AudioDynamicEqualizerContext *s = ctx->priv; |
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ThreadData td; |
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AVFrame *out; |
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if (av_frame_is_writable(in)) { |
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out = in; |
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} else { |
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out = ff_get_audio_buffer(outlink, in->nb_samples); |
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if (!out) { |
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av_frame_free(&in); |
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return AVERROR(ENOMEM); |
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} |
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av_frame_copy_props(out, in); |
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} |
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s->attack_coef = get_coef(s->attack, in->sample_rate); |
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s->release_coef = get_coef(s->release, in->sample_rate); |
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td.in = in; |
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td.out = out; |
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ff_filter_execute(ctx, filter_channels, &td, NULL, |
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FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx))); |
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if (out != in) |
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av_frame_free(&in); |
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return ff_filter_frame(outlink, out); |
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} |
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static av_cold void uninit(AVFilterContext *ctx) |
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{ |
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AudioDynamicEqualizerContext *s = ctx->priv; |
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av_frame_free(&s->state); |
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} |
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#define OFFSET(x) offsetof(AudioDynamicEqualizerContext, x) |
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM |
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static const AVOption adynamicequalizer_options[] = { |
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{ "threshold", "set detection threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 100, FLAGS }, |
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{ "dfrequency", "set detection frequency", OFFSET(dfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS }, |
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{ "dqfactor", "set detection Q factor", OFFSET(dqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS }, |
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{ "tfrequency", "set target frequency", OFFSET(tfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS }, |
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{ "tqfactor", "set target Q factor", OFFSET(tqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS }, |
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{ "attack", "set attack duration", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 2000, FLAGS }, |
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{ "release", "set release duration", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=200}, 1, 2000, FLAGS }, |
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{ "knee", "set knee factor", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 8, FLAGS }, |
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{ "ratio", "set ratio factor", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 20, FLAGS }, |
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{ "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, FLAGS }, |
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{ "range", "set max gain", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 200, FLAGS }, |
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{ "slew", "set slew factor", OFFSET(slew), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 200, FLAGS }, |
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{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, -1, 1, FLAGS, "mode" }, |
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{ "listen", 0, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, "mode" }, |
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{ "cut", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" }, |
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{ "boost", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" }, |
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{ NULL } |
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}; |
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AVFILTER_DEFINE_CLASS(adynamicequalizer); |
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static const AVFilterPad inputs[] = { |
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{ |
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.name = "default", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.filter_frame = filter_frame, |
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.config_props = config_input, |
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}, |
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}; |
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static const AVFilterPad outputs[] = { |
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{ |
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.name = "default", |
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.type = AVMEDIA_TYPE_AUDIO, |
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}, |
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}; |
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const AVFilter ff_af_adynamicequalizer = { |
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.name = "adynamicequalizer", |
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.description = NULL_IF_CONFIG_SMALL("Apply Dynamic Equalization of input audio."), |
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.priv_size = sizeof(AudioDynamicEqualizerContext), |
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.priv_class = &adynamicequalizer_class, |
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.uninit = uninit, |
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FILTER_INPUTS(inputs), |
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FILTER_OUTPUTS(outputs), |
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FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP), |
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL | |
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AVFILTER_FLAG_SLICE_THREADS, |
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.process_command = ff_filter_process_command, |
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}; |
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