mirror of
https://github.com/FFmpeg/FFmpeg.git
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283 lines
8.0 KiB
C
283 lines
8.0 KiB
C
/*
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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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#undef ftype
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#undef SQRT
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#undef TAN
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#undef ONE
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#undef TWO
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#undef ZERO
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#undef FMAX
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#undef FMIN
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#undef CLIP
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#undef SAMPLE_FORMAT
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#undef EPSILON
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#undef FABS
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#if DEPTH == 32
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#define SAMPLE_FORMAT float
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#define SQRT sqrtf
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#define TAN tanf
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#define ONE 1.f
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#define TWO 2.f
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#define ZERO 0.f
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#define FMIN fminf
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#define FMAX fmaxf
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#define CLIP av_clipf
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#define FABS fabsf
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#define ftype float
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#define EPSILON (1.f / (1 << 22))
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#else
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#define SAMPLE_FORMAT double
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#define SQRT sqrt
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#define TAN tan
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#define ONE 1.0
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#define TWO 2.0
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#define ZERO 0.0
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#define FMIN fmin
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#define FMAX fmax
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#define CLIP av_clipd
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#define FABS fabs
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#define ftype double
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#define EPSILON (1.0 / (1LL << 51))
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#endif
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#define fn3(a,b) a##_##b
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#define fn2(a,b) fn3(a,b)
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#define fn(a) fn2(a, SAMPLE_FORMAT)
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static ftype fn(get_svf)(ftype in, const ftype *m, const ftype *a, ftype *b)
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{
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const ftype v0 = in;
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const ftype v3 = v0 - b[1];
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const ftype v1 = a[0] * b[0] + a[1] * v3;
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const ftype v2 = b[1] + a[1] * b[0] + a[2] * v3;
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b[0] = TWO * v1 - b[0];
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b[1] = TWO * 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 int fn(filter_prepare)(AVFilterContext *ctx)
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{
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AudioDynamicEqualizerContext *s = ctx->priv;
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const ftype sample_rate = ctx->inputs[0]->sample_rate;
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const ftype dfrequency = FMIN(s->dfrequency, sample_rate * 0.5);
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const ftype dg = TAN(M_PI * dfrequency / sample_rate);
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const ftype dqfactor = s->dqfactor;
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const int dftype = s->dftype;
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ftype *da = fn(s->da);
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ftype *dm = fn(s->dm);
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ftype k;
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s->attack_coef = get_coef(s->attack, sample_rate);
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s->release_coef = get_coef(s->release, sample_rate);
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switch (dftype) {
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case 0:
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k = ONE / dqfactor;
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da[0] = ONE / (ONE + 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] = ZERO;
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dm[1] = k;
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dm[2] = ZERO;
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break;
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case 1:
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k = ONE / dqfactor;
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da[0] = ONE / (ONE + 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] = ZERO;
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dm[1] = ZERO;
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dm[2] = ONE;
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break;
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case 2:
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k = ONE / dqfactor;
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da[0] = ONE / (ONE + 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] = ZERO;
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dm[1] = -k;
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dm[2] = -ONE;
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break;
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case 3:
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k = ONE / dqfactor;
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da[0] = ONE / (ONE + 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] = ONE;
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dm[1] = -k;
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dm[2] = -TWO;
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break;
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}
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return 0;
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}
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static int fn(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 ftype sample_rate = in->sample_rate;
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const ftype makeup = s->makeup;
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const ftype ratio = s->ratio;
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const ftype range = s->range;
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const ftype tfrequency = FMIN(s->tfrequency, sample_rate * 0.5);
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const int mode = s->mode;
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const int power = (mode == CUT_BELOW || mode == CUT_ABOVE) ? -1 : 1;
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const ftype release = s->release_coef;
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const ftype attack = s->attack_coef;
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const ftype tqfactor = s->tqfactor;
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const ftype itqfactor = ONE / tqfactor;
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const ftype fg = TAN(M_PI * tfrequency / sample_rate);
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const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
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const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
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const int is_disabled = ctx->is_disabled;
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const int detection = s->detection;
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const int tftype = s->tftype;
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const ftype *da = fn(s->da);
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const ftype *dm = fn(s->dm);
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if (detection > 0) {
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for (int ch = start; ch < end; ch++) {
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const ftype *src = (const ftype *)in->extended_data[ch];
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ChannelContext *cc = &s->cc[ch];
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ftype *tstate = fn(cc->tstate);
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for (int n = 0; n < in->nb_samples; n++) {
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ftype detect = fn(get_svf)(src[n], dm, da, tstate);
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fn(cc->threshold) = FMAX(fn(cc->threshold), detect);
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}
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}
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} else if (detection < 0) {
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for (int ch = start; ch < end; ch++) {
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ChannelContext *cc = &s->cc[ch];
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fn(cc->threshold) = s->threshold;
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}
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}
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for (int ch = start; ch < end; ch++) {
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const ftype *src = (const ftype *)in->extended_data[ch];
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ftype *dst = (ftype *)out->extended_data[ch];
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ChannelContext *cc = &s->cc[ch];
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const ftype threshold = fn(cc->threshold);
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ftype *fa = fn(cc->fa), *fm = fn(cc->fm);
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ftype *fstate = fn(cc->fstate);
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ftype *dstate = fn(cc->dstate);
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ftype gain = fn(cc->gain);
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const int init = cc->init;
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for (int n = 0; n < out->nb_samples; n++) {
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ftype detect, v, listen, new_gain = ONE;
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ftype k, g;
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detect = listen = fn(get_svf)(src[n], dm, da, dstate);
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detect = FABS(detect);
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switch (mode) {
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case LISTEN:
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break;
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case CUT_BELOW:
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case BOOST_BELOW:
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if (detect < threshold)
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new_gain = CLIP(ONE + makeup + (threshold - detect) * ratio, ONE, range);
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break;
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case CUT_ABOVE:
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case BOOST_ABOVE:
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if (detect > threshold)
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new_gain = CLIP(ONE + makeup + (detect - threshold) * ratio, ONE, range);
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break;
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}
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if (power < 0)
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new_gain = ONE / new_gain;
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if (mode > LISTEN) {
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ftype delta = new_gain - gain;
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if (delta > EPSILON)
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new_gain = gain + attack * delta;
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else if (delta < -EPSILON)
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new_gain = gain + release * delta;
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}
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if (gain != new_gain || !init) {
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gain = new_gain;
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switch (tftype) {
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case 0:
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k = itqfactor / gain;
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fa[0] = ONE / (ONE + 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] = ONE;
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fm[1] = k * (gain * gain - ONE);
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fm[2] = ZERO;
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break;
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case 1:
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k = itqfactor;
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g = fg / SQRT(gain);
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fa[0] = ONE / (ONE + g * (g + k));
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fa[1] = g * fa[0];
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fa[2] = g * fa[1];
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fm[0] = ONE;
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fm[1] = k * (gain - ONE);
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fm[2] = gain * gain - ONE;
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break;
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case 2:
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k = itqfactor;
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g = fg * SQRT(gain);
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fa[0] = ONE / (ONE + g * (g + k));
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fa[1] = g * fa[0];
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fa[2] = g * fa[1];
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fm[0] = gain * gain;
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fm[1] = k * (ONE - gain) * gain;
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fm[2] = ONE - gain * gain;
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break;
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}
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}
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v = fn(get_svf)(src[n], fm, fa, fstate);
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v = mode == -1 ? listen : v;
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dst[n] = is_disabled ? src[n] : v;
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}
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fn(cc->gain) = gain;
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cc->init = 1;
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}
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return 0;
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}
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