mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-07 11:13:41 +02:00
439 lines
13 KiB
C
439 lines
13 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 FABS
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#undef FLOG
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#undef FEXP
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#undef FLOG2
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#undef FLOG10
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#undef FEXP2
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#undef FEXP10
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#undef EPSILON
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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 FLOG logf
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#define FEXP expf
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#define FLOG2 log2f
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#define FLOG10 log10f
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#define FEXP2 exp2f
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#define FEXP10 ff_exp10f
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#define EPSILON (1.f / (1 << 23))
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#define ftype float
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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 FLOG log
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#define FEXP exp
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#define FLOG2 log2
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#define FLOG10 log10
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#define FEXP2 exp2
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#define FEXP10 ff_exp10
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#define EPSILON (1.0 / (1LL << 53))
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#define ftype double
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#endif
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#define LIN2LOG(x) (20.0 * FLOG10(x))
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#define LOG2LIN(x) (FEXP10(x / 20.0))
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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->threshold_log = LIN2LOG(s->threshold);
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s->dattack_coef = get_coef(s->dattack, sample_rate);
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s->drelease_coef = get_coef(s->drelease, sample_rate);
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s->gattack_coef = s->dattack_coef * 0.25;
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s->grelease_coef = s->drelease_coef * 0.25;
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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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#define PEAKS(empty_value,op,sample, psample)\
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if (!empty && psample == ss[front]) { \
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ss[front] = empty_value; \
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if (back != front) { \
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front--; \
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if (front < 0) \
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front = n - 1; \
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} \
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empty = front == back; \
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} \
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\
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if (!empty && sample op ss[front]) { \
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while (1) { \
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ss[front] = empty_value; \
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if (back == front) { \
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empty = 1; \
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break; \
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} \
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front--; \
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if (front < 0) \
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front = n - 1; \
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} \
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} \
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\
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while (!empty && sample op ss[back]) { \
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ss[back] = empty_value; \
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if (back == front) { \
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empty = 1; \
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break; \
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} \
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back++; \
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if (back >= n) \
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back = 0; \
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} \
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\
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if (!empty) { \
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back--; \
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if (back < 0) \
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back = n - 1; \
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}
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static void fn(queue_sample)(ChannelContext *cc,
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const ftype x,
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const int nb_samples)
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{
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ftype *ss = cc->dqueue;
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ftype *qq = cc->queue;
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int front = cc->front;
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int back = cc->back;
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int empty, n, pos = cc->position;
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ftype px = qq[pos];
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fn(cc->sum) += x;
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fn(cc->log_sum) += FLOG2(x);
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if (cc->size >= nb_samples) {
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fn(cc->sum) -= px;
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fn(cc->log_sum) -= FLOG2(px);
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}
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qq[pos] = x;
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pos++;
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if (pos >= nb_samples)
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pos = 0;
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cc->position = pos;
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if (cc->size < nb_samples)
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cc->size++;
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n = cc->size;
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empty = (front == back) && (ss[front] == ZERO);
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PEAKS(ZERO, >, x, px)
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ss[back] = x;
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cc->front = front;
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cc->back = back;
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}
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static ftype fn(get_peak)(ChannelContext *cc, ftype *score)
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{
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ftype s, *ss = cc->dqueue;
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s = FEXP2(fn(cc->log_sum) / cc->size) / (fn(cc->sum) / cc->size);
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*score = LIN2LOG(s);
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return ss[cc->front];
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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 int isample_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 ftype power = (mode == CUT_BELOW || mode == CUT_ABOVE) ? -ONE : ONE;
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const ftype grelease = s->grelease_coef;
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const ftype gattack = s->gattack_coef;
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const ftype drelease = s->drelease_coef;
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const ftype dattack = s->dattack_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 == DET_ON) {
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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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ftype new_threshold = ZERO;
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if (cc->detection != detection) {
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cc->detection = detection;
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fn(cc->new_threshold_log) = LIN2LOG(EPSILON);
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}
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for (int n = 0; n < in->nb_samples; n++) {
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ftype detect = FABS(fn(get_svf)(src[n], dm, da, tstate));
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new_threshold = FMAX(new_threshold, detect);
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}
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fn(cc->new_threshold_log) = FMAX(fn(cc->new_threshold_log), LIN2LOG(new_threshold));
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}
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} else if (detection == DET_ADAPTIVE) {
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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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ftype score, peak;
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for (int n = 0; n < in->nb_samples; n++) {
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ftype detect = FMAX(FABS(fn(get_svf)(src[n], dm, da, tstate)), EPSILON);
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fn(queue_sample)(cc, detect, isample_rate);
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}
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peak = fn(get_peak)(cc, &score);
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if (score >= -3.5) {
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fn(cc->threshold_log) = LIN2LOG(peak);
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} else if (cc->detection == DET_UNSET) {
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fn(cc->threshold_log) = s->threshold_log;
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}
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cc->detection = detection;
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}
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} else if (detection == DET_DISABLED) {
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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_log) = s->threshold_log;
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cc->detection = detection;
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}
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} else if (detection == DET_OFF) {
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for (int ch = start; ch < end; ch++) {
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ChannelContext *cc = &s->cc[ch];
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if (cc->detection == DET_ON)
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fn(cc->threshold_log) = fn(cc->new_threshold_log);
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else if (cc->detection == DET_UNSET)
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fn(cc->threshold_log) = s->threshold_log;
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cc->detection = detection;
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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_log = fn(cc->threshold_log);
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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 detect = fn(cc->detect);
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ftype lin_gain = fn(cc->lin_gain);
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int init = cc->init;
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for (int n = 0; n < out->nb_samples; n++) {
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ftype new_detect, new_lin_gain = ONE;
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ftype f, v, listen, k, g, ld;
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listen = fn(get_svf)(src[n], dm, da, dstate);
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if (mode > LISTEN) {
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new_detect = FABS(listen);
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f = (new_detect > detect) * dattack + (new_detect <= detect) * drelease;
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detect = f * new_detect + (ONE - f) * detect;
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}
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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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ld = LIN2LOG(detect);
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if (ld < threshold_log) {
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ftype new_log_gain = CLIP(makeup + (threshold_log - ld) * ratio, ZERO, range) * power;
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new_lin_gain = LOG2LIN(new_log_gain);
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}
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break;
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case CUT_ABOVE:
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case BOOST_ABOVE:
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ld = LIN2LOG(detect);
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if (ld > threshold_log) {
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ftype new_log_gain = CLIP(makeup + (ld - threshold_log) * ratio, ZERO, range) * power;
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new_lin_gain = LOG2LIN(new_log_gain);
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}
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break;
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}
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f = (new_lin_gain > lin_gain) * gattack + (new_lin_gain <= lin_gain) * grelease;
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new_lin_gain = f * new_lin_gain + (ONE - f) * lin_gain;
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if (lin_gain != new_lin_gain || !init) {
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init = 1;
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lin_gain = new_lin_gain;
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switch (tftype) {
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case 0:
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k = itqfactor / lin_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 * (lin_gain * lin_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(lin_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 * (lin_gain - ONE);
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fm[2] = lin_gain * lin_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(lin_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] = lin_gain * lin_gain;
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fm[1] = k * (ONE - lin_gain) * lin_gain;
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fm[2] = ONE - lin_gain * lin_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 == LISTEN ? listen : v;
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dst[n] = is_disabled ? src[n] : v;
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}
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fn(cc->detect) = detect;
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fn(cc->lin_gain) = lin_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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