avfilter/af_adynamicequalizer: add precision option

1 files changed, 270 insertions, 0 deletions

diff --git a/libavfilter/adynamicequalizer_template.c b/libavfilter/adynamicequalizer_template.c
new file mode 100644
index 0000000000..ab180a9b9d
--- /dev/null
+++ b/libavfilter/adynamicequalizer_template.c
@@ -0,0 +1,270 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#undef ftype
+#undef SQRT
+#undef TAN
+#undef ONE
+#undef TWO
+#undef ZERO
+#undef FMAX
+#undef FMIN
+#undef CLIP
+#undef SAMPLE_FORMAT
+#undef FABS
+#if DEPTH == 32
+#define SAMPLE_FORMAT float
+#define SQRT sqrtf
+#define TAN tanf
+#define ONE 1.f
+#define TWO 2.f
+#define ZERO 0.f
+#define FMIN fminf
+#define FMAX fmaxf
+#define CLIP av_clipf
+#define FABS fabsf
+#define ftype float
+#else
+#define SAMPLE_FORMAT double
+#define SQRT sqrt
+#define TAN tan
+#define ONE 1.0
+#define TWO 2.0
+#define ZERO 0.0
+#define FMIN fmin
+#define FMAX fmax
+#define CLIP av_clipd
+#define FABS fabs
+#define ftype double
+#endif
+
+#define fn3(a,b)   a##_##b
+#define fn2(a,b)   fn3(a,b)
+#define fn(a)      fn2(a, SAMPLE_FORMAT)
+
+static ftype fn(get_svf)(ftype in, const ftype *m, const ftype *a, ftype *b)
+{
+    const ftype v0 = in;
+    const ftype v3 = v0 - b[1];
+    const ftype v1 = a[0] * b[0] + a[1] * v3;
+    const ftype v2 = b[1] + a[1] * b[0] + a[2] * v3;
+
+    b[0] = TWO * v1 - b[0];
+    b[1] = TWO * v2 - b[1];
+
+    return m[0] * v0 + m[1] * v1 + m[2] * v2;
+}
+
+static int fn(filter_prepare)(AVFilterContext *ctx)
+{
+    AudioDynamicEqualizerContext *s = ctx->priv;
+    const ftype sample_rate = ctx->inputs[0]->sample_rate;
+    const ftype dfrequency = FMIN(s->dfrequency, sample_rate * 0.5);
+    const ftype dg = TAN(M_PI * dfrequency / sample_rate);
+    const ftype dqfactor = s->dqfactor;
+    const int dftype = s->dftype;
+    ftype *da = fn(s->da);
+    ftype *dm = fn(s->dm);
+    ftype k;
+
+    s->attack_coef = get_coef(s->attack, sample_rate);
+    s->release_coef = get_coef(s->release, sample_rate);
+
+    switch (dftype) {
+    case 0:
+        k = ONE / dqfactor;
+
+        da[0] = ONE / (ONE + dg * (dg + k));
+        da[1] = dg * da[0];
+        da[2] = dg * da[1];
+
+        dm[0] = ZERO;
+        dm[1] = k;
+        dm[2] = ZERO;
+        break;
+    case 1:
+        k = ONE / dqfactor;
+
+        da[0] = ONE / (ONE + dg * (dg + k));
+        da[1] = dg * da[0];
+        da[2] = dg * da[1];
+
+        dm[0] = ZERO;
+        dm[1] = ZERO;
+        dm[2] = ONE;
+        break;
+    case 2:
+        k = ONE / dqfactor;
+
+        da[0] = ONE / (ONE + dg * (dg + k));
+        da[1] = dg * da[0];
+        da[2] = dg * da[1];
+
+        dm[0] = ZERO;
+        dm[1] = -k;
+        dm[2] = -ONE;
+        break;
+    case 3:
+        k = ONE / dqfactor;
+
+        da[0] = ONE / (ONE + dg * (dg + k));
+        da[1] = dg * da[0];
+        da[2] = dg * da[1];
+
+        dm[0] = ZERO;
+        dm[1] = -k;
+        dm[2] = -TWO;
+        break;
+    }
+
+    return 0;
+}
+
+static int fn(filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+    AudioDynamicEqualizerContext *s = ctx->priv;
+    ThreadData *td = arg;
+    AVFrame *in = td->in;
+    AVFrame *out = td->out;
+    const ftype sample_rate = in->sample_rate;
+    const ftype makeup = s->makeup;
+    const ftype ratio = s->ratio;
+    const ftype range = s->range;
+    const ftype tfrequency = FMIN(s->tfrequency, sample_rate * 0.5);
+    const ftype release = s->release_coef;
+    const ftype irelease = ONE - release;
+    const ftype attack = s->attack_coef;
+    const ftype iattack = ONE - attack;
+    const ftype tqfactor = s->tqfactor;
+    const ftype itqfactor = ONE / tqfactor;
+    const ftype fg = TAN(M_PI * tfrequency / sample_rate);
+    const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
+    const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
+    const int detection = s->detection;
+    const int direction = s->direction;
+    const int tftype = s->tftype;
+    const int mode = s->mode;
+    const ftype *da = fn(s->da);
+    const ftype *dm = fn(s->dm);
+
+    for (int ch = start; ch < end; ch++) {
+        const ftype *src = (const ftype *)in->extended_data[ch];
+        ftype *dst = (ftype *)out->extended_data[ch];
+        ftype *state = (ftype *)s->state->extended_data[ch];
+        const ftype threshold = detection == 0 ? state[5] : s->threshold;
+
+        if (detection < 0)
+            state[5] = threshold;
+
+        for (int n = 0; n < out->nb_samples; n++) {
+            ftype detect, gain, v, listen;
+            ftype fa[3], fm[3];
+            ftype k, g;
+
+            detect = listen = fn(get_svf)(src[n], dm, da, state);
+            detect = FABS(detect);
+
+            if (detection > 0)
+                state[5] = FMAX(state[5], detect);
+
+            if (direction == 0) {
+                if (detect < threshold) {
+                    if (mode == 0)
+                        detect = ONE / CLIP(ONE + makeup + (threshold - detect) * ratio, ONE, range);
+                    else
+                        detect = CLIP(ONE + makeup + (threshold - detect) * ratio, ONE, range);
+                } else {
+                    detect = ONE;
+                }
+            } else {
+                if (detect > threshold) {
+                    if (mode == 0)
+                        detect = ONE / CLIP(ONE + makeup + (detect - threshold) * ratio, ONE, range);
+                    else
+                        detect = CLIP(ONE + makeup + (detect - threshold) * ratio, ONE, range);
+                } else {
+                    detect = ONE;
+                }
+            }
+
+            if (direction == 0) {
+                if (detect > state[4]) {
+                    detect = iattack * detect + attack * state[4];
+                } else {
+                    detect = irelease * detect + release * state[4];
+                }
+            } else {
+                if (detect < state[4]) {
+                    detect = iattack * detect + attack * state[4];
+                } else {
+                    detect = irelease * detect + release * state[4];
+                }
+            }
+
+            if (state[4] != detect || n == 0) {
+                state[4] = gain = detect;
+
+                switch (tftype) {
+                case 0:
+                    k = ONE / (tqfactor * gain);
+
+                    fa[0] = ONE / (ONE + fg * (fg + k));
+                    fa[1] = fg * fa[0];
+                    fa[2] = fg * fa[1];
+
+                    fm[0] = ONE;
+                    fm[1] = k * (gain * gain - ONE);
+                    fm[2] = ZERO;
+                    break;
+                case 1:
+                    k = itqfactor;
+                    g = fg / SQRT(gain);
+
+                    fa[0] = ONE / (ONE + g * (g + k));
+                    fa[1] = g * fa[0];
+                    fa[2] = g * fa[1];
+
+                    fm[0] = ONE;
+                    fm[1] = k * (gain - ONE);
+                    fm[2] = gain * gain - ONE;
+                    break;
+                case 2:
+                    k = itqfactor;
+                    g = fg / SQRT(gain);
+
+                    fa[0] = ONE / (ONE + g * (g + k));
+                    fa[1] = g * fa[0];
+                    fa[2] = g * fa[1];
+
+                    fm[0] = gain * gain;
+                    fm[1] = k * (ONE - gain) * gain;
+                    fm[2] = ONE - gain * gain;
+                    break;
+                }
+            }
+
+            v = fn(get_svf)(src[n], fm, fa, &state[2]);
+            v = mode == -1 ? listen : v;
+            dst[n] = ctx->is_disabled ? src[n] : v;
+        }
+    }
+
+    return 0;
+}
+
+