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| #undef ftype |
| #undef SQRT |
| #undef TAN |
| #undef ONE |
| #undef TWO |
| #undef ZERO |
| #undef FMAX |
| #undef FMIN |
| #undef CLIP |
| #undef SAMPLE_FORMAT |
| #undef EPSILON |
| #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 |
| #define EPSILON (1.f / (1 << 22)) |
| #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 |
| #define EPSILON (1.0 / (1LL << 51)) |
| #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] = ONE; |
| 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 attack = s->attack_coef; |
| 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; |
| ftype fa[3], fm[3]; |
|
|
| if (detection < 0) |
| state[5] = threshold; |
|
|
| memcpy(fa, state + 8, sizeof(fa)); |
| memcpy(fm, state + 11, sizeof(fm)); |
|
|
| for (int n = 0; n < out->nb_samples; n++) { |
| ftype detect, gain, v, listen; |
| 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 (mode >= 0) { |
| if (direction == 0 && detect < threshold) { |
| detect = CLIP(ONE + makeup + (threshold - detect) * ratio, ONE, range); |
| if (!mode) |
| detect = ONE / detect; |
| } else if (direction == 1 && detect > threshold) { |
| detect = CLIP(ONE + makeup + (detect - threshold) * ratio, ONE, range); |
| if (!mode) |
| detect = ONE / detect; |
| } else { |
| detect = ONE; |
| } |
|
|
| { |
| ftype delta = detect - state[4]; |
|
|
| if (delta > EPSILON) |
| detect = state[4] + attack * delta; |
| else if (delta < -EPSILON) |
| detect = state[4] + release * delta; |
| } |
| } |
|
|
| if (state[4] != detect) { |
| state[4] = gain = detect; |
|
|
| switch (tftype) { |
| case 0: |
| k = itqfactor / 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; |
| } |
|
|
| memcpy(state + 8, fa, sizeof(fa)); |
| memcpy(state + 11, fm, sizeof(fm)); |
| } |
|
|
| return 0; |
| } |
|
|