#pragma once #include "audio_math.h" #include "rms_detector.h" #include namespace trnr { struct hp_filter { float a0, a1, b1; float z1; // filter state }; inline void hp_filter_init(hp_filter& f, float samplerate) { const float cutoff = 100.0f; float w0 = 2.0f * 3.14159265359f * cutoff / samplerate; float alpha = (1.0f - std::tan(w0 / 2.0f)) / (1.0f + std::tan(w0 / 2.0f)); f.a0 = 0.5f * (1.0f + alpha); f.a1 = -0.5f * (1.0f + alpha); f.b1 = alpha; f.z1 = 0.0f; } inline float hp_filter_process(hp_filter& f, float x) { float y = f.a0 * x + f.a1 * f.z1 - f.b1 * f.z1; f.z1 = x; return y; } struct oneknob_comp { // params float amount; // state rms_detector detector; hp_filter filter; float attack_coef; float release_coef; float envelope_level; float sidechain_in; }; inline void oneknob_init(oneknob_comp& c, float samplerate, float window_ms) { rms_init(c.detector, samplerate, window_ms); hp_filter_init(c.filter, samplerate); const float attack_ms = 0.2f; const float release_ms = 150.f; // c.amount = 0.f; c.attack_coef = expf(-1.0f / (attack_ms * 1e-6 * samplerate)); c.release_coef = expf(-1.0f / (release_ms * 1e-3 * samplerate)); c.envelope_level = -60.f; c.sidechain_in = 0.f; } template inline void oneknob_process_block(oneknob_comp& c, sample** audio, int frames) { const float min_user_ratio = 1.0f; const float max_user_ratio = 20.0f; const float threshold_db = -9.f; const float amount = fmaxf(0.0f, fminf(powf(c.amount, 2.f), 1.0f)); // clamp to [0, 1] float ratio = min_user_ratio + amount * (max_user_ratio - min_user_ratio); for (int i = 0; i < frames; ++i) { float rms_value = rms_process(c.detector, c.sidechain_in); float envelope_in = lin_2_db(fmaxf(fabs(rms_value), 1e-20f)); // attack if (envelope_in > c.envelope_level) { c.envelope_level = envelope_in + c.attack_coef * (c.envelope_level - envelope_in); } // release else { c.envelope_level = envelope_in + c.release_coef * (c.envelope_level - envelope_in); } float x = c.envelope_level; float y; if (x < threshold_db) y = x; else y = threshold_db + (x - threshold_db) / ratio; float gain_reduction_db = y - x; float gain_reduction_lin = db_2_lin(gain_reduction_db); audio[0][i] *= gain_reduction_lin; audio[1][i] *= gain_reduction_lin; // feedback compression float sum = sqrtf(0.5f * (audio[0][i] * audio[0][i] + audio[1][i] * audio[1][i])); c.sidechain_in = hp_filter_process(c.filter, sum); } } } // namespace trnr