-// float limiter_att, compressor_att;
-
- // The real compressor.
- if (compressor_enabled) {
- float threshold = pow(10.0f, compressor_threshold_dbfs / 20.0f);
- float ratio = 20.0f;
- float attack_time = 0.005f;
- float release_time = 0.040f;
- float makeup_gain = 2.0f; // +6 dB.
- compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
-// compressor_att = compressor.get_attenuation();
- }
-
- // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
- // Note that since ratio is not infinite, we could go slightly higher than this.
- if (limiter_enabled) {
- float threshold = pow(10.0f, limiter_threshold_dbfs / 20.0f);
- float ratio = 30.0f;
- float attack_time = 0.0f; // Instant.
- float release_time = 0.020f;
- float makeup_gain = 1.0f; // 0 dB.
- limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
-// limiter_att = limiter.get_attenuation();
- }
-
-// printf("limiter=%+5.1f compressor=%+5.1f\n", 20.0*log10(limiter_att), 20.0*log10(compressor_att));
-
- // Upsample 4x to find interpolated peak.
- peak_resampler.inp_data = samples_out.data();
- peak_resampler.inp_count = samples_out.size() / 2;
-
- vector<float> interpolated_samples_out;
- interpolated_samples_out.resize(samples_out.size());
- while (peak_resampler.inp_count > 0) { // About four iterations.
- peak_resampler.out_data = &interpolated_samples_out[0];
- peak_resampler.out_count = interpolated_samples_out.size() / 2;
- peak_resampler.process();
- size_t out_stereo_samples = interpolated_samples_out.size() / 2 - peak_resampler.out_count;
- peak = max<float>(peak, find_peak(interpolated_samples_out.data(), out_stereo_samples * 2));
- peak_resampler.out_data = nullptr;
- }
-
- // At this point, we are most likely close to +0 LU, but all of our
- // measurements have been on raw sample values, not R128 values.
- // So we have a final makeup gain to get us to +0 LU; the gain
- // adjustments required should be relatively small, and also, the
- // offset shouldn't change much (only if the type of audio changes
- // significantly). Thus, we shoot for updating this value basically
- // “whenever we process buffers”, since the R128 calculation isn't exactly
- // something we get out per-sample.
- //
- // Note that there's a feedback loop here, so we choose a very slow filter
- // (half-time of 100 seconds).
- double target_loudness_factor, alpha;
- {
- unique_lock<mutex> lock(compressor_mutex);
- double loudness_lu = r128.loudness_M() - ref_level_lufs;
- double current_makeup_lu = 20.0f * log10(final_makeup_gain);
- target_loudness_factor = pow(10.0f, -loudness_lu / 20.0f);
-
- // If we're outside +/- 5 LU uncorrected, we don't count it as
- // a normal signal (probably silence) and don't change the
- // correction factor; just apply what we already have.
- if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
- alpha = 0.0;
- } else {
- // Formula adapted from
- // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
- const double half_time_s = 100.0;
- const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
- alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);