+ // 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);
+ }
+
+ double m = final_makeup_gain;
+ for (size_t i = 0; i < samples_out.size(); i += 2) {
+ samples_out[i + 0] *= m;
+ samples_out[i + 1] *= m;
+ m += (target_loudness_factor - m) * alpha;
+ }
+ final_makeup_gain = m;
+ }
+