Make the license blurb read-only.

[nageru] / mixer.cpp

diff --git a/mixer.cpp b/mixer.cpp
index dd8cc6013d72559bb08b89c6d5605953ac6b4269..00b3efef5032ef3ecfb0fd0c2100c506009bde96 100644 (file)
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -184,7 +184,7 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
 
        // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
        // and there's a limit to how important the peak meter is.
-       peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16);
+       peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
 
        alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
 }
@@ -526,7 +526,7 @@ void Mixer::thread_func()
                }
 
                if (audio_level_callback != nullptr) {
-                       unique_lock<mutex> lock(r128_mutex);
+                       unique_lock<mutex> lock(compressor_mutex);
                        double loudness_s = r128.loudness_S();
                        double loudness_i = r128.integrated();
                        double loudness_range_low = r128.range_min();
@@ -534,7 +534,7 @@ void Mixer::thread_func()
 
                        audio_level_callback(loudness_s, 20.0 * log10(peak),
                                             loudness_i, loudness_range_low, loudness_range_high,
-                                            last_gain_staging_db);
+                                            gain_staging_db, 20.0 * log10(final_makeup_gain));
                }
 
                for (unsigned card_index = 1; card_index < num_cards; ++card_index) {
@@ -715,14 +715,23 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
        // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
        // entirely arbitrary, but from practical tests with speech, it seems to
        // put ut around -23 LUFS, so it's a reasonable starting point for later use.
-       if (level_compressor_enabled) {
-               float threshold = 0.01f;   // -40 dBFS.
-               float ratio = 20.0f;
-               float attack_time = 0.5f;
-               float release_time = 20.0f;
-               float makeup_gain = pow(10.0f, (ref_level_dbfs - (-40.0f)) / 20.0f);  // +26 dB.
-               level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
-               last_gain_staging_db = 20.0 * log10(level_compressor.get_attenuation() * makeup_gain);
+       {
+               unique_lock<mutex> lock(compressor_mutex);
+               if (level_compressor_enabled) {
+                       float threshold = 0.01f;   // -40 dBFS.
+                       float ratio = 20.0f;
+                       float attack_time = 0.5f;
+                       float release_time = 20.0f;
+                       float makeup_gain = pow(10.0f, (ref_level_dbfs - (-40.0f)) / 20.0f);  // +26 dB.
+                       level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
+                       gain_staging_db = 20.0 * log10(level_compressor.get_attenuation() * makeup_gain);
+               } else {
+                       // Just apply the gain we already had.
+                       float g = pow(10.0f, gain_staging_db / 20.0f);
+                       for (size_t i = 0; i < samples_out.size(); ++i) {
+                               samples_out[i] *= g;
+                       }
+               }
        }
 
 #if 0
@@ -773,12 +782,52 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
                peak = max<float>(peak, find_peak(interpolated_samples_out.data(), out_stereo_samples * 2));
        }
 
+       // 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;
+       }
+
        // Find R128 levels.
        vector<float> left, right;
        deinterleave_samples(samples_out, &left, &right);
        float *ptrs[] = { left.data(), right.data() };
        {
-               unique_lock<mutex> lock(r128_mutex);
+               unique_lock<mutex> lock(compressor_mutex);
                r128.process(left.size(), ptrs);
        }