Include the limiter and the main compressor. Still needs some tuning, though.

[nageru] / mixer.cpp

diff --git a/mixer.cpp b/mixer.cpp
index d90359265b9cec2b722cbc5b141d9ffb1f943e8c..656cdfaf39feb9ef67b79464fffacde678c2e5c1 100644 (file)
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -70,7 +70,9 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
          num_cards(num_cards),
          mixer_surface(create_surface(format)),
          h264_encoder_surface(create_surface(format)),
-         level_compressor(OUTPUT_FREQUENCY)
+         level_compressor(OUTPUT_FREQUENCY),
+         limiter(OUTPUT_FREQUENCY),
+         compressor(OUTPUT_FREQUENCY)
 {
        httpd.start(9095);
 
@@ -149,6 +151,8 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
 
        r128.init(2, OUTPUT_FREQUENCY);
        r128.integr_start();
+
+       locut.init(FILTER_HPF, 2);
 }
 
 Mixer::~Mixer()
@@ -382,7 +386,8 @@ void Mixer::thread_func()
                        double loudness_range_high = r128.range_max();
 
                        audio_level_callback(loudness_s, 20.0 * log10(peak),
-                                            loudness_i, loudness_range_low, loudness_range_high);
+                                            loudness_i, loudness_range_low, loudness_range_high,
+                                            last_gain_staging_db);
                }
 
                for (unsigned card_index = 1; card_index < num_cards; ++card_index) {
@@ -538,33 +543,61 @@ void Mixer::process_audio_one_frame()
                }
        }
 
+       // Cut away everything under 150 Hz; we don't need it for voice,
+       // and it will reduce headroom and confuse the compressor.
+       // (In particular, any hums at 50 or 60 Hz should be dampened.)
+       locut.render(samples_out.data(), samples_out.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
+
        // Apply a level compressor to get the general level right.
        // Basically, if it's over about -40 dBFS, we squeeze it down to that level
        // (or more precisely, near it, since we don't use infinite ratio),
-       // then apply a makeup gain to get it to -12 dBFS. -12 dBFS is, of course,
+       // 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.
-       //
-       // TODO: Hook this up to a UI, so we can see the effects, and/or turn it off
-       // to control the gain manually instead. For now, there's only the #if-ed out
-       // code below.
-       //
-       // TODO: Add the actual compressors/limiters (for taking care of transients)
-       // later in the chain.
-       float threshold = 0.01f;   // -40 dBFS.
-       float ratio = 20.0f;
-       float attack_time = 0.1f;
-       float release_time = 10.0f;
-       float makeup_gain = pow(10.0f, 28.0f / 20.0f);  // +28 dB takes us to -12 dBFS.
-       level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
+       float ref_level_dbfs = -14.0f;
+       {
+               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);
+       }
 
 #if 0
        printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
-               compressor.get_level(), 20.0 * log10(compressor.get_level()),
-               compressor.get_attenuation(), 20.0 * log10(compressor.get_attenuation()),
-               20.0 * log10(compressor.get_level() * compressor.get_attenuation() * makeup_gain));
+               level_compressor.get_level(), 20.0 * log10(level_compressor.get_level()),
+               level_compressor.get_attenuation(), 20.0 * log10(level_compressor.get_attenuation()),
+               20.0 * log10(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
 #endif
 
+//     float limiter_att, compressor_att;
+
+       // Then a limiter at +0 dB (so, -14 dBFS) to take out the worst peaks only.
+       {
+               float threshold = pow(10.0f, (ref_level_dbfs + 0.0f) / 20.0f);  // +0 dB.
+               float ratio = 1000.0f;  // Infinity.
+               float attack_time = 0.001f;
+               float release_time = 0.005f;
+               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();
+       }
+
+       // Finally, the real compressor.
+       {
+               float threshold = pow(10.0f, (ref_level_dbfs - 12.0f) / 20.0f);  // -12 dB.
+               float ratio = 20.0f;
+               float attack_time = 0.005f;
+               float release_time = 0.040f;
+               float makeup_gain = 2.0f;  // +3 dB.
+               compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
+//             compressor_att = compressor.get_attenuation();
+       }
+
+//     printf("limiter=%+5.1f  compressor=%+5.1f\n", 20.0*log10(limiter_att), 20.0*log10(compressor_att));
+
        // Find peak and R128 levels.
        peak = std::max(peak, find_peak(samples_out));
        vector<float> left, right;