Hook up the reset meters button.

[nageru] / mixer.cpp

diff --git a/mixer.cpp b/mixer.cpp
index d1e50f59c112197583467f26d452cc5747af6f6f..a91b3d8cc843c556dd376946eb05be9988e78e3a 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);
 
@@ -544,24 +546,24 @@ 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, 150.0 * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
+       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: 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);
-       last_gain_staging_db = 20.0 * log10(level_compressor.get_attenuation() * 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",
@@ -570,8 +572,34 @@ void Mixer::process_audio_one_frame()
                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));
+       peak = max<float>(peak, find_peak(samples_out));
        vector<float> left, right;
        deinterleave_samples(samples_out, &left, &right);
        float *ptrs[] = { left.data(), right.data() };
@@ -666,6 +694,13 @@ void Mixer::channel_clicked(int preview_num)
        theme->channel_clicked(preview_num);
 }
 
+void Mixer::reset_meters()
+{
+       peak = 0.0f;
+       r128.reset();
+       r128.integr_start();
+}
+
 Mixer::OutputChannel::~OutputChannel()
 {
        if (has_current_frame) {