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);
// 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
// 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 = 0.5f;
- float release_time = 20.0f;
- float makeup_gain = pow(10.0f, 26.0f / 20.0f); // +26 dB takes us to -14 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",
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; // +6 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() };
theme->channel_clicked(preview_num);
}
+void Mixer::reset_meters()
+{
+ peak = 0.0f;
+ r128.reset();
+ r128.integr_start();
+}
+
Mixer::OutputChannel::~OutputChannel()
{
if (has_current_frame) {