for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
locut[bus_index].init(FILTER_HPF, 2);
locut_enabled[bus_index] = global_flags.locut_enabled;
+ eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
+ // Note: EQ_BAND_MID isn't used (see comments in apply_eq()).
+ eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1);
+
gain_staging_db[bus_index] = global_flags.initial_gain_staging_db;
compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
compressor_threshold_dbfs[bus_index] = ref_level_dbfs - 12.0f; // -12 dB.
samples_bus.resize(num_samples * 2);
for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
-
- // Cut away everything under 120 Hz (or whatever the cutoff is);
- // 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.)
- if (locut_enabled[bus_index]) {
- locut[bus_index].render(samples_bus.data(), samples_bus.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
- }
+ apply_eq(bus_index, &samples_bus);
{
lock_guard<mutex> lock(compressor_mutex);
}
}
- float volume = from_db(fader_volume_db[bus_index]);
- if (bus_index == 0) {
- for (unsigned i = 0; i < num_samples * 2; ++i) {
- samples_out[i] = samples_bus[i] * volume;
- }
- } else {
- for (unsigned i = 0; i < num_samples * 2; ++i) {
- samples_out[i] += samples_bus[i] * volume;
- }
- }
-
+ add_bus_to_master(bus_index, samples_bus, &samples_out);
deinterleave_samples(samples_bus, &left, &right);
- measure_bus_levels(bus_index, left, right, volume);
+ measure_bus_levels(bus_index, left, right);
}
{
return samples_out;
}
-void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right, float volume)
+void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
+{
+ constexpr float bass_freq_hz = 200.0f;
+ constexpr float treble_freq_hz = 4700.0f;
+
+ // Cut away everything under 120 Hz (or whatever the cutoff is);
+ // 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.)
+ if (locut_enabled[bus_index]) {
+ locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
+ }
+
+ // Apply the rest of the EQ. Since we only have a simple three-band EQ,
+ // we can implement it with two shelf filters. We use a simple gain to
+ // set the mid-level filter, and then offset the low and high bands
+ // from that if we need to. (We could perhaps have folded the gain into
+ // the next part, but it's so cheap that the trouble isn't worth it.)
+ if (fabs(eq_level_db[bus_index][EQ_BAND_MID]) > 0.01f) {
+ float g = from_db(eq_level_db[bus_index][EQ_BAND_MID]);
+ for (size_t i = 0; i < samples_bus->size(); ++i) {
+ (*samples_bus)[i] *= g;
+ }
+ }
+
+ float bass_adj_db = eq_level_db[bus_index][EQ_BAND_BASS] - eq_level_db[bus_index][EQ_BAND_MID];
+ if (fabs(bass_adj_db) > 0.01f) {
+ eq[bus_index][EQ_BAND_BASS].render(samples_bus->data(), samples_bus->size() / 2,
+ bass_freq_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f, bass_adj_db / 40.0f);
+ }
+
+ float treble_adj_db = eq_level_db[bus_index][EQ_BAND_TREBLE] - eq_level_db[bus_index][EQ_BAND_MID];
+ if (fabs(treble_adj_db) > 0.01f) {
+ eq[bus_index][EQ_BAND_TREBLE].render(samples_bus->data(), samples_bus->size() / 2,
+ treble_freq_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f, treble_adj_db / 40.0f);
+ }
+}
+
+void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
+{
+ assert(samples_bus.size() == samples_out->size());
+ assert(samples_bus.size() % 2 == 0);
+ unsigned num_samples = samples_bus.size() / 2;
+ if (fabs(fader_volume_db[bus_index] - last_fader_volume_db[bus_index]) > 1e-3) {
+ // The volume has changed; do a fade over the course of this frame.
+ // (We might have some numerical issues here, but it seems to sound OK.)
+ // For the purpose of fading here, the silence floor is set to -90 dB
+ // (the fader only goes to -84).
+ float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
+ float volume = from_db(max<float>(fader_volume_db[bus_index], -90.0f));
+
+ float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
+ volume = old_volume;
+ if (bus_index == 0) {
+ for (unsigned i = 0; i < num_samples; ++i) {
+ (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
+ (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
+ volume *= volume_inc;
+ }
+ } else {
+ for (unsigned i = 0; i < num_samples; ++i) {
+ (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
+ (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
+ volume *= volume_inc;
+ }
+ }
+ } else {
+ float volume = from_db(fader_volume_db[bus_index]);
+ if (bus_index == 0) {
+ for (unsigned i = 0; i < num_samples; ++i) {
+ (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
+ (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
+ }
+ } else {
+ for (unsigned i = 0; i < num_samples; ++i) {
+ (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
+ (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
+ }
+ }
+ }
+
+ last_fader_volume_db[bus_index] = fader_volume_db[bus_index];
+}
+
+void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
{
assert(left.size() == right.size());
+ const float volume = from_db(fader_volume_db[bus_index]);
const float peak_levels[2] = {
find_peak(left.data(), left.size()) * volume,
find_peak(right.data(), right.size()) * volume
static constexpr float falloff_db_sec = 15.0f; // dB/sec falloff after hold.
float current_peak;
PeakHistory &history = peak_history[bus_index][channel];
+ history.historic_peak = max(history.historic_peak, peak_levels[channel]);
if (history.age_seconds < hold_sec) {
current_peak = history.last_peak;
} else {
bus_levels[bus_index].current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
bus_levels[bus_index].peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
bus_levels[bus_index].peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
+ bus_levels[bus_index].historic_peak_dbfs = to_db(
+ max(peak_history[bus_index][0].historic_peak,
+ peak_history[bus_index][1].historic_peak));
bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index];
if (compressor_enabled[bus_index]) {
bus_levels[bus_index].compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
lock_guard<timed_mutex> lock(audio_mutex);
return input_mapping;
}
+
+void AudioMixer::reset_peak(unsigned bus_index)
+{
+ lock_guard<timed_mutex> lock(audio_mutex);
+ for (unsigned channel = 0; channel < 2; ++channel) {
+ PeakHistory &history = peak_history[bus_index][channel];
+ history.current_level = 0.0f;
+ history.historic_peak = 0.0f;
+ history.current_peak = 0.0f;
+ history.last_peak = 0.0f;
+ history.age_seconds = 0.0f;
+ }
+}
projects / nageru / blobdiff