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.
- compressor_enabled[bus_index] = global_flags.compressor_enabled;
level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
- level_compressor_enabled[bus_index] = global_flags.gain_staging_auto;
+
+ set_bus_settings(bus_index, get_default_bus_settings());
}
set_limiter_enabled(global_flags.limiter_enabled);
set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
return true;
}
+AudioMixer::BusSettings AudioMixer::get_default_bus_settings()
+{
+ BusSettings settings;
+ settings.fader_volume_db = 0.0f;
+ settings.locut_enabled = global_flags.locut_enabled;
+ for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
+ settings.eq_level_db[band_index] = 0.0f;
+ }
+ settings.gain_staging_db = global_flags.initial_gain_staging_db;
+ settings.level_compressor_enabled = global_flags.gain_staging_auto;
+ settings.compressor_threshold_dbfs = ref_level_dbfs - 12.0f; // -12 dB.
+ settings.compressor_enabled = global_flags.compressor_enabled;
+ return settings;
+}
+
+AudioMixer::BusSettings AudioMixer::get_bus_settings(unsigned bus_index) const
+{
+ lock_guard<timed_mutex> lock(audio_mutex);
+ BusSettings settings;
+ settings.fader_volume_db = fader_volume_db[bus_index];
+ settings.locut_enabled = locut_enabled[bus_index];
+ for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
+ settings.eq_level_db[band_index] = eq_level_db[bus_index][band_index];
+ }
+ settings.gain_staging_db = gain_staging_db[bus_index];
+ settings.level_compressor_enabled = level_compressor_enabled[bus_index];
+ settings.compressor_threshold_dbfs = compressor_threshold_dbfs[bus_index];
+ settings.compressor_enabled = compressor_enabled[bus_index];
+ return settings;
+}
+
+void AudioMixer::set_bus_settings(unsigned bus_index, const AudioMixer::BusSettings &settings)
+{
+ lock_guard<timed_mutex> lock(audio_mutex);
+ fader_volume_db[bus_index] = settings.fader_volume_db;
+ locut_enabled[bus_index] = settings.locut_enabled;
+ for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
+ eq_level_db[bus_index][band_index] = settings.eq_level_db[band_index];
+ }
+ gain_staging_db[bus_index] = settings.gain_staging_db;
+ level_compressor_enabled[bus_index] = settings.level_compressor_enabled;
+ compressor_threshold_dbfs[bus_index] = settings.compressor_threshold_dbfs;
+ compressor_enabled[bus_index] = settings.compressor_enabled;
+}
+
AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
{
switch (device.type) {
return samples_out;
}
+namespace {
+
+void apply_filter_fade(StereoFilter *filter, float *data, unsigned num_samples, float cutoff_hz, float db, float last_db)
+{
+ // A granularity of 32 samples is an okay tradeoff between speed and
+ // smoothness; recalculating the filters is pretty expensive, so it's
+ // good that we don't do this all the time.
+ static constexpr unsigned filter_granularity_samples = 32;
+
+ const float cutoff_linear = cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY;
+ if (fabs(db - last_db) < 1e-3) {
+ // Constant over this frame.
+ if (fabs(db) > 0.01f) {
+ filter->render(data, num_samples, cutoff_linear, 0.5f, db / 40.0f);
+ }
+ } else {
+ // We need to do a fade. (Rounding up avoids division by zero.)
+ unsigned num_blocks = (num_samples + filter_granularity_samples - 1) / filter_granularity_samples;
+ const float inc_db_norm = (db - last_db) / 40.0f / num_blocks;
+ float db_norm = db / 40.0f;
+ for (size_t i = 0; i < num_samples; i += filter_granularity_samples) {
+ size_t samples_this_block = std::min<size_t>(num_samples - i, filter_granularity_samples);
+ filter->render(data + i * 2, samples_this_block, cutoff_linear, 0.5f, db_norm);
+ db_norm += inc_db_norm;
+ }
+ }
+}
+
+} // namespace
+
void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
{
constexpr float bass_freq_hz = 200.0f;
// 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]);
+ //
+ // If any part of the EQ has changed appreciably since last frame,
+ // we fade smoothly during the course of this frame.
+ const float bass_db = eq_level_db[bus_index][EQ_BAND_BASS];
+ const float mid_db = eq_level_db[bus_index][EQ_BAND_MID];
+ const float treble_db = eq_level_db[bus_index][EQ_BAND_TREBLE];
+
+ const float last_bass_db = last_eq_level_db[bus_index][EQ_BAND_BASS];
+ const float last_mid_db = last_eq_level_db[bus_index][EQ_BAND_MID];
+ const float last_treble_db = last_eq_level_db[bus_index][EQ_BAND_TREBLE];
+
+ assert(samples_bus->size() % 2 == 0);
+ const unsigned num_samples = samples_bus->size() / 2;
+
+ if (fabs(mid_db - last_mid_db) < 1e-3) {
+ // Constant over this frame.
+ const float gain = from_db(mid_db);
for (size_t i = 0; i < samples_bus->size(); ++i) {
- (*samples_bus)[i] *= g;
+ (*samples_bus)[i] *= gain;
+ }
+ } else {
+ // We need to do a fade.
+ float gain = from_db(last_mid_db);
+ const float gain_inc = pow(from_db(mid_db - last_mid_db), 1.0 / num_samples);
+ for (size_t i = 0; i < num_samples; ++i) {
+ (*samples_bus)[i * 2 + 0] *= gain;
+ (*samples_bus)[i * 2 + 1] *= gain;
+ gain *= gain_inc;
}
}
- 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);
- }
+ apply_filter_fade(&eq[bus_index][EQ_BAND_BASS], samples_bus->data(), num_samples, bass_freq_hz, bass_db - mid_db, last_bass_db - last_mid_db);
+ apply_filter_fade(&eq[bus_index][EQ_BAND_TREBLE], samples_bus->data(), num_samples, treble_freq_hz, treble_db - mid_db, last_treble_db - last_mid_db);
- 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);
- }
+ last_eq_level_db[bus_index][EQ_BAND_BASS] = bass_db;
+ last_eq_level_db[bus_index][EQ_BAND_MID] = mid_db;
+ last_eq_level_db[bus_index][EQ_BAND_TREBLE] = treble_db;
}
void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)