Implement the EQ in AudioMixer. (No UI yet.)

[nageru] / audio_mixer.cpp

diff --git a/audio_mixer.cpp b/audio_mixer.cpp
index 193d221627cfce48738795d5200476bf972d3c76..678f3d9736b870689e6ec5b6bdd5367f6583c4c5 100644 (file)
--- a/audio_mixer.cpp
+++ b/audio_mixer.cpp
@@ -164,6 +164,10 @@ AudioMixer::AudioMixer(unsigned num_cards)
        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.
@@ -409,14 +413,7 @@ vector<float> AudioMixer::get_output(double pts, unsigned num_samples, Resamplin
        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);
@@ -531,6 +528,44 @@ vector<float> AudioMixer::get_output(double pts, unsigned num_samples, Resamplin
        return samples_out;
 }
 
+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());