Set a default locut cutoff, for the benchmark.

[nageru] / audio_mixer.h

diff --git a/audio_mixer.h b/audio_mixer.h
index 5a3f2af1d6532ac09ded74c31937756415d91c5c..42aa92a70b30ac9e0ad1d5901abd6b003b138472 100644 (file)
--- a/audio_mixer.h
+++ b/audio_mixer.h
@@ -52,6 +52,13 @@ struct DeviceInfo {
        unsigned num_channels;
 };
 
+enum EQBand {
+       EQ_BAND_BASS = 0,
+       EQ_BAND_MID,
+       EQ_BAND_TREBLE,
+       NUM_EQ_BANDS
+};
+
 static inline uint64_t DeviceSpec_to_key(const DeviceSpec &device_spec)
 {
        return (uint64_t(device_spec.type) << 32) | device_spec.index;
@@ -101,6 +108,11 @@ public:
                locut_cutoff_hz = cutoff_hz;
        }
 
+       float get_locut_cutoff() const
+       {
+               return locut_cutoff_hz;
+       }
+
        void set_locut_enabled(unsigned bus, bool enabled)
        {
                locut_enabled[bus] = enabled;
@@ -111,6 +123,18 @@ public:
                return locut_enabled[bus];
        }
 
+       void set_eq(unsigned bus_index, EQBand band, float db_gain)
+       {
+               assert(band >= 0 && band < NUM_EQ_BANDS);
+               eq_level_db[bus_index][band] = db_gain;
+       }
+
+       float get_eq(unsigned bus_index, EQBand band) const
+       {
+               assert(band >= 0 && band < NUM_EQ_BANDS);
+               return eq_level_db[bus_index][band];
+       }
+
        float get_limiter_threshold_dbfs() const
        {
                return limiter_threshold_dbfs;
@@ -201,8 +225,12 @@ public:
                return final_makeup_gain_auto;
        }
 
+       void reset_peak(unsigned bus_index);
+
        struct BusLevel {
-               float loudness_lufs;
+               float current_level_dbfs[2];  // Digital peak of last frame, left and right.
+               float peak_level_dbfs[2];  // Digital peak with hold, left and right.
+               float historic_peak_dbfs;
                float gain_staging_db;
                float compressor_attenuation_db;  // A positive number; 0.0 for no attenuation.
        };
@@ -235,7 +263,9 @@ private:
        void reset_resampler_mutex_held(DeviceSpec device_spec);
        void reset_alsa_mutex_held(DeviceSpec device_spec);
        std::map<DeviceSpec, DeviceInfo> get_devices_mutex_held() const;
+       void apply_eq(unsigned bus_index, std::vector<float> *samples_bus);
        void update_meters(const std::vector<float> &samples);
+       void add_bus_to_master(unsigned bus_index, const std::vector<float> &samples_bus, std::vector<float> *samples_out);
        void measure_bus_levels(unsigned bus_index, const std::vector<float> &left, const std::vector<float> &right);
        void send_audio_level_callback();
 
@@ -249,9 +279,10 @@ private:
        AudioDevice alsa_inputs[MAX_ALSA_CARDS];  // Under audio_mutex.
        std::vector<ALSAInput::Device> available_alsa_cards;
 
-       std::atomic<float> locut_cutoff_hz;
+       std::atomic<float> locut_cutoff_hz{120};
        StereoFilter locut[MAX_BUSES];  // Default cutoff 120 Hz, 24 dB/oct.
        std::atomic<bool> locut_enabled[MAX_BUSES];
+       StereoFilter eq[MAX_BUSES][NUM_EQ_BANDS];  // The one for EQBand::MID isn't actually used (see comments in apply_eq()).
 
        // First compressor; takes us up to about -12 dBFS.
        mutable std::mutex compressor_mutex;
@@ -269,11 +300,23 @@ private:
        std::atomic<float> compressor_threshold_dbfs[MAX_BUSES];
        std::atomic<bool> compressor_enabled[MAX_BUSES];
 
+       // Note: The values here are not in dB.
+       struct PeakHistory {
+               float current_level = 0.0f;  // Peak of the last frame.
+               float historic_peak = 0.0f;  // Highest peak since last reset; no falloff.
+               float current_peak = 0.0f;  // Current peak of the peak meter.
+               float last_peak = 0.0f;
+               float age_seconds = 0.0f;   // Time since "last_peak" was set.
+       };
+       PeakHistory peak_history[MAX_BUSES][2];  // Separate for each channel. Under audio_mutex.
+
        double final_makeup_gain = 1.0;  // Under compressor_mutex. Read/write by the user. Note: Not in dB, we want the numeric precision so that we can change it slowly.
        bool final_makeup_gain_auto = true;  // Under compressor_mutex.
 
        InputMapping input_mapping;  // Under audio_mutex.
        std::atomic<float> fader_volume_db[MAX_BUSES] {{ 0.0f }};
+       float last_fader_volume_db[MAX_BUSES] { 0.0f };  // Under audio_mutex.
+       std::atomic<float> eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{{ 0.0f }}};
 
        audio_level_callback_t audio_level_callback = nullptr;
        mutable std::mutex audio_measure_mutex;
@@ -281,12 +324,6 @@ private:
        CorrelationMeasurer correlation;  // Under audio_measure_mutex.
        Resampler peak_resampler;  // Under audio_measure_mutex.
        std::atomic<float> peak{0.0f};
-
-       // Under audio_measure_mutex. Note that Ebu_r128_proc has a broken
-       // copy constructor (it uses the default, but holds arrays),
-       // so we can't just use raw Ebu_r128_proc elements, but need to use
-       // unique_ptrs.
-       std::vector<std::unique_ptr<Ebu_r128_proc>> bus_r128;
 };
 
 #endif  // !defined(_AUDIO_MIXER_H)