]> git.sesse.net Git - nageru/blobdiff - audio_mixer.h
Implement the EQ in AudioMixer. (No UI yet.)
[nageru] / audio_mixer.h
index 6f7dbe5d0b45850fed81f743b6674eeb176c6c7b..c332671e0bebd6f385d66b8ddc40fd426fd765ad 100644 (file)
@@ -7,20 +7,23 @@
 // all together into one final audio signal.
 //
 // All operations on AudioMixer (except destruction) are thread-safe.
-//
-// TODO: There might be more audio stuff that should be moved here
-// from Mixer.
 
 #include <math.h>
 #include <stdint.h>
 #include <atomic>
+#include <map>
 #include <memory>
 #include <mutex>
+#include <set>
 #include <vector>
+#include <zita-resampler/resampler.h>
 
+#include "alsa_input.h"
 #include "bmusb/bmusb.h"
+#include "correlation_measurer.h"
 #include "db.h"
 #include "defs.h"
+#include "ebu_r128_proc.h"
 #include "filter.h"
 #include "resampling_queue.h"
 #include "stereocompressor.h"
@@ -29,35 +32,73 @@ namespace bmusb {
 struct AudioFormat;
 }  // namespace bmusb
 
-enum class InputSourceType { SILENCE, CAPTURE_CARD };
+enum class InputSourceType { SILENCE, CAPTURE_CARD, ALSA_INPUT };
+struct DeviceSpec {
+       InputSourceType type;
+       unsigned index;
+
+       bool operator== (const DeviceSpec &other) const {
+               return type == other.type && index == other.index;
+       }
+
+       bool operator< (const DeviceSpec &other) const {
+               if (type != other.type)
+                       return type < other.type;
+               return index < other.index;
+       }
+};
+struct DeviceInfo {
+       std::string name;
+       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;
+}
+
+static inline DeviceSpec key_to_DeviceSpec(uint64_t key)
+{
+       return DeviceSpec{ InputSourceType(key >> 32), unsigned(key & 0xffffffff) };
+}
 
 struct InputMapping {
-       struct Input {
+       struct Bus {
                std::string name;
-               InputSourceType input_source_type;
-               unsigned input_source_index;
+               DeviceSpec device;
                int source_channel[2] { -1, -1 };  // Left and right. -1 = none.
        };
 
-       std::vector<Input> inputs;
+       std::vector<Bus> buses;
 };
 
 class AudioMixer {
 public:
        AudioMixer(unsigned num_cards);
-       void reset_card(unsigned card_index);
+       ~AudioMixer();
+       void reset_resampler(DeviceSpec device_spec);
+       void reset_meters();
+
+       // Add audio (or silence) to the given device's queue. Can return false if
+       // the lock wasn't successfully taken; if so, you should simply try again.
+       // (This is to avoid a deadlock where a card hangs on the mutex in add_audio()
+       // while we are trying to shut it down from another thread that also holds
+       // the mutex.) frame_length is in TIMEBASE units.
+       bool add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length);
+       bool add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length);
 
-       // frame_length is in TIMEBASE units.
-       void add_audio(unsigned card_index, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length);
-       void add_silence(unsigned card_index, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length);
        std::vector<float> get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
 
-       // See comments inside get_output().
-       void set_current_loudness(double level_lufs) { loudness_lufs = level_lufs; }
-
-       void set_fader_volume(unsigned card_index, float level_db) { cards[card_index].fader_volume_db = level_db; }
-       std::vector<std::string> get_names() const;
-       void set_name(unsigned card_index, const std::string &name);
+       void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; }
+       std::map<DeviceSpec, DeviceInfo> get_devices() const;
+       void set_name(DeviceSpec device_spec, const std::string &name);
 
        void set_input_mapping(const InputMapping &input_mapping);
        InputMapping get_input_mapping() const;
@@ -67,14 +108,20 @@ public:
                locut_cutoff_hz = cutoff_hz;
        }
 
-       void set_locut_enabled(bool enabled)
+       void set_locut_enabled(unsigned bus, bool enabled)
+       {
+               locut_enabled[bus] = enabled;
+       }
+
+       bool get_locut_enabled(unsigned bus)
        {
-               locut_enabled = enabled;
+               return locut_enabled[bus];
        }
 
-       bool get_locut_enabled() const
+       void set_eq(unsigned bus_index, EQBand band, float db_gain)
        {
-               return locut_enabled;
+               assert(band >= 0 && band < NUM_EQ_BANDS);
+               eq_level_db[bus_index][band] = db_gain;
        }
 
        float get_limiter_threshold_dbfs() const
@@ -82,9 +129,9 @@ public:
                return limiter_threshold_dbfs;
        }
 
-       float get_compressor_threshold_dbfs() const
+       float get_compressor_threshold_dbfs(unsigned bus_index) const
        {
-               return compressor_threshold_dbfs;
+               return compressor_threshold_dbfs[bus_index];
        }
 
        void set_limiter_threshold_dbfs(float threshold_dbfs)
@@ -92,9 +139,9 @@ public:
                limiter_threshold_dbfs = threshold_dbfs;
        }
 
-       void set_compressor_threshold_dbfs(float threshold_dbfs)
+       void set_compressor_threshold_dbfs(unsigned bus_index, float threshold_dbfs)
        {
-               compressor_threshold_dbfs = threshold_dbfs;
+               compressor_threshold_dbfs[bus_index] = threshold_dbfs;
        }
 
        void set_limiter_enabled(bool enabled)
@@ -107,39 +154,39 @@ public:
                return limiter_enabled;
        }
 
-       void set_compressor_enabled(bool enabled)
+       void set_compressor_enabled(unsigned bus_index, bool enabled)
        {
-               compressor_enabled = enabled;
+               compressor_enabled[bus_index] = enabled;
        }
 
-       bool get_compressor_enabled() const
+       bool get_compressor_enabled(unsigned bus_index) const
        {
-               return compressor_enabled;
+               return compressor_enabled[bus_index];
        }
 
-       void set_gain_staging_db(float gain_db)
+       void set_gain_staging_db(unsigned bus_index, float gain_db)
        {
                std::unique_lock<std::mutex> lock(compressor_mutex);
-               level_compressor_enabled = false;
-               gain_staging_db = gain_db;
+               level_compressor_enabled[bus_index] = false;
+               gain_staging_db[bus_index] = gain_db;
        }
 
-       float get_gain_staging_db() const
+       float get_gain_staging_db(unsigned bus_index) const
        {
                std::unique_lock<std::mutex> lock(compressor_mutex);
-               return gain_staging_db;
+               return gain_staging_db[bus_index];
        }
 
-       void set_gain_staging_auto(bool enabled)
+       void set_gain_staging_auto(unsigned bus_index, bool enabled)
        {
                std::unique_lock<std::mutex> lock(compressor_mutex);
-               level_compressor_enabled = enabled;
+               level_compressor_enabled[bus_index] = enabled;
        }
 
-       bool get_gain_staging_auto() const
+       bool get_gain_staging_auto(unsigned bus_index) const
        {
                std::unique_lock<std::mutex> lock(compressor_mutex);
-               return level_compressor_enabled;
+               return level_compressor_enabled[bus_index];
        }
 
        void set_final_makeup_gain_db(float gain_db)
@@ -167,47 +214,105 @@ public:
                return final_makeup_gain_auto;
        }
 
-private:
-       unsigned num_cards;
+       void reset_peak(unsigned bus_index);
 
-       struct CaptureCard {
-               std::atomic<float> fader_volume_db{0.0f};
+       struct BusLevel {
+               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.
+       };
 
-               // Everything below audio_mutex is protected by it.
-               mutable std::mutex audio_mutex;
+       typedef std::function<void(float level_lufs, float peak_db,
+                                  std::vector<BusLevel> bus_levels,
+                                  float global_level_lufs, float range_low_lufs, float range_high_lufs,
+                                  float final_makeup_gain_db,
+                                  float correlation)> audio_level_callback_t;
+       void set_audio_level_callback(audio_level_callback_t callback)
+       {
+               audio_level_callback = callback;
+       }
+
+private:
+       struct AudioDevice {
                std::unique_ptr<ResamplingQueue> resampling_queue;
                int64_t next_local_pts = 0;
                std::string name;
+               unsigned capture_frequency = OUTPUT_FREQUENCY;
+               // Which channels we consider interesting (ie., are part of some input_mapping).
+               std::set<unsigned> interesting_channels;
+               // Only used for ALSA cards, obviously.
+               std::unique_ptr<ALSAInput> alsa_device;
        };
-       CaptureCard cards[MAX_CARDS];
+       AudioDevice *find_audio_device(DeviceSpec device_spec);
+
+       void find_sample_src_from_device(const std::map<DeviceSpec, std::vector<float>> &samples_card, DeviceSpec device_spec, int source_channel, const float **srcptr, unsigned *stride);
+       void fill_audio_bus(const std::map<DeviceSpec, std::vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output);
+       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();
+
+       unsigned num_cards;
+
+       mutable std::timed_mutex audio_mutex;
+
+       AudioDevice video_cards[MAX_VIDEO_CARDS];  // Under audio_mutex.
+
+       // TODO: Figure out a better way to unify these two, as they are sharing indexing.
+       AudioDevice alsa_inputs[MAX_ALSA_CARDS];  // Under audio_mutex.
+       std::vector<ALSAInput::Device> available_alsa_cards;
 
-       StereoFilter locut;  // Default cutoff 120 Hz, 24 dB/oct.
        std::atomic<float> locut_cutoff_hz;
-       std::atomic<bool> locut_enabled{true};
+       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;
-       StereoCompressor level_compressor;  // Under compressor_mutex. Used to set/override gain_staging_db if <level_compressor_enabled>.
-       float gain_staging_db = 0.0f;  // Under compressor_mutex.
-       bool level_compressor_enabled = true;  // Under compressor_mutex.
+       std::unique_ptr<StereoCompressor> level_compressor[MAX_BUSES];  // Under compressor_mutex. Used to set/override gain_staging_db if <level_compressor_enabled>.
+       float gain_staging_db[MAX_BUSES];  // Under compressor_mutex.
+       bool level_compressor_enabled[MAX_BUSES];  // Under compressor_mutex.
 
        static constexpr float ref_level_dbfs = -14.0f;  // Chosen so that we end up around 0 LU in practice.
        static constexpr float ref_level_lufs = -23.0f;  // 0 LU, more or less by definition.
 
-       std::atomic<float> loudness_lufs{ref_level_lufs};
-
        StereoCompressor limiter;
        std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 4.0f};   // 4 dB.
        std::atomic<bool> limiter_enabled{true};
-       StereoCompressor compressor;
-       std::atomic<float> compressor_threshold_dbfs{ref_level_dbfs - 12.0f};  // -12 dB.
-       std::atomic<bool> compressor_enabled{true};
+       std::unique_ptr<StereoCompressor> compressor[MAX_BUSES];
+       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.
 
-       mutable std::mutex mapping_mutex;
-       InputMapping input_mapping;  // Under mapping_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;
+       Ebu_r128_proc r128;  // Under audio_measure_mutex.
+       CorrelationMeasurer correlation;  // Under audio_measure_mutex.
+       Resampler peak_resampler;  // Under audio_measure_mutex.
+       std::atomic<float> peak{0.0f};
 };
 
 #endif  // !defined(_AUDIO_MIXER_H)