git.sesse.net Git - nageru/blob - audio_mixer.h

   1 #ifndef _AUDIO_MIXER_H
   2 #define _AUDIO_MIXER_H 1
   3
   4 // The audio mixer, dealing with extracting the right signals from
   5 // each capture card, resampling signals so that they are in sync,
   6 // processing them with effects (if desired), and then mixing them
   7 // all together into one final audio signal.
   8 //
   9 // All operations on AudioMixer (except destruction) are thread-safe.
  10
  11 #include <math.h>
  12 #include <stdint.h>
  13 #include <atomic>
  14 #include <map>
  15 #include <memory>
  16 #include <mutex>
  17 #include <set>
  18 #include <vector>
  19 #include <zita-resampler/resampler.h>
  20
  21 #include "alsa_input.h"
  22 #include "bmusb/bmusb.h"
  23 #include "correlation_measurer.h"
  24 #include "db.h"
  25 #include "defs.h"
  26 #include "ebu_r128_proc.h"
  27 #include "filter.h"
  28 #include "resampling_queue.h"
  29 #include "stereocompressor.h"
  30
  31 namespace bmusb {
  32 struct AudioFormat;
  33 }  // namespace bmusb
  34
  35 enum class InputSourceType { SILENCE, CAPTURE_CARD, ALSA_INPUT };
  36 struct DeviceSpec {
  37         InputSourceType type;
  38         unsigned index;
  39
  40         bool operator== (const DeviceSpec &other) const {
  41                 return type == other.type && index == other.index;
  42         }
  43
  44         bool operator< (const DeviceSpec &other) const {
  45                 if (type != other.type)
  46                         return type < other.type;
  47                 return index < other.index;
  48         }
  49 };
  50 struct DeviceInfo {
  51         std::string name;
  52         unsigned num_channels;
  53 };
  54
  55 enum EQBand {
  56         EQ_BAND_BASS = 0,
  57         EQ_BAND_MID,
  58         EQ_BAND_TREBLE,
  59         NUM_EQ_BANDS
  60 };
  61
  62 static inline uint64_t DeviceSpec_to_key(const DeviceSpec &device_spec)
  63 {
  64         return (uint64_t(device_spec.type) << 32) | device_spec.index;
  65 }
  66
  67 static inline DeviceSpec key_to_DeviceSpec(uint64_t key)
  68 {
  69         return DeviceSpec{ InputSourceType(key >> 32), unsigned(key & 0xffffffff) };
  70 }
  71
  72 struct InputMapping {
  73         struct Bus {
  74                 std::string name;
  75                 DeviceSpec device;
  76                 int source_channel[2] { -1, -1 };  // Left and right. -1 = none.
  77         };
  78
  79         std::vector<Bus> buses;
  80 };
  81
  82 class AudioMixer {
  83 public:
  84         AudioMixer(unsigned num_cards);
  85         ~AudioMixer();
  86         void reset_resampler(DeviceSpec device_spec);
  87         void reset_meters();
  88
  89         // Add audio (or silence) to the given device's queue. Can return false if
  90         // the lock wasn't successfully taken; if so, you should simply try again.
  91         // (This is to avoid a deadlock where a card hangs on the mutex in add_audio()
  92         // while we are trying to shut it down from another thread that also holds
  93         // the mutex.) frame_length is in TIMEBASE units.
  94         bool add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length);
  95         bool add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length);
  96
  97         std::vector<float> get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
  98
  99         void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; }
 100         std::map<DeviceSpec, DeviceInfo> get_devices() const;
 101         void set_name(DeviceSpec device_spec, const std::string &name);
 102
 103         void set_input_mapping(const InputMapping &input_mapping);
 104         InputMapping get_input_mapping() const;
 105
 106         void set_locut_cutoff(float cutoff_hz)
 107         {
 108                 locut_cutoff_hz = cutoff_hz;
 109         }
 110
 111         float get_locut_cutoff() const
 112         {
 113                 return locut_cutoff_hz;
 114         }
 115
 116         void set_locut_enabled(unsigned bus, bool enabled)
 117         {
 118                 locut_enabled[bus] = enabled;
 119         }
 120
 121         bool get_locut_enabled(unsigned bus)
 122         {
 123                 return locut_enabled[bus];
 124         }
 125
 126         void set_eq(unsigned bus_index, EQBand band, float db_gain)
 127         {
 128                 assert(band >= 0 && band < NUM_EQ_BANDS);
 129                 eq_level_db[bus_index][band] = db_gain;
 130         }
 131
 132         float get_eq(unsigned bus_index, EQBand band) const
 133         {
 134                 assert(band >= 0 && band < NUM_EQ_BANDS);
 135                 return eq_level_db[bus_index][band];
 136         }
 137
 138         float get_limiter_threshold_dbfs() const
 139         {
 140                 return limiter_threshold_dbfs;
 141         }
 142
 143         float get_compressor_threshold_dbfs(unsigned bus_index) const
 144         {
 145                 return compressor_threshold_dbfs[bus_index];
 146         }
 147
 148         void set_limiter_threshold_dbfs(float threshold_dbfs)
 149         {
 150                 limiter_threshold_dbfs = threshold_dbfs;
 151         }
 152
 153         void set_compressor_threshold_dbfs(unsigned bus_index, float threshold_dbfs)
 154         {
 155                 compressor_threshold_dbfs[bus_index] = threshold_dbfs;
 156         }
 157
 158         void set_limiter_enabled(bool enabled)
 159         {
 160                 limiter_enabled = enabled;
 161         }
 162
 163         bool get_limiter_enabled() const
 164         {
 165                 return limiter_enabled;
 166         }
 167
 168         void set_compressor_enabled(unsigned bus_index, bool enabled)
 169         {
 170                 compressor_enabled[bus_index] = enabled;
 171         }
 172
 173         bool get_compressor_enabled(unsigned bus_index) const
 174         {
 175                 return compressor_enabled[bus_index];
 176         }
 177
 178         void set_gain_staging_db(unsigned bus_index, float gain_db)
 179         {
 180                 std::unique_lock<std::mutex> lock(compressor_mutex);
 181                 level_compressor_enabled[bus_index] = false;
 182                 gain_staging_db[bus_index] = gain_db;
 183         }
 184
 185         float get_gain_staging_db(unsigned bus_index) const
 186         {
 187                 std::unique_lock<std::mutex> lock(compressor_mutex);
 188                 return gain_staging_db[bus_index];
 189         }
 190
 191         void set_gain_staging_auto(unsigned bus_index, bool enabled)
 192         {
 193                 std::unique_lock<std::mutex> lock(compressor_mutex);
 194                 level_compressor_enabled[bus_index] = enabled;
 195         }
 196
 197         bool get_gain_staging_auto(unsigned bus_index) const
 198         {
 199                 std::unique_lock<std::mutex> lock(compressor_mutex);
 200                 return level_compressor_enabled[bus_index];
 201         }
 202
 203         void set_final_makeup_gain_db(float gain_db)
 204         {
 205                 std::unique_lock<std::mutex> lock(compressor_mutex);
 206                 final_makeup_gain_auto = false;
 207                 final_makeup_gain = from_db(gain_db);
 208         }
 209
 210         float get_final_makeup_gain_db()
 211         {
 212                 std::unique_lock<std::mutex> lock(compressor_mutex);
 213                 return to_db(final_makeup_gain);
 214         }
 215
 216         void set_final_makeup_gain_auto(bool enabled)
 217         {
 218                 std::unique_lock<std::mutex> lock(compressor_mutex);
 219                 final_makeup_gain_auto = enabled;
 220         }
 221
 222         bool get_final_makeup_gain_auto() const
 223         {
 224                 std::unique_lock<std::mutex> lock(compressor_mutex);
 225                 return final_makeup_gain_auto;
 226         }
 227
 228         void reset_peak(unsigned bus_index);
 229
 230         struct BusLevel {
 231                 float current_level_dbfs[2];  // Digital peak of last frame, left and right.
 232                 float peak_level_dbfs[2];  // Digital peak with hold, left and right.
 233                 float historic_peak_dbfs;
 234                 float gain_staging_db;
 235                 float compressor_attenuation_db;  // A positive number; 0.0 for no attenuation.
 236         };
 237
 238         typedef std::function<void(float level_lufs, float peak_db,
 239                                    std::vector<BusLevel> bus_levels,
 240                                    float global_level_lufs, float range_low_lufs, float range_high_lufs,
 241                                    float final_makeup_gain_db,
 242                                    float correlation)> audio_level_callback_t;
 243         void set_audio_level_callback(audio_level_callback_t callback)
 244         {
 245                 audio_level_callback = callback;
 246         }
 247
 248 private:
 249         struct AudioDevice {
 250                 std::unique_ptr<ResamplingQueue> resampling_queue;
 251                 int64_t next_local_pts = 0;
 252                 std::string name;
 253                 unsigned capture_frequency = OUTPUT_FREQUENCY;
 254                 // Which channels we consider interesting (ie., are part of some input_mapping).
 255                 std::set<unsigned> interesting_channels;
 256                 // Only used for ALSA cards, obviously.
 257                 std::unique_ptr<ALSAInput> alsa_device;
 258         };
 259         AudioDevice *find_audio_device(DeviceSpec device_spec);
 260
 261         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);
 262         void fill_audio_bus(const std::map<DeviceSpec, std::vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output);
 263         void reset_resampler_mutex_held(DeviceSpec device_spec);
 264         void reset_alsa_mutex_held(DeviceSpec device_spec);
 265         std::map<DeviceSpec, DeviceInfo> get_devices_mutex_held() const;
 266         void apply_eq(unsigned bus_index, std::vector<float> *samples_bus);
 267         void update_meters(const std::vector<float> &samples);
 268         void add_bus_to_master(unsigned bus_index, const std::vector<float> &samples_bus, std::vector<float> *samples_out);
 269         void measure_bus_levels(unsigned bus_index, const std::vector<float> &left, const std::vector<float> &right);
 270         void send_audio_level_callback();
 271
 272         unsigned num_cards;
 273
 274         mutable std::timed_mutex audio_mutex;
 275
 276         AudioDevice video_cards[MAX_VIDEO_CARDS];  // Under audio_mutex.
 277
 278         // TODO: Figure out a better way to unify these two, as they are sharing indexing.
 279         AudioDevice alsa_inputs[MAX_ALSA_CARDS];  // Under audio_mutex.
 280         std::vector<ALSAInput::Device> available_alsa_cards;
 281
 282         std::atomic<float> locut_cutoff_hz{120};
 283         StereoFilter locut[MAX_BUSES];  // Default cutoff 120 Hz, 24 dB/oct.
 284         std::atomic<bool> locut_enabled[MAX_BUSES];
 285         StereoFilter eq[MAX_BUSES][NUM_EQ_BANDS];  // The one for EQBand::MID isn't actually used (see comments in apply_eq()).
 286
 287         // First compressor; takes us up to about -12 dBFS.
 288         mutable std::mutex compressor_mutex;
 289         std::unique_ptr<StereoCompressor> level_compressor[MAX_BUSES];  // Under compressor_mutex. Used to set/override gain_staging_db if <level_compressor_enabled>.
 290         float gain_staging_db[MAX_BUSES];  // Under compressor_mutex.
 291         bool level_compressor_enabled[MAX_BUSES];  // Under compressor_mutex.
 292
 293         static constexpr float ref_level_dbfs = -14.0f;  // Chosen so that we end up around 0 LU in practice.
 294         static constexpr float ref_level_lufs = -23.0f;  // 0 LU, more or less by definition.
 295
 296         StereoCompressor limiter;
 297         std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 4.0f};   // 4 dB.
 298         std::atomic<bool> limiter_enabled{true};
 299         std::unique_ptr<StereoCompressor> compressor[MAX_BUSES];
 300         std::atomic<float> compressor_threshold_dbfs[MAX_BUSES];
 301         std::atomic<bool> compressor_enabled[MAX_BUSES];
 302
 303         // Note: The values here are not in dB.
 304         struct PeakHistory {
 305                 float current_level = 0.0f;  // Peak of the last frame.
 306                 float historic_peak = 0.0f;  // Highest peak since last reset; no falloff.
 307                 float current_peak = 0.0f;  // Current peak of the peak meter.
 308                 float last_peak = 0.0f;
 309                 float age_seconds = 0.0f;   // Time since "last_peak" was set.
 310         };
 311         PeakHistory peak_history[MAX_BUSES][2];  // Separate for each channel. Under audio_mutex.
 312
 313         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.
 314         bool final_makeup_gain_auto = true;  // Under compressor_mutex.
 315
 316         InputMapping input_mapping;  // Under audio_mutex.
 317         std::atomic<float> fader_volume_db[MAX_BUSES] {{ 0.0f }};
 318         float last_fader_volume_db[MAX_BUSES] { 0.0f };  // Under audio_mutex.
 319         std::atomic<float> eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{{ 0.0f }}};
 320
 321         audio_level_callback_t audio_level_callback = nullptr;
 322         mutable std::mutex audio_measure_mutex;
 323         Ebu_r128_proc r128;  // Under audio_measure_mutex.
 324         CorrelationMeasurer correlation;  // Under audio_measure_mutex.
 325         Resampler peak_resampler;  // Under audio_measure_mutex.
 326         std::atomic<float> peak{0.0f};
 327 };
 328
 329 #endif  // !defined(_AUDIO_MIXER_H)