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         void reset_resampler(DeviceSpec device_spec);
  86         void reset_meters();
  87
  88         // Add audio (or silence) to the given device's queue. Can return false if
  89         // the lock wasn't successfully taken; if so, you should simply try again.
  90         // (This is to avoid a deadlock where a card hangs on the mutex in add_audio()
  91         // while we are trying to shut it down from another thread that also holds
  92         // the mutex.) frame_length is in TIMEBASE units.
  93         bool add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length);
  94         bool add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length);
  95
  96         std::vector<float> get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
  97
  98         void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; }
  99
 100         // Note: This operation holds all ALSA devices (see ALSAPool::get_devices()).
 101         // You will need to call set_input_mapping() to get the hold state correctly,
 102         // or every card will be held forever.
 103         std::map<DeviceSpec, DeviceInfo> get_devices();
 104
 105         void set_name(DeviceSpec device_spec, const std::string &name);
 106
 107         void set_input_mapping(const InputMapping &input_mapping);
 108         InputMapping get_input_mapping() const;
 109
 110         void set_locut_cutoff(float cutoff_hz)
 111         {
 112                 locut_cutoff_hz = cutoff_hz;
 113         }
 114
 115         float get_locut_cutoff() const
 116         {
 117                 return locut_cutoff_hz;
 118         }
 119
 120         void set_locut_enabled(unsigned bus, bool enabled)
 121         {
 122                 locut_enabled[bus] = enabled;
 123         }
 124
 125         bool get_locut_enabled(unsigned bus)
 126         {
 127                 return locut_enabled[bus];
 128         }
 129
 130         void set_eq(unsigned bus_index, EQBand band, float db_gain)
 131         {
 132                 assert(band >= 0 && band < NUM_EQ_BANDS);
 133                 eq_level_db[bus_index][band] = db_gain;
 134         }
 135
 136         float get_eq(unsigned bus_index, EQBand band) const
 137         {
 138                 assert(band >= 0 && band < NUM_EQ_BANDS);
 139                 return eq_level_db[bus_index][band];
 140         }
 141
 142         float get_limiter_threshold_dbfs() const
 143         {
 144                 return limiter_threshold_dbfs;
 145         }
 146
 147         float get_compressor_threshold_dbfs(unsigned bus_index) const
 148         {
 149                 return compressor_threshold_dbfs[bus_index];
 150         }
 151
 152         void set_limiter_threshold_dbfs(float threshold_dbfs)
 153         {
 154                 limiter_threshold_dbfs = threshold_dbfs;
 155         }
 156
 157         void set_compressor_threshold_dbfs(unsigned bus_index, float threshold_dbfs)
 158         {
 159                 compressor_threshold_dbfs[bus_index] = threshold_dbfs;
 160         }
 161
 162         void set_limiter_enabled(bool enabled)
 163         {
 164                 limiter_enabled = enabled;
 165         }
 166
 167         bool get_limiter_enabled() const
 168         {
 169                 return limiter_enabled;
 170         }
 171
 172         void set_compressor_enabled(unsigned bus_index, bool enabled)
 173         {
 174                 compressor_enabled[bus_index] = enabled;
 175         }
 176
 177         bool get_compressor_enabled(unsigned bus_index) const
 178         {
 179                 return compressor_enabled[bus_index];
 180         }
 181
 182         void set_gain_staging_db(unsigned bus_index, float gain_db)
 183         {
 184                 std::unique_lock<std::mutex> lock(compressor_mutex);
 185                 level_compressor_enabled[bus_index] = false;
 186                 gain_staging_db[bus_index] = gain_db;
 187         }
 188
 189         float get_gain_staging_db(unsigned bus_index) const
 190         {
 191                 std::unique_lock<std::mutex> lock(compressor_mutex);
 192                 return gain_staging_db[bus_index];
 193         }
 194
 195         void set_gain_staging_auto(unsigned bus_index, bool enabled)
 196         {
 197                 std::unique_lock<std::mutex> lock(compressor_mutex);
 198                 level_compressor_enabled[bus_index] = enabled;
 199         }
 200
 201         bool get_gain_staging_auto(unsigned bus_index) const
 202         {
 203                 std::unique_lock<std::mutex> lock(compressor_mutex);
 204                 return level_compressor_enabled[bus_index];
 205         }
 206
 207         void set_final_makeup_gain_db(float gain_db)
 208         {
 209                 std::unique_lock<std::mutex> lock(compressor_mutex);
 210                 final_makeup_gain_auto = false;
 211                 final_makeup_gain = from_db(gain_db);
 212         }
 213
 214         float get_final_makeup_gain_db()
 215         {
 216                 std::unique_lock<std::mutex> lock(compressor_mutex);
 217                 return to_db(final_makeup_gain);
 218         }
 219
 220         void set_final_makeup_gain_auto(bool enabled)
 221         {
 222                 std::unique_lock<std::mutex> lock(compressor_mutex);
 223                 final_makeup_gain_auto = enabled;
 224         }
 225
 226         bool get_final_makeup_gain_auto() const
 227         {
 228                 std::unique_lock<std::mutex> lock(compressor_mutex);
 229                 return final_makeup_gain_auto;
 230         }
 231
 232         void reset_peak(unsigned bus_index);
 233
 234         struct BusLevel {
 235                 float current_level_dbfs[2];  // Digital peak of last frame, left and right.
 236                 float peak_level_dbfs[2];  // Digital peak with hold, left and right.
 237                 float historic_peak_dbfs;
 238                 float gain_staging_db;
 239                 float compressor_attenuation_db;  // A positive number; 0.0 for no attenuation.
 240         };
 241
 242         typedef std::function<void(float level_lufs, float peak_db,
 243                                    std::vector<BusLevel> bus_levels,
 244                                    float global_level_lufs, float range_low_lufs, float range_high_lufs,
 245                                    float final_makeup_gain_db,
 246                                    float correlation)> audio_level_callback_t;
 247         void set_audio_level_callback(audio_level_callback_t callback)
 248         {
 249                 audio_level_callback = callback;
 250         }
 251
 252 private:
 253         struct AudioDevice {
 254                 std::unique_ptr<ResamplingQueue> resampling_queue;
 255                 int64_t next_local_pts = 0;
 256                 std::string name;
 257                 unsigned capture_frequency = OUTPUT_FREQUENCY;
 258                 // Which channels we consider interesting (ie., are part of some input_mapping).
 259                 std::set<unsigned> interesting_channels;
 260         };
 261
 262         const AudioDevice *find_audio_device(DeviceSpec device_spec) const
 263         {
 264                 return const_cast<AudioMixer *>(this)->find_audio_device(device_spec);
 265         }
 266
 267         AudioDevice *find_audio_device(DeviceSpec device_spec);
 268
 269         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);
 270         void fill_audio_bus(const std::map<DeviceSpec, std::vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output);
 271         void reset_resampler_mutex_held(DeviceSpec device_spec);
 272         void apply_eq(unsigned bus_index, std::vector<float> *samples_bus);
 273         void update_meters(const std::vector<float> &samples);
 274         void add_bus_to_master(unsigned bus_index, const std::vector<float> &samples_bus, std::vector<float> *samples_out);
 275         void measure_bus_levels(unsigned bus_index, const std::vector<float> &left, const std::vector<float> &right);
 276         void send_audio_level_callback();
 277         std::vector<DeviceSpec> get_active_devices() const;
 278
 279         unsigned num_cards;
 280
 281         mutable std::timed_mutex audio_mutex;
 282
 283         ALSAPool alsa_pool;
 284         AudioDevice video_cards[MAX_VIDEO_CARDS];  // Under audio_mutex.
 285         AudioDevice alsa_inputs[MAX_ALSA_CARDS];  // Under audio_mutex.
 286
 287         std::atomic<float> locut_cutoff_hz{120};
 288         StereoFilter locut[MAX_BUSES];  // Default cutoff 120 Hz, 24 dB/oct.
 289         std::atomic<bool> locut_enabled[MAX_BUSES];
 290         StereoFilter eq[MAX_BUSES][NUM_EQ_BANDS];  // The one for EQBand::MID isn't actually used (see comments in apply_eq()).
 291
 292         // First compressor; takes us up to about -12 dBFS.
 293         mutable std::mutex compressor_mutex;
 294         std::unique_ptr<StereoCompressor> level_compressor[MAX_BUSES];  // Under compressor_mutex. Used to set/override gain_staging_db if <level_compressor_enabled>.
 295         float gain_staging_db[MAX_BUSES];  // Under compressor_mutex.
 296         bool level_compressor_enabled[MAX_BUSES];  // Under compressor_mutex.
 297
 298         static constexpr float ref_level_dbfs = -14.0f;  // Chosen so that we end up around 0 LU in practice.
 299         static constexpr float ref_level_lufs = -23.0f;  // 0 LU, more or less by definition.
 300
 301         StereoCompressor limiter;
 302         std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 4.0f};   // 4 dB.
 303         std::atomic<bool> limiter_enabled{true};
 304         std::unique_ptr<StereoCompressor> compressor[MAX_BUSES];
 305         std::atomic<float> compressor_threshold_dbfs[MAX_BUSES];
 306         std::atomic<bool> compressor_enabled[MAX_BUSES];
 307
 308         // Note: The values here are not in dB.
 309         struct PeakHistory {
 310                 float current_level = 0.0f;  // Peak of the last frame.
 311                 float historic_peak = 0.0f;  // Highest peak since last reset; no falloff.
 312                 float current_peak = 0.0f;  // Current peak of the peak meter.
 313                 float last_peak = 0.0f;
 314                 float age_seconds = 0.0f;   // Time since "last_peak" was set.
 315         };
 316         PeakHistory peak_history[MAX_BUSES][2];  // Separate for each channel. Under audio_mutex.
 317
 318         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.
 319         bool final_makeup_gain_auto = true;  // Under compressor_mutex.
 320
 321         InputMapping input_mapping;  // Under audio_mutex.
 322         std::atomic<float> fader_volume_db[MAX_BUSES] {{ 0.0f }};
 323         float last_fader_volume_db[MAX_BUSES] { 0.0f };  // Under audio_mutex.
 324         std::atomic<float> eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{{ 0.0f }}};
 325
 326         audio_level_callback_t audio_level_callback = nullptr;
 327         mutable std::mutex audio_measure_mutex;
 328         Ebu_r128_proc r128;  // Under audio_measure_mutex.
 329         CorrelationMeasurer correlation;  // Under audio_measure_mutex.
 330         Resampler peak_resampler;  // Under audio_measure_mutex.
 331         std::atomic<float> peak{0.0f};
 332 };
 333
 334 extern AudioMixer *global_audio_mixer;
 335
 336 #endif  // !defined(_AUDIO_MIXER_H)