2 #define _AUDIO_MIXER_H 1
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.
9 // All operations on AudioMixer (except destruction) are thread-safe.
13 #include <zita-resampler/resampler.h>
24 #include "alsa_pool.h"
25 #include "correlation_measurer.h"
28 #include "ebu_r128_proc.h"
30 #include "input_mapping.h"
31 #include "resampling_queue.h"
32 #include "stereocompressor.h"
34 class DeviceSpecProto;
49 AudioMixer(unsigned num_capture_cards, unsigned num_ffmpeg_inputs);
50 void reset_resampler(DeviceSpec device_spec);
53 // Add audio (or silence) to the given device's queue. Can return false if
54 // the lock wasn't successfully taken; if so, you should simply try again.
55 // (This is to avoid a deadlock where a card hangs on the mutex in add_audio()
56 // while we are trying to shut it down from another thread that also holds
58 bool add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, std::chrono::steady_clock::time_point frame_time);
59 bool add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames);
61 // If a given device is offline for whatever reason and cannot deliver audio
62 // (by means of add_audio() or add_silence()), you can call put it in silence mode,
63 // where it will be taken to only output silence. Note that when taking it _out_
64 // of silence mode, the resampler will be reset, so that old audio will not
65 // affect it. Same true/false behavior as add_audio().
66 bool silence_card(DeviceSpec device_spec, bool silence);
68 std::vector<float> get_output(std::chrono::steady_clock::time_point ts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
70 float get_fader_volume(unsigned bus_index) const { return fader_volume_db[bus_index]; }
71 void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; }
73 bool get_mute(unsigned bus_index) const { return mute[bus_index]; }
74 void set_mute(unsigned bus_index, bool muted) { mute[bus_index] = muted; }
76 // Note: This operation holds all ALSA devices (see ALSAPool::get_devices()).
77 // You will need to call set_input_mapping() to get the hold state correctly,
78 // or every card will be held forever.
79 std::map<DeviceSpec, DeviceInfo> get_devices();
81 // See comments on ALSAPool::get_card_state().
82 ALSAPool::Device::State get_alsa_card_state(unsigned index)
84 return alsa_pool.get_card_state(index);
87 // See comments on ALSAPool::create_dead_card().
88 DeviceSpec create_dead_card(const std::string &name, const std::string &info, unsigned num_channels)
90 unsigned dead_card_index = alsa_pool.create_dead_card(name, info, num_channels);
91 return DeviceSpec{InputSourceType::ALSA_INPUT, dead_card_index};
94 void set_display_name(DeviceSpec device_spec, const std::string &name);
96 // Note: The card should be held (currently this isn't enforced, though).
97 void serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto);
99 enum class MappingMode {
100 // A single bus, only from a video card (no ALSA devices),
101 // only channel 1 and 2, locked to +0 dB. Note that this is
102 // only an UI abstraction around exactly the same audio code
103 // as MULTICHANNEL; it's just less flexible.
106 // Full, arbitrary mappings.
110 // Automatically sets mapping mode to MappingMode::SIMPLE.
111 void set_simple_input(unsigned card_index);
113 // If mapping mode is not representable as a MappingMode::SIMPLE type
114 // mapping, returns numeric_limits<unsigned>::max().
115 unsigned get_simple_input() const;
117 // Implicitly sets mapping mode to MappingMode::MULTICHANNEL.
118 void set_input_mapping(const InputMapping &input_mapping);
120 MappingMode get_mapping_mode() const;
121 InputMapping get_input_mapping() const;
123 unsigned num_buses() const;
125 void set_locut_cutoff(float cutoff_hz)
127 locut_cutoff_hz = cutoff_hz;
130 float get_locut_cutoff() const
132 return locut_cutoff_hz;
135 void set_locut_enabled(unsigned bus, bool enabled)
137 locut_enabled[bus] = enabled;
140 bool get_locut_enabled(unsigned bus)
142 return locut_enabled[bus];
145 bool is_mono(unsigned bus_index);
147 void set_stereo_width(unsigned bus_index, float width)
149 stereo_width[bus_index] = width;
152 float get_stereo_width(unsigned bus_index)
154 return stereo_width[bus_index];
157 void set_eq(unsigned bus_index, EQBand band, float db_gain)
159 assert(band >= 0 && band < NUM_EQ_BANDS);
160 eq_level_db[bus_index][band] = db_gain;
163 float get_eq(unsigned bus_index, EQBand band) const
165 assert(band >= 0 && band < NUM_EQ_BANDS);
166 return eq_level_db[bus_index][band];
169 float get_limiter_threshold_dbfs() const
171 return limiter_threshold_dbfs;
174 float get_compressor_threshold_dbfs(unsigned bus_index) const
176 return compressor_threshold_dbfs[bus_index];
179 void set_limiter_threshold_dbfs(float threshold_dbfs)
181 limiter_threshold_dbfs = threshold_dbfs;
184 void set_compressor_threshold_dbfs(unsigned bus_index, float threshold_dbfs)
186 compressor_threshold_dbfs[bus_index] = threshold_dbfs;
189 void set_limiter_enabled(bool enabled)
191 limiter_enabled = enabled;
194 bool get_limiter_enabled() const
196 return limiter_enabled;
199 void set_compressor_enabled(unsigned bus_index, bool enabled)
201 compressor_enabled[bus_index] = enabled;
204 bool get_compressor_enabled(unsigned bus_index) const
206 return compressor_enabled[bus_index];
209 void set_gain_staging_db(unsigned bus_index, float gain_db)
211 std::lock_guard<std::mutex> lock(compressor_mutex);
212 level_compressor_enabled[bus_index] = false;
213 gain_staging_db[bus_index] = gain_db;
216 float get_gain_staging_db(unsigned bus_index) const
218 std::lock_guard<std::mutex> lock(compressor_mutex);
219 return gain_staging_db[bus_index];
222 void set_gain_staging_auto(unsigned bus_index, bool enabled)
224 std::lock_guard<std::mutex> lock(compressor_mutex);
225 level_compressor_enabled[bus_index] = enabled;
228 bool get_gain_staging_auto(unsigned bus_index) const
230 std::lock_guard<std::mutex> lock(compressor_mutex);
231 return level_compressor_enabled[bus_index];
234 void set_final_makeup_gain_db(float gain_db)
236 std::lock_guard<std::mutex> lock(compressor_mutex);
237 final_makeup_gain_auto = false;
238 final_makeup_gain = from_db(gain_db);
241 float get_final_makeup_gain_db()
243 std::lock_guard<std::mutex> lock(compressor_mutex);
244 return to_db(final_makeup_gain);
247 void set_final_makeup_gain_auto(bool enabled)
249 std::lock_guard<std::mutex> lock(compressor_mutex);
250 final_makeup_gain_auto = enabled;
253 bool get_final_makeup_gain_auto() const
255 std::lock_guard<std::mutex> lock(compressor_mutex);
256 return final_makeup_gain_auto;
259 void reset_peak(unsigned bus_index);
262 float current_level_dbfs[2]; // Digital peak of last frame, left and right.
263 float peak_level_dbfs[2]; // Digital peak with hold, left and right.
264 float historic_peak_dbfs;
265 float gain_staging_db;
266 float compressor_attenuation_db; // A positive number; 0.0 for no attenuation.
269 typedef std::function<void(float level_lufs, float peak_db,
270 std::vector<BusLevel> bus_levels,
271 float global_level_lufs, float range_low_lufs, float range_high_lufs,
272 float final_makeup_gain_db,
273 float correlation)> audio_level_callback_t;
274 void set_audio_level_callback(audio_level_callback_t callback)
276 audio_level_callback = callback;
279 typedef std::function<void()> state_changed_callback_t;
280 void set_state_changed_callback(state_changed_callback_t callback)
282 state_changed_callback = callback;
285 state_changed_callback_t get_state_changed_callback() const
287 return state_changed_callback;
290 void trigger_state_changed_callback()
292 if (state_changed_callback != nullptr) {
293 state_changed_callback();
297 // A combination of all settings for a bus. Useful if you want to get
298 // or store them as a whole without bothering to call all of the get_*
299 // or set_* functions for that bus.
301 float fader_volume_db;
305 float eq_level_db[NUM_EQ_BANDS];
306 float gain_staging_db;
307 bool level_compressor_enabled;
308 float compressor_threshold_dbfs;
309 bool compressor_enabled;
311 static BusSettings get_default_bus_settings();
312 BusSettings get_bus_settings(unsigned bus_index) const;
313 void set_bus_settings(unsigned bus_index, const BusSettings &settings);
317 std::unique_ptr<ResamplingQueue> resampling_queue;
318 std::string display_name;
319 unsigned capture_frequency = OUTPUT_FREQUENCY;
320 // Which channels we consider interesting (ie., are part of some input_mapping).
321 std::set<unsigned> interesting_channels;
322 bool silenced = false;
325 const AudioDevice *find_audio_device(DeviceSpec device_spec) const
327 return const_cast<AudioMixer *>(this)->find_audio_device(device_spec);
330 AudioDevice *find_audio_device(DeviceSpec device_spec);
332 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);
333 void fill_audio_bus(const std::map<DeviceSpec, std::vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float stereo_width, float *output);
334 void reset_resampler_mutex_held(DeviceSpec device_spec);
335 void apply_eq(unsigned bus_index, std::vector<float> *samples_bus);
336 void update_meters(const std::vector<float> &samples);
337 void add_bus_to_master(unsigned bus_index, const std::vector<float> &samples_bus, std::vector<float> *samples_out);
338 void measure_bus_levels(unsigned bus_index, const std::vector<float> &left, const std::vector<float> &right);
339 void send_audio_level_callback();
340 std::vector<DeviceSpec> get_active_devices() const;
341 void set_input_mapping_lock_held(const InputMapping &input_mapping);
343 unsigned num_capture_cards, num_ffmpeg_inputs;
345 mutable std::timed_mutex audio_mutex;
348 AudioDevice video_cards[MAX_VIDEO_CARDS]; // Under audio_mutex.
349 AudioDevice alsa_inputs[MAX_ALSA_CARDS]; // Under audio_mutex.
350 std::unique_ptr<AudioDevice[]> ffmpeg_inputs; // Under audio_mutex.
352 std::atomic<float> locut_cutoff_hz{120};
353 StereoFilter locut[MAX_BUSES]; // Default cutoff 120 Hz, 24 dB/oct.
354 std::atomic<bool> locut_enabled[MAX_BUSES];
355 StereoFilter eq[MAX_BUSES][NUM_EQ_BANDS]; // The one for EQBand::MID isn't actually used (see comments in apply_eq()).
357 // First compressor; takes us up to about -12 dBFS.
358 mutable std::mutex compressor_mutex;
359 std::unique_ptr<StereoCompressor> level_compressor[MAX_BUSES]; // Under compressor_mutex. Used to set/override gain_staging_db if <level_compressor_enabled>.
360 float gain_staging_db[MAX_BUSES]; // Under compressor_mutex.
361 float last_gain_staging_db[MAX_BUSES]; // Under compressor_mutex.
362 bool level_compressor_enabled[MAX_BUSES]; // Under compressor_mutex.
364 static constexpr float ref_level_dbfs = -14.0f; // Chosen so that we end up around 0 LU in practice.
365 static constexpr float ref_level_lufs = -23.0f; // 0 LU, more or less by definition.
367 StereoCompressor limiter;
368 std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 4.0f}; // 4 dB.
369 std::atomic<bool> limiter_enabled{true};
370 std::unique_ptr<StereoCompressor> compressor[MAX_BUSES];
371 std::atomic<float> compressor_threshold_dbfs[MAX_BUSES];
372 std::atomic<bool> compressor_enabled[MAX_BUSES];
374 // Note: The values here are not in dB.
376 float current_level = 0.0f; // Peak of the last frame.
377 float historic_peak = 0.0f; // Highest peak since last reset; no falloff.
378 float current_peak = 0.0f; // Current peak of the peak meter.
379 float last_peak = 0.0f;
380 float age_seconds = 0.0f; // Time since "last_peak" was set.
382 PeakHistory peak_history[MAX_BUSES][2]; // Separate for each channel. Under audio_mutex.
384 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.
385 bool final_makeup_gain_auto = true; // Under compressor_mutex.
387 MappingMode current_mapping_mode; // Under audio_mutex.
388 InputMapping input_mapping; // Under audio_mutex.
389 std::atomic<float> fader_volume_db[MAX_BUSES] {{ 0.0f }};
390 std::atomic<bool> mute[MAX_BUSES] {{ false }};
391 float last_fader_volume_db[MAX_BUSES] { 0.0f }; // Under audio_mutex.
392 std::atomic<float> stereo_width[MAX_BUSES] {{ 0.0f }}; // Default 1.0f (is set in constructor).
393 std::atomic<float> eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{{ 0.0f }}};
394 float last_eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{ 0.0f }};
396 audio_level_callback_t audio_level_callback = nullptr;
397 state_changed_callback_t state_changed_callback = nullptr;
398 mutable std::mutex audio_measure_mutex;
399 Ebu_r128_proc r128; // Under audio_measure_mutex.
400 CorrelationMeasurer correlation; // Under audio_measure_mutex.
401 Resampler peak_resampler; // Under audio_measure_mutex.
402 std::atomic<float> peak{0.0f};
405 std::atomic<double> metric_audio_loudness_short_lufs{0.0 / 0.0};
406 std::atomic<double> metric_audio_loudness_integrated_lufs{0.0 / 0.0};
407 std::atomic<double> metric_audio_loudness_range_low_lufs{0.0 / 0.0};
408 std::atomic<double> metric_audio_loudness_range_high_lufs{0.0 / 0.0};
409 std::atomic<double> metric_audio_peak_dbfs{0.0 / 0.0};
410 std::atomic<double> metric_audio_final_makeup_gain_db{0.0};
411 std::atomic<double> metric_audio_correlation{0.0};
413 // These are all gauges corresponding to the elements of BusLevel.
414 // In a sense, they'd probably do better as histograms, but that's an
415 // awful lot of time series when you have many buses.
417 std::vector<std::pair<std::string, std::string>> labels;
418 std::atomic<double> current_level_dbfs[2]{{0.0/0.0},{0.0/0.0}};
419 std::atomic<double> peak_level_dbfs[2]{{0.0/0.0},{0.0/0.0}};
420 std::atomic<double> historic_peak_dbfs{0.0/0.0};
421 std::atomic<double> gain_staging_db{0.0/0.0};
422 std::atomic<double> compressor_attenuation_db{0.0/0.0};
424 std::unique_ptr<BusMetrics[]> bus_metrics; // One for each bus in <input_mapping>.
427 extern AudioMixer *global_audio_mixer;
429 #endif // !defined(_AUDIO_MIXER_H)