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.
19 #include <zita-resampler/resampler.h>
21 #include "alsa_input.h"
22 #include "bmusb/bmusb.h"
23 #include "correlation_measurer.h"
26 #include "ebu_r128_proc.h"
28 #include "input_mapping.h"
29 #include "resampling_queue.h"
30 #include "stereocompressor.h"
45 AudioMixer(unsigned num_cards);
46 void reset_resampler(DeviceSpec device_spec);
49 // Add audio (or silence) to the given device's queue. Can return false if
50 // the lock wasn't successfully taken; if so, you should simply try again.
51 // (This is to avoid a deadlock where a card hangs on the mutex in add_audio()
52 // while we are trying to shut it down from another thread that also holds
53 // the mutex.) frame_length is in TIMEBASE units.
54 bool add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length);
55 bool add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length);
57 // If a given device is offline for whatever reason and cannot deliver audio
58 // (by means of add_audio() or add_silence()), you can call put it in silence mode,
59 // where it will be taken to only output silence. Note that when taking it _out_
60 // of silence mode, the resampler will be reset, so that old audio will not
61 // affect it. Same true/false behavior as add_audio().
62 bool silence_card(DeviceSpec device_spec, bool silence);
64 std::vector<float> get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
66 void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; }
68 // Note: This operation holds all ALSA devices (see ALSAPool::get_devices()).
69 // You will need to call set_input_mapping() to get the hold state correctly,
70 // or every card will be held forever.
71 std::map<DeviceSpec, DeviceInfo> get_devices();
73 // See comments on ALSAPool::get_card_state().
74 ALSAPool::Device::State get_alsa_card_state(unsigned index)
76 return alsa_pool.get_card_state(index);
79 void set_display_name(DeviceSpec device_spec, const std::string &name);
81 void set_input_mapping(const InputMapping &input_mapping);
82 InputMapping get_input_mapping() const;
84 void set_locut_cutoff(float cutoff_hz)
86 locut_cutoff_hz = cutoff_hz;
89 float get_locut_cutoff() const
91 return locut_cutoff_hz;
94 void set_locut_enabled(unsigned bus, bool enabled)
96 locut_enabled[bus] = enabled;
99 bool get_locut_enabled(unsigned bus)
101 return locut_enabled[bus];
104 void set_eq(unsigned bus_index, EQBand band, float db_gain)
106 assert(band >= 0 && band < NUM_EQ_BANDS);
107 eq_level_db[bus_index][band] = db_gain;
110 float get_eq(unsigned bus_index, EQBand band) const
112 assert(band >= 0 && band < NUM_EQ_BANDS);
113 return eq_level_db[bus_index][band];
116 float get_limiter_threshold_dbfs() const
118 return limiter_threshold_dbfs;
121 float get_compressor_threshold_dbfs(unsigned bus_index) const
123 return compressor_threshold_dbfs[bus_index];
126 void set_limiter_threshold_dbfs(float threshold_dbfs)
128 limiter_threshold_dbfs = threshold_dbfs;
131 void set_compressor_threshold_dbfs(unsigned bus_index, float threshold_dbfs)
133 compressor_threshold_dbfs[bus_index] = threshold_dbfs;
136 void set_limiter_enabled(bool enabled)
138 limiter_enabled = enabled;
141 bool get_limiter_enabled() const
143 return limiter_enabled;
146 void set_compressor_enabled(unsigned bus_index, bool enabled)
148 compressor_enabled[bus_index] = enabled;
151 bool get_compressor_enabled(unsigned bus_index) const
153 return compressor_enabled[bus_index];
156 void set_gain_staging_db(unsigned bus_index, float gain_db)
158 std::unique_lock<std::mutex> lock(compressor_mutex);
159 level_compressor_enabled[bus_index] = false;
160 gain_staging_db[bus_index] = gain_db;
163 float get_gain_staging_db(unsigned bus_index) const
165 std::unique_lock<std::mutex> lock(compressor_mutex);
166 return gain_staging_db[bus_index];
169 void set_gain_staging_auto(unsigned bus_index, bool enabled)
171 std::unique_lock<std::mutex> lock(compressor_mutex);
172 level_compressor_enabled[bus_index] = enabled;
175 bool get_gain_staging_auto(unsigned bus_index) const
177 std::unique_lock<std::mutex> lock(compressor_mutex);
178 return level_compressor_enabled[bus_index];
181 void set_final_makeup_gain_db(float gain_db)
183 std::unique_lock<std::mutex> lock(compressor_mutex);
184 final_makeup_gain_auto = false;
185 final_makeup_gain = from_db(gain_db);
188 float get_final_makeup_gain_db()
190 std::unique_lock<std::mutex> lock(compressor_mutex);
191 return to_db(final_makeup_gain);
194 void set_final_makeup_gain_auto(bool enabled)
196 std::unique_lock<std::mutex> lock(compressor_mutex);
197 final_makeup_gain_auto = enabled;
200 bool get_final_makeup_gain_auto() const
202 std::unique_lock<std::mutex> lock(compressor_mutex);
203 return final_makeup_gain_auto;
206 void reset_peak(unsigned bus_index);
209 float current_level_dbfs[2]; // Digital peak of last frame, left and right.
210 float peak_level_dbfs[2]; // Digital peak with hold, left and right.
211 float historic_peak_dbfs;
212 float gain_staging_db;
213 float compressor_attenuation_db; // A positive number; 0.0 for no attenuation.
216 typedef std::function<void(float level_lufs, float peak_db,
217 std::vector<BusLevel> bus_levels,
218 float global_level_lufs, float range_low_lufs, float range_high_lufs,
219 float final_makeup_gain_db,
220 float correlation)> audio_level_callback_t;
221 void set_audio_level_callback(audio_level_callback_t callback)
223 audio_level_callback = callback;
226 typedef std::function<void()> state_changed_callback_t;
227 void set_state_changed_callback(state_changed_callback_t callback)
229 state_changed_callback = callback;
232 state_changed_callback_t get_state_changed_callback() const
234 return state_changed_callback;
237 void trigger_state_changed_callback()
239 if (state_changed_callback != nullptr) {
240 state_changed_callback();
246 std::unique_ptr<ResamplingQueue> resampling_queue;
247 int64_t next_local_pts = 0;
248 std::string display_name;
249 unsigned capture_frequency = OUTPUT_FREQUENCY;
250 // Which channels we consider interesting (ie., are part of some input_mapping).
251 std::set<unsigned> interesting_channels;
252 bool silenced = false;
255 const AudioDevice *find_audio_device(DeviceSpec device_spec) const
257 return const_cast<AudioMixer *>(this)->find_audio_device(device_spec);
260 AudioDevice *find_audio_device(DeviceSpec device_spec);
262 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);
263 void fill_audio_bus(const std::map<DeviceSpec, std::vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output);
264 void reset_resampler_mutex_held(DeviceSpec device_spec);
265 void apply_eq(unsigned bus_index, std::vector<float> *samples_bus);
266 void update_meters(const std::vector<float> &samples);
267 void add_bus_to_master(unsigned bus_index, const std::vector<float> &samples_bus, std::vector<float> *samples_out);
268 void measure_bus_levels(unsigned bus_index, const std::vector<float> &left, const std::vector<float> &right);
269 void send_audio_level_callback();
270 std::vector<DeviceSpec> get_active_devices() const;
274 mutable std::timed_mutex audio_mutex;
277 AudioDevice video_cards[MAX_VIDEO_CARDS]; // Under audio_mutex.
278 AudioDevice alsa_inputs[MAX_ALSA_CARDS]; // Under audio_mutex.
280 std::atomic<float> locut_cutoff_hz{120};
281 StereoFilter locut[MAX_BUSES]; // Default cutoff 120 Hz, 24 dB/oct.
282 std::atomic<bool> locut_enabled[MAX_BUSES];
283 StereoFilter eq[MAX_BUSES][NUM_EQ_BANDS]; // The one for EQBand::MID isn't actually used (see comments in apply_eq()).
285 // First compressor; takes us up to about -12 dBFS.
286 mutable std::mutex compressor_mutex;
287 std::unique_ptr<StereoCompressor> level_compressor[MAX_BUSES]; // Under compressor_mutex. Used to set/override gain_staging_db if <level_compressor_enabled>.
288 float gain_staging_db[MAX_BUSES]; // Under compressor_mutex.
289 bool level_compressor_enabled[MAX_BUSES]; // Under compressor_mutex.
291 static constexpr float ref_level_dbfs = -14.0f; // Chosen so that we end up around 0 LU in practice.
292 static constexpr float ref_level_lufs = -23.0f; // 0 LU, more or less by definition.
294 StereoCompressor limiter;
295 std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 4.0f}; // 4 dB.
296 std::atomic<bool> limiter_enabled{true};
297 std::unique_ptr<StereoCompressor> compressor[MAX_BUSES];
298 std::atomic<float> compressor_threshold_dbfs[MAX_BUSES];
299 std::atomic<bool> compressor_enabled[MAX_BUSES];
301 // Note: The values here are not in dB.
303 float current_level = 0.0f; // Peak of the last frame.
304 float historic_peak = 0.0f; // Highest peak since last reset; no falloff.
305 float current_peak = 0.0f; // Current peak of the peak meter.
306 float last_peak = 0.0f;
307 float age_seconds = 0.0f; // Time since "last_peak" was set.
309 PeakHistory peak_history[MAX_BUSES][2]; // Separate for each channel. Under audio_mutex.
311 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.
312 bool final_makeup_gain_auto = true; // Under compressor_mutex.
314 InputMapping input_mapping; // Under audio_mutex.
315 std::atomic<float> fader_volume_db[MAX_BUSES] {{ 0.0f }};
316 float last_fader_volume_db[MAX_BUSES] { 0.0f }; // Under audio_mutex.
317 std::atomic<float> eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{{ 0.0f }}};
319 audio_level_callback_t audio_level_callback = nullptr;
320 state_changed_callback_t state_changed_callback = nullptr;
321 mutable std::mutex audio_measure_mutex;
322 Ebu_r128_proc r128; // Under audio_measure_mutex.
323 CorrelationMeasurer correlation; // Under audio_measure_mutex.
324 Resampler peak_resampler; // Under audio_measure_mutex.
325 std::atomic<float> peak{0.0f};
328 extern AudioMixer *global_audio_mixer;
330 #endif // !defined(_AUDIO_MIXER_H)