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 "resampling_queue.h"
29 #include "stereocompressor.h"
35 enum class InputSourceType { SILENCE, CAPTURE_CARD, ALSA_INPUT };
40 bool operator== (const DeviceSpec &other) const {
41 return type == other.type && index == other.index;
44 bool operator< (const DeviceSpec &other) const {
45 if (type != other.type)
46 return type < other.type;
47 return index < other.index;
52 unsigned num_channels;
62 static inline uint64_t DeviceSpec_to_key(const DeviceSpec &device_spec)
64 return (uint64_t(device_spec.type) << 32) | device_spec.index;
67 static inline DeviceSpec key_to_DeviceSpec(uint64_t key)
69 return DeviceSpec{ InputSourceType(key >> 32), unsigned(key & 0xffffffff) };
76 int source_channel[2] { -1, -1 }; // Left and right. -1 = none.
79 std::vector<Bus> buses;
84 AudioMixer(unsigned num_cards);
85 void reset_resampler(DeviceSpec device_spec);
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);
96 std::vector<float> get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
98 void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; }
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();
105 void set_name(DeviceSpec device_spec, const std::string &name);
107 void set_input_mapping(const InputMapping &input_mapping);
108 InputMapping get_input_mapping() const;
110 void set_locut_cutoff(float cutoff_hz)
112 locut_cutoff_hz = cutoff_hz;
115 float get_locut_cutoff() const
117 return locut_cutoff_hz;
120 void set_locut_enabled(unsigned bus, bool enabled)
122 locut_enabled[bus] = enabled;
125 bool get_locut_enabled(unsigned bus)
127 return locut_enabled[bus];
130 void set_eq(unsigned bus_index, EQBand band, float db_gain)
132 assert(band >= 0 && band < NUM_EQ_BANDS);
133 eq_level_db[bus_index][band] = db_gain;
136 float get_eq(unsigned bus_index, EQBand band) const
138 assert(band >= 0 && band < NUM_EQ_BANDS);
139 return eq_level_db[bus_index][band];
142 float get_limiter_threshold_dbfs() const
144 return limiter_threshold_dbfs;
147 float get_compressor_threshold_dbfs(unsigned bus_index) const
149 return compressor_threshold_dbfs[bus_index];
152 void set_limiter_threshold_dbfs(float threshold_dbfs)
154 limiter_threshold_dbfs = threshold_dbfs;
157 void set_compressor_threshold_dbfs(unsigned bus_index, float threshold_dbfs)
159 compressor_threshold_dbfs[bus_index] = threshold_dbfs;
162 void set_limiter_enabled(bool enabled)
164 limiter_enabled = enabled;
167 bool get_limiter_enabled() const
169 return limiter_enabled;
172 void set_compressor_enabled(unsigned bus_index, bool enabled)
174 compressor_enabled[bus_index] = enabled;
177 bool get_compressor_enabled(unsigned bus_index) const
179 return compressor_enabled[bus_index];
182 void set_gain_staging_db(unsigned bus_index, float gain_db)
184 std::unique_lock<std::mutex> lock(compressor_mutex);
185 level_compressor_enabled[bus_index] = false;
186 gain_staging_db[bus_index] = gain_db;
189 float get_gain_staging_db(unsigned bus_index) const
191 std::unique_lock<std::mutex> lock(compressor_mutex);
192 return gain_staging_db[bus_index];
195 void set_gain_staging_auto(unsigned bus_index, bool enabled)
197 std::unique_lock<std::mutex> lock(compressor_mutex);
198 level_compressor_enabled[bus_index] = enabled;
201 bool get_gain_staging_auto(unsigned bus_index) const
203 std::unique_lock<std::mutex> lock(compressor_mutex);
204 return level_compressor_enabled[bus_index];
207 void set_final_makeup_gain_db(float gain_db)
209 std::unique_lock<std::mutex> lock(compressor_mutex);
210 final_makeup_gain_auto = false;
211 final_makeup_gain = from_db(gain_db);
214 float get_final_makeup_gain_db()
216 std::unique_lock<std::mutex> lock(compressor_mutex);
217 return to_db(final_makeup_gain);
220 void set_final_makeup_gain_auto(bool enabled)
222 std::unique_lock<std::mutex> lock(compressor_mutex);
223 final_makeup_gain_auto = enabled;
226 bool get_final_makeup_gain_auto() const
228 std::unique_lock<std::mutex> lock(compressor_mutex);
229 return final_makeup_gain_auto;
232 void reset_peak(unsigned bus_index);
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.
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)
249 audio_level_callback = callback;
254 std::unique_ptr<ResamplingQueue> resampling_queue;
255 int64_t next_local_pts = 0;
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;
262 const AudioDevice *find_audio_device(DeviceSpec device_spec) const
264 return const_cast<AudioMixer *>(this)->find_audio_device(device_spec);
267 AudioDevice *find_audio_device(DeviceSpec device_spec);
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;
281 mutable std::timed_mutex audio_mutex;
284 AudioDevice video_cards[MAX_VIDEO_CARDS]; // Under audio_mutex.
285 AudioDevice alsa_inputs[MAX_ALSA_CARDS]; // Under audio_mutex.
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()).
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.
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.
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];
308 // Note: The values here are not in dB.
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.
316 PeakHistory peak_history[MAX_BUSES][2]; // Separate for each channel. Under audio_mutex.
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.
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 }}};
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};
334 extern AudioMixer *global_audio_mixer;
336 #endif // !defined(_AUDIO_MIXER_H)