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 // If a given device is offline for whatever reason and cannot deliver audio
97 // (by means of add_audio() or add_silence()), you can call put it in silence mode,
98 // where it will be taken to only output silence. Note that when taking it _out_
99 // of silence mode, the resampler will be reset, so that old audio will not
100 // affect it. Same true/false behavior as add_audio().
101 bool silence_card(DeviceSpec device_spec, bool silence);
103 std::vector<float> get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
105 void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; }
107 // Note: This operation holds all ALSA devices (see ALSAPool::get_devices()).
108 // You will need to call set_input_mapping() to get the hold state correctly,
109 // or every card will be held forever.
110 std::map<DeviceSpec, DeviceInfo> get_devices();
112 // See comments on ALSAPool::get_card_state().
113 ALSAPool::Device::State get_alsa_card_state(unsigned index)
115 return alsa_pool.get_card_state(index);
118 void set_name(DeviceSpec device_spec, const std::string &name);
120 void set_input_mapping(const InputMapping &input_mapping);
121 InputMapping get_input_mapping() const;
123 void set_locut_cutoff(float cutoff_hz)
125 locut_cutoff_hz = cutoff_hz;
128 float get_locut_cutoff() const
130 return locut_cutoff_hz;
133 void set_locut_enabled(unsigned bus, bool enabled)
135 locut_enabled[bus] = enabled;
138 bool get_locut_enabled(unsigned bus)
140 return locut_enabled[bus];
143 void set_eq(unsigned bus_index, EQBand band, float db_gain)
145 assert(band >= 0 && band < NUM_EQ_BANDS);
146 eq_level_db[bus_index][band] = db_gain;
149 float get_eq(unsigned bus_index, EQBand band) const
151 assert(band >= 0 && band < NUM_EQ_BANDS);
152 return eq_level_db[bus_index][band];
155 float get_limiter_threshold_dbfs() const
157 return limiter_threshold_dbfs;
160 float get_compressor_threshold_dbfs(unsigned bus_index) const
162 return compressor_threshold_dbfs[bus_index];
165 void set_limiter_threshold_dbfs(float threshold_dbfs)
167 limiter_threshold_dbfs = threshold_dbfs;
170 void set_compressor_threshold_dbfs(unsigned bus_index, float threshold_dbfs)
172 compressor_threshold_dbfs[bus_index] = threshold_dbfs;
175 void set_limiter_enabled(bool enabled)
177 limiter_enabled = enabled;
180 bool get_limiter_enabled() const
182 return limiter_enabled;
185 void set_compressor_enabled(unsigned bus_index, bool enabled)
187 compressor_enabled[bus_index] = enabled;
190 bool get_compressor_enabled(unsigned bus_index) const
192 return compressor_enabled[bus_index];
195 void set_gain_staging_db(unsigned bus_index, float gain_db)
197 std::unique_lock<std::mutex> lock(compressor_mutex);
198 level_compressor_enabled[bus_index] = false;
199 gain_staging_db[bus_index] = gain_db;
202 float get_gain_staging_db(unsigned bus_index) const
204 std::unique_lock<std::mutex> lock(compressor_mutex);
205 return gain_staging_db[bus_index];
208 void set_gain_staging_auto(unsigned bus_index, bool enabled)
210 std::unique_lock<std::mutex> lock(compressor_mutex);
211 level_compressor_enabled[bus_index] = enabled;
214 bool get_gain_staging_auto(unsigned bus_index) const
216 std::unique_lock<std::mutex> lock(compressor_mutex);
217 return level_compressor_enabled[bus_index];
220 void set_final_makeup_gain_db(float gain_db)
222 std::unique_lock<std::mutex> lock(compressor_mutex);
223 final_makeup_gain_auto = false;
224 final_makeup_gain = from_db(gain_db);
227 float get_final_makeup_gain_db()
229 std::unique_lock<std::mutex> lock(compressor_mutex);
230 return to_db(final_makeup_gain);
233 void set_final_makeup_gain_auto(bool enabled)
235 std::unique_lock<std::mutex> lock(compressor_mutex);
236 final_makeup_gain_auto = enabled;
239 bool get_final_makeup_gain_auto() const
241 std::unique_lock<std::mutex> lock(compressor_mutex);
242 return final_makeup_gain_auto;
245 void reset_peak(unsigned bus_index);
248 float current_level_dbfs[2]; // Digital peak of last frame, left and right.
249 float peak_level_dbfs[2]; // Digital peak with hold, left and right.
250 float historic_peak_dbfs;
251 float gain_staging_db;
252 float compressor_attenuation_db; // A positive number; 0.0 for no attenuation.
255 typedef std::function<void(float level_lufs, float peak_db,
256 std::vector<BusLevel> bus_levels,
257 float global_level_lufs, float range_low_lufs, float range_high_lufs,
258 float final_makeup_gain_db,
259 float correlation)> audio_level_callback_t;
260 void set_audio_level_callback(audio_level_callback_t callback)
262 audio_level_callback = callback;
265 typedef std::function<void()> state_changed_callback_t;
266 void set_state_changed_callback(state_changed_callback_t callback)
268 state_changed_callback = callback;
271 state_changed_callback_t get_state_changed_callback() const
273 return state_changed_callback;
276 void trigger_state_changed_callback()
278 if (state_changed_callback != nullptr) {
279 state_changed_callback();
285 std::unique_ptr<ResamplingQueue> resampling_queue;
286 int64_t next_local_pts = 0;
288 unsigned capture_frequency = OUTPUT_FREQUENCY;
289 // Which channels we consider interesting (ie., are part of some input_mapping).
290 std::set<unsigned> interesting_channels;
291 bool silenced = false;
294 const AudioDevice *find_audio_device(DeviceSpec device_spec) const
296 return const_cast<AudioMixer *>(this)->find_audio_device(device_spec);
299 AudioDevice *find_audio_device(DeviceSpec device_spec);
301 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);
302 void fill_audio_bus(const std::map<DeviceSpec, std::vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output);
303 void reset_resampler_mutex_held(DeviceSpec device_spec);
304 void apply_eq(unsigned bus_index, std::vector<float> *samples_bus);
305 void update_meters(const std::vector<float> &samples);
306 void add_bus_to_master(unsigned bus_index, const std::vector<float> &samples_bus, std::vector<float> *samples_out);
307 void measure_bus_levels(unsigned bus_index, const std::vector<float> &left, const std::vector<float> &right);
308 void send_audio_level_callback();
309 std::vector<DeviceSpec> get_active_devices() const;
313 mutable std::timed_mutex audio_mutex;
316 AudioDevice video_cards[MAX_VIDEO_CARDS]; // Under audio_mutex.
317 AudioDevice alsa_inputs[MAX_ALSA_CARDS]; // Under audio_mutex.
319 std::atomic<float> locut_cutoff_hz{120};
320 StereoFilter locut[MAX_BUSES]; // Default cutoff 120 Hz, 24 dB/oct.
321 std::atomic<bool> locut_enabled[MAX_BUSES];
322 StereoFilter eq[MAX_BUSES][NUM_EQ_BANDS]; // The one for EQBand::MID isn't actually used (see comments in apply_eq()).
324 // First compressor; takes us up to about -12 dBFS.
325 mutable std::mutex compressor_mutex;
326 std::unique_ptr<StereoCompressor> level_compressor[MAX_BUSES]; // Under compressor_mutex. Used to set/override gain_staging_db if <level_compressor_enabled>.
327 float gain_staging_db[MAX_BUSES]; // Under compressor_mutex.
328 bool level_compressor_enabled[MAX_BUSES]; // Under compressor_mutex.
330 static constexpr float ref_level_dbfs = -14.0f; // Chosen so that we end up around 0 LU in practice.
331 static constexpr float ref_level_lufs = -23.0f; // 0 LU, more or less by definition.
333 StereoCompressor limiter;
334 std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 4.0f}; // 4 dB.
335 std::atomic<bool> limiter_enabled{true};
336 std::unique_ptr<StereoCompressor> compressor[MAX_BUSES];
337 std::atomic<float> compressor_threshold_dbfs[MAX_BUSES];
338 std::atomic<bool> compressor_enabled[MAX_BUSES];
340 // Note: The values here are not in dB.
342 float current_level = 0.0f; // Peak of the last frame.
343 float historic_peak = 0.0f; // Highest peak since last reset; no falloff.
344 float current_peak = 0.0f; // Current peak of the peak meter.
345 float last_peak = 0.0f;
346 float age_seconds = 0.0f; // Time since "last_peak" was set.
348 PeakHistory peak_history[MAX_BUSES][2]; // Separate for each channel. Under audio_mutex.
350 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.
351 bool final_makeup_gain_auto = true; // Under compressor_mutex.
353 InputMapping input_mapping; // Under audio_mutex.
354 std::atomic<float> fader_volume_db[MAX_BUSES] {{ 0.0f }};
355 float last_fader_volume_db[MAX_BUSES] { 0.0f }; // Under audio_mutex.
356 std::atomic<float> eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{{ 0.0f }}};
358 audio_level_callback_t audio_level_callback = nullptr;
359 state_changed_callback_t state_changed_callback = nullptr;
360 mutable std::mutex audio_measure_mutex;
361 Ebu_r128_proc r128; // Under audio_measure_mutex.
362 CorrelationMeasurer correlation; // Under audio_measure_mutex.
363 Resampler peak_resampler; // Under audio_measure_mutex.
364 std::atomic<float> peak{0.0f};
367 extern AudioMixer *global_audio_mixer;
369 #endif // !defined(_AUDIO_MIXER_H)