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);
86 void reset_resampler(DeviceSpec device_spec);
89 // Add audio (or silence) to the given device's queue. Can return false if
90 // the lock wasn't successfully taken; if so, you should simply try again.
91 // (This is to avoid a deadlock where a card hangs on the mutex in add_audio()
92 // while we are trying to shut it down from another thread that also holds
93 // the mutex.) frame_length is in TIMEBASE units.
94 bool add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length);
95 bool add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length);
97 std::vector<float> get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
99 void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; }
100 std::map<DeviceSpec, DeviceInfo> get_devices() const;
101 void set_name(DeviceSpec device_spec, const std::string &name);
103 void set_input_mapping(const InputMapping &input_mapping);
104 InputMapping get_input_mapping() const;
106 void set_locut_cutoff(float cutoff_hz)
108 locut_cutoff_hz = cutoff_hz;
111 float get_locut_cutoff() const
113 return locut_cutoff_hz;
116 void set_locut_enabled(unsigned bus, bool enabled)
118 locut_enabled[bus] = enabled;
121 bool get_locut_enabled(unsigned bus)
123 return locut_enabled[bus];
126 void set_eq(unsigned bus_index, EQBand band, float db_gain)
128 assert(band >= 0 && band < NUM_EQ_BANDS);
129 eq_level_db[bus_index][band] = db_gain;
132 float get_eq(unsigned bus_index, EQBand band) const
134 assert(band >= 0 && band < NUM_EQ_BANDS);
135 return eq_level_db[bus_index][band];
138 float get_limiter_threshold_dbfs() const
140 return limiter_threshold_dbfs;
143 float get_compressor_threshold_dbfs(unsigned bus_index) const
145 return compressor_threshold_dbfs[bus_index];
148 void set_limiter_threshold_dbfs(float threshold_dbfs)
150 limiter_threshold_dbfs = threshold_dbfs;
153 void set_compressor_threshold_dbfs(unsigned bus_index, float threshold_dbfs)
155 compressor_threshold_dbfs[bus_index] = threshold_dbfs;
158 void set_limiter_enabled(bool enabled)
160 limiter_enabled = enabled;
163 bool get_limiter_enabled() const
165 return limiter_enabled;
168 void set_compressor_enabled(unsigned bus_index, bool enabled)
170 compressor_enabled[bus_index] = enabled;
173 bool get_compressor_enabled(unsigned bus_index) const
175 return compressor_enabled[bus_index];
178 void set_gain_staging_db(unsigned bus_index, float gain_db)
180 std::unique_lock<std::mutex> lock(compressor_mutex);
181 level_compressor_enabled[bus_index] = false;
182 gain_staging_db[bus_index] = gain_db;
185 float get_gain_staging_db(unsigned bus_index) const
187 std::unique_lock<std::mutex> lock(compressor_mutex);
188 return gain_staging_db[bus_index];
191 void set_gain_staging_auto(unsigned bus_index, bool enabled)
193 std::unique_lock<std::mutex> lock(compressor_mutex);
194 level_compressor_enabled[bus_index] = enabled;
197 bool get_gain_staging_auto(unsigned bus_index) const
199 std::unique_lock<std::mutex> lock(compressor_mutex);
200 return level_compressor_enabled[bus_index];
203 void set_final_makeup_gain_db(float gain_db)
205 std::unique_lock<std::mutex> lock(compressor_mutex);
206 final_makeup_gain_auto = false;
207 final_makeup_gain = from_db(gain_db);
210 float get_final_makeup_gain_db()
212 std::unique_lock<std::mutex> lock(compressor_mutex);
213 return to_db(final_makeup_gain);
216 void set_final_makeup_gain_auto(bool enabled)
218 std::unique_lock<std::mutex> lock(compressor_mutex);
219 final_makeup_gain_auto = enabled;
222 bool get_final_makeup_gain_auto() const
224 std::unique_lock<std::mutex> lock(compressor_mutex);
225 return final_makeup_gain_auto;
228 void reset_peak(unsigned bus_index);
231 float current_level_dbfs[2]; // Digital peak of last frame, left and right.
232 float peak_level_dbfs[2]; // Digital peak with hold, left and right.
233 float historic_peak_dbfs;
234 float gain_staging_db;
235 float compressor_attenuation_db; // A positive number; 0.0 for no attenuation.
238 typedef std::function<void(float level_lufs, float peak_db,
239 std::vector<BusLevel> bus_levels,
240 float global_level_lufs, float range_low_lufs, float range_high_lufs,
241 float final_makeup_gain_db,
242 float correlation)> audio_level_callback_t;
243 void set_audio_level_callback(audio_level_callback_t callback)
245 audio_level_callback = callback;
250 std::unique_ptr<ResamplingQueue> resampling_queue;
251 int64_t next_local_pts = 0;
253 unsigned capture_frequency = OUTPUT_FREQUENCY;
254 // Which channels we consider interesting (ie., are part of some input_mapping).
255 std::set<unsigned> interesting_channels;
256 // Only used for ALSA cards, obviously.
257 std::unique_ptr<ALSAInput> alsa_device;
259 AudioDevice *find_audio_device(DeviceSpec device_spec);
261 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);
262 void fill_audio_bus(const std::map<DeviceSpec, std::vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output);
263 void reset_resampler_mutex_held(DeviceSpec device_spec);
264 void reset_alsa_mutex_held(DeviceSpec device_spec);
265 std::map<DeviceSpec, DeviceInfo> get_devices_mutex_held() const;
266 void apply_eq(unsigned bus_index, std::vector<float> *samples_bus);
267 void update_meters(const std::vector<float> &samples);
268 void add_bus_to_master(unsigned bus_index, const std::vector<float> &samples_bus, std::vector<float> *samples_out);
269 void measure_bus_levels(unsigned bus_index, const std::vector<float> &left, const std::vector<float> &right);
270 void send_audio_level_callback();
274 mutable std::timed_mutex audio_mutex;
276 AudioDevice video_cards[MAX_VIDEO_CARDS]; // Under audio_mutex.
278 // TODO: Figure out a better way to unify these two, as they are sharing indexing.
279 AudioDevice alsa_inputs[MAX_ALSA_CARDS]; // Under audio_mutex.
280 std::vector<ALSAInput::Device> available_alsa_cards;
282 std::atomic<float> locut_cutoff_hz{120};
283 StereoFilter locut[MAX_BUSES]; // Default cutoff 120 Hz, 24 dB/oct.
284 std::atomic<bool> locut_enabled[MAX_BUSES];
285 StereoFilter eq[MAX_BUSES][NUM_EQ_BANDS]; // The one for EQBand::MID isn't actually used (see comments in apply_eq()).
287 // First compressor; takes us up to about -12 dBFS.
288 mutable std::mutex compressor_mutex;
289 std::unique_ptr<StereoCompressor> level_compressor[MAX_BUSES]; // Under compressor_mutex. Used to set/override gain_staging_db if <level_compressor_enabled>.
290 float gain_staging_db[MAX_BUSES]; // Under compressor_mutex.
291 bool level_compressor_enabled[MAX_BUSES]; // Under compressor_mutex.
293 static constexpr float ref_level_dbfs = -14.0f; // Chosen so that we end up around 0 LU in practice.
294 static constexpr float ref_level_lufs = -23.0f; // 0 LU, more or less by definition.
296 StereoCompressor limiter;
297 std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 4.0f}; // 4 dB.
298 std::atomic<bool> limiter_enabled{true};
299 std::unique_ptr<StereoCompressor> compressor[MAX_BUSES];
300 std::atomic<float> compressor_threshold_dbfs[MAX_BUSES];
301 std::atomic<bool> compressor_enabled[MAX_BUSES];
303 // Note: The values here are not in dB.
305 float current_level = 0.0f; // Peak of the last frame.
306 float historic_peak = 0.0f; // Highest peak since last reset; no falloff.
307 float current_peak = 0.0f; // Current peak of the peak meter.
308 float last_peak = 0.0f;
309 float age_seconds = 0.0f; // Time since "last_peak" was set.
311 PeakHistory peak_history[MAX_BUSES][2]; // Separate for each channel. Under audio_mutex.
313 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.
314 bool final_makeup_gain_auto = true; // Under compressor_mutex.
316 InputMapping input_mapping; // Under audio_mutex.
317 std::atomic<float> fader_volume_db[MAX_BUSES] {{ 0.0f }};
318 float last_fader_volume_db[MAX_BUSES] { 0.0f }; // Under audio_mutex.
319 std::atomic<float> eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{{ 0.0f }}};
321 audio_level_callback_t audio_level_callback = nullptr;
322 mutable std::mutex audio_measure_mutex;
323 Ebu_r128_proc r128; // Under audio_measure_mutex.
324 CorrelationMeasurer correlation; // Under audio_measure_mutex.
325 Resampler peak_resampler; // Under audio_measure_mutex.
326 std::atomic<float> peak{0.0f};
329 extern AudioMixer *global_audio_mixer;
331 #endif // !defined(_AUDIO_MIXER_H)