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;
55 static inline uint64_t DeviceSpec_to_key(const DeviceSpec &device_spec)
57 return (uint64_t(device_spec.type) << 32) | device_spec.index;
60 static inline DeviceSpec key_to_DeviceSpec(uint64_t key)
62 return DeviceSpec{ InputSourceType(key >> 32), unsigned(key & 0xffffffff) };
69 int source_channel[2] { -1, -1 }; // Left and right. -1 = none.
72 std::vector<Bus> buses;
77 AudioMixer(unsigned num_cards);
79 void reset_resampler(DeviceSpec device_spec);
82 // Add audio (or silence) to the given device's queue. Can return false if
83 // the lock wasn't successfully taken; if so, you should simply try again.
84 // (This is to avoid a deadlock where a card hangs on the mutex in add_audio()
85 // while we are trying to shut it down from another thread that also holds
86 // the mutex.) frame_length is in TIMEBASE units.
87 bool add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length);
88 bool add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length);
90 std::vector<float> get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
92 void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; }
93 std::map<DeviceSpec, DeviceInfo> get_devices() const;
94 void set_name(DeviceSpec device_spec, const std::string &name);
96 void set_input_mapping(const InputMapping &input_mapping);
97 InputMapping get_input_mapping() const;
99 void set_locut_cutoff(float cutoff_hz)
101 locut_cutoff_hz = cutoff_hz;
104 void set_locut_enabled(bool enabled)
106 locut_enabled = enabled;
109 bool get_locut_enabled() const
111 return locut_enabled;
114 float get_limiter_threshold_dbfs() const
116 return limiter_threshold_dbfs;
119 float get_compressor_threshold_dbfs() const
121 return compressor_threshold_dbfs;
124 void set_limiter_threshold_dbfs(float threshold_dbfs)
126 limiter_threshold_dbfs = threshold_dbfs;
129 void set_compressor_threshold_dbfs(float threshold_dbfs)
131 compressor_threshold_dbfs = threshold_dbfs;
134 void set_limiter_enabled(bool enabled)
136 limiter_enabled = enabled;
139 bool get_limiter_enabled() const
141 return limiter_enabled;
144 void set_compressor_enabled(bool enabled)
146 compressor_enabled = enabled;
149 bool get_compressor_enabled() const
151 return compressor_enabled;
154 void set_gain_staging_db(float gain_db)
156 std::unique_lock<std::mutex> lock(compressor_mutex);
157 level_compressor_enabled = false;
158 gain_staging_db = gain_db;
161 float get_gain_staging_db() const
163 std::unique_lock<std::mutex> lock(compressor_mutex);
164 return gain_staging_db;
167 void set_gain_staging_auto(bool enabled)
169 std::unique_lock<std::mutex> lock(compressor_mutex);
170 level_compressor_enabled = enabled;
173 bool get_gain_staging_auto() const
175 std::unique_lock<std::mutex> lock(compressor_mutex);
176 return level_compressor_enabled;
179 void set_final_makeup_gain_db(float gain_db)
181 std::unique_lock<std::mutex> lock(compressor_mutex);
182 final_makeup_gain_auto = false;
183 final_makeup_gain = from_db(gain_db);
186 float get_final_makeup_gain_db()
188 std::unique_lock<std::mutex> lock(compressor_mutex);
189 return to_db(final_makeup_gain);
192 void set_final_makeup_gain_auto(bool enabled)
194 std::unique_lock<std::mutex> lock(compressor_mutex);
195 final_makeup_gain_auto = enabled;
198 bool get_final_makeup_gain_auto() const
200 std::unique_lock<std::mutex> lock(compressor_mutex);
201 return final_makeup_gain_auto;
204 typedef std::function<void(float level_lufs, float peak_db,
205 float global_level_lufs, float range_low_lufs, float range_high_lufs,
206 float gain_staging_db, float final_makeup_gain_db,
207 float correlation)> audio_level_callback_t;
208 void set_audio_level_callback(audio_level_callback_t callback)
210 audio_level_callback = callback;
215 std::unique_ptr<ResamplingQueue> resampling_queue;
216 int64_t next_local_pts = 0;
218 unsigned capture_frequency = OUTPUT_FREQUENCY;
219 // Which channels we consider interesting (ie., are part of some input_mapping).
220 std::set<unsigned> interesting_channels;
221 // Only used for ALSA cards, obviously.
222 std::unique_ptr<ALSAInput> alsa_device;
224 AudioDevice *find_audio_device(DeviceSpec device_spec);
226 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);
227 void fill_audio_bus(const std::map<DeviceSpec, std::vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output);
228 void reset_resampler_mutex_held(DeviceSpec device_spec);
229 void reset_alsa_mutex_held(DeviceSpec device_spec);
230 std::map<DeviceSpec, DeviceInfo> get_devices_mutex_held() const;
231 void update_meters(const std::vector<float> &samples);
232 void send_audio_level_callback();
236 mutable std::timed_mutex audio_mutex;
238 AudioDevice video_cards[MAX_VIDEO_CARDS]; // Under audio_mutex.
240 // TODO: Figure out a better way to unify these two, as they are sharing indexing.
241 AudioDevice alsa_inputs[MAX_ALSA_CARDS]; // Under audio_mutex.
242 std::vector<ALSAInput::Device> available_alsa_cards;
244 StereoFilter locut; // Default cutoff 120 Hz, 24 dB/oct.
245 std::atomic<float> locut_cutoff_hz;
246 std::atomic<bool> locut_enabled{true};
248 // First compressor; takes us up to about -12 dBFS.
249 mutable std::mutex compressor_mutex;
250 StereoCompressor level_compressor; // Under compressor_mutex. Used to set/override gain_staging_db if <level_compressor_enabled>.
251 float gain_staging_db = 0.0f; // Under compressor_mutex.
252 bool level_compressor_enabled = true; // Under compressor_mutex.
254 static constexpr float ref_level_dbfs = -14.0f; // Chosen so that we end up around 0 LU in practice.
255 static constexpr float ref_level_lufs = -23.0f; // 0 LU, more or less by definition.
257 StereoCompressor limiter;
258 std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 4.0f}; // 4 dB.
259 std::atomic<bool> limiter_enabled{true};
260 StereoCompressor compressor;
261 std::atomic<float> compressor_threshold_dbfs{ref_level_dbfs - 12.0f}; // -12 dB.
262 std::atomic<bool> compressor_enabled{true};
264 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.
265 bool final_makeup_gain_auto = true; // Under compressor_mutex.
267 InputMapping input_mapping; // Under audio_mutex.
268 std::atomic<float> fader_volume_db[MAX_BUSES] {{ 0.0f }};
270 audio_level_callback_t audio_level_callback = nullptr;
271 mutable std::mutex audio_measure_mutex;
272 Ebu_r128_proc r128; // Under audio_measure_mutex.
273 CorrelationMeasurer correlation; // Under audio_measure_mutex.
274 Resampler peak_resampler; // Under audio_measure_mutex.
275 std::atomic<float> peak{0.0f};
278 #endif // !defined(_AUDIO_MIXER_H)