X-Git-Url: https://git.sesse.net/?p=nageru;a=blobdiff_plain;f=audio_mixer.h;h=ebe142a74cbc9e6f81d125505022d88588cb00cd;hp=5d65582c939fa191bcf8efad870d6e1c65b65c25;hb=4e3c52ba57c4552a969e71ccdefd9941ce8d6290;hpb=df76248ec883efbeafdbee35d995d688ec4ac69f diff --git a/audio_mixer.h b/audio_mixer.h index 5d65582..ebe142a 100644 --- a/audio_mixer.h +++ b/audio_mixer.h @@ -8,74 +8,45 @@ // // All operations on AudioMixer (except destruction) are thread-safe. -#include +#include #include +#include #include +#include +#include #include #include #include #include +#include #include -#include -#include "alsa_input.h" -#include "bmusb/bmusb.h" +#include "alsa_pool.h" #include "correlation_measurer.h" #include "db.h" #include "defs.h" #include "ebu_r128_proc.h" #include "filter.h" +#include "input_mapping.h" #include "resampling_queue.h" #include "stereocompressor.h" +class DeviceSpecProto; + namespace bmusb { struct AudioFormat; } // namespace bmusb -enum class InputSourceType { SILENCE, CAPTURE_CARD, ALSA_INPUT }; -struct DeviceSpec { - InputSourceType type; - unsigned index; - - bool operator== (const DeviceSpec &other) const { - return type == other.type && index == other.index; - } - - bool operator< (const DeviceSpec &other) const { - if (type != other.type) - return type < other.type; - return index < other.index; - } -}; -struct DeviceInfo { - std::string name; - unsigned num_channels; -}; - -static inline uint64_t DeviceSpec_to_key(const DeviceSpec &device_spec) -{ - return (uint64_t(device_spec.type) << 32) | device_spec.index; -} - -static inline DeviceSpec key_to_DeviceSpec(uint64_t key) -{ - return DeviceSpec{ InputSourceType(key >> 32), unsigned(key & 0xffffffff) }; -} - -struct InputMapping { - struct Bus { - std::string name; - DeviceSpec device; - int source_channel[2] { -1, -1 }; // Left and right. -1 = none. - }; - - std::vector buses; +enum EQBand { + EQ_BAND_BASS = 0, + EQ_BAND_MID, + EQ_BAND_TREBLE, + NUM_EQ_BANDS }; class AudioMixer { public: - AudioMixer(unsigned num_cards); - ~AudioMixer(); + AudioMixer(unsigned num_capture_cards, unsigned num_ffmpeg_inputs); void reset_resampler(DeviceSpec device_spec); void reset_meters(); @@ -84,23 +55,83 @@ public: // (This is to avoid a deadlock where a card hangs on the mutex in add_audio() // while we are trying to shut it down from another thread that also holds // the mutex.) frame_length is in TIMEBASE units. - bool add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length); + bool add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length, std::chrono::steady_clock::time_point frame_time); bool add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length); - std::vector get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy); + // If a given device is offline for whatever reason and cannot deliver audio + // (by means of add_audio() or add_silence()), you can call put it in silence mode, + // where it will be taken to only output silence. Note that when taking it _out_ + // of silence mode, the resampler will be reset, so that old audio will not + // affect it. Same true/false behavior as add_audio(). + bool silence_card(DeviceSpec device_spec, bool silence); + std::vector get_output(std::chrono::steady_clock::time_point ts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy); + + float get_fader_volume(unsigned bus_index) const { return fader_volume_db[bus_index]; } void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; } - std::map get_devices() const; - void set_name(DeviceSpec device_spec, const std::string &name); + bool get_mute(unsigned bus_index) const { return mute[bus_index]; } + void set_mute(unsigned bus_index, bool muted) { mute[bus_index] = muted; } + + // Note: This operation holds all ALSA devices (see ALSAPool::get_devices()). + // You will need to call set_input_mapping() to get the hold state correctly, + // or every card will be held forever. + std::map get_devices(); + + // See comments on ALSAPool::get_card_state(). + ALSAPool::Device::State get_alsa_card_state(unsigned index) + { + return alsa_pool.get_card_state(index); + } + + // See comments on ALSAPool::create_dead_card(). + DeviceSpec create_dead_card(const std::string &name, const std::string &info, unsigned num_channels) + { + unsigned dead_card_index = alsa_pool.create_dead_card(name, info, num_channels); + return DeviceSpec{InputSourceType::ALSA_INPUT, dead_card_index}; + } + + void set_display_name(DeviceSpec device_spec, const std::string &name); + + // Note: The card should be held (currently this isn't enforced, though). + void serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto); + + enum class MappingMode { + // A single bus, only from a video card (no ALSA devices), + // only channel 1 and 2, locked to +0 dB. Note that this is + // only an UI abstraction around exactly the same audio code + // as MULTICHANNEL; it's just less flexible. + SIMPLE, + + // Full, arbitrary mappings. + MULTICHANNEL + }; + + // Automatically sets mapping mode to MappingMode::SIMPLE. + void set_simple_input(unsigned card_index); + + // If mapping mode is not representable as a MappingMode::SIMPLE type + // mapping, returns numeric_limits::max(). + unsigned get_simple_input() const; + + // Implicitly sets mapping mode to MappingMode::MULTICHANNEL. void set_input_mapping(const InputMapping &input_mapping); + + MappingMode get_mapping_mode() const; InputMapping get_input_mapping() const; + unsigned num_buses() const; + void set_locut_cutoff(float cutoff_hz) { locut_cutoff_hz = cutoff_hz; } + float get_locut_cutoff() const + { + return locut_cutoff_hz; + } + void set_locut_enabled(unsigned bus, bool enabled) { locut_enabled[bus] = enabled; @@ -111,6 +142,18 @@ public: return locut_enabled[bus]; } + void set_eq(unsigned bus_index, EQBand band, float db_gain) + { + assert(band >= 0 && band < NUM_EQ_BANDS); + eq_level_db[bus_index][band] = db_gain; + } + + float get_eq(unsigned bus_index, EQBand band) const + { + assert(band >= 0 && band < NUM_EQ_BANDS); + return eq_level_db[bus_index][band]; + } + float get_limiter_threshold_dbfs() const { return limiter_threshold_dbfs; @@ -201,6 +244,8 @@ public: return final_makeup_gain_auto; } + void reset_peak(unsigned bus_index); + struct BusLevel { float current_level_dbfs[2]; // Digital peak of last frame, left and right. float peak_level_dbfs[2]; // Digital peak with hold, left and right. @@ -219,46 +264,88 @@ public: audio_level_callback = callback; } + typedef std::function state_changed_callback_t; + void set_state_changed_callback(state_changed_callback_t callback) + { + state_changed_callback = callback; + } + + state_changed_callback_t get_state_changed_callback() const + { + return state_changed_callback; + } + + void trigger_state_changed_callback() + { + if (state_changed_callback != nullptr) { + state_changed_callback(); + } + } + + // A combination of all settings for a bus. Useful if you want to get + // or store them as a whole without bothering to call all of the get_* + // or set_* functions for that bus. + struct BusSettings { + float fader_volume_db; + bool muted; + bool locut_enabled; + float eq_level_db[NUM_EQ_BANDS]; + float gain_staging_db; + bool level_compressor_enabled; + float compressor_threshold_dbfs; + bool compressor_enabled; + }; + static BusSettings get_default_bus_settings(); + BusSettings get_bus_settings(unsigned bus_index) const; + void set_bus_settings(unsigned bus_index, const BusSettings &settings); + private: struct AudioDevice { std::unique_ptr resampling_queue; - int64_t next_local_pts = 0; - std::string name; + std::string display_name; unsigned capture_frequency = OUTPUT_FREQUENCY; // Which channels we consider interesting (ie., are part of some input_mapping). std::set interesting_channels; - // Only used for ALSA cards, obviously. - std::unique_ptr alsa_device; + bool silenced = false; }; + + const AudioDevice *find_audio_device(DeviceSpec device_spec) const + { + return const_cast(this)->find_audio_device(device_spec); + } + AudioDevice *find_audio_device(DeviceSpec device_spec); void find_sample_src_from_device(const std::map> &samples_card, DeviceSpec device_spec, int source_channel, const float **srcptr, unsigned *stride); void fill_audio_bus(const std::map> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output); void reset_resampler_mutex_held(DeviceSpec device_spec); - void reset_alsa_mutex_held(DeviceSpec device_spec); - std::map get_devices_mutex_held() const; + void apply_eq(unsigned bus_index, std::vector *samples_bus); void update_meters(const std::vector &samples); - void measure_bus_levels(unsigned bus_index, const std::vector &left, const std::vector &right, float volume); + void add_bus_to_master(unsigned bus_index, const std::vector &samples_bus, std::vector *samples_out); + void measure_bus_levels(unsigned bus_index, const std::vector &left, const std::vector &right); void send_audio_level_callback(); + std::vector get_active_devices() const; + void set_input_mapping_lock_held(const InputMapping &input_mapping); - unsigned num_cards; + unsigned num_capture_cards, num_ffmpeg_inputs; mutable std::timed_mutex audio_mutex; + ALSAPool alsa_pool; AudioDevice video_cards[MAX_VIDEO_CARDS]; // Under audio_mutex. - - // TODO: Figure out a better way to unify these two, as they are sharing indexing. AudioDevice alsa_inputs[MAX_ALSA_CARDS]; // Under audio_mutex. - std::vector available_alsa_cards; + std::unique_ptr ffmpeg_inputs; // Under audio_mutex. - std::atomic locut_cutoff_hz; + std::atomic locut_cutoff_hz{120}; StereoFilter locut[MAX_BUSES]; // Default cutoff 120 Hz, 24 dB/oct. std::atomic locut_enabled[MAX_BUSES]; + StereoFilter eq[MAX_BUSES][NUM_EQ_BANDS]; // The one for EQBand::MID isn't actually used (see comments in apply_eq()). // First compressor; takes us up to about -12 dBFS. mutable std::mutex compressor_mutex; std::unique_ptr level_compressor[MAX_BUSES]; // Under compressor_mutex. Used to set/override gain_staging_db if . float gain_staging_db[MAX_BUSES]; // Under compressor_mutex. + float last_gain_staging_db[MAX_BUSES]; // Under compressor_mutex. bool level_compressor_enabled[MAX_BUSES]; // Under compressor_mutex. static constexpr float ref_level_dbfs = -14.0f; // Chosen so that we end up around 0 LU in practice. @@ -279,20 +366,50 @@ private: float last_peak = 0.0f; float age_seconds = 0.0f; // Time since "last_peak" was set. }; - PeakHistory peak_history[MAX_BUSES][2]; // Separate for each channel. + PeakHistory peak_history[MAX_BUSES][2]; // Separate for each channel. Under audio_mutex. 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. bool final_makeup_gain_auto = true; // Under compressor_mutex. + MappingMode current_mapping_mode; // Under audio_mutex. InputMapping input_mapping; // Under audio_mutex. std::atomic fader_volume_db[MAX_BUSES] {{ 0.0f }}; + std::atomic mute[MAX_BUSES] {{ false }}; + float last_fader_volume_db[MAX_BUSES] { 0.0f }; // Under audio_mutex. + std::atomic eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{{ 0.0f }}}; + float last_eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{ 0.0f }}; audio_level_callback_t audio_level_callback = nullptr; + state_changed_callback_t state_changed_callback = nullptr; mutable std::mutex audio_measure_mutex; Ebu_r128_proc r128; // Under audio_measure_mutex. CorrelationMeasurer correlation; // Under audio_measure_mutex. Resampler peak_resampler; // Under audio_measure_mutex. std::atomic peak{0.0f}; + + // Metrics. + std::atomic metric_audio_loudness_short_lufs{0.0 / 0.0}; + std::atomic metric_audio_loudness_integrated_lufs{0.0 / 0.0}; + std::atomic metric_audio_loudness_range_low_lufs{0.0 / 0.0}; + std::atomic metric_audio_loudness_range_high_lufs{0.0 / 0.0}; + std::atomic metric_audio_peak_dbfs{0.0 / 0.0}; + std::atomic metric_audio_final_makeup_gain_db{0.0}; + std::atomic metric_audio_correlation{0.0}; + + // These are all gauges corresponding to the elements of BusLevel. + // In a sense, they'd probably do better as histograms, but that's an + // awful lot of time series when you have many buses. + struct BusMetrics { + std::vector> labels; + std::atomic current_level_dbfs[2]{{0.0/0.0},{0.0/0.0}}; + std::atomic peak_level_dbfs[2]{{0.0/0.0},{0.0/0.0}}; + std::atomic historic_peak_dbfs{0.0/0.0}; + std::atomic gain_staging_db{0.0/0.0}; + std::atomic compressor_attenuation_db{0.0/0.0}; + }; + std::unique_ptr bus_metrics; // One for each bus in . }; +extern AudioMixer *global_audio_mixer; + #endif // !defined(_AUDIO_MIXER_H)