X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=audio_mixer.h;h=2e73ebd7a1e7e68ff29dee3024dbc90b8911b316;hb=c558d684af1bcbbca87207ede74071850c835b7c;hp=88c2f880cac0fee500d087abec37d1ff9eed1230;hpb=c3a08ff6100840205d295a58d6bf340aa20afde0;p=nageru diff --git a/audio_mixer.h b/audio_mixer.h index 88c2f88..2e73ebd 100644 --- a/audio_mixer.h +++ b/audio_mixer.h @@ -52,6 +52,13 @@ struct DeviceInfo { unsigned num_channels; }; +enum EQBand { + EQ_BAND_BASS = 0, + EQ_BAND_MID, + EQ_BAND_TREBLE, + NUM_EQ_BANDS +}; + static inline uint64_t DeviceSpec_to_key(const DeviceSpec &device_spec) { return (uint64_t(device_spec.type) << 32) | device_spec.index; @@ -75,7 +82,6 @@ struct InputMapping { class AudioMixer { public: AudioMixer(unsigned num_cards); - ~AudioMixer(); void reset_resampler(DeviceSpec device_spec); void reset_meters(); @@ -87,10 +93,28 @@ public: bool add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length); bool add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length); + // 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(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy); void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; } - std::map get_devices() const; + + // 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); + } + void set_name(DeviceSpec device_spec, const std::string &name); void set_input_mapping(const InputMapping &input_mapping); @@ -101,6 +125,11 @@ public: 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 +140,18 @@ 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; @@ -221,6 +262,24 @@ 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(); + } + } + private: struct AudioDevice { std::unique_ptr resampling_queue; @@ -229,34 +288,38 @@ private: 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 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; unsigned num_cards; 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::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; @@ -290,8 +353,10 @@ private: InputMapping input_mapping; // Under audio_mutex. std::atomic fader_volume_db[MAX_BUSES] {{ 0.0f }}; float last_fader_volume_db[MAX_BUSES] { 0.0f }; // Under audio_mutex. + std::atomic 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. @@ -299,4 +364,6 @@ private: std::atomic peak{0.0f}; }; +extern AudioMixer *global_audio_mixer; + #endif // !defined(_AUDIO_MIXER_H)