/*************************************************************************/ /* audio_server.h */ /*************************************************************************/ /* This file is part of: */ /* GODOT ENGINE */ /* https://godotengine.org */ /*************************************************************************/ /* Copyright (c) 2007-2021 Juan Linietsky, Ariel Manzur. */ /* Copyright (c) 2014-2021 Godot Engine contributors (cf. AUTHORS.md). */ /* */ /* Permission is hereby granted, free of charge, to any person obtaining */ /* a copy of this software and associated documentation files (the */ /* "Software"), to deal in the Software without restriction, including */ /* without limitation the rights to use, copy, modify, merge, publish, */ /* distribute, sublicense, and/or sell copies of the Software, and to */ /* permit persons to whom the Software is furnished to do so, subject to */ /* the following conditions: */ /* */ /* The above copyright notice and this permission notice shall be */ /* included in all copies or substantial portions of the Software. */ /* */ /* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */ /* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */ /* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/ /* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */ /* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */ /* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */ /* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */ /*************************************************************************/ #ifndef AUDIO_SERVER_H #define AUDIO_SERVER_H #include "core/math/audio_frame.h" #include "core/object/class_db.h" #include "core/os/os.h" #include "core/templates/safe_list.h" #include "core/variant/variant.h" #include "servers/audio/audio_effect.h" #include "servers/audio/audio_filter_sw.h" #include <atomic> class AudioDriverDummy; class AudioStream; class AudioStreamSample; class AudioStreamPlayback; class AudioDriver { static AudioDriver *singleton; uint64_t _last_mix_time; uint64_t _last_mix_frames; #ifdef DEBUG_ENABLED uint64_t prof_ticks; uint64_t prof_time; #endif protected: Vector<int32_t> input_buffer; unsigned int input_position; unsigned int input_size; void audio_server_process(int p_frames, int32_t *p_buffer, bool p_update_mix_time = true); void update_mix_time(int p_frames); void input_buffer_init(int driver_buffer_frames); void input_buffer_write(int32_t sample); #ifdef DEBUG_ENABLED _FORCE_INLINE_ void start_counting_ticks() { prof_ticks = OS::get_singleton()->get_ticks_usec(); } _FORCE_INLINE_ void stop_counting_ticks() { prof_time += OS::get_singleton()->get_ticks_usec() - prof_ticks; } #else _FORCE_INLINE_ void start_counting_ticks() {} _FORCE_INLINE_ void stop_counting_ticks() {} #endif public: double get_time_since_last_mix(); //useful for video -> audio sync double get_time_to_next_mix(); enum SpeakerMode { SPEAKER_MODE_STEREO, SPEAKER_SURROUND_31, SPEAKER_SURROUND_51, SPEAKER_SURROUND_71, }; static AudioDriver *get_singleton(); void set_singleton(); virtual const char *get_name() const = 0; virtual Error init() = 0; virtual void start() = 0; virtual int get_mix_rate() const = 0; virtual SpeakerMode get_speaker_mode() const = 0; virtual Array get_device_list(); virtual String get_device(); virtual void set_device(String device) {} virtual void lock() = 0; virtual void unlock() = 0; virtual void finish() = 0; virtual Error capture_start() { return FAILED; } virtual Error capture_stop() { return FAILED; } virtual void capture_set_device(const String &p_name) {} virtual String capture_get_device() { return "Default"; } virtual Array capture_get_device_list(); // TODO: convert this and get_device_list to PackedStringArray virtual float get_latency() { return 0; } SpeakerMode get_speaker_mode_by_total_channels(int p_channels) const; int get_total_channels_by_speaker_mode(SpeakerMode) const; Vector<int32_t> get_input_buffer() { return input_buffer; } unsigned int get_input_position() { return input_position; } unsigned int get_input_size() { return input_size; } #ifdef DEBUG_ENABLED uint64_t get_profiling_time() const { return prof_time; } void reset_profiling_time() { prof_time = 0; } #endif AudioDriver(); virtual ~AudioDriver() {} }; class AudioDriverManager { enum { MAX_DRIVERS = 10 }; static const int DEFAULT_MIX_RATE = 44100; static const int DEFAULT_OUTPUT_LATENCY = 15; static AudioDriver *drivers[MAX_DRIVERS]; static int driver_count; static AudioDriverDummy dummy_driver; public: static void add_driver(AudioDriver *p_driver); static void initialize(int p_driver); static int get_driver_count(); static AudioDriver *get_driver(int p_driver); }; class AudioBusLayout; class AudioServer : public Object { GDCLASS(AudioServer, Object); public: //re-expose this here, as AudioDriver is not exposed to script enum SpeakerMode { SPEAKER_MODE_STEREO, SPEAKER_SURROUND_31, SPEAKER_SURROUND_51, SPEAKER_SURROUND_71, }; enum { AUDIO_DATA_INVALID_ID = -1, MAX_CHANNELS_PER_BUS = 4, MAX_BUSES_PER_PLAYBACK = 6, LOOKAHEAD_BUFFER_SIZE = 32, }; typedef void (*AudioCallback)(void *p_userdata); private: uint64_t mix_time; int mix_size; uint32_t buffer_size; uint64_t mix_count; uint64_t mix_frames; #ifdef DEBUG_ENABLED uint64_t prof_time; #endif float channel_disable_threshold_db; uint32_t channel_disable_frames; int channel_count; int to_mix; float playback_speed_scale; struct Bus { StringName name; bool solo; bool mute; bool bypass; bool soloed; //Each channel is a stereo pair. struct Channel { bool used; bool active; AudioFrame peak_volume; Vector<AudioFrame> buffer; Vector<Ref<AudioEffectInstance>> effect_instances; uint64_t last_mix_with_audio; Channel() { last_mix_with_audio = 0; used = false; active = false; peak_volume = AudioFrame(AUDIO_MIN_PEAK_DB, AUDIO_MIN_PEAK_DB); } }; Vector<Channel> channels; struct Effect { Ref<AudioEffect> effect; bool enabled; #ifdef DEBUG_ENABLED uint64_t prof_time; #endif }; Vector<Effect> effects; float volume_db; StringName send; int index_cache; }; struct AudioStreamPlaybackBusDetails { bool bus_active[MAX_BUSES_PER_PLAYBACK] = { false, false, false, false, false, false }; StringName bus[MAX_BUSES_PER_PLAYBACK]; AudioFrame volume[MAX_BUSES_PER_PLAYBACK][MAX_CHANNELS_PER_BUS]; }; struct AudioStreamPlaybackListNode { enum PlaybackState { PAUSED = 0, // Paused. Keep this stream playback around though so it can be restarted. PLAYING = 1, // Playing. Fading may still be necessary if volume changes! FADE_OUT_TO_PAUSE = 2, // About to pause. FADE_OUT_TO_DELETION = 3, // About to stop. AWAITING_DELETION = 4, }; // If zero or positive, a place in the stream to seek to during the next mix. SafeNumeric<float> setseek; SafeNumeric<float> pitch_scale; SafeNumeric<float> highshelf_gain; SafeNumeric<float> attenuation_filter_cutoff_hz; // This isn't used unless highshelf_gain is nonzero. AudioFilterSW::Processor filter_process[8]; // Updating this ref after the list node is created breaks consistency guarantees, don't do it! Ref<AudioStreamPlayback> stream_playback; // Playback state determines the fate of a particular AudioStreamListNode during the mix step. Must be atomically replaced. std::atomic<PlaybackState> state = AWAITING_DELETION; // This data should only ever be modified by an atomic replacement of the pointer. std::atomic<AudioStreamPlaybackBusDetails *> bus_details = nullptr; // Previous bus details should only be accessed on the audio thread. AudioStreamPlaybackBusDetails *prev_bus_details = nullptr; // The next few samples are stored here so we have some time to fade audio out if it ends abruptly at the beginning of the next mix. AudioFrame lookahead[LOOKAHEAD_BUFFER_SIZE]; }; SafeList<AudioStreamPlaybackListNode *> playback_list; SafeList<AudioStreamPlaybackBusDetails *> bus_details_graveyard; Vector<Vector<AudioFrame>> temp_buffer; //temp_buffer for each level Vector<AudioFrame> mix_buffer; Vector<Bus *> buses; Map<StringName, Bus *> bus_map; void _update_bus_effects(int p_bus); static AudioServer *singleton; void init_channels_and_buffers(); void _mix_step(); void _mix_step_for_channel(AudioFrame *p_out_buf, AudioFrame *p_source_buf, AudioFrame p_vol_start, AudioFrame p_vol_final, float p_attenuation_filter_cutoff_hz, float p_highshelf_gain, AudioFilterSW::Processor *p_processor_l, AudioFilterSW::Processor *p_processor_r); // Should only be called on the main thread. AudioStreamPlaybackListNode *_find_playback_list_node(Ref<AudioStreamPlayback> p_playback); struct CallbackItem { AudioCallback callback; void *userdata; }; SafeList<CallbackItem *> update_callback_list; SafeList<CallbackItem *> mix_callback_list; SafeList<CallbackItem *> listener_changed_callback_list; friend class AudioDriver; void _driver_process(int p_frames, int32_t *p_buffer); protected: static void _bind_methods(); public: _FORCE_INLINE_ int get_channel_count() const { switch (get_speaker_mode()) { case SPEAKER_MODE_STEREO: return 1; case SPEAKER_SURROUND_31: return 2; case SPEAKER_SURROUND_51: return 3; case SPEAKER_SURROUND_71: return 4; } ERR_FAIL_V(1); } //do not use from outside audio thread bool thread_has_channel_mix_buffer(int p_bus, int p_buffer) const; AudioFrame *thread_get_channel_mix_buffer(int p_bus, int p_buffer); int thread_get_mix_buffer_size() const; int thread_find_bus_index(const StringName &p_name); void set_bus_count(int p_count); int get_bus_count() const; void remove_bus(int p_index); void add_bus(int p_at_pos = -1); void move_bus(int p_bus, int p_to_pos); void set_bus_name(int p_bus, const String &p_name); String get_bus_name(int p_bus) const; int get_bus_index(const StringName &p_bus_name) const; int get_bus_channels(int p_bus) const; void set_bus_volume_db(int p_bus, float p_volume_db); float get_bus_volume_db(int p_bus) const; void set_bus_send(int p_bus, const StringName &p_send); StringName get_bus_send(int p_bus) const; void set_bus_solo(int p_bus, bool p_enable); bool is_bus_solo(int p_bus) const; void set_bus_mute(int p_bus, bool p_enable); bool is_bus_mute(int p_bus) const; void set_bus_bypass_effects(int p_bus, bool p_enable); bool is_bus_bypassing_effects(int p_bus) const; void add_bus_effect(int p_bus, const Ref<AudioEffect> &p_effect, int p_at_pos = -1); void remove_bus_effect(int p_bus, int p_effect); int get_bus_effect_count(int p_bus); Ref<AudioEffect> get_bus_effect(int p_bus, int p_effect); Ref<AudioEffectInstance> get_bus_effect_instance(int p_bus, int p_effect, int p_channel = 0); void swap_bus_effects(int p_bus, int p_effect, int p_by_effect); void set_bus_effect_enabled(int p_bus, int p_effect, bool p_enabled); bool is_bus_effect_enabled(int p_bus, int p_effect) const; float get_bus_peak_volume_left_db(int p_bus, int p_channel) const; float get_bus_peak_volume_right_db(int p_bus, int p_channel) const; bool is_bus_channel_active(int p_bus, int p_channel) const; void set_playback_speed_scale(float p_scale); float get_playback_speed_scale() const; void start_playback_stream(Ref<AudioStreamPlayback> p_playback, StringName p_bus, Vector<AudioFrame> p_volume_db_vector, float p_start_time = 0); void start_playback_stream(Ref<AudioStreamPlayback> p_playback, Map<StringName, Vector<AudioFrame>> p_bus_volumes, float p_start_time = 0); void stop_playback_stream(Ref<AudioStreamPlayback> p_playback); void set_playback_bus_exclusive(Ref<AudioStreamPlayback> p_playback, StringName p_bus, Vector<AudioFrame> p_volumes); void set_playback_bus_volumes_linear(Ref<AudioStreamPlayback> p_playback, Map<StringName, Vector<AudioFrame>> p_bus_volumes); void set_playback_all_bus_volumes_linear(Ref<AudioStreamPlayback> p_playback, Vector<AudioFrame> p_volumes); void set_playback_pitch_scale(Ref<AudioStreamPlayback> p_playback, float p_pitch_scale); void set_playback_paused(Ref<AudioStreamPlayback> p_playback, bool p_paused); void set_playback_highshelf_params(Ref<AudioStreamPlayback> p_playback, float p_gain, float p_attenuation_cutoff_hz); bool is_playback_active(Ref<AudioStreamPlayback> p_playback); float get_playback_position(Ref<AudioStreamPlayback> p_playback); bool is_playback_paused(Ref<AudioStreamPlayback> p_playback); uint64_t get_mix_count() const; void notify_listener_changed(); virtual void init(); virtual void finish(); virtual void update(); virtual void load_default_bus_layout(); /* MISC config */ virtual void lock(); virtual void unlock(); virtual SpeakerMode get_speaker_mode() const; virtual float get_mix_rate() const; virtual float read_output_peak_db() const; static AudioServer *get_singleton(); virtual double get_output_latency() const; virtual double get_time_to_next_mix() const; virtual double get_time_since_last_mix() const; void add_listener_changed_callback(AudioCallback p_callback, void *p_userdata); void remove_listener_changed_callback(AudioCallback p_callback, void *p_userdata); void add_update_callback(AudioCallback p_callback, void *p_userdata); void remove_update_callback(AudioCallback p_callback, void *p_userdata); void add_mix_callback(AudioCallback p_callback, void *p_userdata); void remove_mix_callback(AudioCallback p_callback, void *p_userdata); void set_bus_layout(const Ref<AudioBusLayout> &p_bus_layout); Ref<AudioBusLayout> generate_bus_layout() const; Array get_device_list(); String get_device(); void set_device(String device); Array capture_get_device_list(); String capture_get_device(); void capture_set_device(const String &p_name); AudioServer(); virtual ~AudioServer(); }; VARIANT_ENUM_CAST(AudioServer::SpeakerMode) class AudioBusLayout : public Resource { GDCLASS(AudioBusLayout, Resource); friend class AudioServer; struct Bus { StringName name; bool solo; bool mute; bool bypass; struct Effect { Ref<AudioEffect> effect; bool enabled; }; Vector<Effect> effects; float volume_db; StringName send; Bus() { solo = false; mute = false; bypass = false; volume_db = 0; } }; Vector<Bus> buses; protected: bool _set(const StringName &p_name, const Variant &p_value); bool _get(const StringName &p_name, Variant &r_ret) const; void _get_property_list(List<PropertyInfo> *p_list) const; public: AudioBusLayout(); }; typedef AudioServer AS; #endif // AUDIO_SERVER_H