mirror of
https://github.com/Relintai/pandemonium_engine_minimal.git
synced 2024-11-17 22:17:19 +01:00
450 lines
11 KiB
C++
450 lines
11 KiB
C++
#ifndef AUDIO_SERVER_H
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#define AUDIO_SERVER_H
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/* audio_server.h */
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#include "core/math/audio_frame.h"
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#include "core/object/object.h"
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#include "core/os/os.h"
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#include "core/variant/variant.h"
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#include "servers/audio/audio_effect.h"
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class AudioDriverDummy;
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class AudioStream;
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class AudioStreamSample;
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class AudioDriver {
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static AudioDriver *singleton;
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uint64_t _last_mix_time;
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uint64_t _last_mix_frames;
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#ifdef DEBUG_ENABLED
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uint64_t prof_ticks;
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uint64_t prof_time;
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#endif
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protected:
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Vector<int32_t> input_buffer;
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unsigned int input_position;
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unsigned int input_size;
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void audio_server_process(int p_frames, int32_t *p_buffer, bool p_update_mix_time = true);
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void update_mix_time(int p_frames);
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void input_buffer_init(int driver_buffer_frames);
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void input_buffer_write(int32_t sample);
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#ifdef DEBUG_ENABLED
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_FORCE_INLINE_ void start_counting_ticks() { prof_ticks = OS::get_singleton()->get_ticks_usec(); }
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_FORCE_INLINE_ void stop_counting_ticks() { prof_time += OS::get_singleton()->get_ticks_usec() - prof_ticks; }
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#else
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_FORCE_INLINE_ void start_counting_ticks() {}
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_FORCE_INLINE_ void stop_counting_ticks() {}
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#endif
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public:
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double get_time_since_last_mix(); //useful for video -> audio sync
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double get_time_to_next_mix();
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enum SpeakerMode {
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SPEAKER_MODE_STEREO,
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SPEAKER_SURROUND_31,
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SPEAKER_SURROUND_51,
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SPEAKER_SURROUND_71,
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};
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static AudioDriver *get_singleton();
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void set_singleton();
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virtual const char *get_name() const = 0;
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virtual Error init() = 0;
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virtual void start() = 0;
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virtual int get_mix_rate() const = 0;
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virtual SpeakerMode get_speaker_mode() const = 0;
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virtual Array get_device_list();
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virtual String get_device();
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virtual void set_device(String device) {}
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virtual void lock() = 0;
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virtual void unlock() = 0;
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virtual void finish() = 0;
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virtual Error capture_start() { return FAILED; }
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virtual Error capture_stop() { return FAILED; }
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virtual void capture_set_device(const String &p_name) {}
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virtual String capture_get_device() { return "Default"; }
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virtual Array capture_get_device_list(); // TODO: convert this and get_device_list to PoolStringArray
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virtual float get_latency() { return 0; }
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SpeakerMode get_speaker_mode_by_total_channels(int p_channels) const;
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int get_total_channels_by_speaker_mode(SpeakerMode) const;
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Vector<int32_t> get_input_buffer() { return input_buffer; }
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unsigned int get_input_position() { return input_position; }
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unsigned int get_input_size() { return input_size; }
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#ifdef DEBUG_ENABLED
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uint64_t get_profiling_time() const { return prof_time; }
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void reset_profiling_time() { prof_time = 0; }
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#endif
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AudioDriver();
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virtual ~AudioDriver() {}
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};
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class AudioDriverManager {
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enum {
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MAX_DRIVERS = 10
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};
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public:
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enum MuteFlags {
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// User enables or disables audio, e.g. via button in editor
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MUTE_FLAG_DISABLED = 1 << 0,
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// Whether app is in focus
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MUTE_FLAG_FOCUS_LOSS = 1 << 1,
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// Whether app is paused / resumed
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MUTE_FLAG_PAUSED = 1 << 2,
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// When a section of silence is detected, the audio can be muted
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MUTE_FLAG_SILENCE = 1 << 3,
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};
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private:
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static const int DEFAULT_MIX_RATE = 44100;
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static const int DEFAULT_OUTPUT_LATENCY = 15;
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static AudioDriver *drivers[MAX_DRIVERS];
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static int driver_count;
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static int desired_driver_id;
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static int actual_driver_id;
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static uint32_t _mute_state; // raw flags
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static uint32_t _mute_state_final; // flags after applying mask
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static uint32_t _mute_state_mask;
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static AudioDriverDummy dummy_driver;
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static void _set_driver(int p_driver);
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static void _update_mute_state();
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static void _log(String p_sz, int p_driver_id = -1);
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public:
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static void add_driver(AudioDriver *p_driver);
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static void initialize(int p_driver);
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static int get_driver_count();
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static AudioDriver *get_driver(int p_driver);
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// Various modes flags can be used to mute the audio, depending on the sensitivity in the _mute_state_mask.
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static void set_mute_flag(MuteFlags p_flag, bool p_enabled);
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static bool get_mute_flag(MuteFlags p_flag) { return _mute_state & p_flag; }
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// Whether audio should play, or whether the audio system should be considered muted.
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static bool is_active() { return _mute_state_final == 0; }
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// Audio processing can be throttled down EXCEPT for the case of silence mode, where any sound should
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// wake up the driver.
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static bool is_audio_processing_allowed() { return (_mute_state_final & (~MUTE_FLAG_SILENCE)) == 0; }
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// Sets the sensitivity mask for different mute flags.
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static void set_mute_sensitivity(MuteFlags p_flag, bool p_enabled);
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static bool get_mute_sensitivity(MuteFlags p_flag) { return _mute_state_mask & p_flag; }
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};
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class AudioBusLayout;
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class AudioServer : public Object {
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GDCLASS(AudioServer, Object);
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public:
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//re-expose this here, as AudioDriver is not exposed to script
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enum SpeakerMode {
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SPEAKER_MODE_STEREO,
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SPEAKER_SURROUND_31,
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SPEAKER_SURROUND_51,
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SPEAKER_SURROUND_71,
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};
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enum {
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AUDIO_DATA_INVALID_ID = -1
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};
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typedef void (*AudioCallback)(void *p_userdata);
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private:
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uint64_t mix_time;
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int mix_size;
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uint32_t buffer_size;
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uint64_t mix_count;
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uint64_t mix_frames;
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#ifdef DEBUG_ENABLED
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uint64_t prof_time;
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#endif
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float channel_disable_threshold_db;
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uint32_t channel_disable_frames;
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int channel_count;
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int to_mix;
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float global_rate_scale;
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struct Bus {
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StringName name;
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bool solo;
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bool mute;
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bool bypass;
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bool soloed;
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//Each channel is a stereo pair.
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struct Channel {
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bool used;
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bool active;
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AudioFrame peak_volume;
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Vector<AudioFrame> buffer;
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Vector<Ref<AudioEffectInstance>> effect_instances;
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uint64_t last_mix_with_audio;
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Channel() {
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last_mix_with_audio = 0;
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used = false;
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active = false;
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peak_volume = AudioFrame(AUDIO_MIN_PEAK_DB, AUDIO_MIN_PEAK_DB);
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}
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};
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Vector<Channel> channels;
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struct Effect {
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Ref<AudioEffect> effect;
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bool enabled;
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#ifdef DEBUG_ENABLED
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uint64_t prof_time;
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#endif
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Effect() {
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enabled = false;
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#ifdef DEBUG_ENABLED
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prof_time = 0;
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#endif
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}
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};
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Vector<Effect> effects;
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float volume_db;
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StringName send;
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int index_cache;
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};
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Vector<Vector<AudioFrame>> temp_buffer; //temp_buffer for each level
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Vector<Bus *> buses;
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RBMap<StringName, Bus *> bus_map;
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void _update_bus_effects(int p_bus);
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static AudioServer *singleton;
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// TODO create an audiodata pool to optimize memory
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RBMap<void *, uint32_t> audio_data;
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size_t audio_data_total_mem;
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size_t audio_data_max_mem;
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Mutex audio_data_lock;
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// keep a rough record of when the last sound was output
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// so we can throttle audio during silence
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uint32_t last_sound_played_ms;
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void init_channels_and_buffers();
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void _mix_step();
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struct CallbackItem {
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AudioCallback callback;
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void *userdata;
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bool operator<(const CallbackItem &p_item) const {
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return (callback == p_item.callback ? userdata < p_item.userdata : callback < p_item.callback);
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}
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};
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RBSet<CallbackItem> callbacks;
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RBSet<CallbackItem> update_callbacks;
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friend class AudioDriver;
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void _driver_process(int p_frames, int32_t *p_buffer);
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protected:
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static void _bind_methods();
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public:
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_FORCE_INLINE_ int get_channel_count() const {
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switch (get_speaker_mode()) {
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case SPEAKER_MODE_STEREO:
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return 1;
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case SPEAKER_SURROUND_31:
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return 2;
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case SPEAKER_SURROUND_51:
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return 3;
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case SPEAKER_SURROUND_71:
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return 4;
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}
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ERR_FAIL_V(1);
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}
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//do not use from outside audio thread
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bool thread_has_channel_mix_buffer(int p_bus, int p_buffer) const;
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AudioFrame *thread_get_channel_mix_buffer(int p_bus, int p_buffer);
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int thread_get_mix_buffer_size() const;
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int thread_find_bus_index(const StringName &p_name);
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void set_bus_count(int p_count);
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int get_bus_count() const;
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void remove_bus(int p_index);
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void add_bus(int p_at_pos = -1);
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void move_bus(int p_bus, int p_to_pos);
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void set_bus_name(int p_bus, const String &p_name);
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String get_bus_name(int p_bus) const;
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int get_bus_index(const StringName &p_bus_name) const;
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int get_bus_channels(int p_bus) const;
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void set_bus_volume_db(int p_bus, float p_volume_db);
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float get_bus_volume_db(int p_bus) const;
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void set_bus_send(int p_bus, const StringName &p_send);
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StringName get_bus_send(int p_bus) const;
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void set_bus_solo(int p_bus, bool p_enable);
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bool is_bus_solo(int p_bus) const;
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void set_bus_mute(int p_bus, bool p_enable);
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bool is_bus_mute(int p_bus) const;
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void set_bus_bypass_effects(int p_bus, bool p_enable);
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bool is_bus_bypassing_effects(int p_bus) const;
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void add_bus_effect(int p_bus, const Ref<AudioEffect> &p_effect, int p_at_pos = -1);
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void remove_bus_effect(int p_bus, int p_effect);
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int get_bus_effect_count(int p_bus);
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Ref<AudioEffect> get_bus_effect(int p_bus, int p_effect);
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Ref<AudioEffectInstance> get_bus_effect_instance(int p_bus, int p_effect, int p_channel = 0);
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void swap_bus_effects(int p_bus, int p_effect, int p_by_effect);
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void set_bus_effect_enabled(int p_bus, int p_effect, bool p_enabled);
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bool is_bus_effect_enabled(int p_bus, int p_effect) const;
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float get_bus_peak_volume_left_db(int p_bus, int p_channel) const;
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float get_bus_peak_volume_right_db(int p_bus, int p_channel) const;
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bool is_bus_channel_active(int p_bus, int p_channel) const;
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void set_global_rate_scale(float p_scale);
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float get_global_rate_scale() const;
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virtual void init();
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virtual void finish();
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virtual void update();
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virtual void load_default_bus_layout();
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/* MISC config */
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virtual void lock();
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virtual void unlock();
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virtual SpeakerMode get_speaker_mode() const;
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virtual float get_mix_rate() const;
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virtual float read_output_peak_db() const;
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static AudioServer *get_singleton();
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virtual double get_output_latency() const;
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virtual double get_time_to_next_mix() const;
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virtual double get_time_since_last_mix() const;
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void *audio_data_alloc(uint32_t p_data_len, const uint8_t *p_from_data = nullptr);
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void audio_data_free(void *p_data);
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size_t audio_data_get_total_memory_usage() const;
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size_t audio_data_get_max_memory_usage() const;
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void add_callback(AudioCallback p_callback, void *p_userdata);
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void remove_callback(AudioCallback p_callback, void *p_userdata);
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void add_update_callback(AudioCallback p_callback, void *p_userdata);
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void remove_update_callback(AudioCallback p_callback, void *p_userdata);
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void set_bus_layout(const Ref<AudioBusLayout> &p_bus_layout);
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Ref<AudioBusLayout> generate_bus_layout() const;
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Array get_device_list();
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String get_device();
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void set_device(String device);
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void set_enabled(bool p_enabled);
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bool is_enabled() const;
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Array capture_get_device_list();
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String capture_get_device();
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void capture_set_device(const String &p_name);
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AudioServer();
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virtual ~AudioServer();
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};
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VARIANT_ENUM_CAST(AudioServer::SpeakerMode)
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class AudioBusLayout : public Resource {
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GDCLASS(AudioBusLayout, Resource);
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friend class AudioServer;
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struct Bus {
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StringName name;
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bool solo;
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bool mute;
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bool bypass;
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struct Effect {
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Ref<AudioEffect> effect;
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bool enabled;
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};
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Vector<Effect> effects;
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float volume_db;
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StringName send;
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Bus() {
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solo = false;
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mute = false;
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bypass = false;
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volume_db = 0;
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}
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};
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Vector<Bus> buses;
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protected:
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bool _set(const StringName &p_name, const Variant &p_value);
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bool _get(const StringName &p_name, Variant &r_ret) const;
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void _get_property_list(List<PropertyInfo> *p_list) const;
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public:
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AudioBusLayout();
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};
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typedef AudioServer AS;
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#endif // AUDIO_SERVER_H
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