godot/servers/audio_server.h
Michael Alexsander Silva Dias 1c2ba35074 Add 'global_rate_scale' to the AudioServer
Closes #28953.
2019-06-19 11:36:46 -03:00

441 lines
12 KiB
C++

/*************************************************************************/
/* audio_server.h */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */
/* Copyright (c) 2014-2019 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.h"
#include "core/os/os.h"
#include "core/variant.h"
#include "servers/audio/audio_effect.h"
class AudioDriverDummy;
class AudioStream;
class AudioStreamSample;
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() const; //useful for video -> audio sync
double get_time_to_next_mix() const;
enum SpeakerMode {
SPEAKER_MODE_STEREO,
SPEAKER_SURROUND_31,
SPEAKER_SURROUND_51,
SPEAKER_SURROUND_71,
};
static const int DEFAULT_MIX_RATE = 44100;
static const int DEFAULT_OUTPUT_LATENCY = 15;
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 PoolStringArray
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 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
};
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 global_rate_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(0, 0);
}
};
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;
};
Vector<Vector<AudioFrame> > temp_buffer; //temp_buffer for each level
Vector<Bus *> buses;
Map<StringName, Bus *> bus_map;
void _update_bus_effects(int p_bus);
static AudioServer *singleton;
// TODO create an audiodata pool to optimize memory
Map<void *, uint32_t> audio_data;
size_t audio_data_total_mem;
size_t audio_data_max_mem;
Mutex *audio_data_lock;
float output_latency;
uint64_t output_latency_ticks;
void init_channels_and_buffers();
void _mix_step();
#if 0
struct AudioInBlock {
Ref<AudioStreamSample> audio_stream;
int current_position;
bool loops;
};
Map<StringName, AudioInBlock *> audio_in_block_map;
Vector<AudioInBlock *> audio_in_blocks;
#endif
struct CallbackItem {
AudioCallback callback;
void *userdata;
bool operator<(const CallbackItem &p_item) const {
return (callback == p_item.callback ? userdata < p_item.userdata : callback < p_item.callback);
}
};
Set<CallbackItem> callbacks;
Set<CallbackItem> update_callbacks;
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_global_rate_scale(float p_scale);
float get_global_rate_scale() const;
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 *audio_data_alloc(uint32_t p_data_len, const uint8_t *p_from_data = NULL);
void audio_data_free(void *p_data);
size_t audio_data_get_total_memory_usage() const;
size_t audio_data_get_max_memory_usage() const;
void add_callback(AudioCallback p_callback, void *p_userdata);
void remove_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 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