virtualx-engine/servers/audio_server.h
Emmanouil Papadeas c36460060e
Further refactoring to AudioDriver implementations after .
- Rename all instances of `capture_start()` and `capture_end()` to their new
  names. Fixes .
- More internal renames to match what was started in .
- Use `override` consistently so that such refactoring bugs can be caught.
- Harmonize the order of definition of the overridden virtual methods in each
  audio driver.
- Harmonize prototype for `set_output_device` and `set_input_device`.

Co-authored-by: Rémi Verschelde <rverschelde@gmail.com>
2023-02-09 11:02:00 +01:00

481 lines
15 KiB
C++

/**************************************************************************/
/* audio_server.h */
/**************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/**************************************************************************/
/* Copyright (c) 2014-present Godot Engine contributors (see AUTHORS.md). */
/* Copyright (c) 2007-2014 Juan Linietsky, Ariel Manzur. */
/* */
/* 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 AudioStreamWAV;
class AudioStreamPlayback;
class AudioDriver {
static AudioDriver *singleton;
uint64_t _last_mix_time = 0;
uint64_t _last_mix_frames = 0;
#ifdef DEBUG_ENABLED
uint64_t prof_ticks = 0;
uint64_t prof_time = 0;
#endif
protected:
Vector<int32_t> input_buffer;
unsigned int input_position = 0;
unsigned int input_size = 0;
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 API to implement.
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 float get_latency() { return 0; }
virtual void lock() = 0;
virtual void unlock() = 0;
virtual void finish() = 0;
virtual PackedStringArray get_output_device_list();
virtual String get_output_device();
virtual void set_output_device(const String &p_name) {}
virtual Error input_start() { return FAILED; }
virtual Error input_stop() { return FAILED; }
virtual PackedStringArray get_input_device_list();
virtual String get_input_device() { return "Default"; }
virtual void set_input_device(const String &p_name) {}
//
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 = 64,
};
typedef void (*AudioCallback)(void *p_userdata);
private:
uint64_t mix_time = 0;
int mix_size = 0;
uint32_t buffer_size = 0;
uint64_t mix_count = 0;
uint64_t mix_frames = 0;
#ifdef DEBUG_ENABLED
uint64_t prof_time = 0;
#endif
float channel_disable_threshold_db = 0.0f;
uint32_t channel_disable_frames = 0;
int channel_count = 0;
int to_mix = 0;
float playback_speed_scale = 1.0f;
bool tag_used_audio_streams = false;
struct Bus {
StringName name;
bool solo = false;
bool mute = false;
bool bypass = false;
bool soloed = false;
// Each channel is a stereo pair.
struct Channel {
bool used = false;
bool active = false;
AudioFrame peak_volume = AudioFrame(AUDIO_MIN_PEAK_DB, AUDIO_MIN_PEAK_DB);
Vector<AudioFrame> buffer;
Vector<Ref<AudioEffectInstance>> effect_instances;
uint64_t last_mix_with_audio = 0;
Channel() {}
};
Vector<Channel> channels;
struct Effect {
Ref<AudioEffect> effect;
bool enabled = false;
#ifdef DEBUG_ENABLED
uint64_t prof_time = 0;
#endif
};
Vector<Effect> effects;
float volume_db = 0.0f;
StringName send;
int index_cache = 0;
};
struct AudioStreamPlaybackBusDetails {
bool bus_active[MAX_BUSES_PER_PLAYBACK] = {};
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;
// TODO document if this is necessary.
SafeList<AudioStreamPlaybackBusDetails *> bus_details_graveyard_frame_old;
Vector<Vector<AudioFrame>> temp_buffer; //temp_buffer for each level
Vector<AudioFrame> mix_buffer;
Vector<Bus *> buses;
HashMap<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 = nullptr;
};
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;
// Convenience method.
void start_playback_stream(Ref<AudioStreamPlayback> p_playback, StringName p_bus, Vector<AudioFrame> p_volume_db_vector, float p_start_time = 0, float p_pitch_scale = 1);
// Expose all parameters.
void start_playback_stream(Ref<AudioStreamPlayback> p_playback, HashMap<StringName, Vector<AudioFrame>> p_bus_volumes, float p_start_time = 0, float p_pitch_scale = 1, float p_highshelf_gain = 0, float p_attenuation_cutoff_hz = 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, HashMap<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;
uint64_t get_mixed_frames() 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;
PackedStringArray get_output_device_list();
String get_output_device();
void set_output_device(const String &p_name);
PackedStringArray get_input_device_list();
String get_input_device();
void set_input_device(const String &p_name);
void set_enable_tagging_used_audio_streams(bool p_enable);
AudioServer();
virtual ~AudioServer();
};
VARIANT_ENUM_CAST(AudioServer::SpeakerMode)
class AudioBusLayout : public Resource {
GDCLASS(AudioBusLayout, Resource);
friend class AudioServer;
struct Bus {
StringName name;
bool solo = false;
bool mute = false;
bool bypass = false;
struct Effect {
Ref<AudioEffect> effect;
bool enabled = false;
};
Vector<Effect> effects;
float volume_db = 0.0f;
StringName send;
Bus() {}
};
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