534 lines
18 KiB
C++
534 lines
18 KiB
C++
/**************************************************************************/
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/* audio_server.h */
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/**************************************************************************/
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/* This file is part of: */
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/* GODOT ENGINE */
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/* https://godotengine.org */
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/**************************************************************************/
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/* Copyright (c) 2014-present Godot Engine contributors (see AUTHORS.md). */
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/* Copyright (c) 2007-2014 Juan Linietsky, Ariel Manzur. */
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/* */
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/* Permission is hereby granted, free of charge, to any person obtaining */
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/* a copy of this software and associated documentation files (the */
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/* "Software"), to deal in the Software without restriction, including */
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/* without limitation the rights to use, copy, modify, merge, publish, */
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/* distribute, sublicense, and/or sell copies of the Software, and to */
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/* permit persons to whom the Software is furnished to do so, subject to */
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/* the following conditions: */
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/* */
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/* The above copyright notice and this permission notice shall be */
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/* included in all copies or substantial portions of the Software. */
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/* */
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/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
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/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
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/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. */
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/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
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/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
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/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
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/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
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/**************************************************************************/
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#ifndef AUDIO_SERVER_H
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#define AUDIO_SERVER_H
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#include "core/math/audio_frame.h"
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#include "core/object/class_db.h"
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#include "core/os/os.h"
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#include "core/templates/safe_list.h"
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#include "core/variant/variant.h"
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#include "servers/audio/audio_effect.h"
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#include "servers/audio/audio_filter_sw.h"
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#include <atomic>
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class AudioDriverDummy;
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class AudioSample;
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class AudioStream;
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class AudioStreamWAV;
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class AudioStreamPlayback;
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class AudioSamplePlayback;
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class AudioDriver {
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static AudioDriver *singleton;
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uint64_t _last_mix_time = 0;
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uint64_t _last_mix_frames = 0;
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#ifdef DEBUG_ENABLED
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SafeNumeric<uint64_t> prof_ticks;
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SafeNumeric<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 = 0;
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unsigned int input_size = 0;
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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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int _get_configured_mix_rate();
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#ifdef DEBUG_ENABLED
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_FORCE_INLINE_ void start_counting_ticks() { prof_ticks.set(OS::get_singleton()->get_ticks_usec()); }
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_FORCE_INLINE_ void stop_counting_ticks() { prof_time.add(OS::get_singleton()->get_ticks_usec() - prof_ticks.get()); }
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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 API to implement.
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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 float get_latency() { return 0; }
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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 PackedStringArray get_output_device_list();
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virtual String get_output_device();
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virtual void set_output_device(const String &p_name) {}
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virtual Error input_start() { return FAILED; }
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virtual Error input_stop() { return FAILED; }
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virtual PackedStringArray get_input_device_list();
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virtual String get_input_device() { return "Default"; }
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virtual void set_input_device(const String &p_name) {}
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//
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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.get(); }
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void reset_profiling_time() { prof_time.set(0); }
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#endif
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// Samples handling.
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virtual bool is_stream_registered_as_sample(const Ref<AudioStream> &p_stream) const {
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return false;
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}
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virtual void register_sample(const Ref<AudioSample> &p_sample) {}
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virtual void unregister_sample(const Ref<AudioSample> &p_sample) {}
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virtual void start_sample_playback(const Ref<AudioSamplePlayback> &p_playback);
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virtual void stop_sample_playback(const Ref<AudioSamplePlayback> &p_playback) {}
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virtual void set_sample_playback_pause(const Ref<AudioSamplePlayback> &p_playback, bool p_paused) {}
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virtual bool is_sample_playback_active(const Ref<AudioSamplePlayback> &p_playback) { return false; }
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virtual void update_sample_playback_pitch_scale(const Ref<AudioSamplePlayback> &p_playback, float p_pitch_scale = 0.0f) {}
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virtual void set_sample_playback_bus_volumes_linear(const Ref<AudioSamplePlayback> &p_playback, const HashMap<StringName, Vector<AudioFrame>> &p_bus_volumes) {}
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virtual void set_sample_bus_count(int p_count) {}
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virtual void remove_sample_bus(int p_bus) {}
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virtual void add_sample_bus(int p_at_pos = -1) {}
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virtual void move_sample_bus(int p_bus, int p_to_pos) {}
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virtual void set_sample_bus_send(int p_bus, const StringName &p_send) {}
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virtual void set_sample_bus_volume_db(int p_bus, float p_volume_db) {}
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virtual void set_sample_bus_solo(int p_bus, bool p_enable) {}
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virtual void set_sample_bus_mute(int p_bus, bool p_enable) {}
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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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static AudioDriver *drivers[MAX_DRIVERS];
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static int driver_count;
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static AudioDriverDummy dummy_driver;
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public:
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static const int DEFAULT_MIX_RATE = 44100;
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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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};
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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 PlaybackType {
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PLAYBACK_TYPE_DEFAULT,
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PLAYBACK_TYPE_STREAM,
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PLAYBACK_TYPE_SAMPLE,
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PLAYBACK_TYPE_MAX
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};
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enum {
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AUDIO_DATA_INVALID_ID = -1,
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MAX_CHANNELS_PER_BUS = 4,
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MAX_BUSES_PER_PLAYBACK = 6,
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LOOKAHEAD_BUFFER_SIZE = 64,
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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 = 0;
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int mix_size = 0;
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uint32_t buffer_size = 0;
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uint64_t mix_count = 0;
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uint64_t mix_frames = 0;
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#ifdef DEBUG_ENABLED
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SafeNumeric<uint64_t> prof_time;
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#endif
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float channel_disable_threshold_db = 0.0f;
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uint32_t channel_disable_frames = 0;
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int channel_count = 0;
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int to_mix = 0;
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float playback_speed_scale = 1.0f;
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bool tag_used_audio_streams = false;
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struct Bus {
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StringName name;
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bool solo = false;
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bool mute = false;
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bool bypass = false;
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bool soloed = false;
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// Each channel is a stereo pair.
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struct Channel {
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bool used = false;
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bool active = false;
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AudioFrame peak_volume = AudioFrame(AUDIO_MIN_PEAK_DB, AUDIO_MIN_PEAK_DB);
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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 = 0;
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Channel() {}
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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 = false;
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#ifdef DEBUG_ENABLED
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uint64_t prof_time = 0;
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#endif
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};
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Vector<Effect> effects;
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float volume_db = 0.0f;
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StringName send;
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int index_cache = 0;
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};
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struct AudioStreamPlaybackBusDetails {
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bool bus_active[MAX_BUSES_PER_PLAYBACK] = {};
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StringName bus[MAX_BUSES_PER_PLAYBACK];
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AudioFrame volume[MAX_BUSES_PER_PLAYBACK][MAX_CHANNELS_PER_BUS];
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};
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struct AudioStreamPlaybackListNode {
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enum PlaybackState {
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PAUSED = 0, // Paused. Keep this stream playback around though so it can be restarted.
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PLAYING = 1, // Playing. Fading may still be necessary if volume changes!
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FADE_OUT_TO_PAUSE = 2, // About to pause.
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FADE_OUT_TO_DELETION = 3, // About to stop.
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AWAITING_DELETION = 4,
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};
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// If zero or positive, a place in the stream to seek to during the next mix.
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SafeNumeric<float> setseek;
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SafeNumeric<float> pitch_scale;
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SafeNumeric<float> highshelf_gain;
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SafeNumeric<float> attenuation_filter_cutoff_hz; // This isn't used unless highshelf_gain is nonzero.
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AudioFilterSW::Processor filter_process[8];
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// Updating this ref after the list node is created breaks consistency guarantees, don't do it!
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Ref<AudioStreamPlayback> stream_playback;
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// Playback state determines the fate of a particular AudioStreamListNode during the mix step. Must be atomically replaced.
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std::atomic<PlaybackState> state = AWAITING_DELETION;
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// This data should only ever be modified by an atomic replacement of the pointer.
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std::atomic<AudioStreamPlaybackBusDetails *> bus_details = nullptr;
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// Previous bus details should only be accessed on the audio thread.
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AudioStreamPlaybackBusDetails *prev_bus_details = nullptr;
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// 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.
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AudioFrame lookahead[LOOKAHEAD_BUFFER_SIZE];
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};
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SafeList<AudioStreamPlaybackListNode *> playback_list;
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SafeList<AudioStreamPlaybackBusDetails *> bus_details_graveyard;
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// TODO document if this is necessary.
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SafeList<AudioStreamPlaybackBusDetails *> bus_details_graveyard_frame_old;
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Vector<Vector<AudioFrame>> temp_buffer; //temp_buffer for each level
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Vector<AudioFrame> mix_buffer;
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Vector<Bus *> buses;
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HashMap<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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void init_channels_and_buffers();
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void _mix_step();
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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);
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// Should only be called on the main thread.
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AudioStreamPlaybackListNode *_find_playback_list_node(Ref<AudioStreamPlayback> p_playback);
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struct CallbackItem {
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AudioCallback callback;
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void *userdata = nullptr;
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};
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SafeList<CallbackItem *> update_callback_list;
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SafeList<CallbackItem *> mix_callback_list;
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SafeList<CallbackItem *> listener_changed_callback_list;
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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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LocalVector<Ref<AudioSamplePlayback>> sample_playback_list;
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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_playback_speed_scale(float p_scale);
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float get_playback_speed_scale() const;
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// Convenience method.
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void start_playback_stream(Ref<AudioStreamPlayback> p_playback, const StringName &p_bus, Vector<AudioFrame> p_volume_db_vector, float p_start_time = 0, float p_pitch_scale = 1);
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// Expose all parameters.
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void start_playback_stream(Ref<AudioStreamPlayback> p_playback, const 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);
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void stop_playback_stream(Ref<AudioStreamPlayback> p_playback);
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void set_playback_bus_exclusive(Ref<AudioStreamPlayback> p_playback, const StringName &p_bus, Vector<AudioFrame> p_volumes);
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void set_playback_bus_volumes_linear(Ref<AudioStreamPlayback> p_playback, const HashMap<StringName, Vector<AudioFrame>> &p_bus_volumes);
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void set_playback_all_bus_volumes_linear(Ref<AudioStreamPlayback> p_playback, Vector<AudioFrame> p_volumes);
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void set_playback_pitch_scale(Ref<AudioStreamPlayback> p_playback, float p_pitch_scale);
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void set_playback_paused(Ref<AudioStreamPlayback> p_playback, bool p_paused);
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void set_playback_highshelf_params(Ref<AudioStreamPlayback> p_playback, float p_gain, float p_attenuation_cutoff_hz);
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bool is_playback_active(Ref<AudioStreamPlayback> p_playback);
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float get_playback_position(Ref<AudioStreamPlayback> p_playback);
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bool is_playback_paused(Ref<AudioStreamPlayback> p_playback);
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uint64_t get_mix_count() const;
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uint64_t get_mixed_frames() const;
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void notify_listener_changed();
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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 add_listener_changed_callback(AudioCallback p_callback, void *p_userdata);
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void remove_listener_changed_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 add_mix_callback(AudioCallback p_callback, void *p_userdata);
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void remove_mix_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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PackedStringArray get_output_device_list();
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String get_output_device();
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void set_output_device(const String &p_name);
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PackedStringArray get_input_device_list();
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String get_input_device();
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void set_input_device(const String &p_name);
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|
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void set_enable_tagging_used_audio_streams(bool p_enable);
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|
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#ifdef TOOLS_ENABLED
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virtual void get_argument_options(const StringName &p_function, int p_idx, List<String> *r_options) const override;
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#endif
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PlaybackType get_default_playback_type() const;
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|
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bool is_stream_registered_as_sample(const Ref<AudioStream> &p_stream);
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void register_stream_as_sample(const Ref<AudioStream> &p_stream);
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void unregister_stream_as_sample(const Ref<AudioStream> &p_stream);
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void register_sample(const Ref<AudioSample> &p_sample);
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void unregister_sample(const Ref<AudioSample> &p_sample);
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void start_sample_playback(const Ref<AudioSamplePlayback> &p_playback);
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void stop_sample_playback(const Ref<AudioSamplePlayback> &p_playback);
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|
void set_sample_playback_pause(const Ref<AudioSamplePlayback> &p_playback, bool p_paused);
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bool is_sample_playback_active(const Ref<AudioSamplePlayback> &p_playback);
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void update_sample_playback_pitch_scale(const Ref<AudioSamplePlayback> &p_playback, float p_pitch_scale = 0.0f);
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AudioServer();
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virtual ~AudioServer();
|
|
};
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|
|
|
VARIANT_ENUM_CAST(AudioServer::SpeakerMode)
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VARIANT_ENUM_CAST(AudioServer::PlaybackType)
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|
|
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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 {
|
|
StringName name;
|
|
bool solo = false;
|
|
bool mute = false;
|
|
bool bypass = false;
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|
|
|
struct Effect {
|
|
Ref<AudioEffect> effect;
|
|
bool enabled = false;
|
|
};
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|
|
|
Vector<Effect> effects;
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|
|
|
float volume_db = 0.0f;
|
|
StringName send;
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|
|
|
Bus() {}
|
|
};
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|
|
|
Vector<Bus> buses;
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|
|
|
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();
|
|
};
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|
|
|
typedef AudioServer AS;
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|
|
|
#endif // AUDIO_SERVER_H
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