5300daaff2
Change instances of audio properties 'device' to 'output_device', and instances of audio properties 'capture_device' to 'input_device', as well as their subsequent getter & setter functions. Update the docs to reflect these changes, as well as the 3-to-4 converter for GDScript and CSharp to make proper conversions (only exception is 'device' since that name is too vague and might replace non-AudioServer related instances, such as user comments and variables). This does not change internal references to references like 'Render Client' and 'Capture Client' in WASAPI; such is outside the scope of this commit. This also does not change ALSA's references, considering that it uses 'device' to mean input and output interchangeably. Other references are changed, however where applicable, to be consistent with the new AudioServer methods and property names.
476 lines
15 KiB
C++
476 lines
15 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 AudioStream;
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class AudioStreamWAV;
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class AudioStreamPlayback;
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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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uint64_t prof_ticks = 0;
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uint64_t prof_time = 0;
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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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#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 PackedStringArray get_output_device_list();
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virtual String get_output_device();
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virtual void set_output_device(String output_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 set_input_device(const String &p_name) {}
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virtual String get_input_device() { return "Default"; }
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virtual PackedStringArray get_input_device_list();
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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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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 AudioDriverDummy dummy_driver;
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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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};
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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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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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uint64_t prof_time = 0;
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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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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, 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, 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, StringName p_bus, Vector<AudioFrame> p_volumes);
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void set_playback_bus_volumes_linear(Ref<AudioStreamPlayback> p_playback, 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(String output_device);
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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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void set_enable_tagging_used_audio_streams(bool p_enable);
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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 = false;
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bool mute = false;
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bool bypass = false;
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struct Effect {
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Ref<AudioEffect> effect;
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bool enabled = false;
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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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Bus() {}
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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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