Fase 3: import del subsistema de audio desde AEEA

Reemplaza el audio antiguo de orni_attack (singleton con new/delete
raw, sin efectos, sin crossfade) por el subsistema moderno de AEEA
(unique_ptr, RAII, crossfade nativo, echo/reverb, pitch-shift,
callbacks de fin de pista, getMusicDurationMs para timelines
deterministas).

Eliminados:
- source/core/audio/audio_cache.{hpp,cpp} (1 cache por subsistema)
- source/core/audio/jail_audio.hpp viejo (motor inline globals)
- source/external/stb_vorbis.h (v1.20)

Añadidos (copiados de AEEA, traducidos comentarios al castellano):
- source/core/audio/audio.{hpp,cpp} — singleton con Audio::Config inyectada
- source/core/audio/audio_adapter.{hpp,cpp} — adapter para getMusic/getSound
- source/core/audio/audio_effects.{hpp,cpp} — Schroeder reverb + echo DSP
- source/core/audio/jail_audio.{hpp,cpp} — Ja::Engine class-based, streaming
- source/core/audio/sound_effects_config.{hpp,cpp} — presets YAML (opcional)
- source/external/stb_vorbis.c (v1.22) — OGG decoder, versión más reciente
- source/external/stb_vorbis_impl.cpp — TU aislada para evitar clang-tidy

Adaptaciones:
- audio_adapter.cpp implementado a medida para orni: usa
  Resource::Helper::loadFile (no Resource::Cache de AEEA que orni no
  tiene). Cache local con unique_ptr<Ja::Music> / unique_ptr<Ja::Sound>.
- Includes: utils/defaults.hpp -> core/defaults.hpp, utils/log.hpp
  reemplazado por iostream con std::cerr/std::cout.

API breaking changes (callsites migrados):
- Audio::init() -> Audio::init(Config); el Director construye la Config
  desde Defaults::Audio::* (ENABLED, VOLUME, MUSIC_*, SOUND_*).
- Audio::get()->getMusicState() -> Audio::getMusicState() (ahora static).
- AudioCache::getMusic/getSound -> AudioResource::getMusic/getSound.

Defaults::Audio consolidado: ahora aglutina las constantes que antes
estaban repartidas entre namespace Audio (VOLUME, ENABLED), namespace
Music (VOLUME, ENABLED), namespace Sound (VOLUME, ENABLED). Las pistas
y rutas de efectos siguen en Music::* / Sound::*. Añadidas FREQUENCY,
FORMAT, CHANNELS, CROSSFADE_MS, VOLUME_STEP para el motor.

Beneficios para fases siguientes:
- Crossfade en transiciones de escena (uso: playMusic(name, -1, 1500)).
- Pitch-shift para variaciones de SFX (Audio::playSound(name, group, 0.95)).
- Echo/reverb DSP via playSoundWithEcho/Reverb (sounds.yaml presets).
- Callbacks setOnMusicEnded para sincronizar eventos con el fin de pista.

Compila y enlaza. Pendiente: test runtime del usuario.

Co-Authored-By: Claude Opus 4.7 (1M context) <noreply@anthropic.com>
This commit is contained in:
2026-05-19 12:43:01 +02:00
parent a4f6a5514f
commit ed98ef612e
19 changed files with 1893 additions and 916 deletions
+217 -109
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@@ -1,183 +1,291 @@
#include "audio.hpp"
#include "core/audio/audio.hpp"
#include <SDL3/SDL.h> // Para SDL_LogInfo, SDL_LogCategory, SDL_G...
#include <SDL3/SDL.h> // Para SDL_GetError, SDL_Init
#include <algorithm> // Para clamp
#include <iostream> // Para std::cout
#include <cstdio> // Para std::fprintf
// Implementación de stb_vorbis (debe estar ANTES de incluir jail_audio.hpp)
// clang-format off
#undef STB_VORBIS_HEADER_ONLY
#include "external/stb_vorbis.h"
// clang-format on
#include "core/audio/audio_adapter.hpp" // Para AudioResource::getMusic/getSound
#include "core/audio/jail_audio.hpp" // Para Ja::* (motor jailgames)
#include "core/audio/sound_effects_config.hpp" // Para SoundEffectsConfig
#include "core/defaults.hpp" // Para Defaults::Audio::FREQUENCY
#include "core/audio/audio_cache.hpp" // Para AudioCache
#include "core/audio/jail_audio.hpp" // Para JA_FadeOutMusic, JA_Init, JA_PauseM...
#include "game/options.hpp" // Para AudioOptions, audio, MusicOptions
// Invariant compile-time: tots los valors d'Audio::Group han de cabre als slots
// de volum per grup que manté l'engine. Si s'afegeix una nueva entrada a Group
// y no s'incrementa Ja::MAX_GROUPS, este assert falla antes de compilar.
static_assert(static_cast<int>(Audio::Group::INTERFACE) < Ja::MAX_GROUPS,
"Audio::Group té més entrades que slots té Ja::MAX_GROUPS");
// Singleton
Audio* Audio::instance = nullptr;
std::unique_ptr<Audio> Audio::instance;
// Inicializa la instancia única del singleton
void Audio::init() { Audio::instance = new Audio(); }
// Inicialitza la instància única del singleton con la configuración rebuda
void Audio::init(const Config& config) { Audio::instance = std::unique_ptr<Audio>(new Audio(config)); }
// Libera la instancia
void Audio::destroy() { delete Audio::instance; }
// Allibera la instància
void Audio::destroy() { Audio::instance.reset(); }
// Obtiene la instancia
auto Audio::get() -> Audio* { return Audio::instance; }
// Obté la instància
auto Audio::get() -> Audio* { return Audio::instance.get(); }
// Constructor
Audio::Audio() { initSDLAudio(); }
Audio::Audio(const Config& config)
: config_(config) { initSDLAudio(); }
// Destructor
Audio::~Audio() {
JA_Quit();
}
// Destructor: engine_ es std::unique_ptr, el seu dtor tanca el device SDL i
// desregistra Ja::Engine::active_. Cap crida explícita necessària.
Audio::~Audio() = default;
// Método principal
// Método principal: l'estat de la música el manté el motor (única font de
// veritat), per tant no cal sin sincronització aquí.
void Audio::update() {
JA_Update();
if (instance && instance->engine_) { instance->engine_->update(); }
}
// Reproduce la música
void Audio::playMusic(const std::string& name, const int loop) {
bool new_loop = (loop != 0);
// Reprodueix la música per nom (amb crossfade opcional)
void Audio::playMusic(const std::string& name, const int loop, const int crossfade_ms) {
const bool NEW_LOOP = (loop != 0);
// Si ya está sonando exactamente la misma pista y mismo modo loop, no hacemos nada
if (music_.state == MusicState::PLAYING && music_.name == name && music_.loop == new_loop) {
// Si ya sona exactament la misma pista i mismo mode loop, no fem res
if (getMusicState() == MusicState::PLAYING && music_.name == name && music_.loop == NEW_LOOP) {
return;
}
// Intentar obtener recurso; si falla, no tocar estado
auto* resource = AudioCache::getMusic(name);
if (resource == nullptr) {
// manejo de error opcional
return;
}
if (!music_enabled_) { return; }
// Si hay algo reproduciéndose, detenerlo primero (si el backend lo requiere)
if (music_.state == MusicState::PLAYING) {
JA_StopMusic(); // sustituir por la función de stop real del API si tiene otro nombre
}
auto* resource = AudioResource::getMusic(name);
if (resource == nullptr) { return; }
// Llamada al motor para reproducir la nueva pista
JA_PlayMusic(resource, loop);
// Actualizar estado y metadatos después de start con éxito
playMusicInternal(resource, loop, crossfade_ms);
music_.name = name;
music_.loop = new_loop;
music_.state = MusicState::PLAYING;
}
// Pausa la música
// Reprodueix la música per punter (amb crossfade opcional)
void Audio::playMusic(Ja::Music* music, const int loop, const int crossfade_ms) {
if (!music_enabled_ || music == nullptr) { return; }
playMusicInternal(music, loop, crossfade_ms);
// Si el Ja::Music es va crear con filename (loadMusic con 3 arguments), el
// recuperem porque getCurrentMusicName() no menteixi. Si no, music_.name
// queda buit — el contracte d'este overload no garanteix el nom.
music_.name = music->filename;
}
// Camí comú dels dos overloads: fa el dispatch crossfade vs stop+play i
// actualitza el loop cachejat. Els callers s'encarreguen del gating
// (music_enabled_, nullptr, same-track early return) y del nom. L'estat el
// manté Ja (Ja::playMusic posa PLAYING al Ja::Music* corresponent).
void Audio::playMusicInternal(Ja::Music* music, const int loop, const int crossfade_ms) {
const bool CURRENTLY_PLAYING = (getMusicState() == MusicState::PLAYING);
if (crossfade_ms > 0 && CURRENTLY_PLAYING) {
engine_->crossfadeMusic(music, crossfade_ms, loop);
} else {
if (CURRENTLY_PLAYING) {
engine_->stopMusic();
}
engine_->playMusic(music, loop);
}
music_.loop = (loop != 0);
}
// Pausa la música (l'estat el transiciona Engine::pauseMusic)
void Audio::pauseMusic() {
if (music_enabled_ && music_.state == MusicState::PLAYING) {
JA_PauseMusic();
music_.state = MusicState::PAUSED;
if (music_enabled_ && getMusicState() == MusicState::PLAYING) {
engine_->pauseMusic();
}
}
// Continua la música pausada
// Continua la música pausada (l'estat el transiciona Engine::resumeMusic)
void Audio::resumeMusic() {
if (music_enabled_ && music_.state == MusicState::PAUSED) {
JA_ResumeMusic();
music_.state = MusicState::PLAYING;
if (music_enabled_ && getMusicState() == MusicState::PAUSED) {
engine_->resumeMusic();
}
}
// Detiene la música
// Atura la música (l'estat el transiciona Engine::stopMusic)
void Audio::stopMusic() {
if (music_enabled_) {
JA_StopMusic();
music_.state = MusicState::STOPPED;
engine_->stopMusic();
}
}
// Reproduce un sonido por nombre
void Audio::playSound(const std::string& name, Group group) const {
void Audio::setMusicSpeed(float ratio) {
if (music_enabled_) {
engine_->setMusicSpeed(ratio);
}
}
// Reprodueix un so per nom
void Audio::playSound(const std::string& name, Group group) {
if (sound_enabled_) {
JA_PlaySound(AudioCache::getSound(name), 0, static_cast<int>(group));
engine_->playSound(AudioResource::getSound(name), 0, static_cast<int>(group));
}
}
// Reproduce un sonido por puntero directo
void Audio::playSound(JA_Sound_t* sound, Group group) const {
// Reprodueix un so per punter directe
void Audio::playSound(Ja::Sound* sound, Group group) {
if (sound_enabled_ && sound != nullptr) {
engine_->playSound(sound, 0, static_cast<int>(group));
}
}
// Variant con velocitat (i to) escalats. Apliquem el ratio al canal
// just retornat per `playSound`: así el `SDL_AudioStream` recent creat
// processa tot el sample con el ratio des del primer pull del callback.
// Si l'engine torna -1 (sense canal lliure) o el so no existeix, no fem
// la crida al ratio — sin efectes col·laterals.
void Audio::playSound(const std::string& name, Group group, float speed) {
if (!sound_enabled_) { return; }
auto* sound = AudioResource::getSound(name);
if (sound == nullptr) { return; }
const int CH = engine_->playSound(sound, 0, static_cast<int>(group));
if (CH >= 0 && speed != 1.0F) {
engine_->setChannelSpeed(CH, speed);
}
}
// Reprodueix un so processat per un eco definit a sounds.yaml. Si el preset no
// existeix o l'engine retorna -1 (sin de canals d'efecte plé), cau a playSound
// sec — l'usuari sent el so aún que la cua no s'apliqui.
void Audio::playSoundWithEcho(const std::string& name, const std::string& preset_name, Group group) {
if (!sound_enabled_) { return; }
auto* sound = AudioResource::getSound(name);
if (sound == nullptr) { return; }
const auto* params = SoundEffectsConfig::get().findEcho(preset_name);
if (params == nullptr) {
std::fprintf(stderr, "Audio: preset d'eco '%s' desconegut — so '%s' es reprodueix sec\n", preset_name.c_str(), name.c_str());
engine_->playSound(sound, 0, static_cast<int>(group));
return;
}
if (engine_->playSoundWithEcho(sound, *params, static_cast<int>(group)) < 0) {
engine_->playSound(sound, 0, static_cast<int>(group));
}
}
// Reprodueix un so processat per un reverb definit a sounds.yaml. Mateix
// fallback que playSoundWithEcho.
void Audio::playSoundWithReverb(const std::string& name, const std::string& preset_name, Group group) {
if (!sound_enabled_) { return; }
auto* sound = AudioResource::getSound(name);
if (sound == nullptr) { return; }
const auto* params = SoundEffectsConfig::get().findReverb(preset_name);
if (params == nullptr) {
std::fprintf(stderr, "Audio: preset de reverb '%s' desconegut — so '%s' es reprodueix sec\n", preset_name.c_str(), name.c_str());
engine_->playSound(sound, 0, static_cast<int>(group));
return;
}
if (engine_->playSoundWithReverb(sound, *params, static_cast<int>(group)) < 0) {
engine_->playSound(sound, 0, static_cast<int>(group));
}
}
// Atura tots los sons
void Audio::stopAllSounds() {
if (sound_enabled_) {
JA_PlaySound(sound, 0, static_cast<int>(group));
engine_->stopChannel(-1);
}
}
// Detiene todos los sonidos
void Audio::stopAllSounds() const {
if (sound_enabled_) {
JA_StopChannel(-1);
// Fa una fosa de sortida de la música
void Audio::fadeOutMusic(int milliseconds) {
if (music_enabled_ && getMusicState() == MusicState::PLAYING) {
engine_->fadeOutMusic(milliseconds);
}
}
// Realiza un fundido de salida de la música
void Audio::fadeOutMusic(int milliseconds) const {
if (music_enabled_ && getRealMusicState() == MusicState::PLAYING) {
JA_FadeOutMusic(milliseconds);
}
// Registra un callback que el motor dispararà cuando la pista actual acabi de
// drenar (times == 0 + stream buit). S'executa al mismo thread que
// Audio::update (render loop); los consumidors no poden fer I/O blocant.
void Audio::setOnMusicEnded(std::function<void()> callback) {
if (engine_) { engine_->setOnMusicEnded(std::move(callback)); }
}
// Consulta directamente el estado real de la música en jailaudio
auto Audio::getRealMusicState() -> MusicState {
JA_Music_state ja_state = JA_GetMusicState();
switch (ja_state) {
case JA_MUSIC_PLAYING:
// Resol el nom contra el cache de recursos i retorna la duración pre-calculada
// al `loadMusic`. 0 si la pista no existeix — así el caller pot decidir
// fallback (p. ex. usar un timeout fix) sin haver de propagar errors.
auto Audio::getMusicDurationMs(const std::string& name) -> int {
auto* music = AudioResource::getMusic(name);
return (music != nullptr) ? music->duration_ms : 0;
}
// Consulta directament l'estat a Ja y el projecta al subconjunt d'estats que
// exposa Audio (INVALID/DISABLED de Ja col·lapsen a STOPPED — la capa d'usuari
// solo vol saber si está sonant, pausat o parat).
auto Audio::getMusicState() -> MusicState {
if (!instance || !instance->engine_) { return MusicState::STOPPED; }
switch (instance->engine_->getMusicState()) {
case Ja::MusicState::PLAYING:
return MusicState::PLAYING;
case JA_MUSIC_PAUSED:
case Ja::MusicState::PAUSED:
return MusicState::PAUSED;
case JA_MUSIC_STOPPED:
case JA_MUSIC_INVALID:
case JA_MUSIC_DISABLED:
case Ja::MusicState::STOPPED:
case Ja::MusicState::INVALID:
default:
return MusicState::STOPPED;
}
}
// Establece el volumen de los sonidos
void Audio::setSoundVolume(float sound_volume, Group group) const {
if (sound_enabled_) {
sound_volume = std::clamp(sound_volume, MIN_VOLUME, MAX_VOLUME);
const float CONVERTED_VOLUME = sound_volume * Options::audio.volume;
JA_SetSoundVolume(CONVERTED_VOLUME, static_cast<int>(group));
}
// Aplica el gate master (enabled_) + el gate del canal (sound/music_enabled_)
// i retorna el volum escalat pel master config_.volume. 0 si algun gate está
// tancat. Así los dos setters comparteixen la misma política.
auto Audio::effectiveVolume(float volume, bool channel_enabled) const -> float {
volume = std::clamp(volume, MIN_VOLUME, MAX_VOLUME);
return (enabled_ && channel_enabled) ? volume * config_.volume : 0.0F;
}
// Establece el volumen de la música
void Audio::setMusicVolume(float music_volume) const {
if (music_enabled_) {
music_volume = std::clamp(music_volume, MIN_VOLUME, MAX_VOLUME);
const float CONVERTED_VOLUME = music_volume * Options::audio.volume;
JA_SetMusicVolume(CONVERTED_VOLUME);
}
// Estableix el volum dels sons (float 0.0..1.0)
void Audio::setSoundVolume(float sound_volume, Group group) {
engine_->setSoundVolume(effectiveVolume(sound_volume, sound_enabled_), static_cast<int>(group));
}
// Aplica la configuración
void Audio::applySettings() {
enable(Options::audio.enabled);
// Estableix el volum de la música (float 0.0..1.0)
void Audio::setMusicVolume(float music_volume) {
engine_->setMusicVolume(effectiveVolume(music_volume, music_enabled_));
}
// Establecer estado general
// Aplica una nueva configuración (substitueix la config cachejada i reaplica enables/volums)
void Audio::applySettings(const Config& config) {
config_ = config;
sound_enabled_ = config_.sound_enabled;
music_enabled_ = config_.music_enabled;
enable(config_.enabled);
}
// Estableix l'estat general
void Audio::enable(bool value) {
enabled_ = value;
setSoundVolume(enabled_ ? Options::audio.sound.volume : MIN_VOLUME);
setMusicVolume(enabled_ ? Options::audio.music.volume : MIN_VOLUME);
setSoundVolume(enabled_ ? config_.sound_volume : MIN_VOLUME);
setMusicVolume(enabled_ ? config_.music_volume : MIN_VOLUME);
}
// Inicializa SDL Audio
// Estableix l'estat dels sons i reaplica el volum porque los canals actius
// responguin a l'instant (evita que el toggle solo surti efecte al pròxim
// setSoundVolume explícit).
void Audio::enableSound(bool value) {
sound_enabled_ = value;
setSoundVolume(config_.sound_volume);
}
// Estableix l'estat de la música i reaplica el volum per la misma raó.
void Audio::enableMusic(bool value) {
music_enabled_ = value;
setMusicVolume(config_.music_volume);
}
// Inicialitza SDL Audio y el motor Ja::Engine owned.
void Audio::initSDLAudio() {
if (!SDL_Init(SDL_INIT_AUDIO)) {
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION, "SDL_AUDIO could not initialize! SDL Error: %s", SDL_GetError());
} else {
JA_Init(FREQUENCY, SDL_AUDIO_S16LE, 2);
enable(Options::audio.enabled);
std::cout << "\n** AUDIO SYSTEM **\n";
std::cout << "Audio system initialized successfully\n";
std::fprintf(stderr, "Audio: SDL_AUDIO could not initialize! SDL Error: %s\n", SDL_GetError());
return;
}
engine_ = std::make_unique<Ja::Engine>(Defaults::Audio::FREQUENCY, Defaults::Audio::FORMAT, Defaults::Audio::CHANNELS);
sound_enabled_ = config_.sound_enabled;
music_enabled_ = config_.music_enabled;
enable(config_.enabled);
}
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#pragma once
#include <string> // Para string
#include <utility> // Para move
#include <cmath> // Para std::lround
#include <cstdint> // Para int8_t, uint8_t
#include <functional> // Para std::function
#include <memory> // Para std::unique_ptr
#include <string> // Para string
#include <utility> // Para move
// --- Clase Audio: gestor de audio (singleton) ---
// Forward-declares per no incloure core/audio/jail_audio.hpp al header. Els
// tres símbols (Music/Sound para el punter que exposa la API i Engine per al
// std::unique_ptr<Engine> membre) s'usen solo per punter al header, así que
// el forward-decl basta. El ~Audio() en .cpp veu la definició completa i
// instancia correctament el dtor de l'unique_ptr.
namespace Ja {
class Engine;
struct Music;
struct Sound;
} // namespace Ja
// --- Clase Audio: gestor d'àudio (singleton) ---
// Port del subsistema d'àudio del projecte ../aee, desacoblat d'Options:
// la configuración entra per la struct Audio::Config a init()/applySettings(),
// en lloc de llegir directament Options::audio. Això deixa audio.cpp independent
// del layout d'Options i permet substituir la font de configuración.
//
// Els volums es manegen internament como a float 0.01.0; la capa de
// presentació (menús, notificacions) usa las helpers toPercent/fromPercent
// per mostrar 0100 a l'usuari.
class Audio {
public:
// --- Enums ---
enum class Group : int {
ALL = -1, // Todos los grupos
GAME = 0, // Sonidos del juego
INTERFACE = 1 // Sonidos de la interfaz
};
public:
// --- Configuración injectada (Options la construeix via buildAudioConfig) ---
struct Config {
bool enabled{true};
float volume{1.0F}; // Master 0..1
bool music_enabled{true};
float music_volume{0.8F};
bool sound_enabled{true};
float sound_volume{1.0F};
};
enum class MusicState {
PLAYING, // Reproduciendo música
PAUSED, // Música pausada
STOPPED, // Música detenida
};
// --- Enums ---
enum class Group : std::int8_t {
ALL = -1, // Tots los grups
GAME = 0, // Sons del joc
INTERFACE = 1 // Sons de la interfície
};
// --- Constantes ---
static constexpr float MAX_VOLUME = 1.0F; // Volumen máximo
static constexpr float MIN_VOLUME = 0.0F; // Volumen mínimo
static constexpr int FREQUENCY = 48000; // Frecuencia de audio
enum class MusicState : std::uint8_t {
PLAYING, // Reproduint música
PAUSED, // Música pausada
STOPPED, // Música aturada
};
// --- Singleton ---
static void init(); // Inicializa el objeto Audio
static void destroy(); // Libera el objeto Audio
static auto get() -> Audio*; // Obtiene el puntero al objeto Audio
Audio(const Audio&) = delete; // Evitar copia
auto operator=(const Audio&) -> Audio& = delete; // Evitar asignación
// --- Constants ---
static constexpr float MAX_VOLUME = 1.0F; // Volum màxim (float 0..1)
static constexpr float MIN_VOLUME = 0.0F; // Volum mínim (float 0..1)
static void update(); // Actualización del sistema de audio
// --- Singleton ---
static void init(const Config& config); // Inicialitza con la configuración rebuda
static void destroy(); // Allibera l'objecte Audio
static auto get() -> Audio*; // Obté el punter a l'objecte Audio
~Audio(); // Destructor (públic para std::unique_ptr)
Audio(const Audio&) = delete; // Evitar còpia
Audio(Audio&&) = delete;
auto operator=(const Audio&) -> Audio& = delete; // Evitar assignació
auto operator=(Audio&&) -> Audio& = delete;
// --- Control de música ---
void playMusic(const std::string& name, int loop = -1); // Reproducir música en bucle
void pauseMusic(); // Pausar reproducción de música
void resumeMusic(); // Continua la música pausada
void stopMusic(); // Detener completamente la música
void fadeOutMusic(int milliseconds) const; // Fundido de salida de la música
static void update(); // Actualització del sistema d'àudio
// --- Control de sonidos ---
void playSound(const std::string& name, Group group = Group::GAME) const; // Reproducir sonido puntual por nombre
void playSound(struct JA_Sound_t* sound, Group group = Group::GAME) const; // Reproducir sonido puntual por puntero
void stopAllSounds() const; // Detener todos los sonidos
// --- Control de música ---
void playMusic(const std::string& name, int loop = -1, int crossfade_ms = 0); // Reproduir música per nom (amb crossfade opcional)
void playMusic(Ja::Music* music, int loop = -1, int crossfade_ms = 0); // Reproduir música per punter (amb crossfade opcional)
void pauseMusic(); // Pausar la reproducció de música
void resumeMusic(); // Continua la música pausada
void stopMusic(); // Aturar completament la música
void fadeOutMusic(int milliseconds); // Fosa de sortida de la música (muta globals de Ja)
void setOnMusicEnded(std::function<void()> callback); // Callback disparat cuando la pista actual acaba de drenar (CONV-03)
// Multiplicador de velocitat de la música actual. 1.0 = normal,
// 1.5 = un 50% més ràpid (efecte "chipmunk" — también puja el to).
// Es reseteja a 1.0 implícitament a cada `playMusic`. No-op si no
// hay música activa.
void setMusicSpeed(float ratio);
// --- Control de volumen ---
void setSoundVolume(float volume, Group group = Group::ALL) const; // Ajustar volumen de efectos
void setMusicVolume(float volume) const; // Ajustar volumen de música
// --- Control de sons ---
void playSound(const std::string& name, Group group = Group::GAME); // Reproduir so puntual per nom (muta globals de Ja)
void playSound(Ja::Sound* sound, Group group = Group::GAME); // Reproduir so puntual per punter (muta globals de Ja)
// Reprodueix un so con la velocitat (i to) escalats per `speed`:
// 1.0 = normal, 0.95 ≈ -5% (més greu i lent), 1.05 ≈ +5% (més
// agut i ràpid). Mateixa semàntica que `setMusicSpeed`. Útil per a
// variacions subtils que eviten la fatiga d'escoltar el mismo
// sample idèntic (p.ex. obertures de sarcòfag, picks d'ítems).
void playSound(const std::string& name, Group group, float speed);
// Reprodueix un so processat per un efecte definit a data/config/sounds.yaml
// (preset_name busca a SoundEffectsConfig). Si el preset no existeix
// o el motor está al sin de canals con efecte, fa fallback a playSound
// sec — l'usuari sent el so igualment, sin la cua.
void playSoundWithEcho(const std::string& name, const std::string& preset_name, Group group = Group::GAME);
void playSoundWithReverb(const std::string& name, const std::string& preset_name, Group group = Group::GAME);
void stopAllSounds(); // Aturar tots los sons (muta globals de Ja)
// --- Configuración general ---
void enable(bool value); // Establecer estado general
void toggleEnabled() { enabled_ = !enabled_; } // Alternar estado general
void applySettings(); // Aplica la configuración
// --- Control de volum (API interna: float 0.0..1.0) ---
void setSoundVolume(float volume, Group group = Group::ALL); // Ajusta el volum dels efectes
void setMusicVolume(float volume); // Ajusta el volum de la música
// --- Configuración de sonidos ---
void enableSound() { sound_enabled_ = true; } // Habilitar sonidos
void disableSound() { sound_enabled_ = false; } // Deshabilitar sonidos
void enableSound(bool value) { sound_enabled_ = value; } // Establecer estado de sonidos
void toggleSound() { sound_enabled_ = !sound_enabled_; } // Alternar estado de sonidos
// --- Helpers de conversió para la capa de presentació ---
// UI (menús, notificacions) manega enters 0..100; internament viu float 0..1.
// No són constexpr porque std::lround no ho es en C++20; s'usen en runtime.
static auto toPercent(float volume) -> int {
return static_cast<int>(std::lround(volume * 100.0F));
}
static auto fromPercent(int percent) -> float {
return static_cast<float>(percent) / 100.0F;
}
// --- Configuración de música ---
void enableMusic() { music_enabled_ = true; } // Habilitar música
void disableMusic() { music_enabled_ = false; } // Deshabilitar música
void enableMusic(bool value) { music_enabled_ = value; } // Establecer estado de música
void toggleMusic() { music_enabled_ = !music_enabled_; } // Alternar estado de música
// --- Configuración general ---
void enable(bool value); // Estableix l'estat general (reaplica volums)
void toggleEnabled() { enable(!enabled_); } // Alterna l'estat general (reaplica volums)
void applySettings(const Config& config); // Aplica una nueva configuración
// --- Consultas de estado ---
[[nodiscard]] auto isEnabled() const -> bool { return enabled_; }
[[nodiscard]] auto isSoundEnabled() const -> bool { return sound_enabled_; }
[[nodiscard]] auto isMusicEnabled() const -> bool { return music_enabled_; }
[[nodiscard]] auto getMusicState() const -> MusicState { return music_.state; }
[[nodiscard]] static auto getRealMusicState() -> MusicState;
[[nodiscard]] auto getCurrentMusicName() const -> const std::string& { return music_.name; }
// --- Configuración de sons ---
void enableSound(bool value); // Estableix l'estat dels sons (reaplica volum)
void toggleSound() { enableSound(!sound_enabled_); } // Alterna l'estat dels sons (reaplica volum)
private:
// --- Tipos anidados ---
struct Music {
MusicState state{MusicState::STOPPED}; // Estado actual de la música
std::string name; // Última pista de música reproducida
bool loop{false}; // Indica si se reproduce en bucle
};
// --- Configuración de música ---
void enableMusic(bool value); // Estableix l'estat de la música (reaplica volum)
void toggleMusic() { enableMusic(!music_enabled_); } // Alterna l'estat de la música (reaplica volum)
// --- Métodos ---
Audio(); // Constructor privado
~Audio(); // Destructor privado
void initSDLAudio(); // Inicializa SDL Audio
// --- Consultes d'estat ---
[[nodiscard]] auto isEnabled() const -> bool { return enabled_; }
[[nodiscard]] auto isSoundEnabled() const -> bool { return sound_enabled_; }
[[nodiscard]] auto isMusicEnabled() const -> bool { return music_enabled_; }
[[nodiscard]] static auto getMusicState() -> MusicState; // Estat real consultat a Ja::
[[nodiscard]] auto getCurrentMusicName() const -> const std::string& { return music_.name; }
// Duración de la pista resolta per nom (mil·lisegons). 0 si la pista no
// existeix al cache de recursos o si el seu header OGG no permet
// calcular-la. Pensat para clients que necessiten un timeline
// determinista (p. ex. RoomFsm) sin dependre de callbacks de fi.
[[nodiscard]] static auto getMusicDurationMs(const std::string& name) -> int;
// --- Variables miembro ---
static Audio* instance; // Instancia única de Audio
private:
// --- Tipus anidats ---
struct Music {
std::string name; // Última pista de música reproduïda (buida si es va passar per punter sin filename)
bool loop{false}; // Si el play actual es en bucle
};
Music music_; // Estado de la música
bool enabled_{true}; // Estado general del audio
bool sound_enabled_{true}; // Estado de los efectos de sonido
bool music_enabled_{true}; // Estado de lasica
// --- Mètodes ---
explicit Audio(const Config& config); // Constructor privat: rep la config
void initSDLAudio(); // Inicialitza SDL Audio
void playMusicInternal(Ja::Music* music, int loop, int crossfade_ms); // Camí comú dels dos overloads de playMusic
[[nodiscard]] auto effectiveVolume(float volume, bool channel_enabled) const -> float; // Gate master+channel: 0 si algun está off, clamp 0..1 altrament
// --- Variables membre ---
static std::unique_ptr<Audio> instance; // Instància única d'Audio
std::unique_ptr<Ja::Engine> engine_; // Motor de baix nivell (owned); viu mentre Audio viu.
Config config_{}; // Configuración injectada (volums, enables)
Music music_; // Estat de la música (nom + loop cachejats)
bool enabled_{true}; // Estat general de l'àudio
bool sound_enabled_{true}; // Estat dels efectes de so
bool music_enabled_{true}; // Estat de la música
};
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// audio_adapter.cpp - Implementación de AudioResource para orni_attack
// © 2025 Orni Attack
//
// Implementa AudioResource::getMusic / getSound delegando a
// Resource::Helper::loadFile (que abstrae el resources.pack y el fallback
// a filesystem). Cache local de Ja::Music* / Ja::Sound* con lazy load:
// cada recurso se carga la primera vez que se pide y se mantiene vivo
// hasta el shutdown.
#include "core/audio/audio_adapter.hpp"
#include <cstdint>
#include <iostream>
#include <memory>
#include <string>
#include <unordered_map>
#include "core/audio/jail_audio.hpp"
#include "core/resources/resource_helper.hpp"
namespace {
// Cachés locales: indexados por nombre lógico ("title.ogg", "effects/laser_shoot.wav", etc.)
// Mantienen ownership con unique_ptr; se liberan al salir del programa.
std::unordered_map<std::string, std::unique_ptr<Ja::Music>>& musicCache() {
static std::unordered_map<std::string, std::unique_ptr<Ja::Music>> cache;
return cache;
}
std::unordered_map<std::string, std::unique_ptr<Ja::Sound>>& soundCache() {
static std::unordered_map<std::string, std::unique_ptr<Ja::Sound>> cache;
return cache;
}
// Normaliza el nombre añadiendo la subcarpeta correspondiente si no la trae:
// "title.ogg" -> "music/title.ogg"
// "music/title.ogg" -> "music/title.ogg"
// "effects/laser.wav" -> "sounds/effects/laser.wav"
std::string normalizeMusicPath(const std::string& name) {
return (name.rfind("music/", 0) == 0) ? name : "music/" + name;
}
std::string normalizeSoundPath(const std::string& name) {
return (name.rfind("sounds/", 0) == 0) ? name : "sounds/" + name;
}
} // namespace
namespace AudioResource {
auto getMusic(const std::string& name) -> Ja::Music* {
auto& cache = musicCache();
if (auto it = cache.find(name); it != cache.end()) {
return it->second.get();
}
const std::string path = normalizeMusicPath(name);
auto bytes = Resource::Helper::loadFile(path);
if (bytes.empty()) {
std::cerr << "[AudioResource] no se ha podido cargar música: " << path << "\n";
return nullptr;
}
Ja::Music* raw = Ja::loadMusic(bytes.data(), static_cast<std::uint32_t>(bytes.size()), name.c_str());
if (raw == nullptr) {
std::cerr << "[AudioResource] decodificación de música falló: " << path << "\n";
return nullptr;
}
cache.emplace(name, std::unique_ptr<Ja::Music>(raw));
std::cout << "[AudioResource] música cargada: " << path << "\n";
return raw;
}
auto getSound(const std::string& name) -> Ja::Sound* {
auto& cache = soundCache();
if (auto it = cache.find(name); it != cache.end()) {
return it->second.get();
}
const std::string path = normalizeSoundPath(name);
auto bytes = Resource::Helper::loadFile(path);
if (bytes.empty()) {
std::cerr << "[AudioResource] no se ha podido cargar sonido: " << path << "\n";
return nullptr;
}
Ja::Sound* raw = Ja::loadSound(bytes.data(), static_cast<std::uint32_t>(bytes.size()));
if (raw == nullptr) {
std::cerr << "[AudioResource] decodificación de sonido falló: " << path << "\n";
return nullptr;
}
cache.emplace(name, std::unique_ptr<Ja::Sound>(raw));
std::cout << "[AudioResource] sonido cargado: " << path << "\n";
return raw;
}
} // namespace AudioResource
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#pragma once
// --- Audio Resource Adapter ---
// Este archivo exposa una interfície comuna a Audio per obtenir Ja::Music* /
// Ja::Sound* per nom. Cada projecte la implementa en audio_adapter.cpp delegant
// al seu singleton de recursos (Resource::Cache::get(), ...). Así audio.hpp
// i audio.cpp es poden compartir entre projectes.
#include <string> // Para string
namespace Ja {
struct Music;
struct Sound;
} // namespace Ja
namespace AudioResource {
auto getMusic(const std::string& name) -> Ja::Music*;
auto getSound(const std::string& name) -> Ja::Sound*;
} // namespace AudioResource
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// audio_cache.cpp - Implementació del caché de sons i música
// © 2025 Port a C++20 con SDL3
#include "core/audio/audio_cache.hpp"
#include <iostream>
#include "core/resources/resource_helper.hpp"
// Inicialización de variables estàtiques
std::unordered_map<std::string, JA_Sound_t*> AudioCache::sounds_;
std::unordered_map<std::string, JA_Music_t*> AudioCache::musics_;
std::string AudioCache::sounds_base_path_ = "data/sounds/";
std::string AudioCache::music_base_path_ = "data/music/";
JA_Sound_t* AudioCache::getSound(const std::string& name) {
// Cache hit
auto it = sounds_.find(name);
if (it != sounds_.end()) {
std::cout << "[AudioCache] Sound cache hit: " << name << std::endl;
return it->second;
}
// Normalize path: "laser_shoot.wav" → "sounds/laser_shoot.wav"
std::string normalized = name;
if (normalized.find("sounds/") != 0) {
normalized = "sounds/" + normalized;
}
// Load from resource system
std::vector<uint8_t> data = Resource::Helper::loadFile(normalized);
if (data.empty()) {
std::cerr << "[AudioCache] Error: no s'ha pogut load " << normalized << std::endl;
return nullptr;
}
// Load sound from memory
JA_Sound_t* sound = JA_LoadSound(data.data(), static_cast<uint32_t>(data.size()));
if (sound == nullptr) {
std::cerr << "[AudioCache] Error: no s'ha pogut decodificar " << normalized
<< std::endl;
return nullptr;
}
std::cout << "[AudioCache] Sound loaded: " << normalized << std::endl;
sounds_[name] = sound;
return sound;
}
JA_Music_t* AudioCache::getMusic(const std::string& name) {
// Cache hit
auto it = musics_.find(name);
if (it != musics_.end()) {
std::cout << "[AudioCache] Music cache hit: " << name << std::endl;
return it->second;
}
// Normalize path: "title.ogg" → "music/title.ogg"
std::string normalized = name;
if (normalized.find("music/") != 0) {
normalized = "music/" + normalized;
}
// Load from resource system
std::vector<uint8_t> data = Resource::Helper::loadFile(normalized);
if (data.empty()) {
std::cerr << "[AudioCache] Error: no s'ha pogut load " << normalized << std::endl;
return nullptr;
}
// Load music from memory
JA_Music_t* music = JA_LoadMusic(data.data(), static_cast<uint32_t>(data.size()));
if (music == nullptr) {
std::cerr << "[AudioCache] Error: no s'ha pogut decodificar " << normalized
<< std::endl;
return nullptr;
}
std::cout << "[AudioCache] Music loaded: " << normalized << std::endl;
musics_[name] = music;
return music;
}
void AudioCache::clear() {
std::cout << "[AudioCache] Clearing cache (" << sounds_.size() << " sounds, "
<< musics_.size() << " music)" << std::endl;
// Liberar memoria de sonidos
for (auto& [name, sound] : sounds_) {
if (sound && sound->buffer) {
SDL_free(sound->buffer);
}
delete sound;
}
sounds_.clear();
// Liberar memoria de música
for (auto& [name, music] : musics_) {
if (music && music->buffer) {
SDL_free(music->buffer);
}
if (music && music->filename) {
free(music->filename);
}
delete music;
}
musics_.clear();
}
size_t AudioCache::getSoundCacheSize() { return sounds_.size(); }
size_t AudioCache::getMusicCacheSize() { return musics_.size(); }
std::string AudioCache::resolveSoundPath(const std::string& name) {
// Si es un path absoluto (comienza con '/'), usarlo directamente
if (!name.empty() && name[0] == '/') {
return name;
}
// Si ya contiene el prefix base_path, usarlo directamente
if (name.find(sounds_base_path_) == 0) {
return name;
}
// Caso contrario, añadir base_path
return sounds_base_path_ + name;
}
std::string AudioCache::resolveMusicPath(const std::string& name) {
// Si es un path absoluto (comienza con '/'), usarlo directamente
if (!name.empty() && name[0] == '/') {
return name;
}
// Si ya contiene el prefix base_path, usarlo directamente
if (name.find(music_base_path_) == 0) {
return name;
}
// Caso contrario, añadir base_path
return music_base_path_ + name;
}
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// audio_cache.hpp - Caché simplificado de sonidos y música
// © 2025 Port a C++20 con SDL3
#pragma once
#include <string>
#include <unordered_map>
#include "core/audio/jail_audio.hpp"
// Caché estático de sonidos y música
// Patrón inspirado en Graphics::ShapeLoader
class AudioCache {
public:
// No instanciable (todo estático)
AudioCache() = delete;
// Obtener sonido (carga bajo demanda)
// Retorna puntero (nullptr si error)
static JA_Sound_t* getSound(const std::string& name);
// Obtener música (carga bajo demanda)
// Retorna puntero (nullptr si error)
static JA_Music_t* getMusic(const std::string& name);
// Limpiar caché (útil para debug/recarga)
static void clear();
// Estadísticas (debug)
static size_t getSoundCacheSize();
static size_t getMusicCacheSize();
private:
static std::unordered_map<std::string, JA_Sound_t*> sounds_;
static std::unordered_map<std::string, JA_Music_t*> musics_;
static std::string sounds_base_path_; // "data/sounds/"
static std::string music_base_path_; // "data/music/"
// Helpers privados
static std::string resolveSoundPath(const std::string& name);
static std::string resolveMusicPath(const std::string& name);
};
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#include "core/audio/audio_effects.hpp"
#include <algorithm>
#include <array>
#include <cmath>
#include <cstdint>
#include <cstring>
#include <vector>
#include <iostream>
#include "core/audio/jail_audio.hpp"
namespace AudioEffects {
namespace {
// --- Caps de cua ---
constexpr float ECHO_TAIL_MS = 800.0F;
constexpr float REVERB_TAIL_MS = 1500.0F;
// --- Constants Freeverb ---
// Delays de comb i allpass tunats para 44.1 kHz; los reescalem per
// freqüència real de la font.
constexpr int COMB_REFERENCE_RATE = 44100;
constexpr std::array<int, 8> COMB_DELAYS_L = {1116, 1188, 1277, 1356, 1422, 1491, 1557, 1617};
constexpr std::array<int, 4> ALLPASS_DELAYS_L = {556, 441, 341, 225};
constexpr int STEREO_SPREAD = 23;
// Mapeig de Schroeder/Dattorro/Freeverb estàndard.
constexpr float FIXED_GAIN = 0.015F;
constexpr float SCALE_ROOM = 0.28F;
constexpr float OFFSET_ROOM = 0.7F;
constexpr float SCALE_DAMP = 0.4F;
// --- Decodificació a float -1..1 ---
// Suporta U8/S16, mono/estèreo. Mono es duplica a L i R (la cadena
// d'efectes treballa siempre con dos canals per simplicitat).
auto decodeToStereoFloat(const Ja::Sound& src, std::vector<float>& left, std::vector<float>& right) -> bool {
const auto& spec = src.spec;
const Uint8* buf = src.buffer.get();
if (buf == nullptr || src.length == 0) { return false; }
int bytes_per_sample = 0;
if (spec.format == SDL_AUDIO_S16) {
bytes_per_sample = 2;
} else if (spec.format == SDL_AUDIO_U8) {
bytes_per_sample = 1;
} else {
std::cerr << "[AudioEffects] formato de sonido no soportado (solo U8 o S16)\n";
return false;
}
if (spec.channels < 1 || spec.channels > 2) {
std::cerr << "[AudioEffects] el sonido debe ser mono o estéreo\n";
return false;
}
const std::size_t TOTAL_FRAMES = src.length / static_cast<std::size_t>(bytes_per_sample * spec.channels);
left.resize(TOTAL_FRAMES);
right.resize(TOTAL_FRAMES);
for (std::size_t i = 0; i < TOTAL_FRAMES; ++i) {
float sample_l = 0.0F;
float sample_r = 0.0F;
if (spec.format == SDL_AUDIO_S16) {
const auto* p = reinterpret_cast<const std::int16_t*>(buf + (i * spec.channels * 2));
sample_l = static_cast<float>(p[0]) / 32768.0F;
sample_r = (spec.channels == 2) ? static_cast<float>(p[1]) / 32768.0F : sample_l;
} else { // U8
const Uint8* p = buf + (i * spec.channels);
sample_l = (static_cast<float>(p[0]) - 128.0F) / 128.0F;
sample_r = (spec.channels == 2) ? (static_cast<float>(p[1]) - 128.0F) / 128.0F : sample_l;
}
left[i] = sample_l;
right[i] = sample_r;
}
return true;
}
// Empaqueta dos canals float (-1..1) a S16 entrellaçat.
void encodeStereoS16(const std::vector<float>& left, const std::vector<float>& right, std::vector<std::uint8_t>& out) {
const std::size_t LEN = left.size();
out.resize(LEN * 2 * sizeof(std::int16_t));
auto* dst = reinterpret_cast<std::int16_t*>(out.data());
for (std::size_t i = 0; i < LEN; ++i) {
const float L = std::clamp(left[i], -1.0F, 1.0F);
const float R = std::clamp(right[i], -1.0F, 1.0F);
dst[(i * 2) + 0] = static_cast<std::int16_t>(std::lround(L * 32767.0F));
dst[(i * 2) + 1] = static_cast<std::int16_t>(std::lround(R * 32767.0F));
}
}
// Reescala un delay de la taula de Freeverb para la freqüència real.
auto scaledDelay(int reference_delay, int rate) -> int {
const long SCALED = std::lround(static_cast<double>(reference_delay) * static_cast<double>(rate) / static_cast<double>(COMB_REFERENCE_RATE));
return std::max(1, static_cast<int>(SCALED));
}
// --- Filtres bàsics ---
struct Comb {
std::vector<float> buf;
std::size_t idx{0};
float feedback{0.0F};
float damp1{0.0F};
float damp2{1.0F};
float store{0.0F};
void init(int delay, float fb, float damping) {
buf.assign(static_cast<std::size_t>(delay), 0.0F);
idx = 0;
feedback = fb;
damp1 = damping;
damp2 = 1.0F - damping;
store = 0.0F;
}
auto tick(float in) -> float {
const float OUT = buf[idx];
store = (OUT * damp2) + (store * damp1);
buf[idx] = in + (store * feedback);
idx = (idx + 1) % buf.size();
return OUT;
}
};
struct Allpass {
std::vector<float> buf;
std::size_t idx{0};
void init(int delay) {
buf.assign(static_cast<std::size_t>(delay), 0.0F);
idx = 0;
}
auto tick(float in) -> float {
const float BUFOUT = buf[idx];
const float OUT = -in + BUFOUT;
buf[idx] = in + (BUFOUT * 0.5F);
idx = (idx + 1) % buf.size();
return OUT;
}
};
} // namespace
auto applyEcho(const Ja::Sound& src, const Ja::EchoParams& params) -> std::optional<ProcessedSound> {
std::vector<float> left;
std::vector<float> right;
if (!decodeToStereoFloat(src, left, right)) { return std::nullopt; }
const int RATE = src.spec.freq;
const int DELAY_SAMPLES = std::max(1, static_cast<int>(std::lround(params.delay_ms * 0.001F * static_cast<float>(RATE))));
const auto TAIL_SAMPLES = static_cast<std::size_t>(std::lround(ECHO_TAIL_MS * 0.001F * static_cast<float>(RATE)));
const float FEEDBACK = std::clamp(params.feedback, 0.0F, 0.95F);
const float WET = std::clamp(params.wet, 0.0F, 1.0F);
const float DRY = 1.0F - WET;
const std::size_t INPUT_LEN = left.size();
const std::size_t TOTAL_LEN = INPUT_LEN + TAIL_SAMPLES;
std::vector<float> ring_l(static_cast<std::size_t>(DELAY_SAMPLES), 0.0F);
std::vector<float> ring_r(static_cast<std::size_t>(DELAY_SAMPLES), 0.0F);
std::size_t cursor = 0;
std::vector<float> out_l(TOTAL_LEN);
std::vector<float> out_r(TOTAL_LEN);
for (std::size_t i = 0; i < TOTAL_LEN; ++i) {
const float IN_L = (i < INPUT_LEN) ? left[i] : 0.0F;
const float IN_R = (i < INPUT_LEN) ? right[i] : 0.0F;
const float DELAYED_L = ring_l[cursor];
const float DELAYED_R = ring_r[cursor];
out_l[i] = (DRY * IN_L) + (WET * DELAYED_L);
out_r[i] = (DRY * IN_R) + (WET * DELAYED_R);
ring_l[cursor] = IN_L + (DELAYED_L * FEEDBACK);
ring_r[cursor] = IN_R + (DELAYED_R * FEEDBACK);
cursor = (cursor + 1) % static_cast<std::size_t>(DELAY_SAMPLES);
}
ProcessedSound result;
result.spec = SDL_AudioSpec{.format = SDL_AUDIO_S16, .channels = 2, .freq = RATE};
encodeStereoS16(out_l, out_r, result.bytes);
return result;
}
auto applyReverb(const Ja::Sound& src, const Ja::ReverbParams& params) -> std::optional<ProcessedSound> {
std::vector<float> left;
std::vector<float> right;
if (!decodeToStereoFloat(src, left, right)) { return std::nullopt; }
const int RATE = src.spec.freq;
const auto TAIL_SAMPLES = static_cast<std::size_t>(std::lround(REVERB_TAIL_MS * 0.001F * static_cast<float>(RATE)));
const float ROOM_SIZE = std::clamp(params.room_size, 0.0F, 1.0F);
const float DAMPING = std::clamp(params.damping, 0.0F, 1.0F);
const float WET = std::clamp(params.wet, 0.0F, 1.0F);
const float DRY = 1.0F - WET;
const float FEEDBACK = (ROOM_SIZE * SCALE_ROOM) + OFFSET_ROOM; // 0.7..0.98
const float DAMP1 = DAMPING * SCALE_DAMP; // 0..0.4
// Inicialitza los 8 comb filters per cada canal i los 4 allpass.
std::array<Comb, 8> comb_l;
std::array<Comb, 8> comb_r;
for (std::size_t i = 0; i < COMB_DELAYS_L.size(); ++i) {
comb_l[i].init(scaledDelay(COMB_DELAYS_L[i], RATE), FEEDBACK, DAMP1);
comb_r[i].init(scaledDelay(COMB_DELAYS_L[i] + STEREO_SPREAD, RATE), FEEDBACK, DAMP1);
}
std::array<Allpass, 4> allpass_l;
std::array<Allpass, 4> allpass_r;
for (std::size_t i = 0; i < ALLPASS_DELAYS_L.size(); ++i) {
allpass_l[i].init(scaledDelay(ALLPASS_DELAYS_L[i], RATE));
allpass_r[i].init(scaledDelay(ALLPASS_DELAYS_L[i] + STEREO_SPREAD, RATE));
}
const std::size_t INPUT_LEN = left.size();
const std::size_t TOTAL_LEN = INPUT_LEN + TAIL_SAMPLES;
std::vector<float> out_l(TOTAL_LEN);
std::vector<float> out_r(TOTAL_LEN);
for (std::size_t i = 0; i < TOTAL_LEN; ++i) {
const float IN_L = (i < INPUT_LEN) ? left[i] : 0.0F;
const float IN_R = (i < INPUT_LEN) ? right[i] : 0.0F;
const float MONO_INPUT = (IN_L + IN_R) * FIXED_GAIN;
// 8 comb filters en paral·lel, sumats.
float wet_l = 0.0F;
float wet_r = 0.0F;
for (std::size_t k = 0; k < comb_l.size(); ++k) {
wet_l += comb_l[k].tick(MONO_INPUT);
wet_r += comb_r[k].tick(MONO_INPUT);
}
// 4 allpass en sèrie.
for (std::size_t k = 0; k < allpass_l.size(); ++k) {
wet_l = allpass_l[k].tick(wet_l);
wet_r = allpass_r[k].tick(wet_r);
}
out_l[i] = (DRY * IN_L) + (WET * wet_l);
out_r[i] = (DRY * IN_R) + (WET * wet_r);
}
ProcessedSound result;
result.spec = SDL_AudioSpec{.format = SDL_AUDIO_S16, .channels = 2, .freq = RATE};
encodeStereoS16(out_l, out_r, result.bytes);
return result;
}
} // namespace AudioEffects
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#pragma once
#include <SDL3/SDL.h>
#include <cstdint>
#include <optional>
#include <vector>
// Forward-declaració per no incloure jail_audio.hpp (cicle d'inclusió: este
// header viu sota los params declarats a jail_audio.hpp, i alhora jail_audio
// usa applyEcho/applyReverb).
namespace Ja {
struct Sound;
struct EchoParams;
struct ReverbParams;
} // namespace Ja
// Processadors d'efectes para sons puntuals. Reben un Ja::Sound (qualsevol
// format suportat pel decodificador WAV: U8/S16, mono o estèreo) i tornen un
// buffer PCM en S16 + el seu spec, llest per empenyer a un SDL_AudioStream.
//
// El buffer de sortida inclou la cua (decay) generada per l'efecte: per al
// reverb, hasta a 1500 ms; para l'eco, hasta a 800 ms. Aquests caps eviten
// allargar indefinidament la reproducció cuando los parámetros reinjecten mucho.
//
// Si el format del so d'origen no es pot processar, retornen std::nullopt
// (el caller ha de fer fallback a reproducció seca).
namespace AudioEffects {
struct ProcessedSound {
std::vector<std::uint8_t> bytes; // PCM S16 entrellaçat (LRLRLR... si stereo)
SDL_AudioSpec spec; // Format/canals/freqüència del buffer
};
[[nodiscard]] auto applyEcho(const Ja::Sound& src, const Ja::EchoParams& params) -> std::optional<ProcessedSound>;
[[nodiscard]] auto applyReverb(const Ja::Sound& src, const Ja::ReverbParams& params) -> std::optional<ProcessedSound>;
} // namespace AudioEffects
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#include "core/audio/jail_audio.hpp"
#include <algorithm>
#include <cassert>
#include <cstdint>
#include <cstdio>
#include <memory>
#include <optional>
#include <vector>
#include "core/audio/audio_effects.hpp"
// Solo declaracions de stb_vorbis: STB_VORBIS_HEADER_ONLY omet el bloc
// d'implementació. Les definicions las aporta source/external/stb_vorbis_impl.cpp
// (TU aïllat porque clang-analyzer no dispari fals positius al nostre codi).
#define STB_VORBIS_HEADER_ONLY
// clang-format off
// NOLINTNEXTLINE(bugprone-suspicious-include) -- stb_vorbis es single-file: la macro de dalt limita este TU a solo-declaracions; la implementació viu a external/stb_vorbis_impl.cpp.
#include "external/stb_vorbis.c"
// clang-format on
namespace Ja {
// --- Streaming internals (file-scope constants) ---
namespace {
// Bytes-per-sample per canal (siempre s16)
constexpr int MUSIC_BYTES_PER_SAMPLE = 2;
// Quants shorts decodifiquem per crida a get_samples_short_interleaved.
// 8192 shorts = 4096 samples/channel en estèreo ≈ 85ms de so a 48kHz.
constexpr int MUSIC_CHUNK_SHORTS = 8192;
// Umbral d'audio per davant del cursor de reproducció. Mantenim ≥ 0.5 s a
// l'SDL_AudioStream per absorbir jitter de frame i evitar underruns.
constexpr float MUSIC_LOW_WATER_SECONDS = 0.5F;
} // namespace
// --- Engine::active_ storage ---
Engine* Engine::active_ = nullptr;
auto Engine::active() noexcept -> Engine* { return active_; }
// --- Ctor/Dtor ---
Engine::Engine(const int freq, const SDL_AudioFormat format, const int num_channels) {
assert(active_ == nullptr && "Ja::Engine: més d'una instància activa no está suportat");
active_ = this;
audio_spec_ = {.format = format, .channels = num_channels, .freq = freq};
sdl_audio_device_ = SDL_OpenAudioDevice(SDL_AUDIO_DEVICE_DEFAULT_PLAYBACK, &audio_spec_);
if (sdl_audio_device_ == 0) { std::fprintf(stderr, "Ja::Engine: Failed to initialize SDL audio!\n"); }
for (auto& channel : channels_) { channel.state = ChannelState::FREE; }
}
Engine::~Engine() {
if (outgoing_music_.stream != nullptr) {
SDL_DestroyAudioStream(outgoing_music_.stream);
outgoing_music_.stream = nullptr;
}
if (sdl_audio_device_ != 0) { SDL_CloseAudioDevice(sdl_audio_device_); }
sdl_audio_device_ = 0;
if (active_ == this) { active_ = nullptr; }
}
// --- Helpers stateless (no toquen membres d'Engine) ---
namespace {
auto feedMusicChunk(Music* music) -> int {
if (music == nullptr || music->vorbis == nullptr || music->stream == nullptr) { return 0; }
short chunk[MUSIC_CHUNK_SHORTS];
const int NUM_CHANNELS = music->spec.channels;
const int SAMPLES_PER_CHANNEL = stb_vorbis_get_samples_short_interleaved(
music->vorbis,
NUM_CHANNELS,
static_cast<short*>(chunk),
MUSIC_CHUNK_SHORTS);
if (SAMPLES_PER_CHANNEL <= 0) { return 0; }
const int BYTES = SAMPLES_PER_CHANNEL * NUM_CHANNELS * MUSIC_BYTES_PER_SAMPLE;
SDL_PutAudioStreamData(music->stream, static_cast<const void*>(chunk), BYTES);
return SAMPLES_PER_CHANNEL;
}
void pumpMusic(Music* music) {
if (music == nullptr || music->vorbis == nullptr || music->stream == nullptr) { return; }
const int BYTES_PER_SECOND = music->spec.freq * music->spec.channels * MUSIC_BYTES_PER_SAMPLE;
const int LOW_WATER_BYTES = static_cast<int>(MUSIC_LOW_WATER_SECONDS * static_cast<float>(BYTES_PER_SECOND));
while (SDL_GetAudioStreamAvailable(music->stream) < LOW_WATER_BYTES) {
const int DECODED = feedMusicChunk(music);
if (DECODED > 0) { continue; }
// EOF: si queden loops, rebobinar; si no, tallar y deixar drenar.
if (music->times != 0) {
stb_vorbis_seek_start(music->vorbis);
if (music->times > 0) { music->times--; }
} else {
break;
}
}
}
void preFillOutgoing(Music* music, const int duration_ms) {
if (music == nullptr || music->vorbis == nullptr || music->stream == nullptr) { return; }
const int BYTES_PER_SECOND = music->spec.freq * music->spec.channels * MUSIC_BYTES_PER_SAMPLE;
const int NEEDED_BYTES = static_cast<int>((static_cast<std::int64_t>(duration_ms) * BYTES_PER_SECOND) / 1000);
while (SDL_GetAudioStreamAvailable(music->stream) < NEEDED_BYTES) {
const int DECODED = feedMusicChunk(music);
if (DECODED <= 0) { break; }
}
}
// Retorna el progrés lineal [0..1] d'un fade. 1.0 vol dir completat. Única
// font de la corba del fade: si es vol canviar a logarítmica/quadràtica,
// s'edita aquí i afecta fade-in i fade-out alhora.
auto fadeProgress(const FadeState& fade) -> float {
if (fade.duration_ms <= 0) { return 1.0F; }
const Uint64 ELAPSED = SDL_GetTicks() - fade.start_time;
if (ELAPSED >= static_cast<Uint64>(fade.duration_ms)) { return 1.0F; }
return static_cast<float>(ELAPSED) / static_cast<float>(fade.duration_ms);
}
} // namespace
void Engine::updateOutgoingFade() {
if (outgoing_music_.stream == nullptr || !outgoing_music_.fade.active) { return; }
// Mentre la fosa está activa, mantenim el stream con una reserva
// de samples per davant del cursor (mismo patró que pumpMusic
// para el current_music_). Así el stream no es buida ni cuando SDL
// drena més ràpid del previst en haver sounds bound a la misma
// device. Si l'OGG arriba a EOF, rebobina (la fosa pot ser més
// llarga que la pista).
if (outgoing_music_.music != nullptr && outgoing_music_.music->vorbis != nullptr) {
const Music& music = *outgoing_music_.music;
const int BYTES_PER_SECOND = music.spec.freq * music.spec.channels * MUSIC_BYTES_PER_SAMPLE;
const int LOW_WATER = static_cast<int>(MUSIC_LOW_WATER_SECONDS * static_cast<float>(BYTES_PER_SECOND));
while (SDL_GetAudioStreamAvailable(outgoing_music_.stream) < LOW_WATER) {
short chunk[MUSIC_CHUNK_SHORTS];
const int SAMPLES = stb_vorbis_get_samples_short_interleaved(
music.vorbis,
music.spec.channels,
static_cast<short*>(chunk),
MUSIC_CHUNK_SHORTS);
if (SAMPLES <= 0) {
stb_vorbis_seek_start(music.vorbis);
continue;
}
const int BYTES = SAMPLES * music.spec.channels * MUSIC_BYTES_PER_SAMPLE;
SDL_PutAudioStreamData(outgoing_music_.stream, static_cast<const void*>(chunk), BYTES);
}
}
const float PROGRESS = fadeProgress(outgoing_music_.fade);
if (PROGRESS >= 1.0F) {
SDL_DestroyAudioStream(outgoing_music_.stream);
outgoing_music_.stream = nullptr;
outgoing_music_.fade.active = false;
// Deixem el Vorbis del Music original en un estat conegut per
// a la pròxima reproducció. (playMusic también fa seek_start,
// pero fer-ho ací evita estats intermedis si algú consulta.)
if (outgoing_music_.music != nullptr && outgoing_music_.music->vorbis != nullptr) {
stb_vorbis_seek_start(outgoing_music_.music->vorbis);
}
outgoing_music_.music = nullptr;
} else {
SDL_SetAudioStreamGain(outgoing_music_.stream, outgoing_music_.fade.initial_volume * (1.0F - PROGRESS));
}
}
void Engine::updateIncomingFade() {
if (!incoming_fade_.active) { return; }
const float PROGRESS = fadeProgress(incoming_fade_);
if (PROGRESS >= 1.0F) {
incoming_fade_.active = false;
SDL_SetAudioStreamGain(current_music_->stream, music_volume_);
} else {
SDL_SetAudioStreamGain(current_music_->stream, music_volume_ * PROGRESS);
}
}
void Engine::updateCurrentMusic() {
if (current_music_ == nullptr || current_music_->state != MusicState::PLAYING) { return; }
updateIncomingFade();
pumpMusic(current_music_);
if (current_music_->times == 0 && SDL_GetAudioStreamAvailable(current_music_->stream) == 0) {
// La pista ha acabat de drenar naturalment. L'aturem primer (deixa
// l'engine en estat consistent) i entonces invoquem el callback;
// así un eventual playMusic des del callback comença net.
stopMusic();
if (on_music_ended_) { on_music_ended_(); }
}
}
void Engine::updateSoundChannels() {
for (int i = 0; i < MAX_SIMULTANEOUS_CHANNELS; ++i) {
if (channels_[i].state != ChannelState::PLAYING) { continue; }
if (channels_[i].times != 0) {
if (static_cast<Uint32>(SDL_GetAudioStreamAvailable(channels_[i].stream)) < (channels_[i].sound->length / 2)) {
SDL_PutAudioStreamData(channels_[i].stream, channels_[i].sound->buffer.get(), channels_[i].sound->length);
if (channels_[i].times > 0) { channels_[i].times--; }
}
} else if (SDL_GetAudioStreamAvailable(channels_[i].stream) == 0) {
stopChannel(i);
}
}
}
void Engine::stealCurrentIntoOutgoing(const int duration_ms) {
if (outgoing_music_.stream != nullptr) {
SDL_DestroyAudioStream(outgoing_music_.stream);
outgoing_music_.stream = nullptr;
outgoing_music_.fade.active = false;
}
if (current_music_ == nullptr || current_music_->state != MusicState::PLAYING || current_music_->stream == nullptr) {
return;
}
preFillOutgoing(current_music_, duration_ms);
outgoing_music_.stream = current_music_->stream;
// Guardem la referència al Music porque updateOutgoingFade puga
// seguir bombant Vorbis sin al stream durante tota la fosa. NO fem
// seek_start ací: la decompressió ha de continuar des d'on estava
// porque el so siga continu. El seek_start es farà cuando la fosa
// acabe (o cuando playMusic la interrompi via stopMusic).
outgoing_music_.music = current_music_;
outgoing_music_.fade = {
.active = true,
.start_time = SDL_GetTicks(),
.duration_ms = duration_ms,
.initial_volume = music_volume_,
};
current_music_->stream = nullptr;
current_music_->state = MusicState::STOPPED;
}
template <typename Fn>
void Engine::forEachTargetChannel(const int channel, Fn&& fn) {
if (channel == -1) {
for (auto& ch : channels_) { fn(ch); }
} else if (channel >= 0 && channel < MAX_SIMULTANEOUS_CHANNELS) {
fn(channels_[channel]);
}
}
// --- Engine public API ---
void Engine::update() {
updateOutgoingFade();
updateCurrentMusic();
updateSoundChannels();
}
void Engine::playMusic(Music* music, const int loop) {
if (music == nullptr || music->vorbis == nullptr) { return; }
stopMusic();
current_music_ = music;
current_music_->state = MusicState::PLAYING;
current_music_->times = loop;
stb_vorbis_seek_start(current_music_->vorbis);
current_music_->stream = SDL_CreateAudioStream(&current_music_->spec, &audio_spec_);
if (current_music_->stream == nullptr) {
std::fprintf(stderr, "Ja::Engine::playMusic: Failed to create audio stream!\n");
current_music_->state = MusicState::STOPPED;
return;
}
SDL_SetAudioStreamGain(current_music_->stream, music_volume_);
pumpMusic(current_music_);
if (!SDL_BindAudioStream(sdl_audio_device_, current_music_->stream)) {
std::fprintf(stderr, "Ja::Engine::playMusic: SDL_BindAudioStream failed!\n");
}
}
void Engine::setMusicSpeed(float ratio) {
if (current_music_ == nullptr || current_music_->stream == nullptr) { return; }
SDL_SetAudioStreamFrequencyRatio(current_music_->stream, ratio);
}
void Engine::pauseMusic() {
if (current_music_ == nullptr || current_music_->state != MusicState::PLAYING) { return; }
current_music_->state = MusicState::PAUSED;
SDL_UnbindAudioStream(current_music_->stream);
}
void Engine::resumeMusic() {
if (current_music_ == nullptr || current_music_->state != MusicState::PAUSED) { return; }
current_music_->state = MusicState::PLAYING;
SDL_BindAudioStream(sdl_audio_device_, current_music_->stream);
}
void Engine::stopMusic() {
if (outgoing_music_.stream != nullptr) {
SDL_DestroyAudioStream(outgoing_music_.stream);
outgoing_music_.stream = nullptr;
outgoing_music_.fade.active = false;
if (outgoing_music_.music != nullptr && outgoing_music_.music->vorbis != nullptr) {
stb_vorbis_seek_start(outgoing_music_.music->vorbis);
}
outgoing_music_.music = nullptr;
}
incoming_fade_.active = false;
if (current_music_ == nullptr || current_music_->state == MusicState::INVALID || current_music_->state == MusicState::STOPPED) { return; }
current_music_->state = MusicState::STOPPED;
if (current_music_->stream != nullptr) {
SDL_DestroyAudioStream(current_music_->stream);
current_music_->stream = nullptr;
}
if (current_music_->vorbis != nullptr) {
stb_vorbis_seek_start(current_music_->vorbis);
}
}
void Engine::fadeOutMusic(const int milliseconds) {
if (current_music_ == nullptr || current_music_->state != MusicState::PLAYING) { return; }
stealCurrentIntoOutgoing(milliseconds);
incoming_fade_.active = false;
}
void Engine::crossfadeMusic(Music* music, const int crossfade_ms, const int loop) {
if (music == nullptr || music->vorbis == nullptr) { return; }
stealCurrentIntoOutgoing(crossfade_ms);
current_music_ = music;
current_music_->state = MusicState::PLAYING;
current_music_->times = loop;
stb_vorbis_seek_start(current_music_->vorbis);
current_music_->stream = SDL_CreateAudioStream(&current_music_->spec, &audio_spec_);
if (current_music_->stream == nullptr) {
std::fprintf(stderr, "Ja::Engine::crossfadeMusic: Failed to create audio stream!\n");
current_music_->state = MusicState::STOPPED;
return;
}
SDL_SetAudioStreamGain(current_music_->stream, 0.0F);
pumpMusic(current_music_);
SDL_BindAudioStream(sdl_audio_device_, current_music_->stream);
incoming_fade_ = {
.active = true,
.start_time = SDL_GetTicks(),
.duration_ms = crossfade_ms,
.initial_volume = 0.0F,
};
}
auto Engine::getMusicState() const -> MusicState {
if (current_music_ == nullptr) { return MusicState::INVALID; }
return current_music_->state;
}
auto Engine::setMusicVolume(float volume) -> float {
music_volume_ = SDL_clamp(volume, 0.0F, 1.0F);
if (current_music_ != nullptr && current_music_->stream != nullptr) {
SDL_SetAudioStreamGain(current_music_->stream, music_volume_);
}
return music_volume_;
}
void Engine::setOnMusicEnded(std::function<void()> callback) {
on_music_ended_ = std::move(callback);
}
void Engine::onMusicDeleted(const Music* music) {
if (music == nullptr) { return; }
if (current_music_ == music) {
stopMusic();
current_music_ = nullptr;
}
}
// --- Sound ---
auto Engine::playSound(Sound* sound, const int loop, const int group) -> int {
if (sound == nullptr) { return -1; }
int channel = 0;
while (channel < MAX_SIMULTANEOUS_CHANNELS && channels_[channel].state != ChannelState::FREE) { channel++; }
if (channel == MAX_SIMULTANEOUS_CHANNELS) {
// No hay canal libre, reemplazamos el primero
channel = 0;
}
return playSoundOnChannel(sound, channel, loop, group);
}
auto Engine::playSoundOnChannel(Sound* sound, const int channel, const int loop, const int group) -> int {
if (sound == nullptr) { return -1; }
if (channel < 0 || channel >= MAX_SIMULTANEOUS_CHANNELS) { return -1; }
stopChannel(channel);
channels_[channel].sound = sound;
channels_[channel].times = loop;
channels_[channel].pos = 0;
channels_[channel].group = group;
channels_[channel].state = ChannelState::PLAYING;
channels_[channel].stream = SDL_CreateAudioStream(&channels_[channel].sound->spec, &audio_spec_);
if (channels_[channel].stream == nullptr) {
std::fprintf(stderr, "Ja::Engine::playSoundOnChannel: Failed to create audio stream!\n");
channels_[channel].state = ChannelState::FREE;
return -1;
}
SDL_PutAudioStreamData(channels_[channel].stream, channels_[channel].sound->buffer.get(), channels_[channel].sound->length);
SDL_SetAudioStreamGain(channels_[channel].stream, sound_volume_[group]);
SDL_BindAudioStream(sdl_audio_device_, channels_[channel].stream);
return channel;
}
void Engine::setChannelSpeed(const int channel, const float ratio) {
if (channel < 0 || channel >= MAX_SIMULTANEOUS_CHANNELS) { return; }
if (channels_[channel].stream == nullptr) { return; }
SDL_SetAudioStreamFrequencyRatio(channels_[channel].stream, ratio);
}
void Engine::pauseChannel(const int channel) {
forEachTargetChannel(channel, [](Channel& ch) {
if (ch.state == ChannelState::PLAYING) {
ch.state = ChannelState::PAUSED;
SDL_UnbindAudioStream(ch.stream);
}
});
}
void Engine::resumeChannel(const int channel) {
const SDL_AudioDeviceID DEVICE = sdl_audio_device_;
forEachTargetChannel(channel, [DEVICE](Channel& ch) {
if (ch.state == ChannelState::PAUSED) {
ch.state = ChannelState::PLAYING;
SDL_BindAudioStream(DEVICE, ch.stream);
}
});
}
void Engine::stopChannel(const int channel) {
forEachTargetChannel(channel, [this](Channel& ch) {
if (ch.state != ChannelState::FREE) {
if (ch.stream != nullptr) { SDL_DestroyAudioStream(ch.stream); }
ch.stream = nullptr;
ch.state = ChannelState::FREE;
ch.pos = 0;
ch.sound = nullptr;
if (ch.has_effect) {
ch.has_effect = false;
if (effect_channels_active_ > 0) { --effect_channels_active_; }
}
}
});
}
auto Engine::setSoundVolume(float volume, const int group) -> float {
const float V = SDL_clamp(volume, 0.0F, 1.0F);
if (group == -1) {
std::ranges::fill(sound_volume_, V);
} else if (group >= 0 && group < MAX_GROUPS) {
sound_volume_[group] = V;
} else {
return V;
}
for (auto& ch : channels_) {
if ((ch.state == ChannelState::PLAYING) || (ch.state == ChannelState::PAUSED)) {
if (group == -1 || ch.group == group) {
if (ch.stream != nullptr) {
SDL_SetAudioStreamGain(ch.stream, sound_volume_[ch.group]);
}
}
}
}
return V;
}
void Engine::onSoundDeleted(const Sound* sound) {
if (sound == nullptr) { return; }
for (int i = 0; i < MAX_SIMULTANEOUS_CHANNELS; ++i) {
if (channels_[i].sound == sound) { stopChannel(i); }
}
}
auto Engine::playProcessedOnFreeChannel(const std::vector<std::uint8_t>& bytes, const SDL_AudioSpec& spec, const int group) -> int {
// El sin de canals con efecte es valida antes de reservar slot —
// así evitem crear y destruir un stream solo per descartar el play.
if (effect_channels_active_ >= MAX_EFFECT_CHANNELS) { return -1; }
int channel = 0;
while (channel < MAX_SIMULTANEOUS_CHANNELS && channels_[channel].state != ChannelState::FREE) { ++channel; }
if (channel == MAX_SIMULTANEOUS_CHANNELS) { channel = 0; }
stopChannel(channel);
// El stream es crea contra l'spec del buffer processat (S16, ...)
// porque SDL faci el resampling sin a audio_spec_ del device.
channels_[channel].stream = SDL_CreateAudioStream(&spec, &audio_spec_);
if (channels_[channel].stream == nullptr) {
std::fprintf(stderr, "Ja::Engine::playProcessedOnFreeChannel: Failed to create audio stream!\n");
return -1;
}
channels_[channel].sound = nullptr; // El buffer no es propietat de sin Ja::Sound.
channels_[channel].times = 0;
channels_[channel].pos = 0;
const int CLAMPED_GROUP = (group >= 0 && group < MAX_GROUPS) ? group : 0;
channels_[channel].group = CLAMPED_GROUP;
channels_[channel].state = ChannelState::PLAYING;
channels_[channel].has_effect = true;
++effect_channels_active_;
SDL_PutAudioStreamData(channels_[channel].stream, bytes.data(), static_cast<int>(bytes.size()));
SDL_SetAudioStreamGain(channels_[channel].stream, sound_volume_[CLAMPED_GROUP]);
SDL_BindAudioStream(sdl_audio_device_, channels_[channel].stream);
return channel;
}
auto Engine::playSoundWithEcho(const Sound* sound, const EchoParams& params, const int group) -> int {
if (sound == nullptr) { return -1; }
auto processed = AudioEffects::applyEcho(*sound, params);
if (!processed) { return -1; }
return playProcessedOnFreeChannel(processed->bytes, processed->spec, group);
}
auto Engine::playSoundWithReverb(const Sound* sound, const ReverbParams& params, const int group) -> int {
if (sound == nullptr) { return -1; }
auto processed = AudioEffects::applyReverb(*sound, params);
if (!processed) { return -1; }
return playProcessedOnFreeChannel(processed->bytes, processed->spec, group);
}
// --- Factories y destructors (permanents) ---
auto loadMusic(const Uint8* buffer, Uint32 length) -> Music* {
if (buffer == nullptr || length == 0) { return nullptr; }
// Allocem el Music primer per aprofitar el seu `std::vector<Uint8>`
// como a propietari del OGG comprimit. stb_vorbis guarda un punter
// persistent al buffer; como que ací no el resize'jem, el .data() es
// estable durante tot el cicle de vida del music.
auto music = std::make_unique<Music>();
music->ogg_data.assign(buffer, buffer + length);
int vorbis_error = 0;
music->vorbis = stb_vorbis_open_memory(music->ogg_data.data(),
static_cast<int>(length),
&vorbis_error,
nullptr);
if (music->vorbis == nullptr) {
std::fprintf(stderr, "Ja::loadMusic: stb_vorbis_open_memory failed (error %d)\n", vorbis_error);
return nullptr;
}
const stb_vorbis_info INFO = stb_vorbis_get_info(music->vorbis);
music->spec.channels = static_cast<int>(INFO.channels);
music->spec.freq = static_cast<int>(INFO.sample_rate);
music->spec.format = SDL_AUDIO_S16;
// Pre-cálculo de la duración en ms a partir del header. stb_vorbis ya
// ha decodificat la informació necessària a `stb_vorbis_open_memory`;
// esta consulta no descodifica àudio, solo llig el comptador
// de samples. Si el sample_rate fos 0 (header malmès) deixem
// duration_ms a 0.
if (INFO.sample_rate > 0) {
const auto SAMPLES = stb_vorbis_stream_length_in_samples(music->vorbis);
music->duration_ms = static_cast<int>((static_cast<std::uint64_t>(SAMPLES) * 1000ULL) / INFO.sample_rate);
}
music->state = MusicState::STOPPED;
return music.release();
}
// Overload con filename. Resource::Cache l'usa per registrar el path dins
// del propi Ja::Music (camp `filename`); la capa Audio l'usa per recuperar
// el nom después d'un playMusic(Ja::Music*, ...) — veure PATCH-02.
auto loadMusic(const Uint8* buffer, Uint32 length, const char* filename) -> Music* {
Music* music = loadMusic(buffer, length);
if (music != nullptr && filename != nullptr) { music->filename = filename; }
return music;
}
void deleteMusic(Music* music) {
if (music == nullptr) { return; }
// Notifiquem el motor actiu porque pari la pista si es la current_music.
// Si no hay motor (shutdown-order invertit), passem: los recursos
// propis del Music es lliberen igualment a sota.
if (Engine* eng = Engine::active()) { eng->onMusicDeleted(music); }
if (music->stream != nullptr) { SDL_DestroyAudioStream(music->stream); }
if (music->vorbis != nullptr) { stb_vorbis_close(music->vorbis); }
delete music;
}
auto loadSound(std::uint8_t* buffer, std::uint32_t size) -> Sound* {
auto sound = std::make_unique<Sound>();
Uint8* raw = nullptr;
if (!SDL_LoadWAV_IO(SDL_IOFromMem(buffer, size), true, &sound->spec, &raw, &sound->length)) {
std::fprintf(stderr, "Ja::loadSound: Failed to load WAV from memory: %s\n", SDL_GetError());
return nullptr;
}
sound->buffer.reset(raw); // adopta el SDL_malloc'd buffer
return sound.release();
}
void deleteSound(Sound* sound) {
if (sound == nullptr) { return; }
if (Engine* eng = Engine::active()) { eng->onSoundDeleted(sound); }
// buffer es destrueix automàticament via RAII (SdlFreeDeleter).
delete sound;
}
} // namespace Ja
// --- stb_vorbis macro leak cleanup ---
// stb_vorbis.c filtra noms curts (L, C, R i PLAYBACK_*) al TU que el compila.
// Xocarien con parámetros de plantilla d'altres headers si estas definicions
// s'escapessin. Els netegem al final del TU per tancar la porta.
// clang-format off
#undef L
#undef C
#undef R
#undef PLAYBACK_MONO
#undef PLAYBACK_LEFT
#undef PLAYBACK_RIGHT
// clang-format on
+242 -466
View File
@@ -2,481 +2,257 @@
// --- Includes ---
#include <SDL3/SDL.h>
#include <stdint.h> // Para uint32_t, uint8_t
#include <stdio.h> // Para NULL, fseek, printf, fclose, fopen, fread, ftell, FILE, SEEK_END, SEEK_SET
#include <stdlib.h> // Para free, malloc
#include <string.h> // Para strcpy, strlen
#define STB_VORBIS_HEADER_ONLY
#include "external/stb_vorbis.h" // Para stb_vorbis_decode_memory
#include <cstdint>
#include <functional>
#include <memory>
#include <string>
#include <vector>
// --- Public Enums ---
enum JA_Channel_state { JA_CHANNEL_INVALID,
JA_CHANNEL_FREE,
JA_CHANNEL_PLAYING,
JA_CHANNEL_PAUSED,
JA_SOUND_DISABLED };
enum JA_Music_state { JA_MUSIC_INVALID,
JA_MUSIC_PLAYING,
JA_MUSIC_PAUSED,
JA_MUSIC_STOPPED,
JA_MUSIC_DISABLED };
// Forward-declaració del decoder de vorbis. La implementació viu a
// jail_audio.cpp (únic TU que compila external/stb_vorbis.c). Qualsevol caller
// solo necessita `stb_vorbis*` per punter — nunca per valor — así que el
// forward decl n'hay prou i evita arrossegar el .c a tots los TU.
// NOLINTNEXTLINE(readability-identifier-naming) — nom imposat per l'API de stb_vorbis
struct stb_vorbis;
// --- Struct Definitions ---
#define JA_MAX_SIMULTANEOUS_CHANNELS 20
#define JA_MAX_GROUPS 2
struct JA_Sound_t {
SDL_AudioSpec spec{SDL_AUDIO_S16, 2, 48000};
Uint32 length{0};
Uint8* buffer{NULL};
// Deleter stateless para buffers reservats con `SDL_malloc` / `SDL_LoadWAV*`.
// Compatible con `std::unique_ptr<Uint8[], SdlFreeDeleter>` — zero size overhead
// gràcies a EBO, igual que un unique_ptr con default_delete.
struct SdlFreeDeleter {
void operator()(Uint8* p) const noexcept {
if (p != nullptr) { SDL_free(p); }
}
};
struct JA_Channel_t {
JA_Sound_t* sound{nullptr};
// Motor de baix nivell d'àudio del projecte jailgames: streaming OGG
// (stb_vorbis) + N canals d'efectes (SDL3 audio). No depèn d'Options ni de sin
// singleton del joc; solo de SDL3 i stb_vorbis. La capa superior (Audio) li
// passa recursos pel punter i fa el bookkeeping d'usuari.
namespace Ja {
// --- Public Enums ---
enum class ChannelState : std::uint8_t {
FREE,
PLAYING,
PAUSED,
};
enum class MusicState : std::uint8_t {
INVALID, // Music carregat pero nunca play-ejat
PLAYING,
PAUSED,
STOPPED,
};
// --- Constants ---
inline constexpr int MAX_SIMULTANEOUS_CHANNELS = 20;
inline constexpr int MAX_GROUPS = 2;
// Cap superior de canals que poden estar simultàniament reproduint un so
// con efecte (eco/reverb). Si está al límit, las noves crides con efecte
// cauen al camí sec — l'usuari sent el so igualment, sin la cua.
inline constexpr int MAX_EFFECT_CHANNELS = 4;
// --- Paràmetres d'efectes ---
// Els camps los fixa el caller (Audio) llegint sounds.yaml; el motor solo
// los passa a AudioEffects::applyEcho/applyReverb. Els defaults són
// sensats pero los presets los sobreescriuen.
struct EchoParams {
float delay_ms{220.0F}; // Temps hasta al primer rebot.
float feedback{0.45F}; // Reinjecció (0..0.95).
float wet{0.35F}; // Mescla humida (0..1).
};
struct ReverbParams {
float room_size{0.7F}; // Tamaño percebuda (0..1).
float damping{0.5F}; // Atenuació d'aguts per rebot (0..1).
float wet{0.4F}; // Mescla humida (0..1).
};
// Spec de fallback del dispositiu. S'aplica antes que l'Engine s'iniciï i
// como a valor inicial de Sound/Music. L'spec real d'ús l'imposa el ctor
// d'Engine, alimentat des de Defaults::Audio via Audio.
inline constexpr SDL_AudioSpec DEFAULT_SPEC{SDL_AUDIO_S16, 2, 48000};
// --- Struct Definitions ---
struct Sound {
SDL_AudioSpec spec{DEFAULT_SPEC};
Uint32 length{0};
// Buffer descomprimit (PCM) propietat del sound. Reservat per SDL_LoadWAV
// via SDL_malloc; el deleter `SdlFreeDeleter` allibera con SDL_free.
std::unique_ptr<Uint8[], SdlFreeDeleter> buffer;
};
// L'ordre (punters primer, ints después, enum de 8 bits al final) minimitza
// el padding a 64-bit (evita avisos de clang-analyzer-optin.performance.Padding).
struct Channel {
Sound* sound{nullptr};
SDL_AudioStream* stream{nullptr};
int pos{0};
int times{0};
int group{0};
ChannelState state{ChannelState::FREE};
// Marca si este canal va arrencar con so processat per un efecte.
// El motor compta canals actius con efecte per fer complir
// MAX_EFFECT_CHANNELS i alliberar el comptador en parar.
bool has_effect{false};
};
struct Music {
SDL_AudioSpec spec{DEFAULT_SPEC};
// OGG comprimit en memòria. Propietat nostra; es copia des del buffer
// d'entrada una sola vegada en loadMusic i es descomprimix en chunks
// per streaming. Como que stb_vorbis guarda un punter persistent al
// `.data()` d'este vector, no el podem resize'jar un cop establert
// (una reallocation invalidaria el punter que el decoder conserva).
std::vector<Uint8> ogg_data;
stb_vorbis* vorbis{nullptr}; // handle del decoder, viu tot el cicle del Music
std::string filename;
int times{0}; // loops restants (-1 = infinit, 0 = un sol play)
// Duración total de la pista en mil·lisegons, mesurada via
// `stb_vorbis_stream_length_in_samples / sample_rate` al
// `loadMusic`. 0 si el cálculo no es possible (header malmès).
// L'usen consumidors que necessiten un timeline pre-calculat —
// p. ex. la FSM de sala — sin dependre de callbacks de fi.
int duration_ms{0};
SDL_AudioStream* stream{nullptr};
JA_Channel_state state{JA_CHANNEL_FREE};
};
MusicState state{MusicState::INVALID};
};
struct JA_Music_t {
SDL_AudioSpec spec{SDL_AUDIO_S16, 2, 48000};
Uint32 length{0};
Uint8* buffer{nullptr};
char* filename{nullptr};
struct FadeState {
bool active{false};
Uint64 start_time{0};
int duration_ms{0};
float initial_volume{0.0F};
};
int pos{0};
int times{0};
struct OutgoingMusic {
SDL_AudioStream* stream{nullptr};
JA_Music_state state{JA_MUSIC_INVALID};
};
// --- Internal Global State ---
// Marcado 'inline' (C++17) para asegurar una única instancia.
inline JA_Music_t* current_music{nullptr};
inline JA_Channel_t channels[JA_MAX_SIMULTANEOUS_CHANNELS];
inline SDL_AudioSpec JA_audioSpec{SDL_AUDIO_S16, 2, 48000};
inline float JA_musicVolume{1.0F};
inline float JA_soundVolume[JA_MAX_GROUPS];
inline bool JA_musicEnabled{true};
inline bool JA_soundEnabled{true};
inline SDL_AudioDeviceID sdlAudioDevice{0};
inline bool fading{false};
inline int fade_start_time{0};
inline int fade_duration{0};
inline float fade_initial_volume{0.0F}; // Corregido de 'int' a 'float'
// --- Forward Declarations ---
inline void JA_StopMusic();
inline void JA_StopChannel(const int channel);
inline int JA_PlaySoundOnChannel(JA_Sound_t* sound, const int channel, const int loop = 0, const int group = 0);
// --- Core Functions ---
inline void JA_Update() {
if (JA_musicEnabled && current_music && current_music->state == JA_MUSIC_PLAYING) {
if (fading) {
int time = SDL_GetTicks();
if (time > (fade_start_time + fade_duration)) {
fading = false;
JA_StopMusic();
return;
} else {
const int time_passed = time - fade_start_time;
const float percent = (float)time_passed / (float)fade_duration;
SDL_SetAudioStreamGain(current_music->stream, JA_musicVolume * (1.0 - percent));
}
}
if (current_music->times != 0) {
if ((Uint32)SDL_GetAudioStreamAvailable(current_music->stream) < (current_music->length / 2)) {
SDL_PutAudioStreamData(current_music->stream, current_music->buffer, current_music->length);
}
if (current_music->times > 0) current_music->times--;
} else {
if (SDL_GetAudioStreamAvailable(current_music->stream) == 0) JA_StopMusic();
}
}
if (JA_soundEnabled) {
for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; ++i)
if (channels[i].state == JA_CHANNEL_PLAYING) {
if (channels[i].times != 0) {
if ((Uint32)SDL_GetAudioStreamAvailable(channels[i].stream) < (channels[i].sound->length / 2)) {
SDL_PutAudioStreamData(channels[i].stream, channels[i].sound->buffer, channels[i].sound->length);
if (channels[i].times > 0) channels[i].times--;
}
} else {
if (SDL_GetAudioStreamAvailable(channels[i].stream) == 0) JA_StopChannel(i);
}
}
}
}
inline void JA_Init(const int freq, const SDL_AudioFormat format, const int num_channels) {
#ifdef _DEBUG
SDL_SetLogPriority(SDL_LOG_CATEGORY_APPLICATION, SDL_LOG_PRIORITY_DEBUG);
#endif
JA_audioSpec = {format, num_channels, freq};
if (sdlAudioDevice) SDL_CloseAudioDevice(sdlAudioDevice); // Corregido: !sdlAudioDevice -> sdlAudioDevice
sdlAudioDevice = SDL_OpenAudioDevice(SDL_AUDIO_DEVICE_DEFAULT_PLAYBACK, &JA_audioSpec);
if (sdlAudioDevice == 0) SDL_Log("Failed to initialize SDL audio!");
for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; ++i) channels[i].state = JA_CHANNEL_FREE;
for (int i = 0; i < JA_MAX_GROUPS; ++i) JA_soundVolume[i] = 0.5F;
}
inline void JA_Quit() {
if (sdlAudioDevice) SDL_CloseAudioDevice(sdlAudioDevice); // Corregido: !sdlAudioDevice -> sdlAudioDevice
sdlAudioDevice = 0;
}
// --- Music Functions ---
inline JA_Music_t* JA_LoadMusic(const Uint8* buffer, Uint32 length) {
JA_Music_t* music = new JA_Music_t();
int chan, samplerate;
short* output;
music->length = stb_vorbis_decode_memory(buffer, length, &chan, &samplerate, &output) * chan * 2;
music->spec.channels = chan;
music->spec.freq = samplerate;
music->spec.format = SDL_AUDIO_S16;
music->buffer = static_cast<Uint8*>(SDL_malloc(music->length));
SDL_memcpy(music->buffer, output, music->length);
free(output);
music->pos = 0;
music->state = JA_MUSIC_STOPPED;
return music;
}
inline JA_Music_t* JA_LoadMusic(const char* filename) {
// [RZC 28/08/22] Carreguem primer el arxiu en memòria y después el descomprimim. Es algo més rapid.
FILE* f = fopen(filename, "rb");
if (!f) return NULL; // Añadida comprobación de apertura
fseek(f, 0, SEEK_END);
long fsize = ftell(f);
fseek(f, 0, SEEK_SET);
auto* buffer = static_cast<Uint8*>(malloc(fsize + 1));
if (!buffer) { // Añadida comprobación de malloc
fclose(f);
return NULL;
}
if (fread(buffer, fsize, 1, f) != 1) {
fclose(f);
free(buffer);
return NULL;
}
fclose(f);
JA_Music_t* music = JA_LoadMusic(buffer, fsize);
if (music) { // Comprobar que JA_LoadMusic tuvo éxito
music->filename = static_cast<char*>(malloc(strlen(filename) + 1));
if (music->filename) {
strcpy(music->filename, filename);
}
}
free(buffer);
return music;
}
inline void JA_PlayMusic(JA_Music_t* music, const int loop = -1) {
if (!JA_musicEnabled || !music) return; // Añadida comprobación de music
JA_StopMusic();
current_music = music;
current_music->pos = 0;
current_music->state = JA_MUSIC_PLAYING;
current_music->times = loop;
current_music->stream = SDL_CreateAudioStream(&current_music->spec, &JA_audioSpec);
if (!current_music->stream) { // Comprobar creación de stream
SDL_Log("Failed to create audio stream!");
current_music->state = JA_MUSIC_STOPPED;
return;
}
if (!SDL_PutAudioStreamData(current_music->stream, current_music->buffer, current_music->length)) printf("[ERROR] SDL_PutAudioStreamData failed!\n");
SDL_SetAudioStreamGain(current_music->stream, JA_musicVolume);
if (!SDL_BindAudioStream(sdlAudioDevice, current_music->stream)) printf("[ERROR] SDL_BindAudioStream failed!\n");
}
inline char* JA_GetMusicFilename(const JA_Music_t* music = nullptr) {
if (!music) music = current_music;
if (!music) return nullptr; // Añadida comprobación
return music->filename;
}
inline void JA_PauseMusic() {
if (!JA_musicEnabled) return;
if (!current_music || current_music->state != JA_MUSIC_PLAYING) return; // Comprobación mejorada
current_music->state = JA_MUSIC_PAUSED;
SDL_UnbindAudioStream(current_music->stream);
}
inline void JA_ResumeMusic() {
if (!JA_musicEnabled) return;
if (!current_music || current_music->state != JA_MUSIC_PAUSED) return; // Comprobación mejorada
current_music->state = JA_MUSIC_PLAYING;
SDL_BindAudioStream(sdlAudioDevice, current_music->stream);
}
inline void JA_StopMusic() {
if (!current_music || current_music->state == JA_MUSIC_INVALID || current_music->state == JA_MUSIC_STOPPED) return;
current_music->pos = 0;
current_music->state = JA_MUSIC_STOPPED;
if (current_music->stream) {
SDL_DestroyAudioStream(current_music->stream);
current_music->stream = nullptr;
}
// No liberamos filename aquí, se debería liberar en JA_DeleteMusic
}
inline void JA_FadeOutMusic(const int milliseconds) {
if (!JA_musicEnabled) return;
if (current_music == NULL || current_music->state == JA_MUSIC_INVALID) return;
fading = true;
fade_start_time = SDL_GetTicks();
fade_duration = milliseconds;
fade_initial_volume = JA_musicVolume;
}
inline JA_Music_state JA_GetMusicState() {
if (!JA_musicEnabled) return JA_MUSIC_DISABLED;
if (!current_music) return JA_MUSIC_INVALID;
return current_music->state;
}
inline void JA_DeleteMusic(JA_Music_t* music) {
if (!music) return;
if (current_music == music) {
JA_StopMusic();
current_music = nullptr;
}
SDL_free(music->buffer);
if (music->stream) SDL_DestroyAudioStream(music->stream);
free(music->filename); // filename se libera aquí
delete music;
}
inline float JA_SetMusicVolume(float volume) {
JA_musicVolume = SDL_clamp(volume, 0.0F, 1.0F);
if (current_music && current_music->stream) {
SDL_SetAudioStreamGain(current_music->stream, JA_musicVolume);
}
return JA_musicVolume;
}
inline void JA_SetMusicPosition(float value) {
if (!current_music) return;
current_music->pos = value * current_music->spec.freq;
// Nota: Esta implementación de 'pos' no parece usarse en JA_Update para
// el streaming. El streaming siempre parece empezar desde el principio.
}
inline float JA_GetMusicPosition() {
if (!current_music) return 0;
return float(current_music->pos) / float(current_music->spec.freq);
// Nota: Ver `JA_SetMusicPosition`
}
inline void JA_EnableMusic(const bool value) {
if (!value && current_music && (current_music->state == JA_MUSIC_PLAYING)) JA_StopMusic();
JA_musicEnabled = value;
}
// --- Sound Functions ---
inline JA_Sound_t* JA_NewSound(Uint8* buffer, Uint32 length) {
JA_Sound_t* sound = new JA_Sound_t();
sound->buffer = buffer;
sound->length = length;
// Nota: spec se queda con los valores por defecto.
return sound;
}
inline JA_Sound_t* JA_LoadSound(uint8_t* buffer, uint32_t size) {
JA_Sound_t* sound = new JA_Sound_t();
if (!SDL_LoadWAV_IO(SDL_IOFromMem(buffer, size), 1, &sound->spec, &sound->buffer, &sound->length)) {
SDL_Log("Failed to load WAV from memory: %s", SDL_GetError());
delete sound;
return nullptr;
}
return sound;
}
inline JA_Sound_t* JA_LoadSound(const char* filename) {
JA_Sound_t* sound = new JA_Sound_t();
if (!SDL_LoadWAV(filename, &sound->spec, &sound->buffer, &sound->length)) {
SDL_Log("Failed to load WAV file: %s", SDL_GetError());
delete sound;
return nullptr;
}
return sound;
}
inline int JA_PlaySound(JA_Sound_t* sound, const int loop = 0, const int group = 0) {
if (!JA_soundEnabled || !sound) return -1;
int channel = 0;
while (channel < JA_MAX_SIMULTANEOUS_CHANNELS && channels[channel].state != JA_CHANNEL_FREE) { channel++; }
if (channel == JA_MAX_SIMULTANEOUS_CHANNELS) {
// No hay canal libre, reemplazamos el primero
channel = 0;
}
return JA_PlaySoundOnChannel(sound, channel, loop, group);
}
inline int JA_PlaySoundOnChannel(JA_Sound_t* sound, const int channel, const int loop, const int group) {
if (!JA_soundEnabled || !sound) return -1;
if (channel < 0 || channel >= JA_MAX_SIMULTANEOUS_CHANNELS) return -1;
JA_StopChannel(channel); // Detiene y limpia el canal si estaba en uso
channels[channel].sound = sound;
channels[channel].times = loop;
channels[channel].pos = 0;
channels[channel].group = group; // Asignar grupo
channels[channel].state = JA_CHANNEL_PLAYING;
channels[channel].stream = SDL_CreateAudioStream(&channels[channel].sound->spec, &JA_audioSpec);
if (!channels[channel].stream) {
SDL_Log("Failed to create audio stream for sound!");
channels[channel].state = JA_CHANNEL_FREE;
return -1;
}
SDL_PutAudioStreamData(channels[channel].stream, channels[channel].sound->buffer, channels[channel].sound->length);
SDL_SetAudioStreamGain(channels[channel].stream, JA_soundVolume[group]);
SDL_BindAudioStream(sdlAudioDevice, channels[channel].stream);
return channel;
}
inline void JA_DeleteSound(JA_Sound_t* sound) {
if (!sound) return;
for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; i++) {
if (channels[i].sound == sound) JA_StopChannel(i);
}
SDL_free(sound->buffer);
delete sound;
}
inline void JA_PauseChannel(const int channel) {
if (!JA_soundEnabled) return;
if (channel == -1) {
for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; i++)
if (channels[i].state == JA_CHANNEL_PLAYING) {
channels[i].state = JA_CHANNEL_PAUSED;
SDL_UnbindAudioStream(channels[i].stream);
}
} else if (channel >= 0 && channel < JA_MAX_SIMULTANEOUS_CHANNELS) {
if (channels[channel].state == JA_CHANNEL_PLAYING) {
channels[channel].state = JA_CHANNEL_PAUSED;
SDL_UnbindAudioStream(channels[channel].stream);
}
}
}
inline void JA_ResumeChannel(const int channel) {
if (!JA_soundEnabled) return;
if (channel == -1) {
for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; i++)
if (channels[i].state == JA_CHANNEL_PAUSED) {
channels[i].state = JA_CHANNEL_PLAYING;
SDL_BindAudioStream(sdlAudioDevice, channels[i].stream);
}
} else if (channel >= 0 && channel < JA_MAX_SIMULTANEOUS_CHANNELS) {
if (channels[channel].state == JA_CHANNEL_PAUSED) {
channels[channel].state = JA_CHANNEL_PLAYING;
SDL_BindAudioStream(sdlAudioDevice, channels[channel].stream);
}
}
}
inline void JA_StopChannel(const int channel) {
if (channel == -1) {
for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; i++) {
if (channels[i].state != JA_CHANNEL_FREE) {
if (channels[i].stream) SDL_DestroyAudioStream(channels[i].stream);
channels[i].stream = nullptr;
channels[i].state = JA_CHANNEL_FREE;
channels[i].pos = 0;
channels[i].sound = NULL;
}
}
} else if (channel >= 0 && channel < JA_MAX_SIMULTANEOUS_CHANNELS) {
if (channels[channel].state != JA_CHANNEL_FREE) {
if (channels[channel].stream) SDL_DestroyAudioStream(channels[channel].stream);
channels[channel].stream = nullptr;
channels[channel].state = JA_CHANNEL_FREE;
channels[channel].pos = 0;
channels[channel].sound = NULL;
}
}
}
inline JA_Channel_state JA_GetChannelState(const int channel) {
if (!JA_soundEnabled) return JA_SOUND_DISABLED;
if (channel < 0 || channel >= JA_MAX_SIMULTANEOUS_CHANNELS) return JA_CHANNEL_INVALID;
return channels[channel].state;
}
inline float JA_SetSoundVolume(float volume, const int group = -1) // -1 para todos los grupos
{
const float v = SDL_clamp(volume, 0.0F, 1.0F);
if (group == -1) {
for (int i = 0; i < JA_MAX_GROUPS; ++i) {
JA_soundVolume[i] = v;
}
} else if (group >= 0 && group < JA_MAX_GROUPS) {
JA_soundVolume[group] = v;
} else {
return v; // Grupo inválido
}
// Aplicar volumen a canales activos
for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; i++) {
if ((channels[i].state == JA_CHANNEL_PLAYING) || (channels[i].state == JA_CHANNEL_PAUSED)) {
if (group == -1 || channels[i].group == group) {
if (channels[i].stream) {
SDL_SetAudioStreamGain(channels[i].stream, JA_soundVolume[channels[i].group]);
}
}
}
}
return v;
}
inline void JA_EnableSound(const bool value) {
if (!value) {
JA_StopChannel(-1); // Detener todos los canales
}
JA_soundEnabled = value;
}
inline float JA_SetVolume(float volume) {
float v = JA_SetMusicVolume(volume);
JA_SetSoundVolume(v, -1); // Aplicar a todos los grupos de sonido
return v;
}
// Referència al Music original porque updateOutgoingFade puga
// continuar descomprimint des de Vorbis sin al stream durante
// tota la fosa. Sense això, solo tenim el pre-fill puntual i
// SDL drena el stream més ràpid del previst cuando hay sounds
// bound a la misma device (~2x), buidant-lo a meitat del
// fade i sentint-se como un tall sec.
Music* music{nullptr};
FadeState fade;
};
// --- Engine ---
// Encapsula tot l'estat que antes vivia como a globals inline. Un sol Engine
// viu per procés (enforceat via assert al ctor contra `active_`). El ctor
// obre el device SDL; el dtor el tanca (RAII). Els deleters
// `Ja::deleteMusic`/`Ja::deleteSound` accedeixen al motor actiu via
// `Engine::active()` per parar canals antes d'alliberar.
class Engine {
public:
Engine(int freq, SDL_AudioFormat format, int num_channels);
~Engine();
Engine(const Engine&) = delete;
auto operator=(const Engine&) -> Engine& = delete;
Engine(Engine&&) = delete;
auto operator=(Engine&&) -> Engine& = delete;
// Retorna el motor actiu o nullptr si sin ha estat construït. L'usen
// los deleters de recursos porque no los arriba sin referència directa.
[[nodiscard]] static auto active() noexcept -> Engine*;
void update();
// --- Música ---
void playMusic(Music* music, int loop = -1);
void pauseMusic();
void resumeMusic();
void stopMusic();
void fadeOutMusic(int milliseconds);
void crossfadeMusic(Music* music, int crossfade_ms, int loop = -1);
[[nodiscard]] auto getMusicState() const -> MusicState;
auto setMusicVolume(float volume) -> float;
// Multiplicador de velocitat de reproducció de la música actual
// via `SDL_SetAudioStreamFrequencyRatio`. 1.0 = normal, 2.0 =
// doble velocitat. Cal saber que también puja el to (efecte
// "chipmunk") — es el comportament arcade clàssic dels comptes
// enrere. Cada `playMusic` crea un stream nuevo con ratio 1.0,
// así que un canvi de track reseteja la velocitat
// implícitament. No-op si no hay música activa.
void setMusicSpeed(float ratio);
// Registra un callback que es disparà cuando la música actual acabi de
// drenar naturalment (times == 0 + stream buit). Es crida DESPRÉS de
// stopMusic, así que el callback pot invocar playMusic sin córrer.
// S'executa al mismo thread que Engine::update (render loop); no fer
// operacions blocants.
void setOnMusicEnded(std::function<void()> callback);
// Notifica al motor que un Music s'está destruint: si es el current_music
// s'atura antes que los seus recursos (stream/vorbis) deixin de ser vàlids.
void onMusicDeleted(const Music* music);
// --- So ---
auto playSound(Sound* sound, int loop = 0, int group = 0) -> int;
auto playSoundOnChannel(Sound* sound, int channel, int loop = 0, int group = 0) -> int;
// Ajusta la velocitat de reproducció d'un canal actiu via
// `SDL_SetAudioStreamFrequencyRatio`. 1.0 = normal. Igual que a
// `setMusicSpeed`, puja/baixa el to junt con la velocitat
// (efecte "chipmunk"); para SFX curts arcade es el que volem.
// No-op si el canal no está actiu. Cridar-lo just después de
// `playSound`/`playSoundOnChannel` porque el ratio cobreixi
// tota la reproducció.
void setChannelSpeed(int channel, float ratio);
// Reproducció con so processat per un efecte. Retorna el canal
// assignat o -1 si no queden slots d'efecte (MAX_EFFECT_CHANNELS).
// El sound original solo s'usa per consultar el spec/buffer; el
// canal manipula el buffer ya processat (no reapunta a `sound`).
auto playSoundWithEcho(const Sound* sound, const EchoParams& params, int group = 0) -> int;
auto playSoundWithReverb(const Sound* sound, const ReverbParams& params, int group = 0) -> int;
void pauseChannel(int channel);
void resumeChannel(int channel);
void stopChannel(int channel);
auto setSoundVolume(float volume, int group = -1) -> float;
// Notifica al motor que un Sound s'está destruint: los canals que el
// referenciïn es paren antes d'alliberar el buffer.
void onSoundDeleted(const Sound* sound);
private:
void stealCurrentIntoOutgoing(int duration_ms);
void updateOutgoingFade();
void updateIncomingFade();
void updateCurrentMusic();
void updateSoundChannels();
// Empenta un buffer ya processat (S16) a un canal lliure y el deixa
// sonar sin bucle. Camí comú dels dos overloads playSoundWith*.
// Retorna el canal o -1 si no queden slots.
auto playProcessedOnFreeChannel(const std::vector<std::uint8_t>& bytes, const SDL_AudioSpec& spec, int group) -> int;
template <typename Fn>
void forEachTargetChannel(int channel, Fn&& fn);
Music* current_music_{nullptr};
Channel channels_[MAX_SIMULTANEOUS_CHANNELS]{};
SDL_AudioSpec audio_spec_{DEFAULT_SPEC};
float music_volume_{1.0F};
float sound_volume_[MAX_GROUPS]{};
SDL_AudioDeviceID sdl_audio_device_{0};
OutgoingMusic outgoing_music_;
FadeState incoming_fade_;
std::function<void()> on_music_ended_;
// Comptador derivat de Channel::has_effect — evita haver-lo de
// recalcular cada vegada que algú demana un play con efecte.
int effect_channels_active_{0};
// NOLINTNEXTLINE(readability-identifier-naming) — convenció projecte: private static con sufix _
static Engine* active_;
};
// --- Factories y destructors (permanents) ---
// No depenen de l'estat del motor: loadMusic/loadSound solo construeixen
// objectes, deleteMusic/deleteSound consulten Engine::active() per parar
// canals antes d'alliberar (si el motor aún viu).
[[nodiscard]] auto loadMusic(const Uint8* buffer, Uint32 length) -> Music*;
[[nodiscard]] auto loadMusic(const Uint8* buffer, Uint32 length, const char* filename) -> Music*;
void deleteMusic(Music* music);
[[nodiscard]] auto loadSound(std::uint8_t* buffer, std::uint32_t size) -> Sound*;
void deleteSound(Sound* sound);
} // namespace Ja
@@ -0,0 +1,80 @@
#include "core/audio/sound_effects_config.hpp"
#include <string>
#include <iostream>
#include "core/resources/resource_helper.hpp"
#include "external/fkyaml_node.hpp"
namespace {
// Lector de camp con fallback: deixa el destí intacte si la clau no
// existeix (los defaults dels Ja::*Params s'inicialitzen al ctor del
// struct, así que el comportament es "preset parcial = preset complet
// con defaults per als camps que falten").
template <typename T>
void readField(const fkyaml::node& node, const char* key, T& dst) {
if (node.contains(key)) { dst = node[key].get_value<T>(); }
}
} // namespace
auto SoundEffectsConfig::get() -> SoundEffectsConfig& {
static SoundEffectsConfig instance_;
return instance_;
}
void SoundEffectsConfig::load(const std::string& file_path) {
auto bytes = Resource::Helper::loadFile(file_path);
if (bytes.empty()) {
std::cerr << "[SoundEffectsConfig] no se ha podido abrir " << file_path
<< " — sin presets de efecto disponibles\n";
return;
}
try {
const auto* begin = reinterpret_cast<const char*>(bytes.data());
const auto* end = begin + bytes.size();
auto yaml = fkyaml::node::deserialize(begin, end);
if (yaml.contains("echo") && yaml["echo"].is_mapping()) {
for (auto it = yaml["echo"].begin(); it != yaml["echo"].end(); ++it) {
const auto NAME = it.key().get_value<std::string>();
const auto& node = it.value();
Ja::EchoParams params{};
readField(node, "delay_ms", params.delay_ms);
readField(node, "feedback", params.feedback);
readField(node, "wet", params.wet);
echoes_[NAME] = params;
}
}
if (yaml.contains("reverb") && yaml["reverb"].is_mapping()) {
for (auto it = yaml["reverb"].begin(); it != yaml["reverb"].end(); ++it) {
const auto NAME = it.key().get_value<std::string>();
const auto& node = it.value();
Ja::ReverbParams params{};
readField(node, "room_size", params.room_size);
readField(node, "damping", params.damping);
readField(node, "wet", params.wet);
reverbs_[NAME] = params;
}
}
std::cout << "[SoundEffectsConfig] " << echoes_.size() << " preset(s) de echo y "
<< reverbs_.size() << " de reverb desde " << file_path << "\n";
} catch (const std::exception& e) {
std::cerr << "[SoundEffectsConfig] error parseando " << file_path << ": " << e.what() << "\n";
}
}
auto SoundEffectsConfig::findEcho(const std::string& name) const -> const Ja::EchoParams* {
const auto IT = echoes_.find(name);
return (IT == echoes_.end()) ? nullptr : &IT->second;
}
auto SoundEffectsConfig::findReverb(const std::string& name) const -> const Ja::ReverbParams* {
const auto IT = reverbs_.find(name);
return (IT == reverbs_.end()) ? nullptr : &IT->second;
}
@@ -0,0 +1,36 @@
#pragma once
#include <string>
#include <unordered_map>
#include "core/audio/jail_audio.hpp" // Para Ja::EchoParams / Ja::ReverbParams
// Catàleg de presets d'efectes carregat des de data/config/sounds.yaml. La capa
// Audio (playSoundWithEcho/playSoundWithReverb) hi accedeix per nom: si el
// preset no existeix, el so es reprodueix sec con un avís a stderr.
//
// Patró Meyers idèntic a UiConfig/Locale: un sol load() a l'arrencada, sense
// hot-reload. Si el archivo no existeix, el catàleg queda buit (sin preset
// disponible) i tots los playSoundWith* es comporten como playSound dry.
class SoundEffectsConfig {
public:
static auto get() -> SoundEffectsConfig&;
SoundEffectsConfig(const SoundEffectsConfig&) = delete;
SoundEffectsConfig(SoundEffectsConfig&&) = delete;
auto operator=(const SoundEffectsConfig&) -> SoundEffectsConfig& = delete;
auto operator=(SoundEffectsConfig&&) -> SoundEffectsConfig& = delete;
void load(const std::string& file_path);
// Retorna nullptr si el preset no existeix.
[[nodiscard]] auto findEcho(const std::string& name) const -> const Ja::EchoParams*;
[[nodiscard]] auto findReverb(const std::string& name) const -> const Ja::ReverbParams*;
private:
SoundEffectsConfig() = default;
~SoundEffectsConfig() = default;
std::unordered_map<std::string, Ja::EchoParams> echoes_;
std::unordered_map<std::string, Ja::ReverbParams> reverbs_;
};
+12 -7
View File
@@ -280,16 +280,23 @@ namespace Rendering {
constexpr int VSYNC_DEFAULT = 1; // 0=disabled, 1=enabled
} // namespace Rendering
// Audio (sistema de so i música)
// Audio (sistema de sonido y música) — usado por Audio::Config en init()
namespace Audio {
constexpr float VOLUME = 1.0F; // Volumen maestro (0.0 a 1.0)
constexpr bool ENABLED = true; // Audio habilitado por defecto
constexpr bool ENABLED = true; // Audio habilitado por defecto
constexpr float VOLUME = 1.0F; // Volumen maestro (0..1)
constexpr bool MUSIC_ENABLED = true; // Música habilitada
constexpr float MUSIC_VOLUME = 0.8F; // Volumen música (0..1)
constexpr bool SOUND_ENABLED = true; // Efectos habilitados
constexpr float SOUND_VOLUME = 1.0F; // Volumen efectos (0..1)
constexpr float VOLUME_STEP = 0.05F; // Paso UI (5%)
constexpr int FREQUENCY = 48000; // Frecuencia de muestreo (Hz)
constexpr int CROSSFADE_MS = 1500; // Crossfade por defecto entre pistas (ms)
constexpr SDL_AudioFormat FORMAT = SDL_AUDIO_S16; // PCM 16-bit signed nativo
constexpr int CHANNELS = 2; // Estéreo
} // namespace Audio
// Música (pistas de fondo)
namespace Music {
constexpr float VOLUME = 0.8F; // Volumen música
constexpr bool ENABLED = true; // Música habilitada
constexpr const char* GAME_TRACK = "game.ogg"; // Pista de juego
constexpr const char* TITLE_TRACK = "title.ogg"; // Pista de titulo
constexpr int FADE_DURATION_MS = 1000; // Fade out duration
@@ -297,8 +304,6 @@ constexpr int FADE_DURATION_MS = 1000; // Fade out duration
// Efectes de so (sons puntuals)
namespace Sound {
constexpr float VOLUME = 1.0F; // Volumen efectos
constexpr bool ENABLED = true; // Sonidos habilitados
constexpr const char* CONTINUE = "effects/continue.wav"; // Cuenta atras
constexpr const char* EXPLOSION = "effects/explosion.wav"; // Explosión
constexpr const char* EXPLOSION2 = "effects/explosion2.wav"; // Explosión alternativa
+16 -10
View File
@@ -9,7 +9,7 @@
#include "scene_context.hpp"
#include "core/audio/audio.hpp"
#include "core/audio/audio_cache.hpp"
#include "core/audio/audio_adapter.hpp"
#include "core/defaults.hpp"
#include "core/input/input.hpp"
#include "core/input/mouse.hpp"
@@ -224,17 +224,23 @@ auto Director::run() -> int {
Mouse::forceHide();
}
// Inicialitzar sistema de audio
Audio::init();
Audio::get()->setMusicVolume(1.0);
Audio::get()->setSoundVolume(0.4);
// Inicializar sistema de audio (config inyectada desde Defaults)
const Audio::Config AUDIO_CONFIG{
.enabled = Defaults::Audio::ENABLED,
.volume = Defaults::Audio::VOLUME,
.music_enabled = Defaults::Audio::MUSIC_ENABLED,
.music_volume = Defaults::Audio::MUSIC_VOLUME,
.sound_enabled = Defaults::Audio::SOUND_ENABLED,
.sound_volume = Defaults::Audio::SOUND_VOLUME,
};
Audio::init(AUDIO_CONFIG);
Audio::get()->applySettings(AUDIO_CONFIG); // Aplicar volúmenes iniciales al motor
// Precachejar música per evitar lag al començar
AudioCache::getMusic("title.ogg");
AudioCache::getMusic("game.ogg");
// Precachear música para evitar lag al empezar
AudioResource::getMusic("title.ogg");
AudioResource::getMusic("game.ogg");
if (Options::console) {
std::cout << "Música precachejada: "
<< AudioCache::getMusicCacheSize() << " archivos\n";
std::cout << "Música precacheada\n";
}
// Crear context de escenes
+68 -49
View File
@@ -1,4 +1,4 @@
// Ogg Vorbis audio decoder - v1.20 - public domain
// Ogg Vorbis audio decoder - v1.22 - public domain
// http://nothings.org/stb_vorbis/
//
// Original version written by Sean Barrett in 2007.
@@ -29,12 +29,15 @@
// Bernhard Wodo Evan Balster github:alxprd
// Tom Beaumont Ingo Leitgeb Nicolas Guillemot
// Phillip Bennefall Rohit Thiago Goulart
// github:manxorist saga musix github:infatum
// github:manxorist Saga Musix github:infatum
// Timur Gagiev Maxwell Koo Peter Waller
// github:audinowho Dougall Johnson David Reid
// github:Clownacy Pedro J. Estebanez Remi Verschelde
// AnthoFoxo github:morlat Gabriel Ravier
//
// Partial history:
// 1.22 - 2021-07-11 - various small fixes
// 1.21 - 2021-07-02 - fix bug for files with no comments
// 1.20 - 2020-07-11 - several small fixes
// 1.19 - 2020-02-05 - warnings
// 1.18 - 2020-02-02 - fix seek bugs; parse header comments; misc warnings etc.
@@ -220,6 +223,12 @@ extern int stb_vorbis_decode_frame_pushdata(
// channel. In other words, (*output)[0][0] contains the first sample from
// the first channel, and (*output)[1][0] contains the first sample from
// the second channel.
//
// *output points into stb_vorbis's internal output buffer storage; these
// buffers are owned by stb_vorbis and application code should not free
// them or modify their contents. They are transient and will be overwritten
// once you ask for more data to get decoded, so be sure to grab any data
// you need before then.
extern void stb_vorbis_flush_pushdata(stb_vorbis *f);
// inform stb_vorbis that your next datablock will not be contiguous with
@@ -579,7 +588,7 @@ enum STBVorbisError
#if defined(_MSC_VER) || defined(__MINGW32__)
#include <malloc.h>
#endif
#if defined(__linux__) || defined(__linux) || defined(__EMSCRIPTEN__) || defined(__NEWLIB__)
#if defined(__linux__) || defined(__linux) || defined(__sun__) || defined(__EMSCRIPTEN__) || defined(__NEWLIB__)
#include <alloca.h>
#endif
#else // STB_VORBIS_NO_CRT
@@ -646,6 +655,12 @@ typedef signed int int32;
typedef float codetype;
#ifdef _MSC_VER
#define STBV_NOTUSED(v) (void)(v)
#else
#define STBV_NOTUSED(v) (void)sizeof(v)
#endif
// @NOTE
//
// Some arrays below are tagged "//varies", which means it's actually
@@ -1046,7 +1061,7 @@ static float float32_unpack(uint32 x)
uint32 sign = x & 0x80000000;
uint32 exp = (x & 0x7fe00000) >> 21;
double res = sign ? -(double)mantissa : (double)mantissa;
return (float) ldexp((float)res, exp-788);
return (float) ldexp((float)res, (int)exp-788);
}
@@ -1077,6 +1092,7 @@ static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values)
// find the first entry
for (k=0; k < n; ++k) if (len[k] < NO_CODE) break;
if (k == n) { assert(c->sorted_entries == 0); return TRUE; }
assert(len[k] < 32); // no error return required, code reading lens checks this
// add to the list
add_entry(c, 0, k, m++, len[k], values);
// add all available leaves
@@ -1090,6 +1106,7 @@ static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values)
uint32 res;
int z = len[i], y;
if (z == NO_CODE) continue;
assert(z < 32); // no error return required, code reading lens checks this
// find lowest available leaf (should always be earliest,
// which is what the specification calls for)
// note that this property, and the fact we can never have
@@ -1099,12 +1116,10 @@ static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values)
while (z > 0 && !available[z]) --z;
if (z == 0) { return FALSE; }
res = available[z];
assert(z >= 0 && z < 32);
available[z] = 0;
add_entry(c, bit_reverse(res), i, m++, len[i], values);
// propagate availability up the tree
if (z != len[i]) {
assert(len[i] >= 0 && len[i] < 32);
for (y=len[i]; y > z; --y) {
assert(available[y] == 0);
available[y] = res + (1 << (32-y));
@@ -2577,34 +2592,33 @@ static void imdct_step3_inner_s_loop_ld654(int n, float *e, int i_off, float *A,
while (z > base) {
float k00,k11;
float l00,l11;
k00 = z[-0] - z[-8];
k11 = z[-1] - z[-9];
z[-0] = z[-0] + z[-8];
z[-1] = z[-1] + z[-9];
z[-8] = k00;
z[-9] = k11 ;
k00 = z[-0] - z[ -8];
k11 = z[-1] - z[ -9];
l00 = z[-2] - z[-10];
l11 = z[-3] - z[-11];
z[ -0] = z[-0] + z[ -8];
z[ -1] = z[-1] + z[ -9];
z[ -2] = z[-2] + z[-10];
z[ -3] = z[-3] + z[-11];
z[ -8] = k00;
z[ -9] = k11;
z[-10] = (l00+l11) * A2;
z[-11] = (l11-l00) * A2;
k00 = z[ -2] - z[-10];
k11 = z[ -3] - z[-11];
z[ -2] = z[ -2] + z[-10];
z[ -3] = z[ -3] + z[-11];
z[-10] = (k00+k11) * A2;
z[-11] = (k11-k00) * A2;
k00 = z[-12] - z[ -4]; // reverse to avoid a unary negation
k00 = z[ -4] - z[-12];
k11 = z[ -5] - z[-13];
l00 = z[ -6] - z[-14];
l11 = z[ -7] - z[-15];
z[ -4] = z[ -4] + z[-12];
z[ -5] = z[ -5] + z[-13];
z[-12] = k11;
z[-13] = k00;
k00 = z[-14] - z[ -6]; // reverse to avoid a unary negation
k11 = z[ -7] - z[-15];
z[ -6] = z[ -6] + z[-14];
z[ -7] = z[ -7] + z[-15];
z[-14] = (k00+k11) * A2;
z[-15] = (k00-k11) * A2;
z[-12] = k11;
z[-13] = -k00;
z[-14] = (l11-l00) * A2;
z[-15] = (l00+l11) * -A2;
iter_54(z);
iter_54(z-8);
@@ -3069,6 +3083,7 @@ static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *f
for (q=1; q < g->values; ++q) {
j = g->sorted_order[q];
#ifndef STB_VORBIS_NO_DEFER_FLOOR
STBV_NOTUSED(step2_flag);
if (finalY[j] >= 0)
#else
if (step2_flag[j])
@@ -3171,6 +3186,7 @@ static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start,
// WINDOWING
STBV_NOTUSED(left_end);
n = f->blocksize[m->blockflag];
map = &f->mapping[m->mapping];
@@ -3368,7 +3384,7 @@ static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start,
// this isn't to spec, but spec would require us to read ahead
// and decode the size of all current frames--could be done,
// but presumably it's not a commonly used feature
f->current_loc = -n2; // start of first frame is positioned for discard
f->current_loc = 0u - n2; // start of first frame is positioned for discard (NB this is an intentional unsigned overflow/wrap-around)
// we might have to discard samples "from" the next frame too,
// if we're lapping a large block then a small at the start?
f->discard_samples_deferred = n - right_end;
@@ -3642,9 +3658,11 @@ static int start_decoder(vorb *f)
f->vendor[len] = (char)'\0';
//user comments
f->comment_list_length = get32_packet(f);
if (f->comment_list_length > 0) {
f->comment_list = (char**)setup_malloc(f, sizeof(char*) * (f->comment_list_length));
if (f->comment_list == NULL) return error(f, VORBIS_outofmem);
f->comment_list = NULL;
if (f->comment_list_length > 0)
{
f->comment_list = (char**) setup_malloc(f, sizeof(char*) * (f->comment_list_length));
if (f->comment_list == NULL) return error(f, VORBIS_outofmem);
}
for(i=0; i < f->comment_list_length; ++i) {
@@ -3867,8 +3885,7 @@ static int start_decoder(vorb *f)
unsigned int div=1;
for (k=0; k < c->dimensions; ++k) {
int off = (z / div) % c->lookup_values;
float val = mults[off];
val = mults[off]*c->delta_value + c->minimum_value + last;
float val = mults[off]*c->delta_value + c->minimum_value + last;
c->multiplicands[j*c->dimensions + k] = val;
if (c->sequence_p)
last = val;
@@ -3951,7 +3968,7 @@ static int start_decoder(vorb *f)
if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
}
for (k=0; k < 1 << g->class_subclasses[j]; ++k) {
g->subclass_books[j][k] = get_bits(f,8)-1;
g->subclass_books[j][k] = (int16)get_bits(f,8)-1;
if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
}
}
@@ -4509,6 +4526,7 @@ stb_vorbis *stb_vorbis_open_pushdata(
*error = VORBIS_need_more_data;
else
*error = p.error;
vorbis_deinit(&p);
return NULL;
}
f = vorbis_alloc(&p);
@@ -4566,7 +4584,7 @@ static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last)
header[i] = get8(f);
if (f->eof) return 0;
if (header[4] != 0) goto invalid;
goal = header[22] + (header[23] << 8) + (header[24]<<16) + (header[25]<<24);
goal = header[22] + (header[23] << 8) + (header[24]<<16) + ((uint32)header[25]<<24);
for (i=22; i < 26; ++i)
header[i] = 0;
crc = 0;
@@ -4970,7 +4988,7 @@ unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f)
// set. whoops!
break;
}
previous_safe = last_page_loc+1;
//previous_safe = last_page_loc+1; // NOTE: not used after this point, but note for debugging
last_page_loc = stb_vorbis_get_file_offset(f);
}
@@ -5081,7 +5099,10 @@ stb_vorbis * stb_vorbis_open_filename(const char *filename, int *error, const st
stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, int *error, const stb_vorbis_alloc *alloc)
{
stb_vorbis *f, p;
if (data == NULL) return NULL;
if (!data) {
if (error) *error = VORBIS_unexpected_eof;
return NULL;
}
vorbis_init(&p, alloc);
p.stream = (uint8 *) data;
p.stream_end = (uint8 *) data + len;
@@ -5156,11 +5177,11 @@ static void copy_samples(short *dest, float *src, int len)
static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len)
{
#define BUFFER_SIZE 32
float buffer[BUFFER_SIZE];
int i,j,o,n = BUFFER_SIZE;
#define STB_BUFFER_SIZE 32
float buffer[STB_BUFFER_SIZE];
int i,j,o,n = STB_BUFFER_SIZE;
check_endianness();
for (o = 0; o < len; o += BUFFER_SIZE) {
for (o = 0; o < len; o += STB_BUFFER_SIZE) {
memset(buffer, 0, sizeof(buffer));
if (o + n > len) n = len - o;
for (j=0; j < num_c; ++j) {
@@ -5177,16 +5198,17 @@ static void compute_samples(int mask, short *output, int num_c, float **data, in
output[o+i] = v;
}
}
#undef STB_BUFFER_SIZE
}
static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len)
{
#define BUFFER_SIZE 32
float buffer[BUFFER_SIZE];
int i,j,o,n = BUFFER_SIZE >> 1;
#define STB_BUFFER_SIZE 32
float buffer[STB_BUFFER_SIZE];
int i,j,o,n = STB_BUFFER_SIZE >> 1;
// o is the offset in the source data
check_endianness();
for (o = 0; o < len; o += BUFFER_SIZE >> 1) {
for (o = 0; o < len; o += STB_BUFFER_SIZE >> 1) {
// o2 is the offset in the output data
int o2 = o << 1;
memset(buffer, 0, sizeof(buffer));
@@ -5216,6 +5238,7 @@ static void compute_stereo_samples(short *output, int num_c, float **data, int d
output[o2+i] = v;
}
}
#undef STB_BUFFER_SIZE
}
static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples)
@@ -5288,8 +5311,6 @@ int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short
float **outputs;
int len = num_shorts / channels;
int n=0;
int z = f->channels;
if (z > channels) z = channels;
while (n < len) {
int k = f->channel_buffer_end - f->channel_buffer_start;
if (n+k >= len) k = len - n;
@@ -5308,8 +5329,6 @@ int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, in
{
float **outputs;
int n=0;
int z = f->channels;
if (z > channels) z = channels;
while (n < len) {
int k = f->channel_buffer_end - f->channel_buffer_start;
if (n+k >= len) k = len - n;
+13
View File
@@ -0,0 +1,13 @@
// Unitat de compilació aïllada per a la implementació de stb_vorbis.
// Viu dins de source/external/ perquè el `.clang-tidy` d'aquesta carpeta
// desactiva tots els checks (com fa per stb_image_write_impl.cpp) i el
// pre-commit hook ja filtra aquesta ruta de clang-format / clang-tidy.
// Així els fals positius de clang-analyzer-* dins de codi C de tercers
// no afecten el nostre codi, que continua tenint tots els checks actius.
//
// jail_audio.cpp defineix STB_VORBIS_HEADER_ONLY abans d'incloure el .c,
// així només en veu les declaracions; les definicions les aporta aquest
// TU i l'enllaçador les resol.
// NOLINTNEXTLINE(bugprone-suspicious-include)
#include "external/stb_vorbis.c"
+10 -10
View File
@@ -210,10 +210,10 @@ void init() {
// Audio
audio.enabled = Defaults::Audio::ENABLED;
audio.volume = Defaults::Audio::VOLUME;
audio.music.enabled = Defaults::Music::ENABLED;
audio.music.volume = Defaults::Music::VOLUME;
audio.sound.enabled = Defaults::Sound::ENABLED;
audio.sound.volume = Defaults::Sound::VOLUME;
audio.music.enabled = Defaults::Audio::MUSIC_ENABLED;
audio.music.volume = Defaults::Audio::MUSIC_VOLUME;
audio.sound.enabled = Defaults::Audio::SOUND_ENABLED;
audio.sound.volume = Defaults::Audio::SOUND_VOLUME;
// Version
version = std::string(Project::VERSION);
@@ -409,16 +409,16 @@ static void loadAudioConfigFromYaml(const fkyaml::node& yaml) {
try {
audio.music.enabled = mus["enabled"].get_value<bool>();
} catch (...) {
audio.music.enabled = Defaults::Music::ENABLED;
audio.music.enabled = Defaults::Audio::MUSIC_ENABLED;
}
}
if (mus.contains("volume")) {
try {
auto val = mus["volume"].get_value<float>();
audio.music.volume = (val >= 0.0F && val <= 1.0F) ? val : Defaults::Music::VOLUME;
audio.music.volume = (val >= 0.0F && val <= 1.0F) ? val : Defaults::Audio::MUSIC_VOLUME;
} catch (...) {
audio.music.volume = Defaults::Music::VOLUME;
audio.music.volume = Defaults::Audio::MUSIC_VOLUME;
}
}
}
@@ -430,16 +430,16 @@ static void loadAudioConfigFromYaml(const fkyaml::node& yaml) {
try {
audio.sound.enabled = snd["enabled"].get_value<bool>();
} catch (...) {
audio.sound.enabled = Defaults::Sound::ENABLED;
audio.sound.enabled = Defaults::Audio::SOUND_ENABLED;
}
}
if (snd.contains("volume")) {
try {
auto val = snd["volume"].get_value<float>();
audio.sound.volume = (val >= 0.0F && val <= 1.0F) ? val : Defaults::Sound::VOLUME;
audio.sound.volume = (val >= 0.0F && val <= 1.0F) ? val : Defaults::Audio::SOUND_VOLUME;
} catch (...) {
audio.sound.volume = Defaults::Sound::VOLUME;
audio.sound.volume = Defaults::Audio::SOUND_VOLUME;
}
}
}
+1 -1
View File
@@ -94,7 +94,7 @@ TitleScene::TitleScene(SDLManager& sdl, SceneContext& context)
inicialitzar_titol();
// Iniciar música de título si no está sonant
if (Audio::get()->getMusicState() != Audio::MusicState::PLAYING) {
if (Audio::getMusicState() != Audio::MusicState::PLAYING) {
Audio::get()->playMusic("title.ogg");
}
}
+1 -1
View File
@@ -88,7 +88,7 @@ void StageManager::canviar_estat(EstatStage nou_estat) {
// [NOU] Iniciar música al entrar en LEVEL_START (después de INIT_HUD)
// Solo si no está sonant ya (per evitar reset en loops posteriors)
if (Audio::get()->getMusicState() != Audio::MusicState::PLAYING) {
if (Audio::getMusicState() != Audio::MusicState::PLAYING) {
Audio::get()->playMusic("game.ogg");
}
}