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normalitzat Audio

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This commit is contained in:
Sergio
2026-04-18 11:43:45 +02:00
parent 53c2b345c9
commit 0dbf38f506
7 changed files with 435 additions and 227 deletions
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1
CMakeLists.txt
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@@ -43,6 +43,7 @@ configure_file(${CMAKE_SOURCE_DIR}/source/version.h.in ${CMAKE_BINARY_DIR}/versi
set(APP_SOURCES
# Core - Audio
source/core/audio/audio.cpp
source/core/audio/audio_adapter.cpp
# Core - Input
source/core/input/global_inputs.cpp

113
source/core/audio/audio.cpp
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@@ -1,19 +1,27 @@
#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
// Implementación de stb_vorbis (debe estar ANTES de incluir jail_audio.hpp)
// 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"
#include "external/stb_vorbis.c"
// stb_vorbis.c filtra les macros L, C i R (i PLAYBACK_*) al TU. Les netegem
// perquè xocarien amb noms de paràmetres de plantilla en altres headers.
#undef L
#undef C
#undef R
#undef PLAYBACK_MONO
#undef PLAYBACK_LEFT
#undef PLAYBACK_RIGHT
// clang-format on
#include "core/audio/jail_audio.hpp" // Para JA_FadeOutMusic, JA_Init, JA_PauseM...
#include "core/resources/resource_cache.hpp" // Para Resource
#include "game/options.hpp" // Para AudioOptions, audio, MusicOptions
#include "core/audio/audio_adapter.hpp" // Para AudioResource::getMusic/getSound
#include "core/audio/jail_audio.hpp" // Para JA_*
#include "game/options.hpp" // Para Options::audio
// Singleton
Audio* Audio::instance = nullptr;
@@ -22,7 +30,10 @@ Audio* Audio::instance = nullptr;
void Audio::init() { Audio::instance = new Audio(); }
// Libera la instancia
void Audio::destroy() { delete Audio::instance; }
void Audio::destroy() {
delete Audio::instance;
Audio::instance = nullptr;
}
// Obtiene la instancia
auto Audio::get() -> Audio* { return Audio::instance; }
@@ -38,10 +49,15 @@ Audio::~Audio() {
// Método principal
void Audio::update() {
JA_Update();
// Sincronizar estado: detectar cuando la música se para (ej. fade-out completado)
if (instance && instance->music_.state == MusicState::PLAYING && JA_GetMusicState() != JA_MUSIC_PLAYING) {
instance->music_.state = MusicState::STOPPED;
}
}
// Reproduce la música
void Audio::playMusic(const std::string& name, const int loop) { // NOLINT(readability-convert-member-functions-to-static)
// Reproduce la música por nombre (con crossfade opcional)
void Audio::playMusic(const std::string& name, const int loop, const int crossfade_ms) {
bool new_loop = (loop != 0);
// Si ya está sonando exactamente la misma pista y mismo modo loop, no hacemos nada
@@ -49,29 +65,45 @@ void Audio::playMusic(const std::string& name, const int loop) { // NOLINT(read
return;
}
// Intentar obtener recurso; si falla, no tocar estado
auto* resource = Resource::Cache::get()->getMusic(name);
if (resource == nullptr) {
// manejo de error opcional
return;
if (!music_enabled_) return;
auto* resource = AudioResource::getMusic(name);
if (resource == nullptr) return;
if (crossfade_ms > 0 && music_.state == MusicState::PLAYING) {
JA_CrossfadeMusic(resource, crossfade_ms, loop);
} else {
if (music_.state == MusicState::PLAYING) {
JA_StopMusic();
}
JA_PlayMusic(resource, loop);
}
// 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
}
// Llamada al motor para reproducir la nueva pista
JA_PlayMusic(resource, loop);
// Actualizar estado y metadatos después de iniciar con éxito
music_.name = name;
music_.loop = new_loop;
music_.state = MusicState::PLAYING;
}
// Reproduce la música por puntero (con crossfade opcional)
void Audio::playMusic(JA_Music_t* music, const int loop, const int crossfade_ms) {
if (!music_enabled_ || music == nullptr) return;
if (crossfade_ms > 0 && music_.state == MusicState::PLAYING) {
JA_CrossfadeMusic(music, crossfade_ms, loop);
} else {
if (music_.state == MusicState::PLAYING) {
JA_StopMusic();
}
JA_PlayMusic(music, loop);
}
music_.name.clear(); // nom desconegut quan es passa per punter
music_.loop = (loop != 0);
music_.state = MusicState::PLAYING;
}
// Pausa la música
void Audio::pauseMusic() { // NOLINT(readability-convert-member-functions-to-static)
void Audio::pauseMusic() {
if (music_enabled_ && music_.state == MusicState::PLAYING) {
JA_PauseMusic();
music_.state = MusicState::PAUSED;
@@ -79,7 +111,7 @@ void Audio::pauseMusic() { // NOLINT(readability-convert-member-functions-to-st
}
// Continua la música pausada
void Audio::resumeMusic() { // NOLINT(readability-convert-member-functions-to-static)
void Audio::resumeMusic() {
if (music_enabled_ && music_.state == MusicState::PAUSED) {
JA_ResumeMusic();
music_.state = MusicState::PLAYING;
@@ -87,7 +119,7 @@ void Audio::resumeMusic() { // NOLINT(readability-convert-member-functions-to-s
}
// Detiene la música
void Audio::stopMusic() { // NOLINT(readability-make-member-function-const)
void Audio::stopMusic() {
if (music_enabled_) {
JA_StopMusic();
music_.state = MusicState::STOPPED;
@@ -97,13 +129,13 @@ void Audio::stopMusic() { // NOLINT(readability-make-member-function-const)
// Reproduce un sonido por nombre
void Audio::playSound(const std::string& name, Group group) const {
if (sound_enabled_) {
JA_PlaySound(Resource::Cache::get()->getSound(name), 0, static_cast<int>(group));
JA_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 {
if (sound_enabled_) {
if (sound_enabled_ && sound != nullptr) {
JA_PlaySound(sound, 0, static_cast<int>(group));
}
}
@@ -138,7 +170,7 @@ auto Audio::getRealMusicState() -> MusicState {
}
}
// Establece el volumen de los sonidos
// Establece el volumen de los sonidos (float 0.0..1.0)
void Audio::setSoundVolume(float sound_volume, Group group) const {
if (sound_enabled_) {
sound_volume = std::clamp(sound_volume, MIN_VOLUME, MAX_VOLUME);
@@ -147,7 +179,7 @@ void Audio::setSoundVolume(float sound_volume, Group group) const {
}
}
// Establece el volumen de la música
// Establece el volumen de la música (float 0.0..1.0)
void Audio::setMusicVolume(float music_volume) const {
if (music_enabled_) {
music_volume = std::clamp(music_volume, MIN_VOLUME, MAX_VOLUME);
@@ -172,24 +204,9 @@ void Audio::enable(bool value) {
// Inicializa SDL Audio
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());
std::cout << "SDL_AUDIO could not initialize! SDL Error: " << SDL_GetError() << '\n';
} else {
JA_Init(FREQUENCY, SDL_AUDIO_S16LE, 2);
enable(Options::audio.enabled);
// Aplicar estado de música y sonido guardado en las opciones.
// enable() ya aplica los volúmenes, pero no toca music_enabled_/sound_enabled_.
// Si alguno está desactivado, hay que forzar el volumen a 0 en el backend.
if (!Options::audio.music.enabled) {
setMusicVolume(0.0F); // music_enabled_=true aún → llega a JA
enableMusic(false);
}
if (!Options::audio.sound.enabled) {
setSoundVolume(0.0F); // sound_enabled_=true aún → llega a JA
enableSound(false);
}
std::cout << "\n** AUDIO SYSTEM **\n";
std::cout << "Audio system initialized successfully\n";
}
}
}

41
source/core/audio/audio.hpp
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@@ -1,28 +1,35 @@
#pragma once
#include <cstdint> // Para int8_t, uint8_t
#include <string> // Para string
#include <utility> // Para move
// --- Clase Audio: gestor de audio (singleton) ---
// Implementació canònica, byte-idèntica entre projectes.
// Els volums es manegen internament com a float 0.0–1.0; la capa de
// presentació (menús, notificacions) usa les helpers toPercent/fromPercent
// per mostrar 0–100 a l'usuari.
class Audio {
public:
// --- Enums ---
enum class Group : int {
enum class Group : std::int8_t {
ALL = -1, // Todos los grupos
GAME = 0, // Sonidos del juego
INTERFACE = 1 // Sonidos de la interfaz
};
enum class MusicState {
enum class MusicState : std::uint8_t {
PLAYING, // Reproduciendo música
PAUSED, // Música pausada
STOPPED, // Música detenida
};
// --- 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
static constexpr float MAX_VOLUME = 1.0F; // Volumen máximo (float 0..1)
static constexpr float MIN_VOLUME = 0.0F; // Volumen mínimo (float 0..1)
static constexpr float VOLUME_STEP = 0.05F; // Pas estàndard per a UI (5%)
static constexpr int FREQUENCY = 48000; // Frecuencia de audio
static constexpr int DEFAULT_CROSSFADE_MS = 1500; // Duració del crossfade per defecte (ms)
// --- Singleton ---
static void init(); // Inicializa el objeto Audio
@@ -34,21 +41,31 @@ class Audio {
static void update(); // Actualización del sistema de audio
// --- 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
void playMusic(const std::string& name, int loop = -1, int crossfade_ms = 0); // Reproducir música por nombre (con crossfade opcional)
void playMusic(struct JA_Music_t* music, int loop = -1, int crossfade_ms = 0); // Reproducir música por puntero (con crossfade opcional)
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
// --- 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 volumen ---
// --- Control de volumen (API interna: float 0.0..1.0) ---
void setSoundVolume(float volume, Group group = Group::ALL) const; // Ajustar volumen de efectos
void setMusicVolume(float volume) const; // Ajustar volumen de música
// --- Helpers de conversió per a la capa de presentació ---
// UI (menús, notificacions) manega enters 0..100; internament viu float 0..1.
static constexpr auto toPercent(float volume) -> int {
return static_cast<int>(volume * 100.0F + 0.5F);
}
static constexpr auto fromPercent(int percent) -> float {
return static_cast<float>(percent) / 100.0F;
}
// --- Configuración general ---
void enable(bool value); // Establecer estado general
void toggleEnabled() { enabled_ = !enabled_; } // Alternar estado general
@@ -94,4 +111,4 @@ class Audio {
bool enabled_{true}; // Estado general del audio
bool sound_enabled_{true}; // Estado de los efectos de sonido
bool music_enabled_{true}; // Estado de la música
};
};

13
source/core/audio/audio_adapter.cpp Normal file
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@@ -0,0 +1,13 @@
#include "core/audio/audio_adapter.hpp"
#include "core/resources/resource_cache.hpp"
namespace AudioResource {
JA_Music_t* getMusic(const std::string& name) {
return Resource::Cache::get()->getMusic(name);
}
JA_Sound_t* getSound(const std::string& name) {
return Resource::Cache::get()->getSound(name);
}
} // namespace AudioResource

17
source/core/audio/audio_adapter.hpp Normal file
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@@ -0,0 +1,17 @@
#pragma once
// --- Audio Resource Adapter ---
// Aquest fitxer exposa una interfície comuna a Audio per obtenir JA_Music_t* /
// JA_Sound_t* per nom. Cada projecte la implementa en audio_adapter.cpp
// delegant al seu singleton de recursos (Resource::get(), Resource::Cache::get(),
// etc.). Això permet que audio.hpp/audio.cpp siguin idèntics entre projectes.
#include <string> // Para string
struct JA_Music_t;
struct JA_Sound_t;
namespace AudioResource {
JA_Music_t* getMusic(const std::string& name);
JA_Sound_t* getSound(const std::string& name);
} // namespace AudioResource

360
source/core/audio/jail_audio.hpp
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@@ -3,24 +3,41 @@
// --- 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 <stdio.h> // Para NULL, fseek, fclose, fopen, fread, ftell, FILE, SEEK_END, SEEK_SET
#include <stdlib.h> // Para free, malloc
#include <string.h> // Para strcpy, strlen
#include <iostream> // Para std::cout
#include <memory> // Para std::unique_ptr
#include <string> // Para std::string
#include <vector> // Para std::vector
#define STB_VORBIS_HEADER_ONLY
#include "external/stb_vorbis.h" // Para stb_vorbis_decode_memory
#include "external/stb_vorbis.c" // Para stb_vorbis_open_memory i streaming
// Deleter stateless per a buffers reservats amb `SDL_malloc` / `SDL_LoadWAV*`.
// Compatible amb `std::unique_ptr<Uint8[], SDLFreeDeleter>` — zero size
// overhead gràcies a EBO, igual que un unique_ptr amb default_delete.
struct SDLFreeDeleter {
void operator()(Uint8* p) const noexcept {
if (p) SDL_free(p);
}
};
// --- Public Enums ---
enum JA_Channel_state { JA_CHANNEL_INVALID,
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_SOUND_DISABLED,
};
enum JA_Music_state {
JA_MUSIC_INVALID,
JA_MUSIC_PLAYING,
JA_MUSIC_PAUSED,
JA_MUSIC_STOPPED,
JA_MUSIC_DISABLED };
JA_MUSIC_DISABLED,
};
// --- Struct Definitions ---
#define JA_MAX_SIMULTANEOUS_CHANNELS 20
@@ -29,7 +46,9 @@ enum JA_Music_state { JA_MUSIC_INVALID,
struct JA_Sound_t {
SDL_AudioSpec spec{SDL_AUDIO_S16, 2, 48000};
Uint32 length{0};
Uint8* buffer{NULL};
// Buffer descomprimit (PCM) propietat del sound. Reservat per SDL_LoadWAV
// via SDL_malloc; el deleter `SDLFreeDeleter` allibera amb SDL_free.
std::unique_ptr<Uint8[], SDLFreeDeleter> buffer;
};
struct JA_Channel_t {
@@ -44,21 +63,22 @@ struct JA_Channel_t {
struct JA_Music_t {
SDL_AudioSpec spec{SDL_AUDIO_S16, 2, 48000};
// OGG comprimit en memòria. Propietat nostra; es copia des del fitxer una
// sola vegada en JA_LoadMusic i es descomprimix en chunks per streaming.
Uint8* ogg_data{nullptr};
Uint32 ogg_length{0};
stb_vorbis* vorbis{nullptr}; // Handle del decoder, viu tot el cicle del JA_Music_t
// OGG comprimit en memòria. Propietat nostra; es copia des del buffer
// d'entrada una sola vegada en JA_LoadMusic i es descomprimix en chunks
// per streaming. Com que stb_vorbis guarda un punter persistent al
// `.data()` d'aquest 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 JA_Music_t
char* filename{nullptr};
std::string filename;
int times{0}; // Loops restants (-1 = infinit, 0 = un sol play)
int times{0}; // loops restants (-1 = infinit, 0 = un sol play)
SDL_AudioStream* stream{nullptr};
JA_Music_state state{JA_MUSIC_INVALID};
};
// --- Internal Global State ---
// Marcado 'inline' (C++17) para asegurar una única instancia.
// --- Internal Global State (inline, C++17) ---
inline JA_Music_t* current_music{nullptr};
inline JA_Channel_t channels[JA_MAX_SIMULTANEOUS_CHANNELS];
@@ -70,15 +90,27 @@ 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'
// --- Crossfade / Fade State ---
struct JA_FadeState {
bool active{false};
Uint64 start_time{0};
int duration_ms{0};
float initial_volume{0.0f};
};
struct JA_OutgoingMusic {
SDL_AudioStream* stream{nullptr};
JA_FadeState fade;
};
inline JA_OutgoingMusic outgoing_music;
inline JA_FadeState incoming_fade;
// --- 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);
inline void JA_CrossfadeMusic(JA_Music_t* music, int crossfade_ms, int loop = -1);
// --- Music streaming internals ---
// Bytes-per-sample per canal (sempre s16)
@@ -96,15 +128,15 @@ inline int JA_FeedMusicChunk(JA_Music_t* music) {
if (!music || !music->vorbis || !music->stream) return 0;
short chunk[JA_MUSIC_CHUNK_SHORTS];
const int channels = music->spec.channels;
const int num_channels = music->spec.channels;
const int samples_per_channel = stb_vorbis_get_samples_short_interleaved(
music->vorbis,
channels,
num_channels,
chunk,
JA_MUSIC_CHUNK_SHORTS);
if (samples_per_channel <= 0) return 0;
const int bytes = samples_per_channel * channels * JA_MUSIC_BYTES_PER_SAMPLE;
const int bytes = samples_per_channel * num_channels * JA_MUSIC_BYTES_PER_SAMPLE;
SDL_PutAudioStreamData(music->stream, chunk, bytes);
return samples_per_channel;
}
@@ -131,20 +163,51 @@ inline void JA_PumpMusic(JA_Music_t* music) {
}
}
// Pre-carrega `duration_ms` de so dins l'stream actual abans que l'stream
// siga robat per outgoing_music (crossfade o fade-out). Imprescindible amb
// streaming: l'stream robat no es pot re-alimentar perquè perd la referència
// al seu vorbis decoder. No aplica loop — si el vorbis s'esgota abans, parem.
inline void JA_PreFillOutgoing(JA_Music_t* music, int duration_ms) {
if (!music || !music->vorbis || !music->stream) return;
const int bytes_per_second = music->spec.freq * music->spec.channels * JA_MUSIC_BYTES_PER_SAMPLE;
const int needed_bytes = static_cast<int>((static_cast<int64_t>(duration_ms) * bytes_per_second) / 1000);
while (SDL_GetAudioStreamAvailable(music->stream) < needed_bytes) {
const int decoded = JA_FeedMusicChunk(music);
if (decoded <= 0) break; // EOF: deixem drenar el que hi haja
}
}
// --- Core Functions ---
inline void JA_Update() {
// --- Outgoing music fade-out (crossfade o fade-out a silencio) ---
if (outgoing_music.stream && outgoing_music.fade.active) {
Uint64 now = SDL_GetTicks();
Uint64 elapsed = now - outgoing_music.fade.start_time;
if (elapsed >= (Uint64)outgoing_music.fade.duration_ms) {
SDL_DestroyAudioStream(outgoing_music.stream);
outgoing_music.stream = nullptr;
outgoing_music.fade.active = false;
} else {
float percent = (float)elapsed / (float)outgoing_music.fade.duration_ms;
SDL_SetAudioStreamGain(outgoing_music.stream, outgoing_music.fade.initial_volume * (1.0f - percent));
}
}
// --- Current music ---
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;
// Fade-in (parte de un crossfade)
if (incoming_fade.active) {
Uint64 now = SDL_GetTicks();
Uint64 elapsed = now - incoming_fade.start_time;
if (elapsed >= (Uint64)incoming_fade.duration_ms) {
incoming_fade.active = false;
SDL_SetAudioStreamGain(current_music->stream, JA_musicVolume);
} 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));
float percent = (float)elapsed / (float)incoming_fade.duration_ms;
SDL_SetAudioStreamGain(current_music->stream, JA_musicVolume * percent);
}
}
@@ -156,12 +219,13 @@ inline void JA_Update() {
}
}
// --- Sound channels ---
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);
SDL_PutAudioStreamData(channels[i].stream, channels[i].sound->buffer.get(), channels[i].sound->length);
if (channels[i].times > 0) channels[i].times--;
}
} else {
@@ -172,20 +236,20 @@ inline void JA_Update() {
}
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
if (sdlAudioDevice) SDL_CloseAudioDevice(sdlAudioDevice);
sdlAudioDevice = SDL_OpenAudioDevice(SDL_AUDIO_DEVICE_DEFAULT_PLAYBACK, &JA_audioSpec);
if (sdlAudioDevice == 0) SDL_Log("Failed to initialize SDL audio!");
if (sdlAudioDevice == 0) std::cout << "Failed to initialize SDL audio!" << '\n';
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
if (outgoing_music.stream) {
SDL_DestroyAudioStream(outgoing_music.stream);
outgoing_music.stream = nullptr;
}
if (sdlAudioDevice) SDL_CloseAudioDevice(sdlAudioDevice);
sdlAudioDevice = 0;
}
@@ -194,26 +258,25 @@ inline void JA_Quit() {
inline JA_Music_t* JA_LoadMusic(const Uint8* buffer, Uint32 length) {
if (!buffer || length == 0) return nullptr;
// Còpia del OGG comprimit: stb_vorbis llig de forma persistent aquesta
// memòria mentre el handle estiga viu, així que hem de posseir-la nosaltres.
Uint8* ogg_copy = static_cast<Uint8*>(SDL_malloc(length));
if (!ogg_copy) return nullptr;
SDL_memcpy(ogg_copy, buffer, length);
// Allocem el JA_Music_t primer per aprofitar el seu `std::vector<Uint8>`
// com a propietari del OGG comprimit. stb_vorbis guarda un punter
// persistent al buffer; com que ací no el resize'jem, el .data() és
// estable durant tot el cicle de vida del music.
auto* music = new JA_Music_t();
music->ogg_data.assign(buffer, buffer + length);
int error = 0;
stb_vorbis* vorbis = stb_vorbis_open_memory(ogg_copy, static_cast<int>(length), &error, nullptr);
if (!vorbis) {
SDL_free(ogg_copy);
SDL_Log("JA_LoadMusic: stb_vorbis_open_memory failed (error %d)", error);
music->vorbis = stb_vorbis_open_memory(music->ogg_data.data(),
static_cast<int>(length),
&error,
nullptr);
if (!music->vorbis) {
std::cout << "JA_LoadMusic: stb_vorbis_open_memory failed (error " << error << ")" << '\n';
delete music;
return nullptr;
}
auto* music = new JA_Music_t();
music->ogg_data = ogg_copy;
music->ogg_length = length;
music->vorbis = vorbis;
const stb_vorbis_info info = stb_vorbis_get_info(vorbis);
const stb_vorbis_info info = stb_vorbis_get_info(music->vorbis);
music->spec.channels = info.channels;
music->spec.freq = static_cast<int>(info.sample_rate);
music->spec.format = SDL_AUDIO_S16;
@@ -222,31 +285,36 @@ inline JA_Music_t* JA_LoadMusic(const Uint8* buffer, Uint32 length) {
return music;
}
// Overload amb filename — els callers l'usen per poder comparar la música
// en curs amb JA_GetMusicFilename() i no rearrancar-la si ja és la mateixa.
inline JA_Music_t* JA_LoadMusic(Uint8* buffer, Uint32 length, const char* filename) {
JA_Music_t* music = JA_LoadMusic(static_cast<const Uint8*>(buffer), length);
if (music && filename) music->filename = filename;
return music;
}
inline JA_Music_t* JA_LoadMusic(const char* filename) {
// [RZC 28/08/22] Carreguem primer el arxiu en memòria i després el descomprimim. Es algo més rapid.
// Carreguem primer el arxiu en memòria i després el descomprimim.
FILE* f = fopen(filename, "rb");
if (!f) return NULL; // Añadida comprobación de apertura
if (!f) return nullptr;
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
if (!buffer) {
fclose(f);
return NULL;
return nullptr;
}
if (fread(buffer, fsize, 1, f) != 1) {
fclose(f);
free(buffer);
return NULL;
return nullptr;
}
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);
}
JA_Music_t* music = JA_LoadMusic(static_cast<const Uint8*>(buffer), static_cast<Uint32>(fsize));
if (music) {
music->filename = filename;
}
free(buffer);
@@ -269,7 +337,7 @@ inline void JA_PlayMusic(JA_Music_t* music, const int loop = -1) {
current_music->stream = SDL_CreateAudioStream(&current_music->spec, &JA_audioSpec);
if (!current_music->stream) {
SDL_Log("Failed to create audio stream!");
std::cout << "Failed to create audio stream!" << '\n';
current_music->state = JA_MUSIC_STOPPED;
return;
}
@@ -278,18 +346,20 @@ inline void JA_PlayMusic(JA_Music_t* music, const int loop = -1) {
// Pre-cargem el buffer abans de bindejar per evitar un underrun inicial.
JA_PumpMusic(current_music);
if (!SDL_BindAudioStream(sdlAudioDevice, current_music->stream)) printf("[ERROR] SDL_BindAudioStream failed!\n");
if (!SDL_BindAudioStream(sdlAudioDevice, current_music->stream)) {
std::cout << "[ERROR] SDL_BindAudioStream failed!" << '\n';
}
}
inline char* JA_GetMusicFilename(const JA_Music_t* music = nullptr) {
inline const char* JA_GetMusicFilename(const JA_Music_t* music = nullptr) {
if (!music) music = current_music;
if (!music) return nullptr; // Añadida comprobación
return music->filename;
if (!music || music->filename.empty()) return nullptr;
return music->filename.c_str();
}
inline void JA_PauseMusic() {
if (!JA_musicEnabled) return;
if (!current_music || current_music->state != JA_MUSIC_PLAYING) return; // Comprobación mejorada
if (!current_music || current_music->state != JA_MUSIC_PLAYING) return;
current_music->state = JA_MUSIC_PAUSED;
SDL_UnbindAudioStream(current_music->stream);
@@ -297,13 +367,21 @@ inline void JA_PauseMusic() {
inline void JA_ResumeMusic() {
if (!JA_musicEnabled) return;
if (!current_music || current_music->state != JA_MUSIC_PAUSED) return; // Comprobación mejorada
if (!current_music || current_music->state != JA_MUSIC_PAUSED) return;
current_music->state = JA_MUSIC_PLAYING;
SDL_BindAudioStream(sdlAudioDevice, current_music->stream);
}
inline void JA_StopMusic() {
// Limpiar outgoing crossfade si existe
if (outgoing_music.stream) {
SDL_DestroyAudioStream(outgoing_music.stream);
outgoing_music.stream = nullptr;
outgoing_music.fade.active = false;
}
incoming_fade.active = false;
if (!current_music || current_music->state == JA_MUSIC_INVALID || current_music->state == JA_MUSIC_STOPPED) return;
current_music->state = JA_MUSIC_STOPPED;
@@ -316,17 +394,73 @@ inline void JA_StopMusic() {
if (current_music->vorbis) {
stb_vorbis_seek_start(current_music->vorbis);
}
// No liberem filename aquí; es fa 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;
if (!current_music || current_music->state != JA_MUSIC_PLAYING) return;
fading = true;
fade_start_time = SDL_GetTicks();
fade_duration = milliseconds;
fade_initial_volume = JA_musicVolume;
// Destruir outgoing anterior si existe
if (outgoing_music.stream) {
SDL_DestroyAudioStream(outgoing_music.stream);
outgoing_music.stream = nullptr;
}
// Pre-omplim l'stream amb `milliseconds` de so: un cop robat, ja no
// tindrà accés al vorbis decoder i només podrà drenar el que tinga.
JA_PreFillOutgoing(current_music, milliseconds);
// Robar el stream del current_music al outgoing
outgoing_music.stream = current_music->stream;
outgoing_music.fade = {true, SDL_GetTicks(), milliseconds, JA_musicVolume};
// Dejar current_music sin stream (ya lo tiene outgoing)
current_music->stream = nullptr;
current_music->state = JA_MUSIC_STOPPED;
if (current_music->vorbis) stb_vorbis_seek_start(current_music->vorbis);
incoming_fade.active = false;
}
inline void JA_CrossfadeMusic(JA_Music_t* music, const int crossfade_ms, const int loop) {
if (!JA_musicEnabled || !music || !music->vorbis) return;
// Destruir outgoing anterior si existe (crossfade durante crossfade)
if (outgoing_music.stream) {
SDL_DestroyAudioStream(outgoing_music.stream);
outgoing_music.stream = nullptr;
outgoing_music.fade.active = false;
}
// Robar el stream de la musica actual al outgoing para el fade-out.
// Pre-omplim amb `crossfade_ms` de so perquè no es quede en silenci
// abans d'acabar el fade (l'stream robat ja no pot alimentar-se).
if (current_music && current_music->state == JA_MUSIC_PLAYING && current_music->stream) {
JA_PreFillOutgoing(current_music, crossfade_ms);
outgoing_music.stream = current_music->stream;
outgoing_music.fade = {true, SDL_GetTicks(), crossfade_ms, JA_musicVolume};
current_music->stream = nullptr;
current_music->state = JA_MUSIC_STOPPED;
if (current_music->vorbis) stb_vorbis_seek_start(current_music->vorbis);
}
// Iniciar la nueva pista con gain=0 (el fade-in la sube gradualmente)
current_music = music;
current_music->state = JA_MUSIC_PLAYING;
current_music->times = loop;
stb_vorbis_seek_start(current_music->vorbis);
current_music->stream = SDL_CreateAudioStream(&current_music->spec, &JA_audioSpec);
if (!current_music->stream) {
std::cout << "Failed to create audio stream for crossfade!" << '\n';
current_music->state = JA_MUSIC_STOPPED;
return;
}
SDL_SetAudioStreamGain(current_music->stream, 0.0f);
JA_PumpMusic(current_music); // pre-carrega abans de bindejar
SDL_BindAudioStream(sdlAudioDevice, current_music->stream);
// Configurar fade-in
incoming_fade = {true, SDL_GetTicks(), crossfade_ms, 0.0f};
}
inline JA_Music_state JA_GetMusicState() {
@@ -344,8 +478,8 @@ inline void JA_DeleteMusic(JA_Music_t* music) {
}
if (music->stream) SDL_DestroyAudioStream(music->stream);
if (music->vorbis) stb_vorbis_close(music->vorbis);
SDL_free(music->ogg_data);
free(music->filename); // filename es libera aquí
// ogg_data (std::vector) i filename (std::string) s'alliberen sols
// al destructor de JA_Music_t.
delete music;
}
@@ -358,49 +492,40 @@ inline float JA_SetMusicVolume(float volume) {
}
inline void JA_SetMusicPosition(float /*value*/) {
// No implementat amb el backend de streaming. Mai va arribar a usar-se
// en el codi existent, així que es manté com a stub.
// No implementat amb el backend de streaming.
}
inline float JA_GetMusicPosition() {
// Veure nota a JA_SetMusicPosition.
return 0.0f;
}
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;
auto sound = std::make_unique<JA_Sound_t>();
Uint8* raw = nullptr;
if (!SDL_LoadWAV_IO(SDL_IOFromMem(buffer, size), 1, &sound->spec, &raw, &sound->length)) {
std::cout << "Failed to load WAV from memory: " << SDL_GetError() << '\n';
return nullptr;
}
return sound;
sound->buffer.reset(raw); // adopta el SDL_malloc'd buffer
return sound.release();
}
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;
auto sound = std::make_unique<JA_Sound_t>();
Uint8* raw = nullptr;
if (!SDL_LoadWAV(filename, &sound->spec, &raw, &sound->length)) {
std::cout << "Failed to load WAV file: " << SDL_GetError() << '\n';
return nullptr;
}
return sound;
sound->buffer.reset(raw); // adopta el SDL_malloc'd buffer
return sound.release();
}
inline int JA_PlaySound(JA_Sound_t* sound, const int loop = 0, const int group = 0) {
@@ -420,22 +545,22 @@ inline int JA_PlaySoundOnChannel(JA_Sound_t* sound, const int channel, const int
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
JA_StopChannel(channel);
channels[channel].sound = sound;
channels[channel].times = loop;
channels[channel].pos = 0;
channels[channel].group = group; // Asignar grupo
channels[channel].group = group;
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!");
std::cout << "Failed to create audio stream for sound!" << '\n';
channels[channel].state = JA_CHANNEL_FREE;
return -1;
}
SDL_PutAudioStreamData(channels[channel].stream, channels[channel].sound->buffer, channels[channel].sound->length);
SDL_PutAudioStreamData(channels[channel].stream, channels[channel].sound->buffer.get(), channels[channel].sound->length);
SDL_SetAudioStreamGain(channels[channel].stream, JA_soundVolume[group]);
SDL_BindAudioStream(sdlAudioDevice, channels[channel].stream);
@@ -447,7 +572,7 @@ inline void JA_DeleteSound(JA_Sound_t* sound) {
for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; i++) {
if (channels[i].sound == sound) JA_StopChannel(i);
}
SDL_free(sound->buffer);
// buffer es destrueix automàticament via RAII (SDLFreeDeleter).
delete sound;
}
@@ -493,7 +618,7 @@ inline void JA_StopChannel(const int channel) {
channels[i].stream = nullptr;
channels[i].state = JA_CHANNEL_FREE;
channels[i].pos = 0;
channels[i].sound = NULL;
channels[i].sound = nullptr;
}
}
} else if (channel >= 0 && channel < JA_MAX_SIMULTANEOUS_CHANNELS) {
@@ -502,7 +627,7 @@ inline void JA_StopChannel(const int channel) {
channels[channel].stream = nullptr;
channels[channel].state = JA_CHANNEL_FREE;
channels[channel].pos = 0;
channels[channel].sound = NULL;
channels[channel].sound = nullptr;
}
}
}
@@ -514,8 +639,7 @@ inline JA_Channel_state JA_GetChannelState(const int channel) {
return channels[channel].state;
}
inline float JA_SetSoundVolume(float volume, const int group = -1) // -1 para todos los grupos
{
inline float JA_SetSoundVolume(float volume, const int group = -1) {
const float v = SDL_clamp(volume, 0.0f, 1.0f);
if (group == -1) {
@@ -525,10 +649,10 @@ inline float JA_SetSoundVolume(float volume, const int group = -1) // -1 para t
} else if (group >= 0 && group < JA_MAX_GROUPS) {
JA_soundVolume[group] = v;
} else {
return v; // Grupo inválido
return v;
}
// Aplicar volumen a canales activos
// Aplicar volum als canals actius.
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) {
@@ -543,13 +667,13 @@ inline float JA_SetSoundVolume(float volume, const int group = -1) // -1 para t
inline void JA_EnableSound(const bool value) {
if (!value) {
JA_StopChannel(-1); // Detener todos los canales
JA_StopChannel(-1);
}
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
JA_SetSoundVolume(v, -1);
return v;
}
}

117
source/external/stb_vorbis.h → source/external/stb_vorbis.c vendored
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;