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Diffstat (limited to 'deps/raylib/src/raudio.c')
| -rw-r--r-- | deps/raylib/src/raudio.c | 2879 |
1 files changed, 2879 insertions, 0 deletions
diff --git a/deps/raylib/src/raudio.c b/deps/raylib/src/raudio.c new file mode 100644 index 0000000..15859a6 --- /dev/null +++ b/deps/raylib/src/raudio.c @@ -0,0 +1,2879 @@ +/********************************************************************************************** +* +* raudio v1.1 - A simple and easy-to-use audio library based on miniaudio +* +* FEATURES: +* - Manage audio device (init/close) +* - Manage raw audio context +* - Manage mixing channels +* - Load and unload audio files +* - Format wave data (sample rate, size, channels) +* - Play/Stop/Pause/Resume loaded audio +* +* CONFIGURATION: +* #define SUPPORT_MODULE_RAUDIO +* raudio module is included in the build +* +* #define RAUDIO_STANDALONE +* Define to use the module as standalone library (independently of raylib). +* Required types and functions are defined in the same module. +* +* #define SUPPORT_FILEFORMAT_WAV +* #define SUPPORT_FILEFORMAT_OGG +* #define SUPPORT_FILEFORMAT_MP3 +* #define SUPPORT_FILEFORMAT_QOA +* #define SUPPORT_FILEFORMAT_FLAC +* #define SUPPORT_FILEFORMAT_XM +* #define SUPPORT_FILEFORMAT_MOD +* Selected desired fileformats to be supported for loading. Some of those formats are +* supported by default, to remove support, just comment unrequired #define in this module +* +* DEPENDENCIES: +* miniaudio.h - Audio device management lib (https://github.com/mackron/miniaudio) +* stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/) +* dr_wav.h - WAV audio files loading (http://github.com/mackron/dr_libs) +* dr_mp3.h - MP3 audio file loading (https://github.com/mackron/dr_libs) +* dr_flac.h - FLAC audio file loading (https://github.com/mackron/dr_libs) +* jar_xm.h - XM module file loading +* jar_mod.h - MOD audio file loading +* +* CONTRIBUTORS: +* David Reid (github: @mackron) (Nov. 2017): +* - Complete port to miniaudio library +* +* Joshua Reisenauer (github: @kd7tck) (2015): +* - XM audio module support (jar_xm) +* - MOD audio module support (jar_mod) +* - Mixing channels support +* - Raw audio context support +* +* +* LICENSE: zlib/libpng +* +* Copyright (c) 2013-2024 Ramon Santamaria (@raysan5) +* +* This software is provided "as-is", without any express or implied warranty. In no event +* will the authors be held liable for any damages arising from the use of this software. +* +* Permission is granted to anyone to use this software for any purpose, including commercial +* applications, and to alter it and redistribute it freely, subject to the following restrictions: +* +* 1. The origin of this software must not be misrepresented; you must not claim that you +* wrote the original software. If you use this software in a product, an acknowledgment +* in the product documentation would be appreciated but is not required. +* +* 2. Altered source versions must be plainly marked as such, and must not be misrepresented +* as being the original software. +* +* 3. This notice may not be removed or altered from any source distribution. +* +**********************************************************************************************/ + +#if defined(RAUDIO_STANDALONE) + #include "raudio.h" +#else + #include "raylib.h" // Declares module functions + + // Check if config flags have been externally provided on compilation line + #if !defined(EXTERNAL_CONFIG_FLAGS) + #include "config.h" // Defines module configuration flags + #endif + #include "utils.h" // Required for: fopen() Android mapping +#endif + +#if defined(SUPPORT_MODULE_RAUDIO) + +#if defined(_WIN32) +// To avoid conflicting windows.h symbols with raylib, some flags are defined +// WARNING: Those flags avoid inclusion of some Win32 headers that could be required +// by user at some point and won't be included... +//------------------------------------------------------------------------------------- + +// If defined, the following flags inhibit definition of the indicated items. +#define NOGDICAPMASKS // CC_*, LC_*, PC_*, CP_*, TC_*, RC_ +#define NOVIRTUALKEYCODES // VK_* +#define NOWINMESSAGES // WM_*, EM_*, LB_*, CB_* +#define NOWINSTYLES // WS_*, CS_*, ES_*, LBS_*, SBS_*, CBS_* +#define NOSYSMETRICS // SM_* +#define NOMENUS // MF_* +#define NOICONS // IDI_* +#define NOKEYSTATES // MK_* +#define NOSYSCOMMANDS // SC_* +#define NORASTEROPS // Binary and Tertiary raster ops +#define NOSHOWWINDOW // SW_* +#define OEMRESOURCE // OEM Resource values +#define NOATOM // Atom Manager routines +#define NOCLIPBOARD // Clipboard routines +#define NOCOLOR // Screen colors +#define NOCTLMGR // Control and Dialog routines +#define NODRAWTEXT // DrawText() and DT_* +#define NOGDI // All GDI defines and routines +#define NOKERNEL // All KERNEL defines and routines +#define NOUSER // All USER defines and routines +//#define NONLS // All NLS defines and routines +#define NOMB // MB_* and MessageBox() +#define NOMEMMGR // GMEM_*, LMEM_*, GHND, LHND, associated routines +#define NOMETAFILE // typedef METAFILEPICT +#define NOMINMAX // Macros min(a,b) and max(a,b) +#define NOMSG // typedef MSG and associated routines +#define NOOPENFILE // OpenFile(), OemToAnsi, AnsiToOem, and OF_* +#define NOSCROLL // SB_* and scrolling routines +#define NOSERVICE // All Service Controller routines, SERVICE_ equates, etc. +#define NOSOUND // Sound driver routines +#define NOTEXTMETRIC // typedef TEXTMETRIC and associated routines +#define NOWH // SetWindowsHook and WH_* +#define NOWINOFFSETS // GWL_*, GCL_*, associated routines +#define NOCOMM // COMM driver routines +#define NOKANJI // Kanji support stuff. +#define NOHELP // Help engine interface. +#define NOPROFILER // Profiler interface. +#define NODEFERWINDOWPOS // DeferWindowPos routines +#define NOMCX // Modem Configuration Extensions + +// Type required before windows.h inclusion +typedef struct tagMSG *LPMSG; + +#include <windows.h> // Windows functionality (miniaudio) + +// Type required by some unused function... +typedef struct tagBITMAPINFOHEADER { + DWORD biSize; + LONG biWidth; + LONG biHeight; + WORD biPlanes; + WORD biBitCount; + DWORD biCompression; + DWORD biSizeImage; + LONG biXPelsPerMeter; + LONG biYPelsPerMeter; + DWORD biClrUsed; + DWORD biClrImportant; +} BITMAPINFOHEADER, *PBITMAPINFOHEADER; + +#include <objbase.h> // Component Object Model (COM) header +#include <mmreg.h> // Windows Multimedia, defines some WAVE structs +#include <mmsystem.h> // Windows Multimedia, used by Windows GDI, defines DIBINDEX macro + +// Some required types defined for MSVC/TinyC compiler +#if defined(_MSC_VER) || defined(__TINYC__) + #include "propidl.h" +#endif +#endif + +#define MA_MALLOC RL_MALLOC +#define MA_FREE RL_FREE + +#define MA_NO_JACK +#define MA_NO_WAV +#define MA_NO_FLAC +#define MA_NO_MP3 +#define MA_NO_RESOURCE_MANAGER +#define MA_NO_NODE_GRAPH +#define MA_NO_ENGINE +#define MA_NO_GENERATION + +// Threading model: Default: [0] COINIT_MULTITHREADED: COM calls objects on any thread (free threading) +#define MA_COINIT_VALUE 2 // [2] COINIT_APARTMENTTHREADED: Each object has its own thread (apartment model) + +#define MINIAUDIO_IMPLEMENTATION +//#define MA_DEBUG_OUTPUT +#include "external/miniaudio.h" // Audio device initialization and management +#undef PlaySound // Win32 API: windows.h > mmsystem.h defines PlaySound macro + +#include <stdlib.h> // Required for: malloc(), free() +#include <stdio.h> // Required for: FILE, fopen(), fclose(), fread() +#include <string.h> // Required for: strcmp() [Used in IsFileExtension(), LoadWaveFromMemory(), LoadMusicStreamFromMemory()] + +#if defined(RAUDIO_STANDALONE) + #ifndef TRACELOG + #define TRACELOG(level, ...) printf(__VA_ARGS__) + #endif + + // Allow custom memory allocators + #ifndef RL_MALLOC + #define RL_MALLOC(sz) malloc(sz) + #endif + #ifndef RL_CALLOC + #define RL_CALLOC(n,sz) calloc(n,sz) + #endif + #ifndef RL_REALLOC + #define RL_REALLOC(ptr,sz) realloc(ptr,sz) + #endif + #ifndef RL_FREE + #define RL_FREE(ptr) free(ptr) + #endif +#endif + +#if defined(SUPPORT_FILEFORMAT_WAV) + #define DRWAV_MALLOC RL_MALLOC + #define DRWAV_REALLOC RL_REALLOC + #define DRWAV_FREE RL_FREE + + #define DR_WAV_IMPLEMENTATION + #include "external/dr_wav.h" // WAV loading functions +#endif + +#if defined(SUPPORT_FILEFORMAT_OGG) + // TODO: Remap stb_vorbis malloc()/free() calls to RL_MALLOC/RL_FREE + #include "external/stb_vorbis.c" // OGG loading functions +#endif + +#if defined(SUPPORT_FILEFORMAT_MP3) + #define DRMP3_MALLOC RL_MALLOC + #define DRMP3_REALLOC RL_REALLOC + #define DRMP3_FREE RL_FREE + + #define DR_MP3_IMPLEMENTATION + #include "external/dr_mp3.h" // MP3 loading functions +#endif + +#if defined(SUPPORT_FILEFORMAT_QOA) + #define QOA_MALLOC RL_MALLOC + #define QOA_FREE RL_FREE + + #if defined(_MSC_VER) // Disable some MSVC warning + #pragma warning(push) + #pragma warning(disable : 4018) + #pragma warning(disable : 4267) + #pragma warning(disable : 4244) + #endif + + #define QOA_IMPLEMENTATION + #include "external/qoa.h" // QOA loading and saving functions + #include "external/qoaplay.c" // QOA stream playing helper functions + + #if defined(_MSC_VER) + #pragma warning(pop) // Disable MSVC warning suppression + #endif +#endif + +#if defined(SUPPORT_FILEFORMAT_FLAC) + #define DRFLAC_MALLOC RL_MALLOC + #define DRFLAC_REALLOC RL_REALLOC + #define DRFLAC_FREE RL_FREE + + #define DR_FLAC_IMPLEMENTATION + #define DR_FLAC_NO_WIN32_IO + #include "external/dr_flac.h" // FLAC loading functions +#endif + +#if defined(SUPPORT_FILEFORMAT_XM) + #define JARXM_MALLOC RL_MALLOC + #define JARXM_FREE RL_FREE + + #if defined(_MSC_VER) // Disable some MSVC warning + #pragma warning(push) + #pragma warning(disable : 4244) + #endif + + #define JAR_XM_IMPLEMENTATION + #include "external/jar_xm.h" // XM loading functions + + #if defined(_MSC_VER) + #pragma warning(pop) // Disable MSVC warning suppression + #endif +#endif + +#if defined(SUPPORT_FILEFORMAT_MOD) + #define JARMOD_MALLOC RL_MALLOC + #define JARMOD_FREE RL_FREE + + #define JAR_MOD_IMPLEMENTATION + #include "external/jar_mod.h" // MOD loading functions +#endif + +//---------------------------------------------------------------------------------- +// Defines and Macros +//---------------------------------------------------------------------------------- +#ifndef AUDIO_DEVICE_FORMAT + #define AUDIO_DEVICE_FORMAT ma_format_f32 // Device output format (float-32bit) +#endif +#ifndef AUDIO_DEVICE_CHANNELS + #define AUDIO_DEVICE_CHANNELS 2 // Device output channels: stereo +#endif +#ifndef AUDIO_DEVICE_SAMPLE_RATE + #define AUDIO_DEVICE_SAMPLE_RATE 0 // Device output sample rate +#endif + +#ifndef MAX_AUDIO_BUFFER_POOL_CHANNELS + #define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 // Audio pool channels +#endif + +//---------------------------------------------------------------------------------- +// Types and Structures Definition +//---------------------------------------------------------------------------------- +#if defined(RAUDIO_STANDALONE) +// Trace log level +// NOTE: Organized by priority level +typedef enum { + LOG_ALL = 0, // Display all logs + LOG_TRACE, // Trace logging, intended for internal use only + LOG_DEBUG, // Debug logging, used for internal debugging, it should be disabled on release builds + LOG_INFO, // Info logging, used for program execution info + LOG_WARNING, // Warning logging, used on recoverable failures + LOG_ERROR, // Error logging, used on unrecoverable failures + LOG_FATAL, // Fatal logging, used to abort program: exit(EXIT_FAILURE) + LOG_NONE // Disable logging +} TraceLogLevel; +#endif + +// Music context type +// NOTE: Depends on data structure provided by the library +// in charge of reading the different file types +typedef enum { + MUSIC_AUDIO_NONE = 0, // No audio context loaded + MUSIC_AUDIO_WAV, // WAV audio context + MUSIC_AUDIO_OGG, // OGG audio context + MUSIC_AUDIO_FLAC, // FLAC audio context + MUSIC_AUDIO_MP3, // MP3 audio context + MUSIC_AUDIO_QOA, // QOA audio context + MUSIC_MODULE_XM, // XM module audio context + MUSIC_MODULE_MOD // MOD module audio context +} MusicContextType; + +// NOTE: Different logic is used when feeding data to the playback device +// depending on whether data is streamed (Music vs Sound) +typedef enum { + AUDIO_BUFFER_USAGE_STATIC = 0, + AUDIO_BUFFER_USAGE_STREAM +} AudioBufferUsage; + +// Audio buffer struct +struct rAudioBuffer { + ma_data_converter converter; // Audio data converter + + AudioCallback callback; // Audio buffer callback for buffer filling on audio threads + rAudioProcessor *processor; // Audio processor + + float volume; // Audio buffer volume + float pitch; // Audio buffer pitch + float pan; // Audio buffer pan (0.0f to 1.0f) + + bool playing; // Audio buffer state: AUDIO_PLAYING + bool paused; // Audio buffer state: AUDIO_PAUSED + bool looping; // Audio buffer looping, default to true for AudioStreams + int usage; // Audio buffer usage mode: STATIC or STREAM + + bool isSubBufferProcessed[2]; // SubBuffer processed (virtual double buffer) + unsigned int sizeInFrames; // Total buffer size in frames + unsigned int frameCursorPos; // Frame cursor position + unsigned int framesProcessed; // Total frames processed in this buffer (required for play timing) + + unsigned char *data; // Data buffer, on music stream keeps filling + + rAudioBuffer *next; // Next audio buffer on the list + rAudioBuffer *prev; // Previous audio buffer on the list +}; + +// Audio processor struct +// NOTE: Useful to apply effects to an AudioBuffer +struct rAudioProcessor { + AudioCallback process; // Processor callback function + rAudioProcessor *next; // Next audio processor on the list + rAudioProcessor *prev; // Previous audio processor on the list +}; + +#define AudioBuffer rAudioBuffer // HACK: To avoid CoreAudio (macOS) symbol collision + +// Audio data context +typedef struct AudioData { + struct { + ma_context context; // miniaudio context data + ma_device device; // miniaudio device + ma_mutex lock; // miniaudio mutex lock + bool isReady; // Check if audio device is ready + size_t pcmBufferSize; // Pre-allocated buffer size + void *pcmBuffer; // Pre-allocated buffer to read audio data from file/memory + } System; + struct { + AudioBuffer *first; // Pointer to first AudioBuffer in the list + AudioBuffer *last; // Pointer to last AudioBuffer in the list + int defaultSize; // Default audio buffer size for audio streams + } Buffer; + rAudioProcessor *mixedProcessor; +} AudioData; + +//---------------------------------------------------------------------------------- +// Global Variables Definition +//---------------------------------------------------------------------------------- +static AudioData AUDIO = { // Global AUDIO context + + // NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number + // After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds and a + // standard double-buffering system, a 4096 samples buffer has been chosen, it should be enough + // In case of music-stalls, just increase this number + .Buffer.defaultSize = 0, + .mixedProcessor = NULL +}; + +//---------------------------------------------------------------------------------- +// Module specific Functions Declaration +//---------------------------------------------------------------------------------- +static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage); + +// Reads audio data from an AudioBuffer object in internal/device formats +static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount); +static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount); + +static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount); +static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer); + +static bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer); +static void StopAudioBufferInLockedState(AudioBuffer *buffer); +static void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount); + +#if defined(RAUDIO_STANDALONE) +static bool IsFileExtension(const char *fileName, const char *ext); // Check file extension +static const char *GetFileExtension(const char *fileName); // Get pointer to extension for a filename string (includes the dot: .png) +static const char *GetFileName(const char *filePath); // Get pointer to filename for a path string +static const char *GetFileNameWithoutExt(const char *filePath); // Get filename string without extension (uses static string) + +static unsigned char *LoadFileData(const char *fileName, int *dataSize); // Load file data as byte array (read) +static bool SaveFileData(const char *fileName, void *data, int dataSize); // Save data to file from byte array (write) +static bool SaveFileText(const char *fileName, char *text); // Save text data to file (write), string must be '\0' terminated +#endif + +//---------------------------------------------------------------------------------- +// AudioBuffer management functions declaration +// NOTE: Those functions are not exposed by raylib... for the moment +//---------------------------------------------------------------------------------- +AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage); +void UnloadAudioBuffer(AudioBuffer *buffer); + +bool IsAudioBufferPlaying(AudioBuffer *buffer); +void PlayAudioBuffer(AudioBuffer *buffer); +void StopAudioBuffer(AudioBuffer *buffer); +void PauseAudioBuffer(AudioBuffer *buffer); +void ResumeAudioBuffer(AudioBuffer *buffer); +void SetAudioBufferVolume(AudioBuffer *buffer, float volume); +void SetAudioBufferPitch(AudioBuffer *buffer, float pitch); +void SetAudioBufferPan(AudioBuffer *buffer, float pan); +void TrackAudioBuffer(AudioBuffer *buffer); +void UntrackAudioBuffer(AudioBuffer *buffer); + + +//---------------------------------------------------------------------------------- +// Module Functions Definition - Audio Device initialization and Closing +//---------------------------------------------------------------------------------- +// Initialize audio device +void InitAudioDevice(void) +{ + // Init audio context + ma_context_config ctxConfig = ma_context_config_init(); + ma_log_callback_init(OnLog, NULL); + + ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context); + if (result != MA_SUCCESS) + { + TRACELOG(LOG_WARNING, "AUDIO: Failed to initialize context"); + return; + } + + // Init audio device + // NOTE: Using the default device. Format is floating point because it simplifies mixing + ma_device_config config = ma_device_config_init(ma_device_type_playback); + config.playback.pDeviceID = NULL; // NULL for the default playback AUDIO.System.device + config.playback.format = AUDIO_DEVICE_FORMAT; + config.playback.channels = AUDIO_DEVICE_CHANNELS; + config.capture.pDeviceID = NULL; // NULL for the default capture AUDIO.System.device + config.capture.format = ma_format_s16; + config.capture.channels = 1; + config.sampleRate = AUDIO_DEVICE_SAMPLE_RATE; + config.dataCallback = OnSendAudioDataToDevice; + config.pUserData = NULL; + + result = ma_device_init(&AUDIO.System.context, &config, &AUDIO.System.device); + if (result != MA_SUCCESS) + { + TRACELOG(LOG_WARNING, "AUDIO: Failed to initialize playback device"); + ma_context_uninit(&AUDIO.System.context); + return; + } + + // Mixing happens on a separate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may + // want to look at something a bit smarter later on to keep everything real-time, if that's necessary + if (ma_mutex_init(&AUDIO.System.lock) != MA_SUCCESS) + { + TRACELOG(LOG_WARNING, "AUDIO: Failed to create mutex for mixing"); + ma_device_uninit(&AUDIO.System.device); + ma_context_uninit(&AUDIO.System.context); + return; + } + + // Keep the device running the whole time. May want to consider doing something a bit smarter and only have the device running + // while there's at least one sound being played + result = ma_device_start(&AUDIO.System.device); + if (result != MA_SUCCESS) + { + TRACELOG(LOG_WARNING, "AUDIO: Failed to start playback device"); + ma_device_uninit(&AUDIO.System.device); + ma_context_uninit(&AUDIO.System.context); + return; + } + + TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully"); + TRACELOG(LOG_INFO, " > Backend: miniaudio | %s", ma_get_backend_name(AUDIO.System.context.backend)); + TRACELOG(LOG_INFO, " > Format: %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat)); + TRACELOG(LOG_INFO, " > Channels: %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels); + TRACELOG(LOG_INFO, " > Sample rate: %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate); + TRACELOG(LOG_INFO, " > Periods size: %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods); + + AUDIO.System.isReady = true; +} + +// Close the audio device for all contexts +void CloseAudioDevice(void) +{ + if (AUDIO.System.isReady) + { + ma_mutex_uninit(&AUDIO.System.lock); + ma_device_uninit(&AUDIO.System.device); + ma_context_uninit(&AUDIO.System.context); + + AUDIO.System.isReady = false; + RL_FREE(AUDIO.System.pcmBuffer); + AUDIO.System.pcmBuffer = NULL; + AUDIO.System.pcmBufferSize = 0; + + TRACELOG(LOG_INFO, "AUDIO: Device closed successfully"); + } + else TRACELOG(LOG_WARNING, "AUDIO: Device could not be closed, not currently initialized"); +} + +// Check if device has been initialized successfully +bool IsAudioDeviceReady(void) +{ + return AUDIO.System.isReady; +} + +// Set master volume (listener) +void SetMasterVolume(float volume) +{ + ma_device_set_master_volume(&AUDIO.System.device, volume); +} + +// Get master volume (listener) +float GetMasterVolume(void) +{ + float volume = 0.0f; + ma_device_get_master_volume(&AUDIO.System.device, &volume); + return volume; +} + +//---------------------------------------------------------------------------------- +// Module Functions Definition - Audio Buffer management +//---------------------------------------------------------------------------------- + +// Initialize a new audio buffer (filled with silence) +AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage) +{ + AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer)); + + if (audioBuffer == NULL) + { + TRACELOG(LOG_WARNING, "AUDIO: Failed to allocate memory for buffer"); + return NULL; + } + + if (sizeInFrames > 0) audioBuffer->data = RL_CALLOC(sizeInFrames*channels*ma_get_bytes_per_sample(format), 1); + + // Audio data runs through a format converter + ma_data_converter_config converterConfig = ma_data_converter_config_init(format, AUDIO_DEVICE_FORMAT, channels, AUDIO_DEVICE_CHANNELS, sampleRate, AUDIO.System.device.sampleRate); + converterConfig.allowDynamicSampleRate = true; + + ma_result result = ma_data_converter_init(&converterConfig, NULL, &audioBuffer->converter); + + if (result != MA_SUCCESS) + { + TRACELOG(LOG_WARNING, "AUDIO: Failed to create data conversion pipeline"); + RL_FREE(audioBuffer); + return NULL; + } + + // Init audio buffer values + audioBuffer->volume = 1.0f; + audioBuffer->pitch = 1.0f; + audioBuffer->pan = 0.5f; + + audioBuffer->callback = NULL; + audioBuffer->processor = NULL; + + audioBuffer->playing = false; + audioBuffer->paused = false; + audioBuffer->looping = false; + + audioBuffer->usage = usage; + audioBuffer->frameCursorPos = 0; + audioBuffer->sizeInFrames = sizeInFrames; + + // Buffers should be marked as processed by default so that a call to + // UpdateAudioStream() immediately after initialization works correctly + audioBuffer->isSubBufferProcessed[0] = true; + audioBuffer->isSubBufferProcessed[1] = true; + + // Track audio buffer to linked list next position + TrackAudioBuffer(audioBuffer); + + return audioBuffer; +} + +// Delete an audio buffer +void UnloadAudioBuffer(AudioBuffer *buffer) +{ + if (buffer != NULL) + { + UntrackAudioBuffer(buffer); + ma_data_converter_uninit(&buffer->converter, NULL); + RL_FREE(buffer->data); + RL_FREE(buffer); + } +} + +// Check if an audio buffer is playing from a program state without lock +bool IsAudioBufferPlaying(AudioBuffer *buffer) +{ + bool result = false; + ma_mutex_lock(&AUDIO.System.lock); + result = IsAudioBufferPlayingInLockedState(buffer); + ma_mutex_unlock(&AUDIO.System.lock); + return result; +} + +// Play an audio buffer +// NOTE: Buffer is restarted to the start +// Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained +void PlayAudioBuffer(AudioBuffer *buffer) +{ + if (buffer != NULL) + { + ma_mutex_lock(&AUDIO.System.lock); + buffer->playing = true; + buffer->paused = false; + buffer->frameCursorPos = 0; + ma_mutex_unlock(&AUDIO.System.lock); + } +} + +// Stop an audio buffer from a program state without lock +void StopAudioBuffer(AudioBuffer *buffer) +{ + ma_mutex_lock(&AUDIO.System.lock); + StopAudioBufferInLockedState(buffer); + ma_mutex_unlock(&AUDIO.System.lock); +} + +// Pause an audio buffer +void PauseAudioBuffer(AudioBuffer *buffer) +{ + if (buffer != NULL) + { + ma_mutex_lock(&AUDIO.System.lock); + buffer->paused = true; + ma_mutex_unlock(&AUDIO.System.lock); + } +} + +// Resume an audio buffer +void ResumeAudioBuffer(AudioBuffer *buffer) +{ + if (buffer != NULL) + { + ma_mutex_lock(&AUDIO.System.lock); + buffer->paused = false; + ma_mutex_unlock(&AUDIO.System.lock); + } +} + +// Set volume for an audio buffer +void SetAudioBufferVolume(AudioBuffer *buffer, float volume) +{ + if (buffer != NULL) + { + ma_mutex_lock(&AUDIO.System.lock); + buffer->volume = volume; + ma_mutex_unlock(&AUDIO.System.lock); + } +} + +// Set pitch for an audio buffer +void SetAudioBufferPitch(AudioBuffer *buffer, float pitch) +{ + if ((buffer != NULL) && (pitch > 0.0f)) + { + ma_mutex_lock(&AUDIO.System.lock); + // Pitching is just an adjustment of the sample rate + // Note that this changes the duration of the sound: + // - higher pitches will make the sound faster + // - lower pitches make it slower + ma_uint32 outputSampleRate = (ma_uint32)((float)buffer->converter.sampleRateOut/pitch); + ma_data_converter_set_rate(&buffer->converter, buffer->converter.sampleRateIn, outputSampleRate); + + buffer->pitch = pitch; + ma_mutex_unlock(&AUDIO.System.lock); + } +} + +// Set pan for an audio buffer +void SetAudioBufferPan(AudioBuffer *buffer, float pan) +{ + if (pan < 0.0f) pan = 0.0f; + else if (pan > 1.0f) pan = 1.0f; + + if (buffer != NULL) + { + ma_mutex_lock(&AUDIO.System.lock); + buffer->pan = pan; + ma_mutex_unlock(&AUDIO.System.lock); + } +} + +// Track audio buffer to linked list next position +void TrackAudioBuffer(AudioBuffer *buffer) +{ + ma_mutex_lock(&AUDIO.System.lock); + { + if (AUDIO.Buffer.first == NULL) AUDIO.Buffer.first = buffer; + else + { + AUDIO.Buffer.last->next = buffer; + buffer->prev = AUDIO.Buffer.last; + } + + AUDIO.Buffer.last = buffer; + } + ma_mutex_unlock(&AUDIO.System.lock); +} + +// Untrack audio buffer from linked list +void UntrackAudioBuffer(AudioBuffer *buffer) +{ + ma_mutex_lock(&AUDIO.System.lock); + { + if (buffer->prev == NULL) AUDIO.Buffer.first = buffer->next; + else buffer->prev->next = buffer->next; + + if (buffer->next == NULL) AUDIO.Buffer.last = buffer->prev; + else buffer->next->prev = buffer->prev; + + buffer->prev = NULL; + buffer->next = NULL; + } + ma_mutex_unlock(&AUDIO.System.lock); +} + +//---------------------------------------------------------------------------------- +// Module Functions Definition - Sounds loading and playing (.WAV) +//---------------------------------------------------------------------------------- + +// Load wave data from file +Wave LoadWave(const char *fileName) +{ + Wave wave = { 0 }; + + // Loading file to memory + int dataSize = 0; + unsigned char *fileData = LoadFileData(fileName, &dataSize); + + // Loading wave from memory data + if (fileData != NULL) wave = LoadWaveFromMemory(GetFileExtension(fileName), fileData, dataSize); + + UnloadFileData(fileData); + + return wave; +} + +// Load wave from memory buffer, fileType refers to extension: i.e. ".wav" +// WARNING: File extension must be provided in lower-case +Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int dataSize) +{ + Wave wave = { 0 }; + + if (false) { } +#if defined(SUPPORT_FILEFORMAT_WAV) + else if ((strcmp(fileType, ".wav") == 0) || (strcmp(fileType, ".WAV") == 0)) + { + drwav wav = { 0 }; + bool success = drwav_init_memory(&wav, fileData, dataSize, NULL); + + if (success) + { + wave.frameCount = (unsigned int)wav.totalPCMFrameCount; + wave.sampleRate = wav.sampleRate; + wave.sampleSize = 16; + wave.channels = wav.channels; + wave.data = (short *)RL_MALLOC(wave.frameCount*wave.channels*sizeof(short)); + + // NOTE: We are forcing conversion to 16bit sample size on reading + drwav_read_pcm_frames_s16(&wav, wav.totalPCMFrameCount, wave.data); + } + else TRACELOG(LOG_WARNING, "WAVE: Failed to load WAV data"); + + drwav_uninit(&wav); + } +#endif +#if defined(SUPPORT_FILEFORMAT_OGG) + else if ((strcmp(fileType, ".ogg") == 0) || (strcmp(fileType, ".OGG") == 0)) + { + stb_vorbis *oggData = stb_vorbis_open_memory((unsigned char *)fileData, dataSize, NULL, NULL); + + if (oggData != NULL) + { + stb_vorbis_info info = stb_vorbis_get_info(oggData); + + wave.sampleRate = info.sample_rate; + wave.sampleSize = 16; // By default, ogg data is 16 bit per sample (short) + wave.channels = info.channels; + wave.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggData); // NOTE: It returns frames! + wave.data = (short *)RL_MALLOC(wave.frameCount*wave.channels*sizeof(short)); + + // NOTE: Get the number of samples to process (be careful! we ask for number of shorts, not bytes!) + stb_vorbis_get_samples_short_interleaved(oggData, info.channels, (short *)wave.data, wave.frameCount*wave.channels); + stb_vorbis_close(oggData); + } + else TRACELOG(LOG_WARNING, "WAVE: Failed to load OGG data"); + } +#endif +#if defined(SUPPORT_FILEFORMAT_MP3) + else if ((strcmp(fileType, ".mp3") == 0) || (strcmp(fileType, ".MP3") == 0)) + { + drmp3_config config = { 0 }; + unsigned long long int totalFrameCount = 0; + + // NOTE: We are forcing conversion to 32bit float sample size on reading + wave.data = drmp3_open_memory_and_read_pcm_frames_f32(fileData, dataSize, &config, &totalFrameCount, NULL); + wave.sampleSize = 32; + + if (wave.data != NULL) + { + wave.channels = config.channels; + wave.sampleRate = config.sampleRate; + wave.frameCount = (int)totalFrameCount; + } + else TRACELOG(LOG_WARNING, "WAVE: Failed to load MP3 data"); + + } +#endif +#if defined(SUPPORT_FILEFORMAT_QOA) + else if ((strcmp(fileType, ".qoa") == 0) || (strcmp(fileType, ".QOA") == 0)) + { + qoa_desc qoa = { 0 }; + + // NOTE: Returned sample data is always 16 bit? + wave.data = qoa_decode(fileData, dataSize, &qoa); + wave.sampleSize = 16; + + if (wave.data != NULL) + { + wave.channels = qoa.channels; + wave.sampleRate = qoa.samplerate; + wave.frameCount = qoa.samples; + } + else TRACELOG(LOG_WARNING, "WAVE: Failed to load QOA data"); + + } +#endif +#if defined(SUPPORT_FILEFORMAT_FLAC) + else if ((strcmp(fileType, ".flac") == 0) || (strcmp(fileType, ".FLAC") == 0)) + { + unsigned long long int totalFrameCount = 0; + + // NOTE: We are forcing conversion to 16bit sample size on reading + wave.data = drflac_open_memory_and_read_pcm_frames_s16(fileData, dataSize, &wave.channels, &wave.sampleRate, &totalFrameCount, NULL); + wave.sampleSize = 16; + + if (wave.data != NULL) wave.frameCount = (unsigned int)totalFrameCount; + else TRACELOG(LOG_WARNING, "WAVE: Failed to load FLAC data"); + } +#endif + else TRACELOG(LOG_WARNING, "WAVE: Data format not supported"); + + TRACELOG(LOG_INFO, "WAVE: Data loaded successfully (%i Hz, %i bit, %i channels)", wave.sampleRate, wave.sampleSize, wave.channels); + + return wave; +} + +// Checks if wave data is valid (data loaded and parameters) +bool IsWaveValid(Wave wave) +{ + bool result = false; + + if ((wave.data != NULL) && // Validate wave data available + (wave.frameCount > 0) && // Validate frame count + (wave.sampleRate > 0) && // Validate sample rate is supported + (wave.sampleSize > 0) && // Validate sample size is supported + (wave.channels > 0)) result = true; // Validate number of channels supported + + return result; +} + +// Load sound from file +// NOTE: The entire file is loaded to memory to be played (no-streaming) +Sound LoadSound(const char *fileName) +{ + Wave wave = LoadWave(fileName); + + Sound sound = LoadSoundFromWave(wave); + + UnloadWave(wave); // Sound is loaded, we can unload wave + + return sound; +} + +// Load sound from wave data +// NOTE: Wave data must be unallocated manually +Sound LoadSoundFromWave(Wave wave) +{ + Sound sound = { 0 }; + + if (wave.data != NULL) + { + // When using miniaudio we need to do our own mixing + // To simplify this we need convert the format of each sound to be consistent with + // the format used to open the playback AUDIO.System.device. We can do this two ways: + // + // 1) Convert the whole sound in one go at load time (here) + // 2) Convert the audio data in chunks at mixing time + // + // First option has been selected, format conversion is done on the loading stage + // The downside is that it uses more memory if the original sound is u8 or s16 + ma_format formatIn = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32)); + ma_uint32 frameCountIn = wave.frameCount; + + ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate); + if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed to get frame count for format conversion"); + + AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, frameCount, AUDIO_BUFFER_USAGE_STATIC); + if (audioBuffer == NULL) + { + TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer"); + return sound; // early return to avoid dereferencing the audioBuffer null pointer + } + + frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate); + if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed format conversion"); + + sound.frameCount = frameCount; + sound.stream.sampleRate = AUDIO.System.device.sampleRate; + sound.stream.sampleSize = 32; + sound.stream.channels = AUDIO_DEVICE_CHANNELS; + sound.stream.buffer = audioBuffer; + } + + return sound; +} + +// Clone sound from existing sound data, clone does not own wave data +// NOTE: Wave data must be unallocated manually and will be shared across all clones +Sound LoadSoundAlias(Sound source) +{ + Sound sound = { 0 }; + + if (source.stream.buffer->data != NULL) + { + AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, 0, AUDIO_BUFFER_USAGE_STATIC); + + if (audioBuffer == NULL) + { + TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer"); + return sound; // Early return to avoid dereferencing the audioBuffer null pointer + } + + audioBuffer->sizeInFrames = source.stream.buffer->sizeInFrames; + audioBuffer->volume = source.stream.buffer->volume; + audioBuffer->data = source.stream.buffer->data; + + sound.frameCount = source.frameCount; + sound.stream.sampleRate = AUDIO.System.device.sampleRate; + sound.stream.sampleSize = 32; + sound.stream.channels = AUDIO_DEVICE_CHANNELS; + sound.stream.buffer = audioBuffer; + } + + return sound; +} + + +// Checks if a sound is valid (data loaded and buffers initialized) +bool IsSoundValid(Sound sound) +{ + bool result = false; + + if ((sound.frameCount > 0) && // Validate frame count + (sound.stream.buffer != NULL) && // Validate stream buffer + (sound.stream.sampleRate > 0) && // Validate sample rate is supported + (sound.stream.sampleSize > 0) && // Validate sample size is supported + (sound.stream.channels > 0)) result = true; // Validate number of channels supported + + return result; +} + +// Unload wave data +void UnloadWave(Wave wave) +{ + RL_FREE(wave.data); + //TRACELOG(LOG_INFO, "WAVE: Unloaded wave data from RAM"); +} + +// Unload sound +void UnloadSound(Sound sound) +{ + UnloadAudioBuffer(sound.stream.buffer); + //TRACELOG(LOG_INFO, "SOUND: Unloaded sound data from RAM"); +} + +void UnloadSoundAlias(Sound alias) +{ + // Untrack and unload just the sound buffer, not the sample data, it is shared with the source for the alias + if (alias.stream.buffer != NULL) + { + UntrackAudioBuffer(alias.stream.buffer); + ma_data_converter_uninit(&alias.stream.buffer->converter, NULL); + RL_FREE(alias.stream.buffer); + } +} + +// Update sound buffer with new data +void UpdateSound(Sound sound, const void *data, int frameCount) +{ + if (sound.stream.buffer != NULL) + { + StopAudioBuffer(sound.stream.buffer); + + memcpy(sound.stream.buffer->data, data, frameCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.formatIn, sound.stream.buffer->converter.channelsIn)); + } +} + +// Export wave data to file +bool ExportWave(Wave wave, const char *fileName) +{ + bool success = false; + + if (false) { } +#if defined(SUPPORT_FILEFORMAT_WAV) + else if (IsFileExtension(fileName, ".wav")) + { + drwav wav = { 0 }; + drwav_data_format format = { 0 }; + format.container = drwav_container_riff; + if (wave.sampleSize == 32) format.format = DR_WAVE_FORMAT_IEEE_FLOAT; + else format.format = DR_WAVE_FORMAT_PCM; + format.channels = wave.channels; + format.sampleRate = wave.sampleRate; + format.bitsPerSample = wave.sampleSize; + + void *fileData = NULL; + size_t fileDataSize = 0; + success = drwav_init_memory_write(&wav, &fileData, &fileDataSize, &format, NULL); + if (success) success = (int)drwav_write_pcm_frames(&wav, wave.frameCount, wave.data); + drwav_result result = drwav_uninit(&wav); + + if (result == DRWAV_SUCCESS) success = SaveFileData(fileName, (unsigned char *)fileData, (unsigned int)fileDataSize); + + drwav_free(fileData, NULL); + } +#endif +#if defined(SUPPORT_FILEFORMAT_QOA) + else if (IsFileExtension(fileName, ".qoa")) + { + if (wave.sampleSize == 16) + { + qoa_desc qoa = { 0 }; + qoa.channels = wave.channels; + qoa.samplerate = wave.sampleRate; + qoa.samples = wave.frameCount; + + int bytesWritten = qoa_write(fileName, wave.data, &qoa); + if (bytesWritten > 0) success = true; + } + else TRACELOG(LOG_WARNING, "AUDIO: Wave data must be 16 bit per sample for QOA format export"); + } +#endif + else if (IsFileExtension(fileName, ".raw")) + { + // Export raw sample data (without header) + // NOTE: It's up to the user to track wave parameters + success = SaveFileData(fileName, wave.data, wave.frameCount*wave.channels*wave.sampleSize/8); + } + + if (success) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave data exported successfully", fileName); + else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave data", fileName); + + return success; +} + +// Export wave sample data to code (.h) +bool ExportWaveAsCode(Wave wave, const char *fileName) +{ + bool success = false; + +#ifndef TEXT_BYTES_PER_LINE + #define TEXT_BYTES_PER_LINE 20 +#endif + + int waveDataSize = wave.frameCount*wave.channels*wave.sampleSize/8; + + // NOTE: Text data buffer size is estimated considering wave data size in bytes + // and requiring 12 char bytes for every byte; the actual size varies, but + // the longest possible char being appended is "%.4ff,\n ", which is 12 bytes. + char *txtData = (char *)RL_CALLOC(waveDataSize*12 + 2000, sizeof(char)); + + int byteCount = 0; + byteCount += sprintf(txtData + byteCount, "\n//////////////////////////////////////////////////////////////////////////////////\n"); + byteCount += sprintf(txtData + byteCount, "// //\n"); + byteCount += sprintf(txtData + byteCount, "// WaveAsCode exporter v1.1 - Wave data exported as an array of bytes //\n"); + byteCount += sprintf(txtData + byteCount, "// //\n"); + byteCount += sprintf(txtData + byteCount, "// more info and bugs-report: github.com/raysan5/raylib //\n"); + byteCount += sprintf(txtData + byteCount, "// feedback and support: ray[at]raylib.com //\n"); + byteCount += sprintf(txtData + byteCount, "// //\n"); + byteCount += sprintf(txtData + byteCount, "// Copyright (c) 2018-2024 Ramon Santamaria (@raysan5) //\n"); + byteCount += sprintf(txtData + byteCount, "// //\n"); + byteCount += sprintf(txtData + byteCount, "//////////////////////////////////////////////////////////////////////////////////\n\n"); + + // Get file name from path and convert variable name to uppercase + char varFileName[256] = { 0 }; + strcpy(varFileName, GetFileNameWithoutExt(fileName)); + for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; } + + // Add wave information + byteCount += sprintf(txtData + byteCount, "// Wave data information\n"); + byteCount += sprintf(txtData + byteCount, "#define %s_FRAME_COUNT %u\n", varFileName, wave.frameCount); + byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_RATE %u\n", varFileName, wave.sampleRate); + byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_SIZE %u\n", varFileName, wave.sampleSize); + byteCount += sprintf(txtData + byteCount, "#define %s_CHANNELS %u\n\n", varFileName, wave.channels); + + // Write wave data as an array of values + // Wave data is exported as byte array for 8/16bit and float array for 32bit float data + // NOTE: Frame data exported is channel-interlaced: frame01[sampleChannel1, sampleChannel2, ...], frame02[], frame03[] + if (wave.sampleSize == 32) + { + byteCount += sprintf(txtData + byteCount, "static float %s_DATA[%i] = {\n", varFileName, waveDataSize/4); + for (int i = 1; i < waveDataSize/4; i++) byteCount += sprintf(txtData + byteCount, ((i%TEXT_BYTES_PER_LINE == 0)? "%.4ff,\n " : "%.4ff, "), ((float *)wave.data)[i - 1]); + byteCount += sprintf(txtData + byteCount, "%.4ff };\n", ((float *)wave.data)[waveDataSize/4 - 1]); + } + else + { + byteCount += sprintf(txtData + byteCount, "static unsigned char %s_DATA[%i] = { ", varFileName, waveDataSize); + for (int i = 1; i < waveDataSize; i++) byteCount += sprintf(txtData + byteCount, ((i%TEXT_BYTES_PER_LINE == 0)? "0x%x,\n " : "0x%x, "), ((unsigned char *)wave.data)[i - 1]); + byteCount += sprintf(txtData + byteCount, "0x%x };\n", ((unsigned char *)wave.data)[waveDataSize - 1]); + } + + // NOTE: Text data length exported is determined by '\0' (NULL) character + success = SaveFileText(fileName, txtData); + + RL_FREE(txtData); + + if (success != 0) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave as code exported successfully", fileName); + else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave as code", fileName); + + return success; +} + +// Play a sound +void PlaySound(Sound sound) +{ + PlayAudioBuffer(sound.stream.buffer); +} + +// Pause a sound +void PauseSound(Sound sound) +{ + PauseAudioBuffer(sound.stream.buffer); +} + +// Resume a paused sound +void ResumeSound(Sound sound) +{ + ResumeAudioBuffer(sound.stream.buffer); +} + +// Stop reproducing a sound +void StopSound(Sound sound) +{ + StopAudioBuffer(sound.stream.buffer); +} + +// Check if a sound is playing +bool IsSoundPlaying(Sound sound) +{ + bool result = false; + + if (IsAudioBufferPlaying(sound.stream.buffer)) result = true; + + return result; +} + +// Set volume for a sound +void SetSoundVolume(Sound sound, float volume) +{ + SetAudioBufferVolume(sound.stream.buffer, volume); +} + +// Set pitch for a sound +void SetSoundPitch(Sound sound, float pitch) +{ + SetAudioBufferPitch(sound.stream.buffer, pitch); +} + +// Set pan for a sound +void SetSoundPan(Sound sound, float pan) +{ + SetAudioBufferPan(sound.stream.buffer, pan); +} + +// Convert wave data to desired format +void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) +{ + ma_format formatIn = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32)); + ma_format formatOut = ((sampleSize == 8)? ma_format_u8 : ((sampleSize == 16)? ma_format_s16 : ma_format_f32)); + + ma_uint32 frameCountIn = wave->frameCount; + ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate); + + if (frameCount == 0) + { + TRACELOG(LOG_WARNING, "WAVE: Failed to get frame count for format conversion"); + return; + } + + void *data = RL_MALLOC(frameCount*channels*(sampleSize/8)); + + frameCount = (ma_uint32)ma_convert_frames(data, frameCount, formatOut, channels, sampleRate, wave->data, frameCountIn, formatIn, wave->channels, wave->sampleRate); + if (frameCount == 0) + { + TRACELOG(LOG_WARNING, "WAVE: Failed format conversion"); + return; + } + + wave->frameCount = frameCount; + wave->sampleSize = sampleSize; + wave->sampleRate = sampleRate; + wave->channels = channels; + + RL_FREE(wave->data); + wave->data = data; +} + +// Copy a wave to a new wave +Wave WaveCopy(Wave wave) +{ + Wave newWave = { 0 }; + + newWave.data = RL_MALLOC(wave.frameCount*wave.channels*wave.sampleSize/8); + + if (newWave.data != NULL) + { + // NOTE: Size must be provided in bytes + memcpy(newWave.data, wave.data, wave.frameCount*wave.channels*wave.sampleSize/8); + + newWave.frameCount = wave.frameCount; + newWave.sampleRate = wave.sampleRate; + newWave.sampleSize = wave.sampleSize; + newWave.channels = wave.channels; + } + + return newWave; +} + +// Crop a wave to defined frames range +// NOTE: Security check in case of out-of-range +void WaveCrop(Wave *wave, int initFrame, int finalFrame) +{ + if ((initFrame >= 0) && (initFrame < finalFrame) && ((unsigned int)finalFrame <= wave->frameCount)) + { + int frameCount = finalFrame - initFrame; + + void *data = RL_MALLOC(frameCount*wave->channels*wave->sampleSize/8); + + memcpy(data, (unsigned char *)wave->data + (initFrame*wave->channels*wave->sampleSize/8), frameCount*wave->channels*wave->sampleSize/8); + + RL_FREE(wave->data); + wave->data = data; + wave->frameCount = (unsigned int)frameCount; + } + else TRACELOG(LOG_WARNING, "WAVE: Crop range out of bounds"); +} + +// Load samples data from wave as a floats array +// NOTE 1: Returned sample values are normalized to range [-1..1] +// NOTE 2: Sample data allocated should be freed with UnloadWaveSamples() +float *LoadWaveSamples(Wave wave) +{ + float *samples = (float *)RL_MALLOC(wave.frameCount*wave.channels*sizeof(float)); + + // NOTE: sampleCount is the total number of interlaced samples (including channels) + + for (unsigned int i = 0; i < wave.frameCount*wave.channels; i++) + { + if (wave.sampleSize == 8) samples[i] = (float)(((unsigned char *)wave.data)[i] - 128)/128.0f; + else if (wave.sampleSize == 16) samples[i] = (float)(((short *)wave.data)[i])/32768.0f; + else if (wave.sampleSize == 32) samples[i] = ((float *)wave.data)[i]; + } + + return samples; +} + +// Unload samples data loaded with LoadWaveSamples() +void UnloadWaveSamples(float *samples) +{ + RL_FREE(samples); +} + +//---------------------------------------------------------------------------------- +// Module Functions Definition - Music loading and stream playing +//---------------------------------------------------------------------------------- + +// Load music stream from file +Music LoadMusicStream(const char *fileName) +{ + Music music = { 0 }; + bool musicLoaded = false; + + if (false) { } +#if defined(SUPPORT_FILEFORMAT_WAV) + else if (IsFileExtension(fileName, ".wav")) + { + drwav *ctxWav = RL_CALLOC(1, sizeof(drwav)); + bool success = drwav_init_file(ctxWav, fileName, NULL); + + if (success) + { + music.ctxType = MUSIC_AUDIO_WAV; + music.ctxData = ctxWav; + int sampleSize = ctxWav->bitsPerSample; + if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream() + + music.stream = LoadAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels); + music.frameCount = (unsigned int)ctxWav->totalPCMFrameCount; + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + else + { + RL_FREE(ctxWav); + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_OGG) + else if (IsFileExtension(fileName, ".ogg")) + { + // Open ogg audio stream + stb_vorbis *ctxOgg = stb_vorbis_open_filename(fileName, NULL, NULL); + + if (ctxOgg != NULL) + { + music.ctxType = MUSIC_AUDIO_OGG; + music.ctxData = ctxOgg; + stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData); // Get Ogg file info + + // OGG bit rate defaults to 16 bit, it's enough for compressed format + music.stream = LoadAudioStream(info.sample_rate, 16, info.channels); + + // WARNING: It seems this function returns length in frames, not samples, so we multiply by channels + music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData); + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + else + { + stb_vorbis_close(ctxOgg); + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_MP3) + else if (IsFileExtension(fileName, ".mp3")) + { + drmp3 *ctxMp3 = RL_CALLOC(1, sizeof(drmp3)); + int result = drmp3_init_file(ctxMp3, fileName, NULL); + + if (result > 0) + { + music.ctxType = MUSIC_AUDIO_MP3; + music.ctxData = ctxMp3; + music.stream = LoadAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels); + music.frameCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3); + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + else + { + RL_FREE(ctxMp3); + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_QOA) + else if (IsFileExtension(fileName, ".qoa")) + { + qoaplay_desc *ctxQoa = qoaplay_open(fileName); + + if (ctxQoa != NULL) + { + music.ctxType = MUSIC_AUDIO_QOA; + music.ctxData = ctxQoa; + // NOTE: We are loading samples are 32bit float normalized data, so, + // we configure the output audio stream to also use float 32bit + music.stream = LoadAudioStream(ctxQoa->info.samplerate, 32, ctxQoa->info.channels); + music.frameCount = ctxQoa->info.samples; + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + else{} //No uninit required + } +#endif +#if defined(SUPPORT_FILEFORMAT_FLAC) + else if (IsFileExtension(fileName, ".flac")) + { + drflac *ctxFlac = drflac_open_file(fileName, NULL); + + if (ctxFlac != NULL) + { + music.ctxType = MUSIC_AUDIO_FLAC; + music.ctxData = ctxFlac; + int sampleSize = ctxFlac->bitsPerSample; + if (ctxFlac->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream() + music.stream = LoadAudioStream(ctxFlac->sampleRate, sampleSize, ctxFlac->channels); + music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount; + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + else + { + drflac_free(ctxFlac, NULL); + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_XM) + else if (IsFileExtension(fileName, ".xm")) + { + jar_xm_context_t *ctxXm = NULL; + int result = jar_xm_create_context_from_file(&ctxXm, AUDIO.System.device.sampleRate, fileName); + + if (result == 0) // XM AUDIO.System.context created successfully + { + music.ctxType = MUSIC_MODULE_XM; + music.ctxData = ctxXm; + jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops + + unsigned int bits = 32; + if (AUDIO_DEVICE_FORMAT == ma_format_s16) bits = 16; + else if (AUDIO_DEVICE_FORMAT == ma_format_u8) bits = 8; + + // NOTE: Only stereo is supported for XM + music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, AUDIO_DEVICE_CHANNELS); + music.frameCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm); // NOTE: Always 2 channels (stereo) + music.looping = true; // Looping enabled by default + jar_xm_reset(ctxXm); // Make sure we start at the beginning of the song + musicLoaded = true; + } + else + { + jar_xm_free_context(ctxXm); + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_MOD) + else if (IsFileExtension(fileName, ".mod")) + { + jar_mod_context_t *ctxMod = RL_CALLOC(1, sizeof(jar_mod_context_t)); + jar_mod_init(ctxMod); + int result = jar_mod_load_file(ctxMod, fileName); + + if (result > 0) + { + music.ctxType = MUSIC_MODULE_MOD; + music.ctxData = ctxMod; + // NOTE: Only stereo is supported for MOD + music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, 16, AUDIO_DEVICE_CHANNELS); + music.frameCount = (unsigned int)jar_mod_max_samples(ctxMod); // NOTE: Always 2 channels (stereo) + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + else + { + jar_mod_unload(ctxMod); + RL_FREE(ctxMod); + } + } +#endif + else TRACELOG(LOG_WARNING, "STREAM: [%s] File format not supported", fileName); + + if (!musicLoaded) + { + TRACELOG(LOG_WARNING, "FILEIO: [%s] Music file could not be opened", fileName); + } + else + { + // Show some music stream info + TRACELOG(LOG_INFO, "FILEIO: [%s] Music file loaded successfully", fileName); + TRACELOG(LOG_INFO, " > Sample rate: %i Hz", music.stream.sampleRate); + TRACELOG(LOG_INFO, " > Sample size: %i bits", music.stream.sampleSize); + TRACELOG(LOG_INFO, " > Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi"); + TRACELOG(LOG_INFO, " > Total frames: %i", music.frameCount); + } + + return music; +} + +// Load music stream from memory buffer, fileType refers to extension: i.e. ".wav" +// WARNING: File extension must be provided in lower-case +Music LoadMusicStreamFromMemory(const char *fileType, const unsigned char *data, int dataSize) +{ + Music music = { 0 }; + bool musicLoaded = false; + + if (false) { } +#if defined(SUPPORT_FILEFORMAT_WAV) + else if ((strcmp(fileType, ".wav") == 0) || (strcmp(fileType, ".WAV") == 0)) + { + drwav *ctxWav = RL_CALLOC(1, sizeof(drwav)); + + bool success = drwav_init_memory(ctxWav, (const void *)data, dataSize, NULL); + + if (success) + { + music.ctxType = MUSIC_AUDIO_WAV; + music.ctxData = ctxWav; + int sampleSize = ctxWav->bitsPerSample; + if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream() + + music.stream = LoadAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels); + music.frameCount = (unsigned int)ctxWav->totalPCMFrameCount; + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + else { + drwav_uninit(ctxWav); + RL_FREE(ctxWav); + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_OGG) + else if ((strcmp(fileType, ".ogg") == 0) || (strcmp(fileType, ".OGG") == 0)) + { + // Open ogg audio stream + stb_vorbis* ctxOgg = stb_vorbis_open_memory((const unsigned char *)data, dataSize, NULL, NULL); + + if (ctxOgg != NULL) + { + music.ctxType = MUSIC_AUDIO_OGG; + music.ctxData = ctxOgg; + stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData); // Get Ogg file info + + // OGG bit rate defaults to 16 bit, it's enough for compressed format + music.stream = LoadAudioStream(info.sample_rate, 16, info.channels); + + // WARNING: It seems this function returns length in frames, not samples, so we multiply by channels + music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData); + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + else + { + stb_vorbis_close(ctxOgg); + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_MP3) + else if ((strcmp(fileType, ".mp3") == 0) || (strcmp(fileType, ".MP3") == 0)) + { + drmp3 *ctxMp3 = RL_CALLOC(1, sizeof(drmp3)); + int success = drmp3_init_memory(ctxMp3, (const void*)data, dataSize, NULL); + + if (success) + { + music.ctxType = MUSIC_AUDIO_MP3; + music.ctxData = ctxMp3; + music.stream = LoadAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels); + music.frameCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3); + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + else + { + drmp3_uninit(ctxMp3); + RL_FREE(ctxMp3); + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_QOA) + else if ((strcmp(fileType, ".qoa") == 0) || (strcmp(fileType, ".QOA") == 0)) + { + qoaplay_desc *ctxQoa = NULL; + if ((data != NULL) && (dataSize > 0)) + { + ctxQoa = qoaplay_open_memory(data, dataSize); + } + + if (ctxQoa != NULL) + { + music.ctxType = MUSIC_AUDIO_QOA; + music.ctxData = ctxQoa; + // NOTE: We are loading samples are 32bit float normalized data, so, + // we configure the output audio stream to also use float 32bit + music.stream = LoadAudioStream(ctxQoa->info.samplerate, 32, ctxQoa->info.channels); + music.frameCount = ctxQoa->info.samples; + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + else{} //No uninit required + } +#endif +#if defined(SUPPORT_FILEFORMAT_FLAC) + else if ((strcmp(fileType, ".flac") == 0) || (strcmp(fileType, ".FLAC") == 0)) + { + drflac *ctxFlac = drflac_open_memory((const void*)data, dataSize, NULL); + + if (ctxFlac != NULL) + { + music.ctxType = MUSIC_AUDIO_FLAC; + music.ctxData = ctxFlac; + int sampleSize = ctxFlac->bitsPerSample; + if (ctxFlac->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream() + music.stream = LoadAudioStream(ctxFlac->sampleRate, sampleSize, ctxFlac->channels); + music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount; + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + else + { + drflac_free(ctxFlac, NULL); + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_XM) + else if ((strcmp(fileType, ".xm") == 0) || (strcmp(fileType, ".XM") == 0)) + { + jar_xm_context_t *ctxXm = NULL; + int result = jar_xm_create_context_safe(&ctxXm, (const char *)data, dataSize, AUDIO.System.device.sampleRate); + if (result == 0) // XM AUDIO.System.context created successfully + { + music.ctxType = MUSIC_MODULE_XM; + music.ctxData = ctxXm; + jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops + + unsigned int bits = 32; + if (AUDIO_DEVICE_FORMAT == ma_format_s16) bits = 16; + else if (AUDIO_DEVICE_FORMAT == ma_format_u8) bits = 8; + + // NOTE: Only stereo is supported for XM + music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, 2); + music.frameCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm); // NOTE: Always 2 channels (stereo) + music.looping = true; // Looping enabled by default + jar_xm_reset(ctxXm); // Make sure we start at the beginning of the song + + musicLoaded = true; + } + else + { + jar_xm_free_context(ctxXm); + } + } +#endif +#if defined(SUPPORT_FILEFORMAT_MOD) + else if ((strcmp(fileType, ".mod") == 0) || (strcmp(fileType, ".MOD") == 0)) + { + jar_mod_context_t *ctxMod = (jar_mod_context_t *)RL_MALLOC(sizeof(jar_mod_context_t)); + int result = 0; + + jar_mod_init(ctxMod); + + // Copy data to allocated memory for default UnloadMusicStream + unsigned char *newData = (unsigned char *)RL_MALLOC(dataSize); + int it = dataSize/sizeof(unsigned char); + for (int i = 0; i < it; i++) newData[i] = data[i]; + + // Memory loaded version for jar_mod_load_file() + if (dataSize && (dataSize < 32*1024*1024)) + { + ctxMod->modfilesize = dataSize; + ctxMod->modfile = newData; + if (jar_mod_load(ctxMod, (void *)ctxMod->modfile, dataSize)) result = dataSize; + } + + if (result > 0) + { + music.ctxType = MUSIC_MODULE_MOD; + music.ctxData = ctxMod; + + // NOTE: Only stereo is supported for MOD + music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, 16, 2); + music.frameCount = (unsigned int)jar_mod_max_samples(ctxMod); // NOTE: Always 2 channels (stereo) + music.looping = true; // Looping enabled by default + musicLoaded = true; + } + else + { + jar_mod_unload(ctxMod); + RL_FREE(ctxMod); + } + } +#endif + else TRACELOG(LOG_WARNING, "STREAM: Data format not supported"); + + if (!musicLoaded) + { + TRACELOG(LOG_WARNING, "FILEIO: Music data could not be loaded"); + } + else + { + // Show some music stream info + TRACELOG(LOG_INFO, "FILEIO: Music data loaded successfully"); + TRACELOG(LOG_INFO, " > Sample rate: %i Hz", music.stream.sampleRate); + TRACELOG(LOG_INFO, " > Sample size: %i bits", music.stream.sampleSize); + TRACELOG(LOG_INFO, " > Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi"); + TRACELOG(LOG_INFO, " > Total frames: %i", music.frameCount); + } + + return music; +} + +// Checks if a music stream is valid (context and buffers initialized) +bool IsMusicValid(Music music) +{ + return ((music.ctxData != NULL) && // Validate context loaded + (music.frameCount > 0) && // Validate audio frame count + (music.stream.sampleRate > 0) && // Validate sample rate is supported + (music.stream.sampleSize > 0) && // Validate sample size is supported + (music.stream.channels > 0)); // Validate number of channels supported +} + +// Unload music stream +void UnloadMusicStream(Music music) +{ + UnloadAudioStream(music.stream); + + if (music.ctxData != NULL) + { + if (false) { } +#if defined(SUPPORT_FILEFORMAT_WAV) + else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData); +#endif +#if defined(SUPPORT_FILEFORMAT_OGG) + else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData); +#endif +#if defined(SUPPORT_FILEFORMAT_MP3) + else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); } +#endif +#if defined(SUPPORT_FILEFORMAT_QOA) + else if (music.ctxType == MUSIC_AUDIO_QOA) qoaplay_close((qoaplay_desc *)music.ctxData); +#endif +#if defined(SUPPORT_FILEFORMAT_FLAC) + else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL); +#endif +#if defined(SUPPORT_FILEFORMAT_XM) + else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData); +#endif +#if defined(SUPPORT_FILEFORMAT_MOD) + else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); } +#endif + } +} + +// Start music playing (open stream) from beginning +void PlayMusicStream(Music music) +{ + PlayAudioStream(music.stream); +} + +// Pause music playing +void PauseMusicStream(Music music) +{ + PauseAudioStream(music.stream); +} + +// Resume music playing +void ResumeMusicStream(Music music) +{ + ResumeAudioStream(music.stream); +} + +// Stop music playing (close stream) +void StopMusicStream(Music music) +{ + StopAudioStream(music.stream); + + switch (music.ctxType) + { +#if defined(SUPPORT_FILEFORMAT_WAV) + case MUSIC_AUDIO_WAV: drwav_seek_to_first_pcm_frame((drwav *)music.ctxData); break; +#endif +#if defined(SUPPORT_FILEFORMAT_OGG) + case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break; +#endif +#if defined(SUPPORT_FILEFORMAT_MP3) + case MUSIC_AUDIO_MP3: drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData); break; +#endif +#if defined(SUPPORT_FILEFORMAT_QOA) + case MUSIC_AUDIO_QOA: qoaplay_rewind((qoaplay_desc *)music.ctxData); break; +#endif +#if defined(SUPPORT_FILEFORMAT_FLAC) + case MUSIC_AUDIO_FLAC: drflac__seek_to_first_frame((drflac *)music.ctxData); break; +#endif +#if defined(SUPPORT_FILEFORMAT_XM) + case MUSIC_MODULE_XM: jar_xm_reset((jar_xm_context_t *)music.ctxData); break; +#endif +#if defined(SUPPORT_FILEFORMAT_MOD) + case MUSIC_MODULE_MOD: jar_mod_seek_start((jar_mod_context_t *)music.ctxData); break; +#endif + default: break; + } +} + +// Seek music to a certain position (in seconds) +void SeekMusicStream(Music music, float position) +{ + // Seeking is not supported in module formats + if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD)) return; + + unsigned int positionInFrames = (unsigned int)(position*music.stream.sampleRate); + + switch (music.ctxType) + { +#if defined(SUPPORT_FILEFORMAT_WAV) + case MUSIC_AUDIO_WAV: drwav_seek_to_pcm_frame((drwav *)music.ctxData, positionInFrames); break; +#endif +#if defined(SUPPORT_FILEFORMAT_OGG) + case MUSIC_AUDIO_OGG: stb_vorbis_seek_frame((stb_vorbis *)music.ctxData, positionInFrames); break; +#endif +#if defined(SUPPORT_FILEFORMAT_MP3) + case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, positionInFrames); break; +#endif +#if defined(SUPPORT_FILEFORMAT_QOA) + case MUSIC_AUDIO_QOA: + { + int qoaFrame = positionInFrames/QOA_FRAME_LEN; + qoaplay_seek_frame((qoaplay_desc *)music.ctxData, qoaFrame); // Seeks to QOA frame, not PCM frame + + // We need to compute QOA frame number and update positionInFrames + positionInFrames = ((qoaplay_desc *)music.ctxData)->sample_position; + } break; +#endif +#if defined(SUPPORT_FILEFORMAT_FLAC) + case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, positionInFrames); break; +#endif + default: break; + } + + ma_mutex_lock(&AUDIO.System.lock); + music.stream.buffer->framesProcessed = positionInFrames; + ma_mutex_unlock(&AUDIO.System.lock); +} + +// Update (re-fill) music buffers if data already processed +void UpdateMusicStream(Music music) +{ + if (music.stream.buffer == NULL) return; + + ma_mutex_lock(&AUDIO.System.lock); + + unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2; + + // On first call of this function we lazily pre-allocated a temp buffer to read audio files/memory data in + int frameSize = music.stream.channels*music.stream.sampleSize/8; + unsigned int pcmSize = subBufferSizeInFrames*frameSize; + + if (AUDIO.System.pcmBufferSize < pcmSize) + { + RL_FREE(AUDIO.System.pcmBuffer); + AUDIO.System.pcmBuffer = RL_CALLOC(1, pcmSize); + AUDIO.System.pcmBufferSize = pcmSize; + } + + // Check both sub-buffers to check if they require refilling + for (int i = 0; i < 2; i++) + { + if (!music.stream.buffer->isSubBufferProcessed[i]) continue; // No refilling required, move to next sub-buffer + + unsigned int framesLeft = music.frameCount - music.stream.buffer->framesProcessed; // Frames left to be processed + unsigned int framesToStream = 0; // Total frames to be streamed + + if ((framesLeft >= subBufferSizeInFrames) || music.looping) framesToStream = subBufferSizeInFrames; + else framesToStream = framesLeft; + + int frameCountStillNeeded = framesToStream; + int frameCountReadTotal = 0; + + switch (music.ctxType) + { + #if defined(SUPPORT_FILEFORMAT_WAV) + case MUSIC_AUDIO_WAV: + { + if (music.stream.sampleSize == 16) + { + while (true) + { + int frameCountRead = (int)drwav_read_pcm_frames_s16((drwav *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize)); + frameCountReadTotal += frameCountRead; + frameCountStillNeeded -= frameCountRead; + if (frameCountStillNeeded == 0) break; + else drwav_seek_to_first_pcm_frame((drwav *)music.ctxData); + } + } + else if (music.stream.sampleSize == 32) + { + while (true) + { + int frameCountRead = (int)drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize)); + frameCountReadTotal += frameCountRead; + frameCountStillNeeded -= frameCountRead; + if (frameCountStillNeeded == 0) break; + else drwav_seek_to_first_pcm_frame((drwav *)music.ctxData); + } + } + } break; + #endif + #if defined(SUPPORT_FILEFORMAT_OGG) + case MUSIC_AUDIO_OGG: + { + while (true) + { + int frameCountRead = stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize), frameCountStillNeeded*music.stream.channels); + frameCountReadTotal += frameCountRead; + frameCountStillNeeded -= frameCountRead; + if (frameCountStillNeeded == 0) break; + else stb_vorbis_seek_start((stb_vorbis *)music.ctxData); + } + } break; + #endif + #if defined(SUPPORT_FILEFORMAT_MP3) + case MUSIC_AUDIO_MP3: + { + while (true) + { + int frameCountRead = (int)drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize)); + frameCountReadTotal += frameCountRead; + frameCountStillNeeded -= frameCountRead; + if (frameCountStillNeeded == 0) break; + else drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData); + } + } break; + #endif + #if defined(SUPPORT_FILEFORMAT_QOA) + case MUSIC_AUDIO_QOA: + { + unsigned int frameCountRead = qoaplay_decode((qoaplay_desc *)music.ctxData, (float *)AUDIO.System.pcmBuffer, framesToStream); + frameCountReadTotal += frameCountRead; + /* + while (true) + { + int frameCountRead = (int)qoaplay_decode((qoaplay_desc *)music.ctxData, (float *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize), frameCountStillNeeded); + frameCountReadTotal += frameCountRead; + frameCountStillNeeded -= frameCountRead; + if (frameCountStillNeeded == 0) break; + else qoaplay_rewind((qoaplay_desc *)music.ctxData); + } + */ + } break; + #endif + #if defined(SUPPORT_FILEFORMAT_FLAC) + case MUSIC_AUDIO_FLAC: + { + while (true) + { + int frameCountRead = (int)drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize)); + frameCountReadTotal += frameCountRead; + frameCountStillNeeded -= frameCountRead; + if (frameCountStillNeeded == 0) break; + else drflac__seek_to_first_frame((drflac *)music.ctxData); + } + } break; + #endif + #if defined(SUPPORT_FILEFORMAT_XM) + case MUSIC_MODULE_XM: + { + // NOTE: Internally we consider 2 channels generation, so sampleCount/2 + if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)AUDIO.System.pcmBuffer, framesToStream); + else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream); + else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)AUDIO.System.pcmBuffer, framesToStream); + //jar_xm_reset((jar_xm_context_t *)music.ctxData); + + } break; + #endif + #if defined(SUPPORT_FILEFORMAT_MOD) + case MUSIC_MODULE_MOD: + { + // NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2 + jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream, 0); + //jar_mod_seek_start((jar_mod_context_t *)music.ctxData); + + } break; + #endif + default: break; + } + + UpdateAudioStreamInLockedState(music.stream, AUDIO.System.pcmBuffer, framesToStream); + + music.stream.buffer->framesProcessed = music.stream.buffer->framesProcessed%music.frameCount; + + if (framesLeft <= subBufferSizeInFrames) + { + if (!music.looping) + { + ma_mutex_unlock(&AUDIO.System.lock); + // Streaming is ending, we filled latest frames from input + StopMusicStream(music); + return; + } + } + } + + ma_mutex_unlock(&AUDIO.System.lock); +} + +// Check if any music is playing +bool IsMusicStreamPlaying(Music music) +{ + return IsAudioStreamPlaying(music.stream); +} + +// Set volume for music +void SetMusicVolume(Music music, float volume) +{ + SetAudioStreamVolume(music.stream, volume); +} + +// Set pitch for music +void SetMusicPitch(Music music, float pitch) +{ + SetAudioBufferPitch(music.stream.buffer, pitch); +} + +// Set pan for a music +void SetMusicPan(Music music, float pan) +{ + SetAudioBufferPan(music.stream.buffer, pan); +} + +// Get music time length (in seconds) +float GetMusicTimeLength(Music music) +{ + float totalSeconds = 0.0f; + + totalSeconds = (float)music.frameCount/music.stream.sampleRate; + + return totalSeconds; +} + +// Get current music time played (in seconds) +float GetMusicTimePlayed(Music music) +{ + float secondsPlayed = 0.0f; + if (music.stream.buffer != NULL) + { +#if defined(SUPPORT_FILEFORMAT_XM) + if (music.ctxType == MUSIC_MODULE_XM) + { + uint64_t framesPlayed = 0; + + jar_xm_get_position(music.ctxData, NULL, NULL, NULL, &framesPlayed); + secondsPlayed = (float)framesPlayed/music.stream.sampleRate; + } + else +#endif + { + ma_mutex_lock(&AUDIO.System.lock); + //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels; + int framesProcessed = (int)music.stream.buffer->framesProcessed; + int subBufferSize = (int)music.stream.buffer->sizeInFrames/2; + int framesInFirstBuffer = music.stream.buffer->isSubBufferProcessed[0]? 0 : subBufferSize; + int framesInSecondBuffer = music.stream.buffer->isSubBufferProcessed[1]? 0 : subBufferSize; + int framesSentToMix = music.stream.buffer->frameCursorPos%subBufferSize; + int framesPlayed = (framesProcessed - framesInFirstBuffer - framesInSecondBuffer + framesSentToMix)%(int)music.frameCount; + if (framesPlayed < 0) framesPlayed += music.frameCount; + secondsPlayed = (float)framesPlayed/music.stream.sampleRate; + ma_mutex_unlock(&AUDIO.System.lock); + } + } + + return secondsPlayed; +} + +// Load audio stream (to stream audio pcm data) +AudioStream LoadAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels) +{ + AudioStream stream = { 0 }; + + stream.sampleRate = sampleRate; + stream.sampleSize = sampleSize; + stream.channels = channels; + + ma_format formatIn = ((stream.sampleSize == 8)? ma_format_u8 : ((stream.sampleSize == 16)? ma_format_s16 : ma_format_f32)); + + // The size of a streaming buffer must be at least double the size of a period + unsigned int periodSize = AUDIO.System.device.playback.internalPeriodSizeInFrames; + + // If the buffer is not set, compute one that would give us a buffer good enough for a decent frame rate + unsigned int subBufferSize = (AUDIO.Buffer.defaultSize == 0)? AUDIO.System.device.sampleRate/30 : AUDIO.Buffer.defaultSize; + + if (subBufferSize < periodSize) subBufferSize = periodSize; + + // Create a double audio buffer of defined size + stream.buffer = LoadAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM); + + if (stream.buffer != NULL) + { + stream.buffer->looping = true; // Always loop for streaming buffers + TRACELOG(LOG_INFO, "STREAM: Initialized successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo"); + } + else TRACELOG(LOG_WARNING, "STREAM: Failed to load audio buffer, stream could not be created"); + + return stream; +} + +// Checks if an audio stream is valid (buffers initialized) +bool IsAudioStreamValid(AudioStream stream) +{ + return ((stream.buffer != NULL) && // Validate stream buffer + (stream.sampleRate > 0) && // Validate sample rate is supported + (stream.sampleSize > 0) && // Validate sample size is supported + (stream.channels > 0)); // Validate number of channels supported +} + +// Unload audio stream and free memory +void UnloadAudioStream(AudioStream stream) +{ + UnloadAudioBuffer(stream.buffer); + + TRACELOG(LOG_INFO, "STREAM: Unloaded audio stream data from RAM"); +} + +// Update audio stream buffers with data +// NOTE 1: Only updates one buffer of the stream source: dequeue -> update -> queue +// NOTE 2: To dequeue a buffer it needs to be processed: IsAudioStreamProcessed() +void UpdateAudioStream(AudioStream stream, const void *data, int frameCount) +{ + ma_mutex_lock(&AUDIO.System.lock); + UpdateAudioStreamInLockedState(stream, data, frameCount); + ma_mutex_unlock(&AUDIO.System.lock); +} + +// Check if any audio stream buffers requires refill +bool IsAudioStreamProcessed(AudioStream stream) +{ + if (stream.buffer == NULL) return false; + + bool result = false; + ma_mutex_lock(&AUDIO.System.lock); + result = stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]; + ma_mutex_unlock(&AUDIO.System.lock); + return result; +} + +// Play audio stream +void PlayAudioStream(AudioStream stream) +{ + PlayAudioBuffer(stream.buffer); +} + +// Play audio stream +void PauseAudioStream(AudioStream stream) +{ + PauseAudioBuffer(stream.buffer); +} + +// Resume audio stream playing +void ResumeAudioStream(AudioStream stream) +{ + ResumeAudioBuffer(stream.buffer); +} + +// Check if audio stream is playing +bool IsAudioStreamPlaying(AudioStream stream) +{ + return IsAudioBufferPlaying(stream.buffer); +} + +// Stop audio stream +void StopAudioStream(AudioStream stream) +{ + StopAudioBuffer(stream.buffer); +} + +// Set volume for audio stream (1.0 is max level) +void SetAudioStreamVolume(AudioStream stream, float volume) +{ + SetAudioBufferVolume(stream.buffer, volume); +} + +// Set pitch for audio stream (1.0 is base level) +void SetAudioStreamPitch(AudioStream stream, float pitch) +{ + SetAudioBufferPitch(stream.buffer, pitch); +} + +// Set pan for audio stream +void SetAudioStreamPan(AudioStream stream, float pan) +{ + SetAudioBufferPan(stream.buffer, pan); +} + +// Default size for new audio streams +void SetAudioStreamBufferSizeDefault(int size) +{ + AUDIO.Buffer.defaultSize = size; +} + +// Audio thread callback to request new data +void SetAudioStreamCallback(AudioStream stream, AudioCallback callback) +{ + if (stream.buffer != NULL) + { + ma_mutex_lock(&AUDIO.System.lock); + stream.buffer->callback = callback; + ma_mutex_unlock(&AUDIO.System.lock); + } +} + +// Add processor to audio stream. Contrary to buffers, the order of processors is important +// The new processor must be added at the end. As there aren't supposed to be a lot of processors attached to +// a given stream, we iterate through the list to find the end. That way we don't need a pointer to the last element +void AttachAudioStreamProcessor(AudioStream stream, AudioCallback process) +{ + ma_mutex_lock(&AUDIO.System.lock); + + rAudioProcessor *processor = (rAudioProcessor *)RL_CALLOC(1, sizeof(rAudioProcessor)); + processor->process = process; + + rAudioProcessor *last = stream.buffer->processor; + + while (last && last->next) + { + last = last->next; + } + if (last) + { + processor->prev = last; + last->next = processor; + } + else stream.buffer->processor = processor; + + ma_mutex_unlock(&AUDIO.System.lock); +} + +// Remove processor from audio stream +void DetachAudioStreamProcessor(AudioStream stream, AudioCallback process) +{ + ma_mutex_lock(&AUDIO.System.lock); + + rAudioProcessor *processor = stream.buffer->processor; + + while (processor) + { + rAudioProcessor *next = processor->next; + rAudioProcessor *prev = processor->prev; + + if (processor->process == process) + { + if (stream.buffer->processor == processor) stream.buffer->processor = next; + if (prev) prev->next = next; + if (next) next->prev = prev; + + RL_FREE(processor); + } + + processor = next; + } + + ma_mutex_unlock(&AUDIO.System.lock); +} + +// Add processor to audio pipeline. Order of processors is important +// Works the same way as {Attach,Detach}AudioStreamProcessor() functions, except +// these two work on the already mixed output just before sending it to the sound hardware +void AttachAudioMixedProcessor(AudioCallback process) +{ + ma_mutex_lock(&AUDIO.System.lock); + + rAudioProcessor *processor = (rAudioProcessor *)RL_CALLOC(1, sizeof(rAudioProcessor)); + processor->process = process; + + rAudioProcessor *last = AUDIO.mixedProcessor; + + while (last && last->next) + { + last = last->next; + } + if (last) + { + processor->prev = last; + last->next = processor; + } + else AUDIO.mixedProcessor = processor; + + ma_mutex_unlock(&AUDIO.System.lock); +} + +// Remove processor from audio pipeline +void DetachAudioMixedProcessor(AudioCallback process) +{ + ma_mutex_lock(&AUDIO.System.lock); + + rAudioProcessor *processor = AUDIO.mixedProcessor; + + while (processor) + { + rAudioProcessor *next = processor->next; + rAudioProcessor *prev = processor->prev; + + if (processor->process == process) + { + if (AUDIO.mixedProcessor == processor) AUDIO.mixedProcessor = next; + if (prev) prev->next = next; + if (next) next->prev = prev; + + RL_FREE(processor); + } + + processor = next; + } + + ma_mutex_unlock(&AUDIO.System.lock); +} + + +//---------------------------------------------------------------------------------- +// Module specific Functions Definition +//---------------------------------------------------------------------------------- + +// Log callback function +static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage) +{ + TRACELOG(LOG_WARNING, "miniaudio: %s", pMessage); // All log messages from miniaudio are errors +} + +// Reads audio data from an AudioBuffer object in internal format +static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount) +{ + // Using audio buffer callback + if (audioBuffer->callback) + { + audioBuffer->callback(framesOut, frameCount); + audioBuffer->framesProcessed += frameCount; + + return frameCount; + } + + ma_uint32 subBufferSizeInFrames = (audioBuffer->sizeInFrames > 1)? audioBuffer->sizeInFrames/2 : audioBuffer->sizeInFrames; + ma_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames; + + if (currentSubBufferIndex > 1) return 0; + + // Another thread can update the processed state of buffers, so + // we just take a copy here to try and avoid potential synchronization problems + bool isSubBufferProcessed[2] = { 0 }; + isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0]; + isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1]; + + ma_uint32 frameSizeInBytes = ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn); + + // Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0 + ma_uint32 framesRead = 0; + while (1) + { + // We break from this loop differently depending on the buffer's usage + // - For static buffers, we simply fill as much data as we can + // - For streaming buffers we only fill half of the buffer that are processed + // Unprocessed halves must keep their audio data in-tact + if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC) + { + if (framesRead >= frameCount) break; + } + else + { + if (isSubBufferProcessed[currentSubBufferIndex]) break; + } + + ma_uint32 totalFramesRemaining = (frameCount - framesRead); + if (totalFramesRemaining == 0) break; + + ma_uint32 framesRemainingInOutputBuffer; + if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC) + { + framesRemainingInOutputBuffer = audioBuffer->sizeInFrames - audioBuffer->frameCursorPos; + } + else + { + ma_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames*currentSubBufferIndex; + framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer); + } + + ma_uint32 framesToRead = totalFramesRemaining; + if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer; + + memcpy((unsigned char *)framesOut + (framesRead*frameSizeInBytes), audioBuffer->data + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes); + audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead)%audioBuffer->sizeInFrames; + framesRead += framesToRead; + + // If we've read to the end of the buffer, mark it as processed + if (framesToRead == framesRemainingInOutputBuffer) + { + audioBuffer->isSubBufferProcessed[currentSubBufferIndex] = true; + isSubBufferProcessed[currentSubBufferIndex] = true; + + currentSubBufferIndex = (currentSubBufferIndex + 1)%2; + + // We need to break from this loop if we're not looping + if (!audioBuffer->looping) + { + StopAudioBufferInLockedState(audioBuffer); + break; + } + } + } + + // Zero-fill excess + ma_uint32 totalFramesRemaining = (frameCount - framesRead); + if (totalFramesRemaining > 0) + { + memset((unsigned char *)framesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes); + + // For static buffers we can fill the remaining frames with silence for safety, but we don't want + // to report those frames as "read". The reason for this is that the caller uses the return value + // to know whether a non-looping sound has finished playback + if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining; + } + + return framesRead; +} + +// Reads audio data from an AudioBuffer object in device format, returned data will be in a format appropriate for mixing +static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount) +{ + // What's going on here is that we're continuously converting data from the AudioBuffer's internal format to the mixing format, which + // should be defined by the output format of the data converter. We do this until frameCount frames have been output. The important + // detail to remember here is that we never, ever attempt to read more input data than is required for the specified number of output + // frames. This can be achieved with ma_data_converter_get_required_input_frame_count() + ma_uint8 inputBuffer[4096] = { 0 }; + ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn); + + ma_uint32 totalOutputFramesProcessed = 0; + while (totalOutputFramesProcessed < frameCount) + { + ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed; + ma_uint64 inputFramesToProcessThisIteration = 0; + + (void)ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration, &inputFramesToProcessThisIteration); + if (inputFramesToProcessThisIteration > inputBufferFrameCap) + { + inputFramesToProcessThisIteration = inputBufferFrameCap; + } + + float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.channelsOut); + + /* At this point we can convert the data to our mixing format. */ + ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration); /* Safe cast. */ + ma_uint64 outputFramesProcessedThisIteration = outputFramesToProcessThisIteration; + ma_data_converter_process_pcm_frames(&audioBuffer->converter, inputBuffer, &inputFramesProcessedThisIteration, runningFramesOut, &outputFramesProcessedThisIteration); + + totalOutputFramesProcessed += (ma_uint32)outputFramesProcessedThisIteration; /* Safe cast. */ + + if (inputFramesProcessedThisIteration < inputFramesToProcessThisIteration) + { + break; /* Ran out of input data. */ + } + + /* This should never be hit, but will add it here for safety. Ensures we get out of the loop when no input nor output frames are processed. */ + if (inputFramesProcessedThisIteration == 0 && outputFramesProcessedThisIteration == 0) + { + break; + } + } + + return totalOutputFramesProcessed; +} + +// Sending audio data to device callback function +// This function will be called when miniaudio needs more data +// NOTE: All the mixing takes place here +static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount) +{ + (void)pDevice; + + // Mixing is basically just an accumulation, we need to initialize the output buffer to 0 + memset(pFramesOut, 0, frameCount*pDevice->playback.channels*ma_get_bytes_per_sample(pDevice->playback.format)); + + // Using a mutex here for thread-safety which makes things not real-time + // This is unlikely to be necessary for this project, but may want to consider how you might want to avoid this + ma_mutex_lock(&AUDIO.System.lock); + { + for (AudioBuffer *audioBuffer = AUDIO.Buffer.first; audioBuffer != NULL; audioBuffer = audioBuffer->next) + { + // Ignore stopped or paused sounds + if (!audioBuffer->playing || audioBuffer->paused) continue; + + ma_uint32 framesRead = 0; + + while (1) + { + if (framesRead >= frameCount) break; + + // Just read as much data as we can from the stream + ma_uint32 framesToRead = (frameCount - framesRead); + + while (framesToRead > 0) + { + float tempBuffer[1024] = { 0 }; // Frames for stereo + + ma_uint32 framesToReadRightNow = framesToRead; + if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS) + { + framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS; + } + + ma_uint32 framesJustRead = ReadAudioBufferFramesInMixingFormat(audioBuffer, tempBuffer, framesToReadRightNow); + if (framesJustRead > 0) + { + float *framesOut = (float *)pFramesOut + (framesRead*AUDIO.System.device.playback.channels); + float *framesIn = tempBuffer; + + // Apply processors chain if defined + rAudioProcessor *processor = audioBuffer->processor; + while (processor) + { + processor->process(framesIn, framesJustRead); + processor = processor->next; + } + + MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer); + + framesToRead -= framesJustRead; + framesRead += framesJustRead; + } + + if (!audioBuffer->playing) + { + framesRead = frameCount; + break; + } + + // If we weren't able to read all the frames we requested, break + if (framesJustRead < framesToReadRightNow) + { + if (!audioBuffer->looping) + { + StopAudioBufferInLockedState(audioBuffer); + break; + } + else + { + // Should never get here, but just for safety, + // move the cursor position back to the start and continue the loop + audioBuffer->frameCursorPos = 0; + continue; + } + } + } + + // If for some reason we weren't able to read every frame we'll need to break from the loop + // Not doing this could theoretically put us into an infinite loop + if (framesToRead > 0) break; + } + } + } + + rAudioProcessor *processor = AUDIO.mixedProcessor; + while (processor) + { + processor->process(pFramesOut, frameCount); + processor = processor->next; + } + + ma_mutex_unlock(&AUDIO.System.lock); +} + +// Main mixing function, pretty simple in this project, just an accumulation +// NOTE: framesOut is both an input and an output, it is initially filled with zeros outside of this function +static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer) +{ + const float localVolume = buffer->volume; + const ma_uint32 channels = AUDIO.System.device.playback.channels; + + if (channels == 2) // We consider panning + { + const float left = buffer->pan; + const float right = 1.0f - left; + + // Fast sine approximation in [0..1] for pan law: y = 0.5f*x*(3 - x*x); + const float levels[2] = { localVolume*0.5f*left*(3.0f - left*left), localVolume*0.5f*right*(3.0f - right*right) }; + + float *frameOut = framesOut; + const float *frameIn = framesIn; + + for (ma_uint32 frame = 0; frame < frameCount; frame++) + { + frameOut[0] += (frameIn[0]*levels[0]); + frameOut[1] += (frameIn[1]*levels[1]); + + frameOut += 2; + frameIn += 2; + } + } + else // We do not consider panning + { + for (ma_uint32 frame = 0; frame < frameCount; frame++) + { + for (ma_uint32 c = 0; c < channels; c++) + { + float *frameOut = framesOut + (frame*channels); + const float *frameIn = framesIn + (frame*channels); + + // Output accumulates input multiplied by volume to provided output (usually 0) + frameOut[c] += (frameIn[c]*localVolume); + } + } + } +} + +// Check if an audio buffer is playing, assuming the audio system mutex has been locked +static bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer) +{ + bool result = false; + + if (buffer != NULL) result = (buffer->playing && !buffer->paused); + + return result; +} + +// Stop an audio buffer, assuming the audio system mutex has been locked +static void StopAudioBufferInLockedState(AudioBuffer *buffer) +{ + if (buffer != NULL) + { + if (IsAudioBufferPlayingInLockedState(buffer)) + { + buffer->playing = false; + buffer->paused = false; + buffer->frameCursorPos = 0; + buffer->framesProcessed = 0; + buffer->isSubBufferProcessed[0] = true; + buffer->isSubBufferProcessed[1] = true; + } + } +} + +// Update audio stream, assuming the audio system mutex has been locked +static void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount) +{ + if (stream.buffer != NULL) + { + if (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]) + { + ma_uint32 subBufferToUpdate = 0; + + if (stream.buffer->isSubBufferProcessed[0] && stream.buffer->isSubBufferProcessed[1]) + { + // Both buffers are available for updating + // Update the first one and make sure the cursor is moved back to the front + subBufferToUpdate = 0; + stream.buffer->frameCursorPos = 0; + } + else + { + // Just update whichever sub-buffer is processed + subBufferToUpdate = (stream.buffer->isSubBufferProcessed[0])? 0 : 1; + } + + ma_uint32 subBufferSizeInFrames = stream.buffer->sizeInFrames/2; + unsigned char *subBuffer = stream.buffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate); + + // Total frames processed in buffer is always the complete size, filled with 0 if required + stream.buffer->framesProcessed += subBufferSizeInFrames; + + // Does this API expect a whole buffer to be updated in one go? + // Assuming so, but if not will need to change this logic + if (subBufferSizeInFrames >= (ma_uint32)frameCount) + { + ma_uint32 framesToWrite = (ma_uint32)frameCount; + + ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8); + memcpy(subBuffer, data, bytesToWrite); + + // Any leftover frames should be filled with zeros + ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite; + + if (leftoverFrameCount > 0) memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8)); + + stream.buffer->isSubBufferProcessed[subBufferToUpdate] = false; + } + else TRACELOG(LOG_WARNING, "STREAM: Attempting to write too many frames to buffer"); + } + else TRACELOG(LOG_WARNING, "STREAM: Buffer not available for updating"); + } +} + +// Some required functions for audio standalone module version +#if defined(RAUDIO_STANDALONE) +// Check file extension +static bool IsFileExtension(const char *fileName, const char *ext) +{ + bool result = false; + const char *fileExt; + + if ((fileExt = strrchr(fileName, '.')) != NULL) + { + if (strcmp(fileExt, ext) == 0) result = true; + } + + return result; +} + +// Get pointer to extension for a filename string (includes the dot: .png) +static const char *GetFileExtension(const char *fileName) +{ + const char *dot = strrchr(fileName, '.'); + + if (!dot || dot == fileName) return NULL; + + return dot; +} + +// String pointer reverse break: returns right-most occurrence of charset in s +static const char *strprbrk(const char *s, const char *charset) +{ + const char *latestMatch = NULL; + for (; s = strpbrk(s, charset), s != NULL; latestMatch = s++) { } + return latestMatch; +} + +// Get pointer to filename for a path string +static const char *GetFileName(const char *filePath) +{ + const char *fileName = NULL; + if (filePath != NULL) fileName = strprbrk(filePath, "\\/"); + + if (!fileName) return filePath; + + return fileName + 1; +} + +// Get filename string without extension (uses static string) +static const char *GetFileNameWithoutExt(const char *filePath) +{ + #define MAX_FILENAMEWITHOUTEXT_LENGTH 256 + + static char fileName[MAX_FILENAMEWITHOUTEXT_LENGTH] = { 0 }; + memset(fileName, 0, MAX_FILENAMEWITHOUTEXT_LENGTH); + + if (filePath != NULL) strcpy(fileName, GetFileName(filePath)); // Get filename with extension + + int size = (int)strlen(fileName); // Get size in bytes + + for (int i = 0; (i < size) && (i < MAX_FILENAMEWITHOUTEXT_LENGTH); i++) + { + if (fileName[i] == '.') + { + // NOTE: We break on first '.' found + fileName[i] = '\0'; + break; + } + } + + return fileName; +} + +// Load data from file into a buffer +static unsigned char *LoadFileData(const char *fileName, int *dataSize) +{ + unsigned char *data = NULL; + *dataSize = 0; + + if (fileName != NULL) + { + FILE *file = fopen(fileName, "rb"); + + if (file != NULL) + { + // WARNING: On binary streams SEEK_END could not be found, + // using fseek() and ftell() could not work in some (rare) cases + fseek(file, 0, SEEK_END); + int size = ftell(file); + fseek(file, 0, SEEK_SET); + + if (size > 0) + { + data = (unsigned char *)RL_MALLOC(size*sizeof(unsigned char)); + + // NOTE: fread() returns number of read elements instead of bytes, so we read [1 byte, size elements] + unsigned int count = (unsigned int)fread(data, sizeof(unsigned char), size, file); + *dataSize = count; + + if (count != size) TRACELOG(LOG_WARNING, "FILEIO: [%s] File partially loaded", fileName); + else TRACELOG(LOG_INFO, "FILEIO: [%s] File loaded successfully", fileName); + } + else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to read file", fileName); + + fclose(file); + } + else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open file", fileName); + } + else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid"); + + return data; +} + +// Save data to file from buffer +static bool SaveFileData(const char *fileName, void *data, int dataSize) +{ + if (fileName != NULL) + { + FILE *file = fopen(fileName, "wb"); + + if (file != NULL) + { + unsigned int count = (unsigned int)fwrite(data, sizeof(unsigned char), dataSize, file); + + if (count == 0) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to write file", fileName); + else if (count != dataSize) TRACELOG(LOG_WARNING, "FILEIO: [%s] File partially written", fileName); + else TRACELOG(LOG_INFO, "FILEIO: [%s] File saved successfully", fileName); + + fclose(file); + } + else + { + TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open file", fileName); + return false; + } + } + else + { + TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid"); + return false; + } + + return true; +} + +// Save text data to file (write), string must be '\0' terminated +static bool SaveFileText(const char *fileName, char *text) +{ + if (fileName != NULL) + { + FILE *file = fopen(fileName, "wt"); + + if (file != NULL) + { + int count = fprintf(file, "%s", text); + + if (count == 0) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to write text file", fileName); + else TRACELOG(LOG_INFO, "FILEIO: [%s] Text file saved successfully", fileName); + + fclose(file); + } + else + { + TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open text file", fileName); + return false; + } + } + else + { + TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid"); + return false; + } + + return true; +} +#endif + +#undef AudioBuffer + +#endif // SUPPORT_MODULE_RAUDIO |