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captain-amygdala.pistorm/raylib/raudio.c
beeanyew 31f58d05d3 [MEGA-WIP] Raylib-based RTG output 
NOTE: A working keyboard must be attached to the Raspberry Pi while testing this, otherwise it's impossible to actually quit the emulator.
raylib takes possession of the SSH keyboard for some reason, which makes it so you can't Ctrl+C out of the emulator over SSH, you must Ctrl+C or press Q on the Pi keyboard.

A mostly working RTG implementation using raylib instead of SDL2.0
Greatly decreases the rendering overhead for 8bpp modes and gets rid of the need for hardware ARGB888 texture format support.
RTG will be initialized using the resolution of the Raspberry Pi, and onbly the 320x200/320x240 modes are currently scaled to the full vertical area of the screen.
2021-05-01 19:43:26 +02:00

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/**********************************************************************************************
*
* raudio v1.0 - 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 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_XM
* #define SUPPORT_FILEFORMAT_MOD
* #define SUPPORT_FILEFORMAT_FLAC
* #define SUPPORT_FILEFORMAT_MP3
* 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/dr-soft/miniaudio)
* stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
* 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) 2013-2021 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"
#include <stdarg.h> // Required for: va_list, va_start(), vfprintf(), va_end()
#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(_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>
// 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>
#include <mmreg.h>
#include <mmsystem.h>
// 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 MINIAUDIO_IMPLEMENTATION
//#define MA_DEBUG_OUTPUT
#include "external/miniaudio.h" // miniaudio library
#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()
#if defined(RAUDIO_STANDALONE)
#include <string.h> // Required for: strcmp() [Used in IsFileExtension()]
#if !defined(TRACELOG)
#define TRACELOG(level, ...) (void)0
#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_OGG)
// TODO: Remap malloc()/free() calls to RL_MALLOC/RL_FREE
#define STB_VORBIS_IMPLEMENTATION
#include "external/stb_vorbis.h" // OGG loading functions
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
#define JARXM_MALLOC RL_MALLOC
#define JARXM_FREE RL_FREE
#define JAR_XM_IMPLEMENTATION
#include "external/jar_xm.h" // XM loading functions
#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
#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_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_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(_MSC_VER)
#undef bool
#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 channels: stereo
#endif
#ifndef MAX_AUDIO_BUFFER_POOL_CHANNELS
#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 // Audio pool channels
#endif
#ifndef DEFAULT_AUDIO_BUFFER_SIZE
#define DEFAULT_AUDIO_BUFFER_SIZE 4096 // Default audio buffer size
#endif
//----------------------------------------------------------------------------------
// Types and Structures Definition
//----------------------------------------------------------------------------------
// 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,
MUSIC_AUDIO_WAV,
MUSIC_AUDIO_OGG,
MUSIC_AUDIO_FLAC,
MUSIC_AUDIO_MP3,
MUSIC_MODULE_XM,
MUSIC_MODULE_MOD
} MusicContextType;
#if defined(RAUDIO_STANDALONE)
typedef enum {
LOG_ALL,
LOG_TRACE,
LOG_DEBUG,
LOG_INFO,
LOG_WARNING,
LOG_ERROR,
LOG_FATAL,
LOG_NONE
} TraceLogLevel;
#endif
// NOTE: Different logic is used when feeding data to the playback device
// depending on whether or not data is streamed (Music vs Sound)
typedef enum {
AUDIO_BUFFER_USAGE_STATIC = 0,
AUDIO_BUFFER_USAGE_STREAM
} AudioBufferUsage;
// Audio buffer structure
struct rAudioBuffer {
ma_data_converter converter; // Audio data converter
float volume; // Audio buffer volume
float pitch; // Audio buffer pitch
bool playing; // Audio buffer state: AUDIO_PLAYING
bool paused; // Audio buffer state: AUDIO_PAUSED
bool looping; // Audio buffer looping, always 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 totalFramesProcessed; // 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
};
#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
} 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;
struct {
unsigned int poolCounter; // AudioBuffer pointers pool counter
AudioBuffer *pool[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // Multichannel AudioBuffer pointers pool
unsigned int channels[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // AudioBuffer pool channels
} MultiChannel;
} 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
};
//----------------------------------------------------------------------------------
// Module specific Functions Declaration
//----------------------------------------------------------------------------------
static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message);
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, float localVolume);
#if defined(SUPPORT_FILEFORMAT_WAV)
static Wave LoadWAV(const unsigned char *fileData, unsigned int fileSize); // Load WAV file
static int SaveWAV(Wave wave, const char *fileName); // Save wave data as WAV file
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
static Wave LoadOGG(const unsigned char *fileData, unsigned int fileSize); // Load OGG file
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
static Wave LoadFLAC(const unsigned char *fileData, unsigned int fileSize); // Load FLAC file
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
static Wave LoadMP3(const unsigned char *fileData, unsigned int fileSize); // Load MP3 file
#endif
#if defined(RAUDIO_STANDALONE)
static bool IsFileExtension(const char *fileName, const char *ext); // Check file extension
static unsigned char *LoadFileData(const char *fileName, unsigned int *bytesRead); // Load file data as byte array (read)
static bool SaveFileData(const char *fileName, void *data, unsigned int bytesToWrite); // 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 TrackAudioBuffer(AudioBuffer *buffer);
void UntrackAudioBuffer(AudioBuffer *buffer);
int GetAudioStreamBufferSizeDefault();
//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Device initialization and Closing
//----------------------------------------------------------------------------------
// Initialize audio device
void InitAudioDevice(void)
{
// TODO: Load AUDIO context memory dynamically?
// Init audio context
ma_context_config ctxConfig = ma_context_config_init();
ctxConfig.logCallback = OnLog;
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;
}
// 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;
}
// Mixing happens on a seperate 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;
}
// Init dummy audio buffers pool for multichannel sound playing
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
{
// WARNING: An empty audioBuffer is created (data = 0)
// AudioBuffer data just points to loaded sound data
AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, 0, AUDIO_BUFFER_USAGE_STATIC);
}
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)
{
// Unload dummy audio buffers pool
// WARNING: They can be pointing to already unloaded data
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
{
//UnloadAudioBuffer(AUDIO.MultiChannel.pool[i]);
if (AUDIO.MultiChannel.pool[i] != NULL)
{
ma_data_converter_uninit(&AUDIO.MultiChannel.pool[i]->converter);
UntrackAudioBuffer(AUDIO.MultiChannel.pool[i]);
//RL_FREE(buffer->data); // Already unloaded by UnloadSound()
RL_FREE(AUDIO.MultiChannel.pool[i]);
}
}
ma_mutex_uninit(&AUDIO.System.lock);
ma_device_uninit(&AUDIO.System.device);
ma_context_uninit(&AUDIO.System.context);
AUDIO.System.isReady = false;
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);
}
//----------------------------------------------------------------------------------
// 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.resampling.allowDynamicSampleRate = true; // Required for pitch shifting
ma_result result = ma_data_converter_init(&converterConfig, &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->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)
{
ma_data_converter_uninit(&buffer->converter);
UntrackAudioBuffer(buffer);
RL_FREE(buffer->data);
RL_FREE(buffer);
}
}
// Check if an audio buffer is playing
bool IsAudioBufferPlaying(AudioBuffer *buffer)
{
bool result = false;
if (buffer != NULL) result = (buffer->playing && !buffer->paused);
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)
{
buffer->playing = true;
buffer->paused = false;
buffer->frameCursorPos = 0;
}
}
// Stop an audio buffer
void StopAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL)
{
if (IsAudioBufferPlaying(buffer))
{
buffer->playing = false;
buffer->paused = false;
buffer->frameCursorPos = 0;
buffer->totalFramesProcessed = 0;
buffer->isSubBufferProcessed[0] = true;
buffer->isSubBufferProcessed[1] = true;
}
}
}
// Pause an audio buffer
void PauseAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL) buffer->paused = true;
}
// Resume an audio buffer
void ResumeAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL) buffer->paused = false;
}
// Set volume for an audio buffer
void SetAudioBufferVolume(AudioBuffer *buffer, float volume)
{
if (buffer != NULL) buffer->volume = volume;
}
// Set pitch for an audio buffer
void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
{
if ((buffer != NULL) && (pitch > 0.0f))
{
// 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.config.sampleRateOut/pitch);
ma_data_converter_set_rate(&buffer->converter, buffer->converter.config.sampleRateIn, outputSampleRate);
buffer->pitch = pitch;
}
}
// 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
unsigned int fileSize = 0;
unsigned char *fileData = LoadFileData(fileName, &fileSize);
if (fileData != NULL)
{
// Loading wave from memory data
wave = LoadWaveFromMemory(GetFileExtension(fileName), fileData, fileSize);
RL_FREE(fileData);
}
return wave;
}
// Load wave from memory buffer, fileType refers to extension: i.e. ".wav"
Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int dataSize)
{
Wave wave = { 0 };
char fileExtLower[16] = { 0 };
strcpy(fileExtLower, TextToLower(fileType));
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if (TextIsEqual(fileExtLower, ".wav")) wave = LoadWAV(fileData, dataSize);
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (TextIsEqual(fileExtLower, ".ogg")) wave = LoadOGG(fileData, dataSize);
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (TextIsEqual(fileExtLower, ".flac")) wave = LoadFLAC(fileData, dataSize);
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (TextIsEqual(fileExtLower, ".mp3")) wave = LoadMP3(fileData, dataSize);
#endif
else TRACELOG(LOG_WARNING, "WAVE: File format not supported");
return wave;
}
// 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.sampleCount/wave.channels;
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.sampleCount = frameCount*AUDIO_DEVICE_CHANNELS;
sound.stream.sampleRate = AUDIO.System.device.sampleRate;
sound.stream.sampleSize = 32;
sound.stream.channels = AUDIO_DEVICE_CHANNELS;
sound.stream.buffer = audioBuffer;
}
return sound;
}
// Unload wave data
void UnloadWave(Wave wave)
{
if (wave.data != NULL) 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, "WAVE: Unloaded sound data from RAM");
}
// Update sound buffer with new data
void UpdateSound(Sound sound, const void *data, int samplesCount)
{
if (sound.stream.buffer != NULL)
{
StopAudioBuffer(sound.stream.buffer);
// TODO: May want to lock/unlock this since this data buffer is read at mixing time
memcpy(sound.stream.buffer->data, data, samplesCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.config.formatIn, sound.stream.buffer->converter.config.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")) success = SaveWAV(wave, fileName);
#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.sampleCount*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.sampleCount*wave.channels*wave.sampleSize/8;
// NOTE: Text data buffer size is estimated considering wave data size in bytes
// and requiring 6 char bytes for every byte: "0x00, "
char *txtData = (char *)RL_CALLOC(6*waveDataSize + 2000, sizeof(char));
int bytesCount = 0;
bytesCount += sprintf(txtData + bytesCount, "\n//////////////////////////////////////////////////////////////////////////////////\n");
bytesCount += sprintf(txtData + bytesCount, "// //\n");
bytesCount += sprintf(txtData + bytesCount, "// WaveAsCode exporter v1.0 - Wave data exported as an array of bytes //\n");
bytesCount += sprintf(txtData + bytesCount, "// //\n");
bytesCount += sprintf(txtData + bytesCount, "// more info and bugs-report: github.com/raysan5/raylib //\n");
bytesCount += sprintf(txtData + bytesCount, "// feedback and support: ray[at]raylib.com //\n");
bytesCount += sprintf(txtData + bytesCount, "// //\n");
bytesCount += sprintf(txtData + bytesCount, "// Copyright (c) 2018 Ramon Santamaria (@raysan5) //\n");
bytesCount += sprintf(txtData + bytesCount, "// //\n");
bytesCount += sprintf(txtData + bytesCount, "//////////////////////////////////////////////////////////////////////////////////\n\n");
char varFileName[256] = { 0 };
#if !defined(RAUDIO_STANDALONE)
// Get file name from path and convert variable name to uppercase
strcpy(varFileName, GetFileNameWithoutExt(fileName));
for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; }
#else
strcpy(varFileName, fileName);
#endif
bytesCount += sprintf(txtData + bytesCount, "// Wave data information\n");
bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_COUNT %u\n", varFileName, wave.sampleCount);
bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_RATE %u\n", varFileName, wave.sampleRate);
bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_SIZE %u\n", varFileName, wave.sampleSize);
bytesCount += sprintf(txtData + bytesCount, "#define %s_CHANNELS %u\n\n", varFileName, wave.channels);
// Write byte data as hexadecimal text
bytesCount += sprintf(txtData + bytesCount, "static unsigned char %s_DATA[%i] = { ", varFileName, waveDataSize);
for (int i = 0; i < waveDataSize - 1; i++) bytesCount += sprintf(txtData + bytesCount, ((i%TEXT_BYTES_PER_LINE == 0)? "0x%x,\n" : "0x%x, "), ((unsigned char *)wave.data)[i]);
bytesCount += sprintf(txtData + bytesCount, "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);
return success;
}
// Play a sound
void PlaySound(Sound sound)
{
PlayAudioBuffer(sound.stream.buffer);
}
// Play a sound in the multichannel buffer pool
void PlaySoundMulti(Sound sound)
{
int index = -1;
unsigned int oldAge = 0;
int oldIndex = -1;
// find the first non playing pool entry
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
{
if (AUDIO.MultiChannel.channels[i] > oldAge)
{
oldAge = AUDIO.MultiChannel.channels[i];
oldIndex = i;
}
if (!IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i]))
{
index = i;
break;
}
}
// If no none playing pool members can be index choose the oldest
if (index == -1)
{
TRACELOG(LOG_WARNING, "SOUND: Buffer pool is already full, count: %i", AUDIO.MultiChannel.poolCounter);
if (oldIndex == -1)
{
// Shouldn't be able to get here... but just in case something odd happens!
TRACELOG(LOG_WARNING, "SOUND: Buffer pool could not determine oldest buffer not playing sound");
return;
}
index = oldIndex;
// Just in case...
StopAudioBuffer(AUDIO.MultiChannel.pool[index]);
}
// Experimentally mutex lock doesn't seem to be needed this makes sense
// as pool[index] isn't playing and the only stuff we're copying
// shouldn't be changing...
AUDIO.MultiChannel.channels[index] = AUDIO.MultiChannel.poolCounter;
AUDIO.MultiChannel.poolCounter++;
AUDIO.MultiChannel.pool[index]->volume = sound.stream.buffer->volume;
AUDIO.MultiChannel.pool[index]->pitch = sound.stream.buffer->pitch;
AUDIO.MultiChannel.pool[index]->looping = sound.stream.buffer->looping;
AUDIO.MultiChannel.pool[index]->usage = sound.stream.buffer->usage;
AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[0] = false;
AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[1] = false;
AUDIO.MultiChannel.pool[index]->sizeInFrames = sound.stream.buffer->sizeInFrames;
AUDIO.MultiChannel.pool[index]->data = sound.stream.buffer->data;
PlayAudioBuffer(AUDIO.MultiChannel.pool[index]);
}
// Stop any sound played with PlaySoundMulti()
void StopSoundMulti(void)
{
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) StopAudioBuffer(AUDIO.MultiChannel.pool[i]);
}
// Get number of sounds playing in the multichannel buffer pool
int GetSoundsPlaying(void)
{
int counter = 0;
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
{
if (IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i])) counter++;
}
return counter;
}
// 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)
{
return IsAudioBufferPlaying(sound.stream.buffer);
}
// 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);
}
// 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->sampleCount/wave->channels;
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->sampleCount = frameCount*channels;
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.sampleCount*wave.sampleSize/8);
if (newWave.data != NULL)
{
// NOTE: Size must be provided in bytes
memcpy(newWave.data, wave.data, wave.sampleCount*wave.sampleSize/8);
newWave.sampleCount = wave.sampleCount;
newWave.sampleRate = wave.sampleRate;
newWave.sampleSize = wave.sampleSize;
newWave.channels = wave.channels;
}
return newWave;
}
// Crop a wave to defined samples range
// NOTE: Security check in case of out-of-range
void WaveCrop(Wave *wave, int initSample, int finalSample)
{
if ((initSample >= 0) && (initSample < finalSample) &&
(finalSample > 0) && ((unsigned int)finalSample < wave->sampleCount))
{
int sampleCount = finalSample - initSample;
void *data = RL_MALLOC(sampleCount*wave->sampleSize/8);
memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->sampleSize/8);
RL_FREE(wave->data);
wave->data = data;
}
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.sampleCount*sizeof(float));
// NOTE: sampleCount is the total number of interlaced samples (including channels)
for (unsigned int i = 0; i < wave.sampleCount; i++)
{
if (wave.sampleSize == 8) samples[i] = (float)(((unsigned char *)wave.data)[i] - 127)/256.0f;
else if (wave.sampleSize == 16) samples[i] = (float)(((short *)wave.data)[i])/32767.0f;
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 (.OGG)
//----------------------------------------------------------------------------------
// 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);
music.ctxType = MUSIC_AUDIO_WAV;
music.ctxData = ctxWav;
if (success)
{
int sampleSize = ctxWav->bitsPerSample;
if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream()
music.stream = InitAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels);
music.sampleCount = (unsigned int)ctxWav->totalPCMFrameCount*ctxWav->channels;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (IsFileExtension(fileName, ".ogg"))
{
// Open ogg audio stream
music.ctxType = MUSIC_AUDIO_OGG;
music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);
if (music.ctxData != NULL)
{
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 = InitAudioStream(info.sample_rate, 16, info.channels);
// WARNING: It seems this function returns length in frames, not samples, so we multiply by channels
music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (IsFileExtension(fileName, ".flac"))
{
music.ctxType = MUSIC_AUDIO_FLAC;
music.ctxData = drflac_open_file(fileName, NULL);
if (music.ctxData != NULL)
{
drflac *ctxFlac = (drflac *)music.ctxData;
music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music.sampleCount = (unsigned int)ctxFlac->totalPCMFrameCount*ctxFlac->channels;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
}
#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);
music.ctxType = MUSIC_AUDIO_MP3;
music.ctxData = ctxMp3;
if (result > 0)
{
music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
}
#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);
music.ctxType = MUSIC_MODULE_XM;
music.ctxData = ctxXm;
if (result == 0) // XM AUDIO.System.context created successfully
{
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 = InitAudioStream(AUDIO.System.device.sampleRate, bits, AUDIO_DEVICE_CHANNELS);
music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2; // 2 channels
music.looping = true; // Looping enabled by default
jar_xm_reset(ctxXm); // make sure we start at the beginning of the song
musicLoaded = true;
}
}
#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);
music.ctxType = MUSIC_MODULE_MOD;
music.ctxData = ctxMod;
if (result > 0)
{
// NOTE: Only stereo is supported for MOD
music.stream = InitAudioStream(AUDIO.System.device.sampleRate, 16, AUDIO_DEVICE_CHANNELS);
music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2; // 2 channels
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
}
#endif
else TRACELOG(LOG_WARNING, "STREAM: [%s] Fileformat not supported", fileName);
if (!musicLoaded)
{
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_FLAC)
else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
#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_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
music.ctxData = NULL;
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 successfully loaded:", fileName);
TRACELOG(LOG_INFO, " > Total samples: %i", music.sampleCount);
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");
}
return music;
}
// extension including period ".mod"
Music LoadMusicStreamFromMemory(const char *fileType, unsigned char* data, int dataSize)
{
Music music = { 0 };
bool musicLoaded = false;
char fileExtLower[16] = { 0 };
strcpy(fileExtLower, TextToLower(fileType));
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if (TextIsEqual(fileExtLower, ".wav"))
{
drwav *ctxWav = RL_CALLOC(1, sizeof(drwav));
bool success = drwav_init_memory(ctxWav, (const void*)data, dataSize, NULL);
music.ctxType = MUSIC_AUDIO_WAV;
music.ctxData = ctxWav;
if (success)
{
int sampleSize = ctxWav->bitsPerSample;
if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream()
music.stream = InitAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels);
music.sampleCount = (unsigned int)ctxWav->totalPCMFrameCount*ctxWav->channels;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (TextIsEqual(fileExtLower, ".flac"))
{
music.ctxType = MUSIC_AUDIO_FLAC;
music.ctxData = drflac_open_memory((const void*)data, dataSize, NULL);
if (music.ctxData != NULL)
{
drflac *ctxFlac = (drflac *)music.ctxData;
music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music.sampleCount = (unsigned int)ctxFlac->totalPCMFrameCount*ctxFlac->channels;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (TextIsEqual(fileExtLower, ".mp3"))
{
drmp3 *ctxMp3 = RL_CALLOC(1, sizeof(drmp3));
int success = drmp3_init_memory(ctxMp3, (const void*)data, dataSize, NULL);
music.ctxType = MUSIC_AUDIO_MP3;
music.ctxData = ctxMp3;
if (success)
{
music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (TextIsEqual(fileExtLower, ".ogg"))
{
// Open ogg audio stream
music.ctxType = MUSIC_AUDIO_OGG;
//music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);
music.ctxData = stb_vorbis_open_memory((const unsigned char*)data, dataSize, NULL, NULL);
if (music.ctxData != NULL)
{
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 = InitAudioStream(info.sample_rate, 16, info.channels);
// WARNING: It seems this function returns length in frames, not samples, so we multiply by channels
music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
else if (TextIsEqual(fileExtLower, ".xm"))
{
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;
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 = InitAudioStream(AUDIO.System.device.sampleRate, bits, 2);
music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2; // 2 channels
music.looping = true; // Looping enabled by default
jar_xm_reset(ctxXm); // make sure we start at the beginning of the song
music.ctxData = ctxXm;
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
else if (TextIsEqual(fileExtLower, ".mod"))
{
jar_mod_context_t *ctxMod = 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 = 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;
// NOTE: Only stereo is supported for MOD
music.stream = InitAudioStream(AUDIO.System.device.sampleRate, 16, 2);
music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2; // 2 channels
music.looping = true; // Looping enabled by default
musicLoaded = true;
music.ctxData = ctxMod;
musicLoaded = true;
}
}
#endif
else TRACELOG(LOG_WARNING, "STREAM: [%s] Fileformat not supported", fileType);
if (!musicLoaded)
{
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)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_MP3)
else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(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_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
music.ctxData = NULL;
TRACELOG(LOG_WARNING, "FILEIO: [%s] Music memory could not be opened", fileType);
}
else
{
// Show some music stream info
TRACELOG(LOG_INFO, "FILEIO: [%s] Music memory successfully loaded:", fileType);
TRACELOG(LOG_INFO, " > Total samples: %i", music.sampleCount);
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");
}
return music;
}
// Unload music stream
void UnloadMusicStream(Music music)
{
CloseAudioStream(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_FLAC)
else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
#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_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)
void PlayMusicStream(Music music)
{
if (music.stream.buffer != NULL)
{
// For music streams, we need to make sure we maintain the frame cursor position
// This is a hack for this section of code in UpdateMusicStream()
// NOTE: In case window is minimized, music stream is stopped, just make sure to
// play again on window restore: if (IsMusicPlaying(music)) PlayMusicStream(music);
ma_uint32 frameCursorPos = music.stream.buffer->frameCursorPos;
PlayAudioStream(music.stream); // WARNING: This resets the cursor position.
music.stream.buffer->frameCursorPos = frameCursorPos;
}
}
// 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_pcm_frame((drwav *)music.ctxData, 0); 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_FLAC)
case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, 0); break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0); 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;
}
}
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(Music music)
{
if (music.stream.buffer == NULL)
return;
if (music.ctxType == MUSIC_MODULE_XM)
jar_xm_set_max_loop_count(music.ctxData, music.looping ? 0 : 1);
bool streamEnding = false;
unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2;
// NOTE: Using dynamic allocation because it could require more than 16KB
void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1);
int samplesCount = 0; // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts
// TODO: Get the sampleLeft using totalFramesProcessed... but first, get total frames processed correctly...
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
int sampleLeft = music.sampleCount - (music.stream.buffer->totalFramesProcessed*music.stream.channels);
if (music.ctxType == MUSIC_MODULE_XM && music.looping) sampleLeft = subBufferSizeInFrames*4;
while (IsAudioStreamProcessed(music.stream))
{
if ((sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels;
else samplesCount = sampleLeft;
switch (music.ctxType)
{
#if defined(SUPPORT_FILEFORMAT_WAV)
case MUSIC_AUDIO_WAV:
{
// NOTE: Returns the number of samples to process (not required)
if (music.stream.sampleSize == 16) drwav_read_pcm_frames_s16((drwav *)music.ctxData, samplesCount/music.stream.channels, (short *)pcm);
else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
case MUSIC_AUDIO_OGG:
{
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, samplesCount);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_AUDIO_FLAC:
{
// NOTE: Returns the number of samples to process (not required)
drflac_read_pcm_frames_s16((drflac *)music.ctxData, samplesCount, (short *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3:
{
// NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed
drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
case MUSIC_MODULE_XM:
{
switch (AUDIO_DEVICE_FORMAT)
{
case ma_format_f32:
// NOTE: Internally this function considers 2 channels generation, so samplesCount/2
jar_xm_generate_samples((jar_xm_context_t*)music.ctxData, (float*)pcm, samplesCount / 2);
break;
case ma_format_s16:
// NOTE: Internally this function considers 2 channels generation, so samplesCount/2
jar_xm_generate_samples_16bit((jar_xm_context_t*)music.ctxData, (short*)pcm, samplesCount / 2);
break;
case ma_format_u8:
// NOTE: Internally this function considers 2 channels generation, so samplesCount/2
jar_xm_generate_samples_8bit((jar_xm_context_t*)music.ctxData, (char*)pcm, samplesCount / 2);
break;
}
} 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 *)pcm, samplesCount/2, 0);
} break;
#endif
default: break;
}
UpdateAudioStream(music.stream, pcm, samplesCount);
if ((music.ctxType == MUSIC_MODULE_XM) || music.ctxType == MUSIC_MODULE_MOD)
{
if (samplesCount > 1) sampleLeft -= samplesCount/2;
else sampleLeft -= samplesCount;
}
else sampleLeft -= samplesCount;
if (sampleLeft <= 0)
{
streamEnding = true;
break;
}
}
// Free allocated pcm data
RL_FREE(pcm);
// Reset audio stream for looping
if (streamEnding)
{
StopMusicStream(music); // Stop music (and reset)
if (music.looping) PlayMusicStream(music); // Play again
}
else
{
// NOTE: In case window is minimized, music stream is stopped,
// just make sure to play again on window restore
if (IsMusicPlaying(music)) PlayMusicStream(music);
}
}
// Check if any music is playing
bool IsMusicPlaying(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);
}
// Get music time length (in seconds)
float GetMusicTimeLength(Music music)
{
float totalSeconds = 0.0f;
totalSeconds = (float)music.sampleCount/(music.stream.sampleRate*music.stream.channels);
return totalSeconds;
}
// Get current music time played (in seconds)
float GetMusicTimePlayed(Music music)
{
#if defined(SUPPORT_FILEFORMAT_XM)
if (music.ctxType == MUSIC_MODULE_XM)
{
uint64_t samples = 0;
jar_xm_get_position(music.ctxData, NULL, NULL, NULL, &samples);
samples = samples % (music.sampleCount);
return (float)(samples)/(music.stream.sampleRate*music.stream.channels);
}
#endif
float secondsPlayed = 0.0f;
if (music.stream.buffer != NULL)
{
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
unsigned int samplesPlayed = music.stream.buffer->totalFramesProcessed*music.stream.channels;
secondsPlayed = (float)samplesPlayed/(music.stream.sampleRate*music.stream.channels);
}
return secondsPlayed;
}
// Init audio stream (to stream audio pcm data)
AudioStream InitAudioStream(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;
unsigned int subBufferSize = GetAudioStreamBufferSizeDefault();
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;
}
// Close audio stream and free memory
void CloseAudioStream(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: unqueue -> update -> queue
// NOTE 2: To unqueue a buffer it needs to be processed: IsAudioStreamProcessed()
void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
{
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);
// TODO: Get total frames processed on this buffer... DOES NOT WORK.
stream.buffer->totalFramesProcessed += 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)samplesCount/stream.channels)
{
ma_uint32 framesToWrite = subBufferSizeInFrames;
if (framesToWrite > ((ma_uint32)samplesCount/stream.channels)) framesToWrite = (ma_uint32)samplesCount/stream.channels;
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");
}
}
// Check if any audio stream buffers requires refill
bool IsAudioStreamProcessed(AudioStream stream)
{
if (stream.buffer == NULL) return false;
return (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]);
}
// 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);
}
// Default size for new audio streams
void SetAudioStreamBufferSizeDefault(int size)
{
AUDIO.Buffer.defaultSize = size;
}
int GetAudioStreamBufferSizeDefault()
{
// if the buffer is not set, compute one that would give us a buffer good enough for a decent frame rate
if (AUDIO.Buffer.defaultSize == 0)
AUDIO.Buffer.defaultSize = AUDIO.System.device.sampleRate/30;
return AUDIO.Buffer.defaultSize;
}
//----------------------------------------------------------------------------------
// Module specific Functions Definition
//----------------------------------------------------------------------------------
// Log callback function
static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message)
{
(void)pContext;
(void)pDevice;
TRACELOG(LOG_WARNING, "miniaudio: %s", message); // 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)
{
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];
isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0];
isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1];
ma_uint32 frameSizeInBytes = ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.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 the halves 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)
{
StopAudioBuffer(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 or not 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];
ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn);
ma_uint32 totalOutputFramesProcessed = 0;
while (totalOutputFramesProcessed < frameCount)
{
ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed;
ma_uint64 inputFramesToProcessThisIteration = ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration);
if (inputFramesToProcessThisIteration > inputBufferFrameCap)
{
inputFramesToProcessThisIteration = inputBufferFrameCap;
}
float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.config.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
// 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]; // 512 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;
MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume);
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)
{
StopAudioBuffer(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;
}
}
}
ma_mutex_unlock(&AUDIO.System.lock);
}
// This is the main mixing function. Mixing is pretty simple in this project - it's just an accumulation.
// NOTE: framesOut is both an input and an output. It will be initially filled with zeros outside of this function.
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume)
{
for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
{
for (ma_uint32 iChannel = 0; iChannel < AUDIO.System.device.playback.channels; ++iChannel)
{
float *frameOut = framesOut + (iFrame*AUDIO.System.device.playback.channels);
const float *frameIn = framesIn + (iFrame*AUDIO.System.device.playback.channels);
frameOut[iChannel] += (frameIn[iChannel]*localVolume);
}
}
}
#if defined(SUPPORT_FILEFORMAT_WAV)
// Load WAV file data into Wave structure
// NOTE: Using dr_wav library
static Wave LoadWAV(const unsigned char *fileData, unsigned int fileSize)
{
Wave wave = { 0 };
drwav wav = { 0 };
bool success = drwav_init_memory(&wav, fileData, fileSize, NULL);
if (success)
{
wave.sampleCount = (unsigned int)wav.totalPCMFrameCount*wav.channels;
wave.sampleRate = wav.sampleRate;
wave.sampleSize = 16; // NOTE: We are forcing conversion to 16bit
wave.channels = wav.channels;
wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short));
drwav_read_pcm_frames_s16(&wav, wav.totalPCMFrameCount, wave.data);
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load WAV data");
drwav_uninit(&wav);
return wave;
}
// Save wave data as WAV file
// NOTE: Using dr_wav library
static int SaveWAV(Wave wave, const char *fileName)
{
int success = false;
drwav wav = { 0 };
drwav_data_format format = { 0 };
format.container = drwav_container_riff;
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.sampleCount/wave.channels, 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);
return success;
}
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
// Load OGG file data into Wave structure
// NOTE: Using stb_vorbis library
static Wave LoadOGG(const unsigned char *fileData, unsigned int fileSize)
{
Wave wave = { 0 };
stb_vorbis *oggData = stb_vorbis_open_memory((unsigned char *)fileData, fileSize, NULL, NULL);
if (oggData != NULL)
{
stb_vorbis_info info = stb_vorbis_get_info(oggData);
wave.sampleRate = info.sample_rate;
wave.sampleSize = 16; // 16 bit per sample (short)
wave.channels = info.channels;
wave.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggData)*info.channels; // Independent by channel
float totalSeconds = stb_vorbis_stream_length_in_seconds(oggData);
if (totalSeconds > 10) TRACELOG(LOG_WARNING, "WAVE: OGG audio length larger than 10 seconds (%f sec.), that's a big file in memory, consider music streaming", totalSeconds);
wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short));
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
stb_vorbis_get_samples_short_interleaved(oggData, info.channels, (short *)wave.data, wave.sampleCount);
TRACELOG(LOG_INFO, "WAVE: OGG data loaded successfully (%i Hz, %i bit, %s)", wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
stb_vorbis_close(oggData);
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load OGG data");
return wave;
}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
// Load FLAC file data into Wave structure
// NOTE: Using dr_flac library
static Wave LoadFLAC(const unsigned char *fileData, unsigned int fileSize)
{
Wave wave = { 0 };
// Decode the entire FLAC file in one go
unsigned long long int totalFrameCount = 0;
wave.data = drflac_open_memory_and_read_pcm_frames_s16(fileData, fileSize, &wave.channels, &wave.sampleRate, &totalFrameCount, NULL);
if (wave.data != NULL)
{
wave.sampleCount = (unsigned int)totalFrameCount*wave.channels;
wave.sampleSize = 16;
TRACELOG(LOG_INFO, "WAVE: FLAC data loaded successfully (%i Hz, %i bit, %s)", wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load FLAC data");
return wave;
}
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
// Load MP3 file data into Wave structure
// NOTE: Using dr_mp3 library
static Wave LoadMP3(const unsigned char *fileData, unsigned int fileSize)
{
Wave wave = { 0 };
drmp3_config config = { 0 };
// Decode the entire MP3 file in one go
unsigned long long int totalFrameCount = 0;
wave.data = drmp3_open_memory_and_read_pcm_frames_f32(fileData, fileSize, &config, &totalFrameCount, NULL);
if (wave.data != NULL)
{
wave.channels = config.channels;
wave.sampleRate = config.sampleRate;
wave.sampleCount = (int)totalFrameCount*wave.channels;
wave.sampleSize = 32;
// NOTE: Only support up to 2 channels (mono, stereo)
// TODO: Really?
if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: MP3 channels number (%i) not supported", wave.channels);
TRACELOG(LOG_INFO, "WAVE: MP3 file loaded successfully (%i Hz, %i bit, %s)", wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load MP3 data");
return wave;
}
#endif
// 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;
}
// Load data from file into a buffer
static unsigned char *LoadFileData(const char *fileName, unsigned int *bytesRead)
{
unsigned char *data = NULL;
*bytesRead = 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);
*bytesRead = 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, unsigned int bytesToWrite)
{
if (fileName != NULL)
{
FILE *file = fopen(fileName, "wb");
if (file != NULL)
{
unsigned int count = (unsigned int)fwrite(data, sizeof(unsigned char), bytesToWrite, file);
if (count == 0) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to write file", fileName);
else if (count != bytesToWrite) 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);
}
else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid");
}
// 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);
}
else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid");
}
#endif
#undef AudioBuffer
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