mirror
/
Shipwright
mirror of https://github.com/HarbourMasters/Shipwright.git
1
0
Fork 0
Code Issues Projects Releases Wiki Activity
Shipwright/OTRGui/libs/raylib/src/raudio.c

2413 lines
92 KiB
C
Raw Blame History

/**********************************************************************************************
*
* 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 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_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/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) 2013-2022 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(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 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, ...) (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 stb_vorbis 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 sample rate
#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, // 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_MODULE_XM, // XM module audio context
MUSIC_MODULE_MOD // MOD module audio context
} MusicContextType;
#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
// 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
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, 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 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
};
#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, AudioBuffer *buffer);
#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 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 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();
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 audio buffer 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; // 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->pan = 0.5f;
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->framesProcessed = 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;
}
}
// 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) buffer->pan = pan;
}
// 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);
// Loading wave from memory data
if (fileData != NULL) wave = LoadWaveFromMemory(GetFileExtension(fileName), fileData, fileSize);
RL_FREE(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)
{
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)
{
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_FLAC)
else if (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
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (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
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;
}
// 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;
}
// 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");
}
// Update sound buffer with new data
void UpdateSound(Sound sound, const void *data, int sampleCount)
{
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, sampleCount*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"))
{
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
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 6 char bytes for every byte: "0x00, "
char *txtData = (char *)RL_CALLOC(waveDataSize*6 + 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) 2018-2022 Ramon Santamaria (@raysan5) //\n");
byteCount += sprintf(txtData + byteCount, "// //\n");
byteCount += sprintf(txtData + byteCount, "//////////////////////////////////////////////////////////////////////////////////\n\n");
char fileNameLower[256] = { 0 };
char fileNameUpper[256] = { 0 };
for (int i = 0; fileName[i] != '.'; i++) { fileNameLower[i] = fileName[i]; } // Get filename without extension
for (int i = 0; fileNameLower[i] != '\0'; i++) if (fileNameLower[i] >= 'a' && fileNameLower[i] <= 'z') { fileNameUpper[i] = fileNameLower[i] - 32; }
byteCount += sprintf(txtData + byteCount, "// Wave data information\n");
byteCount += sprintf(txtData + byteCount, "#define %s_FRAME_COUNT %u\n", fileNameUpper, wave.frameCount);
byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_RATE %u\n", fileNameUpper, wave.sampleRate);
byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_SIZE %u\n", fileNameUpper, wave.sampleSize);
byteCount += sprintf(txtData + byteCount, "#define %s_CHANNELS %u\n\n", fileNameUpper, 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 %sData[%i] = {\n", fileNameLower, 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 %sData[%i] = { ", fileNameLower, 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);
}
// 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++;
SetAudioBufferVolume(AUDIO.MultiChannel.pool[index], sound.stream.buffer->volume);
SetAudioBufferPitch(AUDIO.MultiChannel.pool[index], sound.stream.buffer->pitch);
SetAudioBufferPan(AUDIO.MultiChannel.pool[index], sound.stream.buffer->pan);
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);
}
// 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 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->frameCount*wave->channels)))
{
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.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] - 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 = LoadAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels);
music.frameCount = (unsigned int)ctxWav->totalPCMFrameCount;
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 = 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;
}
}
#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 = LoadAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
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 = 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;
}
}
#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 = 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;
}
}
#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 = 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;
}
}
#endif
else TRACELOG(LOG_WARNING, "STREAM: [%s] File format 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 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)
{
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 = LoadAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels);
music.frameCount = (unsigned int)ctxWav->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (strcmp(fileType, ".flac") == 0)
{
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 = LoadAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (strcmp(fileType, ".mp3") == 0)
{
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 = 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;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (strcmp(fileType, ".ogg") == 0)
{
// 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 = 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;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
else if (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;
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
music.ctxData = ctxXm;
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
else if (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;
// 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;
music.ctxData = ctxMod;
musicLoaded = true;
}
}
#endif
else TRACELOG(LOG_WARNING, "STREAM: Data format not supported");
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: 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;
}
// 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_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 (IsMusicStreamPlaying(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;
}
}
// 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_FLAC)
case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)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
default: break;
}
music.stream.buffer->framesProcessed = positionInFrames;
}
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(Music music)
{
if (music.stream.buffer == NULL) return;
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 frameCountToStream = 0; // Total size of data in frames to be streamed
// TODO: Get the framesLeft using framesProcessed... 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;
unsigned int framesLeft = music.frameCount - music.stream.buffer->framesProcessed;
while (IsAudioStreamProcessed(music.stream))
{
if (framesLeft >= subBufferSizeInFrames) frameCountToStream = subBufferSizeInFrames;
else frameCountToStream = framesLeft;
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, frameCountToStream, (short *)pcm);
else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountToStream, (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, frameCountToStream*music.stream.channels);
} 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, frameCountToStream*music.stream.channels, (short *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3:
{
drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountToStream, (float *)pcm);
} 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 *)pcm, frameCountToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, frameCountToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)pcm, frameCountToStream);
} 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, frameCountToStream, 0);
} break;
#endif
default: break;
}
UpdateAudioStream(music.stream, pcm, frameCountToStream);
framesLeft -= frameCountToStream;
if (framesLeft <= 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 (IsMusicStreamPlaying(music)) PlayMusicStream(music);
}
}
// 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_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
unsigned int framesPlayed = music.stream.buffer->framesProcessed;
secondsPlayed = (float)framesPlayed/music.stream.sampleRate;
}
}
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;
}
// 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: unqueue -> update -> queue
// NOTE 2: To unqueue a buffer it needs to be processed: IsAudioStreamProcessed()
void UpdateAudioStream(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);
// TODO: Get total frames processed on this buffer... DOES NOT WORK.
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 = subBufferSizeInFrames;
if (framesToWrite > (ma_uint32)frameCount) 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");
}
}
// 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);
}
// 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;
}
//----------------------------------------------------------------------------------
// 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] = { 0 };
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] = { 0 };
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
// 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;
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)
{
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, AudioBuffer *buffer)
{
const float localVolume = buffer->volume;
const ma_uint32 nChannels = AUDIO.System.device.playback.channels;
if (nChannels == 2)
{
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 iFrame = 0; iFrame < frameCount; ++iFrame)
{
frameOut[0] += (frameIn[0]*levels[0]);
frameOut[1] += (frameIn[1]*levels[1]);
frameOut += 2;
frameIn += 2;
}
}
else // pan is kinda meaningless
{
for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
{
for (ma_uint32 iChannel = 0; iChannel < nChannels; ++iChannel)
{
float *frameOut = framesOut + (iFrame * nChannels);
const float *frameIn = framesIn + (iFrame * nChannels);
frameOut[iChannel] += (frameIn[iChannel] * localVolume);
}
}
}
}
// 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;
}
// 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
#endif // SUPPORT_MODULE_RAUDIO
Reference in New Issue View Git Blame Copy Permalink