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raylib/src/audio.c

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/**********************************************************************************************
*
* raylib.audio - Basic funtionality to work with audio
*
* FEATURES:
* - Manage audio device (init/close)
* - Load and unload audio files
* - Format wave data (sample rate, size, channels)
* - Play/Stop/Pause/Resume loaded audio
* - Manage mixing channels
* - Manage raw audio context
*
* CONFIGURATION:
*
* #define AUDIO_STANDALONE
* Define to use the module as standalone library (independently of raylib).
* Required types and functions are defined in the same module.
*
* #define USE_OPENAL_BACKEND
* Use OpenAL Soft audio backend usage
*
* #define SUPPORT_FILEFORMAT_WAV
* #define SUPPORT_FILEFORMAT_OGG
* #define SUPPORT_FILEFORMAT_XM
* #define SUPPORT_FILEFORMAT_MOD
* #define SUPPORT_FILEFORMAT_FLAC
* 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
*
* LIMITATIONS (only OpenAL Soft):
* Only up to two channels supported: MONO and STEREO (for additional channels, use AL_EXT_MCFORMATS)
* Only the following sample sizes supported: 8bit PCM, 16bit PCM, 32-bit float PCM (using AL_EXT_FLOAT32)
*
* DEPENDENCIES:
* mini_al - Audio device/context management (https://github.com/dr-soft/mini_al)
* stb_vorbis - OGG audio files loading (http://www.nothings.org/stb_vorbis/)
* jar_xm - XM module file loading
* jar_mod - MOD audio file loading
* dr_flac - FLAC audio file loading
*
* *OpenAL Soft - Audio device management, still used on HTML5 and OSX platforms
*
* CONTRIBUTORS:
* David Reid (github: @mackron) (Nov. 2017):
* - Complete port to mini_al 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) 2014-2018 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.
*
**********************************************************************************************/
// Default configuration flags (supported features)
//-------------------------------------------------
#define SUPPORT_FILEFORMAT_WAV
#define SUPPORT_FILEFORMAT_OGG
#define SUPPORT_FILEFORMAT_XM
#define SUPPORT_FILEFORMAT_MOD
//-------------------------------------------------
#if !defined(USE_OPENAL_BACKEND)
#define USE_MINI_AL 1 // Set to 1 to use mini_al; 0 to use OpenAL.
#endif
#if defined(AUDIO_STANDALONE)
#include "audio.h"
#include <stdarg.h> // Required for: va_list, va_start(), vfprintf(), va_end()
#else
#include "raylib.h"
#include "utils.h" // Required for: fopen() Android mapping
#endif
#include "external/mini_al.h" // Implemented in mini_al.c. Cannot implement this here because it conflicts with Win32 APIs such as CloseWindow(), etc.
#if !defined(USE_MINI_AL) || (USE_MINI_AL == 0)
#if defined(__APPLE__)
#include "OpenAL/al.h" // OpenAL basic header
#include "OpenAL/alc.h" // OpenAL context header (like OpenGL, OpenAL requires a context to work)
#else
#include "AL/al.h" // OpenAL basic header
#include "AL/alc.h" // OpenAL context header (like OpenGL, OpenAL requires a context to work)
//#include "AL/alext.h" // OpenAL extensions header, required for AL_EXT_FLOAT32 and AL_EXT_MCFORMATS
#endif
// OpenAL extension: AL_EXT_FLOAT32 - Support for 32bit float samples
// OpenAL extension: AL_EXT_MCFORMATS - Support for multi-channel formats (Quad, 5.1, 6.1, 7.1)
#endif
#include <stdlib.h> // Required for: malloc(), free()
#include <string.h> // Required for: strcmp(), strncmp()
#include <stdio.h> // Required for: FILE, fopen(), fclose(), fread()
#if defined(SUPPORT_FILEFORMAT_OGG)
//#define STB_VORBIS_HEADER_ONLY
#include "external/stb_vorbis.h" // OGG loading functions
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
#define JAR_XM_IMPLEMENTATION
#include "external/jar_xm.h" // XM loading functions
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
#define JAR_MOD_IMPLEMENTATION
#include "external/jar_mod.h" // MOD loading functions
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
#define DR_FLAC_IMPLEMENTATION
#define DR_FLAC_NO_WIN32_IO
#include "external/dr_flac.h" // FLAC loading functions
#endif
#ifdef _MSC_VER
#undef bool
#endif
//----------------------------------------------------------------------------------
// Defines and Macros
//----------------------------------------------------------------------------------
#define MAX_STREAM_BUFFERS 2 // Number of buffers for each audio stream
// 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 double-buffering system, I concluded that a 4096 samples buffer should be enough
// In case of music-stalls, just increase this number
#define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. 16bit, Mono: 8Kb)
// Support uncompressed PCM data in 32-bit float IEEE format
// NOTE: This definition is included in "AL/alext.h", but some OpenAL implementations
// could not provide the extensions header (Android), so its defined here
#if !defined(AL_EXT_float32)
#define AL_EXT_float32 1
#define AL_FORMAT_MONO_FLOAT32 0x10010
#define AL_FORMAT_STEREO_FLOAT32 0x10011
#endif
//----------------------------------------------------------------------------------
// Types and Structures Definition
//----------------------------------------------------------------------------------
typedef enum {
MUSIC_AUDIO_OGG = 0,
MUSIC_AUDIO_FLAC,
MUSIC_MODULE_XM,
MUSIC_MODULE_MOD
} MusicContextType;
// Music type (file streaming from memory)
typedef struct MusicData {
MusicContextType ctxType; // Type of music context (OGG, XM, MOD)
#if defined(SUPPORT_FILEFORMAT_OGG)
stb_vorbis *ctxOgg; // OGG audio context
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
drflac *ctxFlac; // FLAC audio context
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
jar_xm_context_t *ctxXm; // XM chiptune context
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
jar_mod_context_t ctxMod; // MOD chiptune context
#endif
AudioStream stream; // Audio stream (double buffering)
int loopCount; // Loops count (times music repeats), -1 means infinite loop
unsigned int totalSamples; // Total number of samples
unsigned int samplesLeft; // Number of samples left to end
} MusicData;
#if defined(AUDIO_STANDALONE)
typedef enum {
LOG_INFO = 0,
LOG_ERROR,
LOG_WARNING,
LOG_DEBUG,
LOG_OTHER
} TraceLogType;
#endif
//----------------------------------------------------------------------------------
// Global Variables Definition
//----------------------------------------------------------------------------------
// ...
//----------------------------------------------------------------------------------
// Module specific Functions Declaration
//----------------------------------------------------------------------------------
#if defined(SUPPORT_FILEFORMAT_WAV)
static Wave LoadWAV(const char *fileName); // Load WAV file
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
static Wave LoadOGG(const char *fileName); // Load OGG file
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
static Wave LoadFLAC(const char *fileName); // Load FLAC file
#endif
#if defined(AUDIO_STANDALONE)
bool IsFileExtension(const char *fileName, const char *ext); // Check file extension
void TraceLog(int msgType, const char *text, ...); // Show trace log messages (LOG_INFO, LOG_WARNING, LOG_ERROR, LOG_DEBUG)
#endif
//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Device initialization and Closing
//----------------------------------------------------------------------------------
#if USE_MINI_AL
#define DEVICE_FORMAT mal_format_f32
#define DEVICE_CHANNELS 2
#define DEVICE_SAMPLE_RATE 44100
typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioBufferUsage;
// Audio buffer structure
typedef struct AudioBuffer AudioBuffer;
struct AudioBuffer {
mal_dsp dsp; // For format conversion.
float volume;
float pitch;
bool playing;
bool paused;
bool looping; // Always true for AudioStreams.
AudioBufferUsage usage; // Slightly different logic is used when feeding data to the playback device depending on whether or not data is streamed.
bool isSubBufferProcessed[2];
unsigned int frameCursorPos;
unsigned int bufferSizeInFrames;
AudioBuffer *next;
AudioBuffer *prev;
unsigned char buffer[1];
};
void StopAudioBuffer(AudioBuffer *audioBuffer);
static mal_context context;
static mal_device device;
static mal_bool32 isAudioInitialized = MAL_FALSE;
static float masterVolume = 1;
static mal_mutex audioLock;
static AudioBuffer *firstAudioBuffer = NULL; // Audio buffers are tracked in a linked list.
static AudioBuffer *lastAudioBuffer = NULL;
static void TrackAudioBuffer(AudioBuffer* audioBuffer)
{
mal_mutex_lock(&audioLock);
{
if (firstAudioBuffer == NULL) firstAudioBuffer = audioBuffer;
else
{
lastAudioBuffer->next = audioBuffer;
audioBuffer->prev = lastAudioBuffer;
}
lastAudioBuffer = audioBuffer;
}
mal_mutex_unlock(&audioLock);
}
static void UntrackAudioBuffer(AudioBuffer* audioBuffer)
{
mal_mutex_lock(&audioLock);
{
if (audioBuffer->prev == NULL) firstAudioBuffer = audioBuffer->next;
else audioBuffer->prev->next = audioBuffer->next;
if (audioBuffer->next == NULL) lastAudioBuffer = audioBuffer->prev;
else audioBuffer->next->prev = audioBuffer->prev;
audioBuffer->prev = NULL;
audioBuffer->next = NULL;
}
mal_mutex_unlock(&audioLock);
}
static void OnLog_MAL(mal_context *pContext, mal_device *pDevice, const char *message)
{
(void)pContext;
(void)pDevice;
TraceLog(LOG_ERROR, message); // All log messages from mini_al are errors.
}
// This is the main mixing function. Mixing is pretty simple in this project - it's just an accumulation.
//
// framesOut is both an input and an output. It will be initially filled with zeros outside of this function.
static void MixFrames(float* framesOut, const float* framesIn, mal_uint32 frameCount, float localVolume)
{
for (mal_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
{
for (mal_uint32 iChannel = 0; iChannel < device.channels; ++iChannel)
{
float *frameOut = framesOut + (iFrame*device.channels);
const float *frameIn = framesIn + (iFrame*device.channels);
frameOut[iChannel] += frameIn[iChannel]*masterVolume*localVolume;
}
}
}
static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameCount, void *pFramesOut)
{
// This is where all of the mixing takes place.
(void)pDevice;
// Mixing is basically just an accumulation. We need to initialize the output buffer to 0.
memset(pFramesOut, 0, frameCount*pDevice->channels*mal_get_sample_size_in_bytes(pDevice->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.
mal_mutex_lock(&audioLock);
{
for (AudioBuffer* audioBuffer = firstAudioBuffer; audioBuffer != NULL; audioBuffer = audioBuffer->next)
{
// Ignore stopped or paused sounds.
if (!audioBuffer->playing || audioBuffer->paused) continue;
mal_uint32 framesRead = 0;
for (;;)
{
if (framesRead > frameCount)
{
TraceLog(LOG_DEBUG, "Mixed too many frames from audio buffer");
break;
}
if (framesRead == frameCount) break;
// Just read as much data as we can from the stream.
mal_uint32 framesToRead = (frameCount - framesRead);
while (framesToRead > 0)
{
float tempBuffer[1024]; // 512 frames for stereo.
mal_uint32 framesToReadRightNow = framesToRead;
if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS)
{
framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS;
}
// If we're not looping, we need to make sure we flush the internal buffers of the DSP pipeline to ensure we get the
// last few samples.
mal_bool32 flushDSP = !audioBuffer->looping;
mal_uint32 framesJustRead = mal_dsp_read_frames_ex(&audioBuffer->dsp, framesToReadRightNow, tempBuffer, flushDSP);
if (framesJustRead > 0)
{
float *framesOut = (float *)pFramesOut + (framesRead*device.channels);
float *framesIn = tempBuffer;
MixFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume);
framesToRead -= framesJustRead;
framesRead += framesJustRead;
}
// 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;
}
}
}
mal_mutex_unlock(&audioLock);
return frameCount; // We always output the same number of frames that were originally requested.
}
#endif
// Initialize audio device
void InitAudioDevice(void)
{
#if USE_MINI_AL
// Context.
mal_context_config contextConfig = mal_context_config_init(OnLog_MAL);
mal_result result = mal_context_init(NULL, 0, &contextConfig, &context);
if (result != MAL_SUCCESS)
{
TraceLog(LOG_ERROR, "Failed to initialize audio context");
return;
}
// Device. Using the default device. Format is floating point because it simplifies mixing.
mal_device_config deviceConfig = mal_device_config_init(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, OnSendAudioDataToDevice);
result = mal_device_init(&context, mal_device_type_playback, NULL, &deviceConfig, NULL, &device);
if (result != MAL_SUCCESS)
{
TraceLog(LOG_ERROR, "Failed to initialize audio playback device");
mal_context_uninit(&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 = mal_device_start(&device);
if (result != MAL_SUCCESS)
{
TraceLog(LOG_ERROR, "Failed to start audio playback device");
mal_device_uninit(&device);
mal_context_uninit(&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 (mal_mutex_init(&context, &audioLock) != MAL_SUCCESS)
{
TraceLog(LOG_ERROR, "Failed to create mutex for audio mixing");
mal_device_uninit(&device);
mal_context_uninit(&context);
return;
}
TraceLog(LOG_INFO, "Audio device initialized successfully: %s", device.name);
TraceLog(LOG_INFO, "Audio backend: mini_al / %s", mal_get_backend_name(context.backend));
TraceLog(LOG_INFO, "Audio format: %s -> %s", mal_get_format_name(device.format), mal_get_format_name(device.internalFormat));
TraceLog(LOG_INFO, "Audio channels: %d -> %d", device.channels, device.internalChannels);
TraceLog(LOG_INFO, "Audio sample rate: %d -> %d", device.sampleRate, device.internalSampleRate);
TraceLog(LOG_INFO, "Audio buffer size: %d", device.bufferSizeInFrames);
isAudioInitialized = MAL_TRUE;
#else
// Open and initialize a device with default settings
ALCdevice *device = alcOpenDevice(NULL);
if (!device) TraceLog(LOG_ERROR, "Audio device could not be opened");
else
{
ALCcontext *context = alcCreateContext(device, NULL);
if ((context == NULL) || (alcMakeContextCurrent(context) == ALC_FALSE))
{
if (context != NULL) alcDestroyContext(context);
alcCloseDevice(device);
TraceLog(LOG_ERROR, "Could not initialize audio context");
}
else
{
TraceLog(LOG_INFO, "Audio device and context initialized successfully: %s", alcGetString(device, ALC_DEVICE_SPECIFIER));
// Listener definition (just for 2D)
alListener3f(AL_POSITION, 0.0f, 0.0f, 0.0f);
alListener3f(AL_VELOCITY, 0.0f, 0.0f, 0.0f);
alListener3f(AL_ORIENTATION, 0.0f, 0.0f, -1.0f);
alListenerf(AL_GAIN, 1.0f);
if (alIsExtensionPresent("AL_EXT_float32")) TraceLog(LOG_INFO, "[EXTENSION] AL_EXT_float32 supported");
else TraceLog(LOG_INFO, "[EXTENSION] AL_EXT_float32 not supported");
}
}
#endif
}
// Close the audio device for all contexts
void CloseAudioDevice(void)
{
#if USE_MINI_AL
if (!isAudioInitialized)
{
TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized");
return;
}
mal_mutex_uninit(&audioLock);
mal_device_uninit(&device);
mal_context_uninit(&context);
#else
ALCdevice *device;
ALCcontext *context = alcGetCurrentContext();
if (context == NULL) TraceLog(LOG_WARNING, "Could not get current audio context for closing");
device = alcGetContextsDevice(context);
alcMakeContextCurrent(NULL);
alcDestroyContext(context);
alcCloseDevice(device);
#endif
TraceLog(LOG_INFO, "Audio device closed successfully");
}
// Check if device has been initialized successfully
bool IsAudioDeviceReady(void)
{
#if USE_MINI_AL
return isAudioInitialized;
#else
ALCcontext *context = alcGetCurrentContext();
if (context == NULL) return false;
else
{
ALCdevice *device = alcGetContextsDevice(context);
if (device == NULL) return false;
else return true;
}
#endif
}
// Set master volume (listener)
void SetMasterVolume(float volume)
{
if (volume < 0.0f) volume = 0.0f;
else if (volume > 1.0f) volume = 1.0f;
#if USE_MINI_AL
masterVolume = 1;
#else
alListenerf(AL_GAIN, volume);
#endif
}
//----------------------------------------------------------------------------------
// Audio Buffer
//----------------------------------------------------------------------------------
#if USE_MINI_AL
static mal_uint32 AudioBuffer_OnDSPRead(mal_dsp* pDSP, mal_uint32 frameCount, void* pFramesOut, void* pUserData)
{
AudioBuffer *audioBuffer = (AudioBuffer *)pUserData;
mal_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2;
mal_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames;
if (currentSubBufferIndex > 1)
{
TraceLog(LOG_DEBUG, "Frame cursor position moved too far forward in audio stream");
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];
mal_uint32 frameSizeInBytes = mal_get_sample_size_in_bytes(audioBuffer->dsp.config.formatIn)*audioBuffer->dsp.config.channelsIn;
// Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0.
mal_uint32 framesRead = 0;
for (;;)
{
// 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;
}
mal_uint32 totalFramesRemaining = (frameCount - framesRead);
if (totalFramesRemaining == 0) break;
mal_uint32 framesRemainingInOutputBuffer;
if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
{
framesRemainingInOutputBuffer = audioBuffer->bufferSizeInFrames - audioBuffer->frameCursorPos;
}
else
{
mal_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames * currentSubBufferIndex;
framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer);
}
mal_uint32 framesToRead = totalFramesRemaining;
if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer;
memcpy((unsigned char *)pFramesOut + (framesRead*frameSizeInBytes), audioBuffer->buffer + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes);
audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead) % audioBuffer->bufferSizeInFrames;
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.
mal_uint32 totalFramesRemaining = (frameCount - framesRead);
if (totalFramesRemaining > 0)
{
memset((unsigned char*)pFramesOut + (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;
}
// Create a new audio buffer. Initially filled with silence.
AudioBuffer *CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint32 sampleRate, mal_uint32 bufferSizeInFrames, AudioBufferUsage usage)
{
AudioBuffer *audioBuffer = (AudioBuffer *)calloc(sizeof(*audioBuffer) + (bufferSizeInFrames*channels*mal_get_sample_size_in_bytes(format)), 1);
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "CreateAudioBuffer() : Failed to allocate memory for audio buffer");
return NULL;
}
// We run audio data through a format converter.
mal_dsp_config dspConfig;
memset(&dspConfig, 0, sizeof(dspConfig));
dspConfig.formatIn = format;
dspConfig.formatOut = DEVICE_FORMAT;
dspConfig.channelsIn = channels;
dspConfig.channelsOut = DEVICE_CHANNELS;
dspConfig.sampleRateIn = sampleRate;
dspConfig.sampleRateOut = DEVICE_SAMPLE_RATE;
mal_result resultMAL = mal_dsp_init(&dspConfig, AudioBuffer_OnDSPRead, audioBuffer, &audioBuffer->dsp);
if (resultMAL != MAL_SUCCESS)
{
TraceLog(LOG_ERROR, "LoadSoundFromWave() : Failed to create data conversion pipeline");
free(audioBuffer);
return NULL;
}
audioBuffer->volume = 1;
audioBuffer->pitch = 1;
audioBuffer->playing = 0;
audioBuffer->paused = 0;
audioBuffer->looping = 0;
audioBuffer->usage = usage;
audioBuffer->bufferSizeInFrames = bufferSizeInFrames;
audioBuffer->frameCursorPos = 0;
// 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;
TrackAudioBuffer(audioBuffer);
return audioBuffer;
}
// Delete an audio buffer.
void DeleteAudioBuffer(AudioBuffer *audioBuffer)
{
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
return;
}
UntrackAudioBuffer(audioBuffer);
free(audioBuffer);
}
// Check if an audio buffer is playing.
bool IsAudioBufferPlaying(AudioBuffer *audioBuffer)
{
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
return false;
}
return audioBuffer->playing && !audioBuffer->paused;
}
// Play an audio buffer.
//
// This will restart the buffer from the start. Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position
// should be maintained.
void PlayAudioBuffer(AudioBuffer *audioBuffer)
{
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
return;
}
audioBuffer->playing = true;
audioBuffer->paused = false;
audioBuffer->frameCursorPos = 0;
}
// Stop an audio buffer.
void StopAudioBuffer(AudioBuffer *audioBuffer)
{
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
return;
}
// Don't do anything if the audio buffer is already stopped.
if (!IsAudioBufferPlaying(audioBuffer)) return;
audioBuffer->playing = false;
audioBuffer->paused = false;
audioBuffer->frameCursorPos = 0;
audioBuffer->isSubBufferProcessed[0] = true;
audioBuffer->isSubBufferProcessed[1] = true;
}
// Pause an audio buffer.
void PauseAudioBuffer(AudioBuffer *audioBuffer)
{
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
return;
}
audioBuffer->paused = true;
}
// Resume an audio buffer.
void ResumeAudioBuffer(AudioBuffer *audioBuffer)
{
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
return;
}
audioBuffer->paused = false;
}
// Set volume for an audio buffer.
void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume)
{
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
return;
}
audioBuffer->volume = volume;
}
// Set pitch for an audio buffer.
void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch)
{
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
return;
}
audioBuffer->pitch = pitch;
// 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.
mal_uint32 newOutputSampleRate = (mal_uint32)((((float)audioBuffer->dsp.config.sampleRateOut / (float)audioBuffer->dsp.config.sampleRateIn) / pitch) * audioBuffer->dsp.config.sampleRateIn);
mal_dsp_set_output_sample_rate(&audioBuffer->dsp, newOutputSampleRate);
}
#endif
//----------------------------------------------------------------------------------
// Module Functions Definition - Sounds loading and playing (.WAV)
//----------------------------------------------------------------------------------
// Load wave data from file
Wave LoadWave(const char *fileName)
{
Wave wave = { 0 };
if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName);
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (IsFileExtension(fileName, ".ogg")) wave = LoadOGG(fileName);
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (IsFileExtension(fileName, ".flac")) wave = LoadFLAC(fileName);
#endif
else TraceLog(LOG_WARNING, "[%s] Audio fileformat not supported, it can't be loaded", fileName);
return wave;
}
// Load wave data from raw array data
Wave LoadWaveEx(void *data, int sampleCount, int sampleRate, int sampleSize, int channels)
{
Wave wave;
wave.data = data;
wave.sampleCount = sampleCount;
wave.sampleRate = sampleRate;
wave.sampleSize = sampleSize;
wave.channels = channels;
// NOTE: Copy wave data to work with, user is responsible of input data to free
Wave cwave = WaveCopy(wave);
WaveFormat(&cwave, sampleRate, sampleSize, channels);
return cwave;
}
// 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)
{
#if USE_MINI_AL
// When using mini_al 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 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.
//
// I have decided on the first option because it offloads work required for the format conversion to the to the loading stage. The
// downside to this is that it uses more memory if the original sound is u8 or s16.
mal_format formatIn = ((wave.sampleSize == 8) ? mal_format_u8 : ((wave.sampleSize == 16) ? mal_format_s16 : mal_format_f32));
mal_uint32 frameCountIn = wave.sampleCount; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so.
mal_uint32 frameCount = mal_convert_frames(NULL, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, formatIn, wave.channels, wave.sampleRate, frameCountIn);
if (frameCount == 0) TraceLog(LOG_ERROR, "LoadSoundFromWave() : Failed to get frame count for format conversion");
AudioBuffer* audioBuffer = CreateAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC);
if (audioBuffer == NULL) TraceLog(LOG_ERROR, "LoadSoundFromWave() : Failed to create audio buffer");
frameCount = mal_convert_frames(audioBuffer->buffer, audioBuffer->dsp.config.formatIn, audioBuffer->dsp.config.channelsIn, audioBuffer->dsp.config.sampleRateIn, wave.data, formatIn, wave.channels, wave.sampleRate, frameCountIn);
if (frameCount == 0) TraceLog(LOG_ERROR, "LoadSoundFromWave() : Format conversion failed");
sound.audioBuffer = audioBuffer;
#else
ALenum format = 0;
// The OpenAL format is worked out by looking at the number of channels and the sample size (bits per sample)
if (wave.channels == 1)
{
switch (wave.sampleSize)
{
case 8: format = AL_FORMAT_MONO8; break;
case 16: format = AL_FORMAT_MONO16; break;
case 32: format = AL_FORMAT_MONO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32
default: TraceLog(LOG_WARNING, "Wave sample size not supported: %i", wave.sampleSize); break;
}
}
else if (wave.channels == 2)
{
switch (wave.sampleSize)
{
case 8: format = AL_FORMAT_STEREO8; break;
case 16: format = AL_FORMAT_STEREO16; break;
case 32: format = AL_FORMAT_STEREO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32
default: TraceLog(LOG_WARNING, "Wave sample size not supported: %i", wave.sampleSize); break;
}
}
else TraceLog(LOG_WARNING, "Wave number of channels not supported: %i", wave.channels);
// Create an audio source
ALuint source;
alGenSources(1, &source); // Generate pointer to audio source
alSourcef(source, AL_PITCH, 1.0f);
alSourcef(source, AL_GAIN, 1.0f);
alSource3f(source, AL_POSITION, 0.0f, 0.0f, 0.0f);
alSource3f(source, AL_VELOCITY, 0.0f, 0.0f, 0.0f);
alSourcei(source, AL_LOOPING, AL_FALSE);
// Convert loaded data to OpenAL buffer
//----------------------------------------
ALuint buffer;
alGenBuffers(1, &buffer); // Generate pointer to buffer
unsigned int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8; // Size in bytes
// Upload sound data to buffer
alBufferData(buffer, format, wave.data, dataSize, wave.sampleRate);
// Attach sound buffer to source
alSourcei(source, AL_BUFFER, buffer);
TraceLog(LOG_INFO, "[SND ID %i][BUFR ID %i] Sound data loaded successfully (%i Hz, %i bit, %s)", source, buffer, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo");
sound.source = source;
sound.buffer = buffer;
sound.format = format;
#endif
}
return sound;
}
// Unload wave data
void UnloadWave(Wave wave)
{
if (wave.data != NULL) free(wave.data);
TraceLog(LOG_INFO, "Unloaded wave data from RAM");
}
// Unload sound
void UnloadSound(Sound sound)
{
#if USE_MINI_AL
DeleteAudioBuffer((AudioBuffer *)sound.audioBuffer);
#else
alSourceStop(sound.source);
alDeleteSources(1, &sound.source);
alDeleteBuffers(1, &sound.buffer);
#endif
TraceLog(LOG_INFO, "[SND ID %i][BUFR ID %i] Unloaded sound data from RAM", sound.source, sound.buffer);
}
// Update sound buffer with new data
// NOTE: data must match sound.format
void UpdateSound(Sound sound, const void *data, int samplesCount)
{
#if USE_MINI_AL
AudioBuffer *audioBuffer = (AudioBuffer *)sound.audioBuffer;
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer");
return;
}
StopAudioBuffer(audioBuffer);
// TODO: May want to lock/unlock this since this data buffer is read at mixing time.
memcpy(audioBuffer->buffer, data, samplesCount*audioBuffer->dsp.config.channelsIn*mal_get_sample_size_in_bytes(audioBuffer->dsp.config.formatIn));
#else
ALint sampleRate, sampleSize, channels;
alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate);
alGetBufferi(sound.buffer, AL_BITS, &sampleSize); // It could also be retrieved from sound.format
alGetBufferi(sound.buffer, AL_CHANNELS, &channels); // It could also be retrieved from sound.format
TraceLog(LOG_DEBUG, "UpdateSound() : AL_FREQUENCY: %i", sampleRate);
TraceLog(LOG_DEBUG, "UpdateSound() : AL_BITS: %i", sampleSize);
TraceLog(LOG_DEBUG, "UpdateSound() : AL_CHANNELS: %i", channels);
unsigned int dataSize = samplesCount*channels*sampleSize/8; // Size of data in bytes
alSourceStop(sound.source); // Stop sound
alSourcei(sound.source, AL_BUFFER, 0); // Unbind buffer from sound to update
//alDeleteBuffers(1, &sound.buffer); // Delete current buffer data
//alGenBuffers(1, &sound.buffer); // Generate new buffer
// Upload new data to sound buffer
alBufferData(sound.buffer, sound.format, data, dataSize, sampleRate);
// Attach sound buffer to source again
alSourcei(sound.source, AL_BUFFER, sound.buffer);
#endif
}
// Play a sound
void PlaySound(Sound sound)
{
#if USE_MINI_AL
PlayAudioBuffer((AudioBuffer *)sound.audioBuffer);
#else
alSourcePlay(sound.source); // Play the sound
#endif
//TraceLog(LOG_INFO, "Playing sound");
// Find the current position of the sound being played
// NOTE: Only work when the entire file is in a single buffer
//int byteOffset;
//alGetSourcei(sound.source, AL_BYTE_OFFSET, &byteOffset);
//
//int sampleRate;
//alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate); // AL_CHANNELS, AL_BITS (bps)
//float seconds = (float)byteOffset/sampleRate; // Number of seconds since the beginning of the sound
//or
//float result;
//alGetSourcef(sound.source, AL_SEC_OFFSET, &result); // AL_SAMPLE_OFFSET
}
// Pause a sound
void PauseSound(Sound sound)
{
#if USE_MINI_AL
PauseAudioBuffer((AudioBuffer *)sound.audioBuffer);
#else
alSourcePause(sound.source);
#endif
}
// Resume a paused sound
void ResumeSound(Sound sound)
{
#if USE_MINI_AL
ResumeAudioBuffer((AudioBuffer *)sound.audioBuffer);
#else
ALenum state;
alGetSourcei(sound.source, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED) alSourcePlay(sound.source);
#endif
}
// Stop reproducing a sound
void StopSound(Sound sound)
{
#if USE_MINI_AL
StopAudioBuffer((AudioBuffer *)sound.audioBuffer);
#else
alSourceStop(sound.source);
#endif
}
// Check if a sound is playing
bool IsSoundPlaying(Sound sound)
{
#if USE_MINI_AL
return IsAudioBufferPlaying((AudioBuffer *)sound.audioBuffer);
#else
bool playing = false;
ALint state;
alGetSourcei(sound.source, AL_SOURCE_STATE, &state);
if (state == AL_PLAYING) playing = true;
return playing;
#endif
}
// Set volume for a sound
void SetSoundVolume(Sound sound, float volume)
{
#if USE_MINI_AL
SetAudioBufferVolume((AudioBuffer *)sound.audioBuffer, volume);
#else
alSourcef(sound.source, AL_GAIN, volume);
#endif
}
// Set pitch for a sound
void SetSoundPitch(Sound sound, float pitch)
{
#if USE_MINI_AL
SetAudioBufferPitch((AudioBuffer *)sound.audioBuffer, pitch);
#else
alSourcef(sound.source, AL_PITCH, pitch);
#endif
}
// Convert wave data to desired format
void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
{
#if USE_MINI_AL
mal_format formatIn = ((wave->sampleSize == 8) ? mal_format_u8 : ((wave->sampleSize == 16) ? mal_format_s16 : mal_format_f32));
mal_format formatOut = (( sampleSize == 8) ? mal_format_u8 : (( sampleSize == 16) ? mal_format_s16 : mal_format_f32));
mal_uint32 frameCountIn = wave->sampleCount; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so.
mal_uint32 frameCount = mal_convert_frames(NULL, formatOut, channels, sampleRate, NULL, formatIn, wave->channels, wave->sampleRate, frameCountIn);
if (frameCount == 0)
{
TraceLog(LOG_ERROR, "WaveFormat() : Failed to get frame count for format conversion.");
return;
}
void *data = malloc(frameCount*channels*(sampleSize/8));
frameCount = mal_convert_frames(data, formatOut, channels, sampleRate, wave->data, formatIn, wave->channels, wave->sampleRate, frameCountIn);
if (frameCount == 0)
{
TraceLog(LOG_ERROR, "WaveFormat() : Format conversion failed.");
return;
}
wave->sampleCount = frameCount;
wave->sampleSize = sampleSize;
wave->sampleRate = sampleRate;
wave->channels = channels;
free(wave->data);
wave->data = data;
#else
// Format sample rate
// NOTE: Only supported 22050 <--> 44100
if (wave->sampleRate != sampleRate)
{
// TODO: Resample wave data (upsampling or downsampling)
// NOTE 1: To downsample, you have to drop samples or average them.
// NOTE 2: To upsample, you have to interpolate new samples.
wave->sampleRate = sampleRate;
}
// Format sample size
// NOTE: Only supported 8 bit <--> 16 bit <--> 32 bit
if (wave->sampleSize != sampleSize)
{
void *data = malloc(wave->sampleCount*wave->channels*sampleSize/8);
for (int i = 0; i < wave->sampleCount; i++)
{
for (int j = 0; j < wave->channels; j++)
{
if (sampleSize == 8)
{
if (wave->sampleSize == 16) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float)(((short *)wave->data)[wave->channels*i + j])/32767.0f)*256);
else if (wave->sampleSize == 32) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float *)wave->data)[wave->channels*i + j]*127.0f + 127);
}
else if (sampleSize == 16)
{
if (wave->sampleSize == 8) ((short *)data)[wave->channels*i + j] = (short)(((float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f)*32767);
else if (wave->sampleSize == 32) ((short *)data)[wave->channels*i + j] = (short)((((float *)wave->data)[wave->channels*i + j])*32767);
}
else if (sampleSize == 32)
{
if (wave->sampleSize == 8) ((float *)data)[wave->channels*i + j] = (float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f;
else if (wave->sampleSize == 16) ((float *)data)[wave->channels*i + j] = (float)(((short *)wave->data)[wave->channels*i + j])/32767.0f;
}
}
}
wave->sampleSize = sampleSize;
free(wave->data);
wave->data = data;
}
// Format channels (interlaced mode)
// NOTE: Only supported mono <--> stereo
if (wave->channels != channels)
{
void *data = malloc(wave->sampleCount*wave->sampleSize/8*channels);
if ((wave->channels == 1) && (channels == 2)) // mono ---> stereo (duplicate mono information)
{
for (int i = 0; i < wave->sampleCount; i++)
{
for (int j = 0; j < channels; j++)
{
if (wave->sampleSize == 8) ((unsigned char *)data)[channels*i + j] = ((unsigned char *)wave->data)[i];
else if (wave->sampleSize == 16) ((short *)data)[channels*i + j] = ((short *)wave->data)[i];
else if (wave->sampleSize == 32) ((float *)data)[channels*i + j] = ((float *)wave->data)[i];
}
}
}
else if ((wave->channels == 2) && (channels == 1)) // stereo ---> mono (mix stereo channels)
{
for (int i = 0, j = 0; i < wave->sampleCount; i++, j += 2)
{
if (wave->sampleSize == 8) ((unsigned char *)data)[i] = (((unsigned char *)wave->data)[j] + ((unsigned char *)wave->data)[j + 1])/2;
else if (wave->sampleSize == 16) ((short *)data)[i] = (((short *)wave->data)[j] + ((short *)wave->data)[j + 1])/2;
else if (wave->sampleSize == 32) ((float *)data)[i] = (((float *)wave->data)[j] + ((float *)wave->data)[j + 1])/2.0f;
}
}
// TODO: Add/remove additional interlaced channels
wave->channels = channels;
free(wave->data);
wave->data = data;
}
#endif
}
// Copy a wave to a new wave
Wave WaveCopy(Wave wave)
{
Wave newWave = { 0 };
newWave.data = malloc(wave.sampleCount*wave.sampleSize/8*wave.channels);
if (newWave.data != NULL)
{
// NOTE: Size must be provided in bytes
memcpy(newWave.data, wave.data, wave.sampleCount*wave.channels*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) && (finalSample < wave->sampleCount))
{
int sampleCount = finalSample - initSample;
void *data = malloc(sampleCount*wave->sampleSize/8*wave->channels);
memcpy(data, (unsigned char*)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8);
free(wave->data);
wave->data = data;
}
else TraceLog(LOG_WARNING, "Wave crop range out of bounds");
}
// Get samples data from wave as a floats array
// NOTE: Returned sample values are normalized to range [-1..1]
float *GetWaveData(Wave wave)
{
float *samples = (float *)malloc(wave.sampleCount*wave.channels*sizeof(float));
for (int i = 0; i < wave.sampleCount; i++)
{
for (int j = 0; j < wave.channels; j++)
{
if (wave.sampleSize == 8) samples[wave.channels*i + j] = (float)(((unsigned char *)wave.data)[wave.channels*i + j] - 127)/256.0f;
else if (wave.sampleSize == 16) samples[wave.channels*i + j] = (float)((short *)wave.data)[wave.channels*i + j]/32767.0f;
else if (wave.sampleSize == 32) samples[wave.channels*i + j] = ((float *)wave.data)[wave.channels*i + j];
}
}
return samples;
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Music loading and stream playing (.OGG)
//----------------------------------------------------------------------------------
// Load music stream from file
Music LoadMusicStream(const char *fileName)
{
Music music = (MusicData *)malloc(sizeof(MusicData));
if (IsFileExtension(fileName, ".ogg"))
{
// Open ogg audio stream
music->ctxOgg = stb_vorbis_open_filename(fileName, NULL, NULL);
if (music->ctxOgg == NULL) TraceLog(LOG_WARNING, "[%s] OGG audio file could not be opened", fileName);
else
{
stb_vorbis_info info = stb_vorbis_get_info(music->ctxOgg); // 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);
music->totalSamples = (unsigned int)stb_vorbis_stream_length_in_samples(music->ctxOgg); // Independent by channel
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_AUDIO_OGG;
music->loopCount = -1; // Infinite loop by default
TraceLog(LOG_DEBUG, "[%s] FLAC total samples: %i", fileName, music->totalSamples);
TraceLog(LOG_DEBUG, "[%s] OGG sample rate: %i", fileName, info.sample_rate);
TraceLog(LOG_DEBUG, "[%s] OGG channels: %i", fileName, info.channels);
TraceLog(LOG_DEBUG, "[%s] OGG memory required: %i", fileName, info.temp_memory_required);
}
}
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (IsFileExtension(fileName, ".flac"))
{
music->ctxFlac = drflac_open_file(fileName);
if (music->ctxFlac == NULL) TraceLog(LOG_WARNING, "[%s] FLAC audio file could not be opened", fileName);
else
{
music->stream = InitAudioStream(music->ctxFlac->sampleRate, music->ctxFlac->bitsPerSample, music->ctxFlac->channels);
music->totalSamples = (unsigned int)music->ctxFlac->totalSampleCount/music->ctxFlac->channels;
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_AUDIO_FLAC;
music->loopCount = -1; // Infinite loop by default
TraceLog(LOG_DEBUG, "[%s] FLAC total samples: %i", fileName, music->totalSamples);
TraceLog(LOG_DEBUG, "[%s] FLAC sample rate: %i", fileName, music->ctxFlac->sampleRate);
TraceLog(LOG_DEBUG, "[%s] FLAC bits per sample: %i", fileName, music->ctxFlac->bitsPerSample);
TraceLog(LOG_DEBUG, "[%s] FLAC channels: %i", fileName, music->ctxFlac->channels);
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
else if (IsFileExtension(fileName, ".xm"))
{
int result = jar_xm_create_context_from_file(&music->ctxXm, 48000, fileName);
if (!result) // XM context created successfully
{
jar_xm_set_max_loop_count(music->ctxXm, 0); // Set infinite number of loops
// NOTE: Only stereo is supported for XM
music->stream = InitAudioStream(48000, 16, 2);
music->totalSamples = (unsigned int)jar_xm_get_remaining_samples(music->ctxXm);
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_MODULE_XM;
music->loopCount = -1; // Infinite loop by default
TraceLog(LOG_DEBUG, "[%s] XM number of samples: %i", fileName, music->totalSamples);
TraceLog(LOG_DEBUG, "[%s] XM track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f);
}
else TraceLog(LOG_WARNING, "[%s] XM file could not be opened", fileName);
}
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
else if (IsFileExtension(fileName, ".mod"))
{
jar_mod_init(&music->ctxMod);
if (jar_mod_load_file(&music->ctxMod, fileName))
{
music->stream = InitAudioStream(48000, 16, 2);
music->totalSamples = (unsigned int)jar_mod_max_samples(&music->ctxMod);
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_MODULE_MOD;
music->loopCount = -1; // Infinite loop by default
TraceLog(LOG_DEBUG, "[%s] MOD number of samples: %i", fileName, music->samplesLeft);
TraceLog(LOG_DEBUG, "[%s] MOD track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f);
}
else TraceLog(LOG_WARNING, "[%s] MOD file could not be opened", fileName);
}
#endif
else TraceLog(LOG_WARNING, "[%s] Audio fileformat not supported, it can't be loaded", fileName);
return music;
}
// Unload music stream
void UnloadMusicStream(Music music)
{
CloseAudioStream(music->stream);
if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close(music->ctxOgg);
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (music->ctxType == MUSIC_AUDIO_FLAC) drflac_free(music->ctxFlac);
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
else if (music->ctxType == MUSIC_MODULE_XM) jar_xm_free_context(music->ctxXm);
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
else if (music->ctxType == MUSIC_MODULE_MOD) jar_mod_unload(&music->ctxMod);
#endif
free(music);
}
// Start music playing (open stream)
void PlayMusicStream(Music music)
{
#if USE_MINI_AL
AudioBuffer *audioBuffer = (AudioBuffer *)music->stream.audioBuffer;
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "PlayMusicStream() : No audio buffer");
return;
}
// For music streams, we need to make sure we maintain the frame cursor position. This is 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);
mal_uint32 frameCursorPos = audioBuffer->frameCursorPos;
PlayAudioStream(music->stream); // <-- This resets the cursor position.
audioBuffer->frameCursorPos = frameCursorPos;
#else
alSourcePlay(music->stream.source);
#endif
}
// Pause music playing
void PauseMusicStream(Music music)
{
#if USE_MINI_AL
PauseAudioStream(music->stream);
#else
alSourcePause(music->stream.source);
#endif
}
// Resume music playing
void ResumeMusicStream(Music music)
{
#if USE_MINI_AL
ResumeAudioStream(music->stream);
#else
ALenum state;
alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED)
{
TraceLog(LOG_INFO, "[AUD ID %i] Resume music stream playing", music->stream.source);
alSourcePlay(music->stream.source);
}
#endif
}
// Stop music playing (close stream)
// TODO: To clear a buffer, make sure they have been already processed!
void StopMusicStream(Music music)
{
#if USE_MINI_AL
StopAudioStream(music->stream);
#else
alSourceStop(music->stream.source);
/*
// Clear stream buffers
// WARNING: Queued buffers must have been processed before unqueueing and reloaded with data!!!
void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, 1);
for (int i = 0; i < MAX_STREAM_BUFFERS; i++)
{
//UpdateAudioStream(music->stream, pcm, AUDIO_BUFFER_SIZE); // Update one buffer at a time
alBufferData(music->stream.buffers[i], music->stream.format, pcm, AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, music->stream.sampleRate);
}
free(pcm);
*/
#endif
// Restart music context
switch (music->ctxType)
{
case MUSIC_AUDIO_OGG: stb_vorbis_seek_start(music->ctxOgg); break;
#if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_MODULE_FLAC: /* TODO: Restart FLAC context */ break;
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
case MUSIC_MODULE_XM: /* TODO: Restart XM context */ break;
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
case MUSIC_MODULE_MOD: jar_mod_seek_start(&music->ctxMod); break;
#endif
default: break;
}
music->samplesLeft = music->totalSamples;
}
// Update (re-fill) music buffers if data already processed
// TODO: Make sure buffers are ready for update... check music state
void UpdateMusicStream(Music music)
{
#if USE_MINI_AL
bool streamEnding = false;
unsigned int subBufferSizeInFrames = ((AudioBuffer *)music->stream.audioBuffer)->bufferSizeInFrames/2;
// NOTE: Using dynamic allocation because it could require more than 16KB
void *pcm = calloc(subBufferSizeInFrames*music->stream.sampleSize/8*music->stream.channels, 1);
int samplesCount = 0; // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts
while (IsAudioBufferProcessed(music->stream))
{
if (music->samplesLeft >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames;
else samplesCount = music->samplesLeft;
// TODO: Really don't like ctxType thingy...
switch (music->ctxType)
{
case MUSIC_AUDIO_OGG:
{
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, samplesCount*music->stream.channels);
} break;
#if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_AUDIO_FLAC:
{
// NOTE: Returns the number of samples to process
unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, samplesCount*music->stream.channels, (short *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
case MUSIC_MODULE_XM: jar_xm_generate_samples_16bit(music->ctxXm, pcm, samplesCount); break;
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, samplesCount, 0); break;
#endif
default: break;
}
UpdateAudioStream(music->stream, pcm, samplesCount);
music->samplesLeft -= samplesCount;
if (music->samplesLeft <= 0)
{
streamEnding = true;
break;
}
}
// Free allocated pcm data
free(pcm);
// Reset audio stream for looping
if (streamEnding)
{
StopMusicStream(music); // Stop music (and reset)
// Decrease loopCount to stop when required
if (music->loopCount > 0)
{
music->loopCount--; // Decrease loop count
PlayMusicStream(music); // Play again
}
else
{
if (music->loopCount == -1) PlayMusicStream(music);
}
}
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);
}
#else
ALenum state;
ALint processed = 0;
alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state); // Get music stream state
alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed); // Get processed buffers
if (processed > 0)
{
bool streamEnding = false;
// NOTE: Using dynamic allocation because it could require more than 16KB
void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, 1);
int numBuffersToProcess = processed;
int samplesCount = 0; // Total size of data steamed in L+R samples for xm floats,
// individual L or R for ogg shorts
for (int i = 0; i < numBuffersToProcess; i++)
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE) samplesCount = AUDIO_BUFFER_SIZE;
else samplesCount = music->samplesLeft;
// TODO: Really don't like ctxType thingy...
switch (music->ctxType)
{
case MUSIC_AUDIO_OGG:
{
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, samplesCount*music->stream.channels);
} break;
#if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_AUDIO_FLAC:
{
// NOTE: Returns the number of samples to process
unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, samplesCount*music->stream.channels, (short *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
case MUSIC_MODULE_XM: jar_xm_generate_samples_16bit(music->ctxXm, pcm, samplesCount); break;
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, samplesCount, 0); break;
#endif
default: break;
}
UpdateAudioStream(music->stream, pcm, samplesCount);
music->samplesLeft -= samplesCount;
if (music->samplesLeft <= 0)
{
streamEnding = true;
break;
}
}
// Free allocated pcm data
free(pcm);
// Reset audio stream for looping
if (streamEnding)
{
StopMusicStream(music); // Stop music (and reset)
// Decrease loopCount to stop when required
if (music->loopCount > 0)
{
music->loopCount--; // Decrease loop count
PlayMusicStream(music); // Play again
}
else
{
if (music->loopCount == -1)
{
PlayMusicStream(music);
}
}
}
else
{
// NOTE: In case window is minimized, music stream is stopped,
// just make sure to play again on window restore
if (state != AL_PLAYING) PlayMusicStream(music);
}
}
#endif
}
// Check if any music is playing
bool IsMusicPlaying(Music music)
{
#if USE_MINI_AL
return IsAudioStreamPlaying(music->stream);
#else
bool playing = false;
ALint state;
alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state);
if (state == AL_PLAYING) playing = true;
return playing;
#endif
}
// Set volume for music
void SetMusicVolume(Music music, float volume)
{
#if USE_MINI_AL
SetAudioStreamVolume(music->stream, volume);
#else
alSourcef(music->stream.source, AL_GAIN, volume);
#endif
}
// Set pitch for music
void SetMusicPitch(Music music, float pitch)
{
#if USE_MINI_AL
SetAudioStreamPitch(music->stream, pitch);
#else
alSourcef(music->stream.source, AL_PITCH, pitch);
#endif
}
// Set music loop count (loop repeats)
// NOTE: If set to -1, means infinite loop
void SetMusicLoopCount(Music music, int count)
{
music->loopCount = count;
}
// Get music time length (in seconds)
float GetMusicTimeLength(Music music)
{
float totalSeconds = (float)music->totalSamples/music->stream.sampleRate;
return totalSeconds;
}
// Get current music time played (in seconds)
float GetMusicTimePlayed(Music music)
{
float secondsPlayed = 0.0f;
unsigned int samplesPlayed = music->totalSamples - music->samplesLeft;
secondsPlayed = (float)samplesPlayed/music->stream.sampleRate;
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;
// Only mono and stereo channels are supported, more channels require AL_EXT_MCFORMATS extension
if ((channels > 0) && (channels < 3)) stream.channels = channels;
else
{
TraceLog(LOG_WARNING, "Init audio stream: Number of channels not supported: %i", channels);
stream.channels = 1; // Fallback to mono channel
}
#if USE_MINI_AL
mal_format formatIn = ((stream.sampleSize == 8) ? mal_format_u8 : ((stream.sampleSize == 16) ? mal_format_s16 : mal_format_f32));
// The size of a streaming buffer must be at least double the size of a period.
unsigned int periodSize = device.bufferSizeInFrames / device.periods;
unsigned int subBufferSize = AUDIO_BUFFER_SIZE;
if (subBufferSize < periodSize) subBufferSize = periodSize;
AudioBuffer *audioBuffer = CreateAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM);
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "InitAudioStream() : Failed to create audio buffer");
return stream;
}
audioBuffer->looping = true; // Always loop for streaming buffers.
stream.audioBuffer = audioBuffer;
#else
// Setup OpenAL format
if (stream.channels == 1)
{
switch (sampleSize)
{
case 8: stream.format = AL_FORMAT_MONO8; break;
case 16: stream.format = AL_FORMAT_MONO16; break;
case 32: stream.format = AL_FORMAT_MONO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32
default: TraceLog(LOG_WARNING, "Init audio stream: Sample size not supported: %i", sampleSize); break;
}
}
else if (stream.channels == 2)
{
switch (sampleSize)
{
case 8: stream.format = AL_FORMAT_STEREO8; break;
case 16: stream.format = AL_FORMAT_STEREO16; break;
case 32: stream.format = AL_FORMAT_STEREO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32
default: TraceLog(LOG_WARNING, "Init audio stream: Sample size not supported: %i", sampleSize); break;
}
}
// Create an audio source
alGenSources(1, &stream.source);
alSourcef(stream.source, AL_PITCH, 1.0f);
alSourcef(stream.source, AL_GAIN, 1.0f);
alSource3f(stream.source, AL_POSITION, 0.0f, 0.0f, 0.0f);
alSource3f(stream.source, AL_VELOCITY, 0.0f, 0.0f, 0.0f);
// Create Buffers (double buffering)
alGenBuffers(MAX_STREAM_BUFFERS, stream.buffers);
// Initialize buffer with zeros by default
// NOTE: Using dynamic allocation because it requires more than 16KB
void *pcm = calloc(AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, 1);
for (int i = 0; i < MAX_STREAM_BUFFERS; i++)
{
alBufferData(stream.buffers[i], stream.format, pcm, AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, stream.sampleRate);
}
free(pcm);
alSourceQueueBuffers(stream.source, MAX_STREAM_BUFFERS, stream.buffers);
#endif
TraceLog(LOG_INFO, "[AUD ID %i] Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.source, stream.sampleRate, stream.sampleSize, (stream.channels == 1) ? "Mono" : "Stereo");
return stream;
}
// Close audio stream and free memory
void CloseAudioStream(AudioStream stream)
{
#if USE_MINI_AL
DeleteAudioBuffer((AudioBuffer *)stream.audioBuffer);
#else
// Stop playing channel
alSourceStop(stream.source);
// Flush out all queued buffers
int queued = 0;
alGetSourcei(stream.source, AL_BUFFERS_QUEUED, &queued);
ALuint buffer = 0;
while (queued > 0)
{
alSourceUnqueueBuffers(stream.source, 1, &buffer);
queued--;
}
// Delete source and buffers
alDeleteSources(1, &stream.source);
alDeleteBuffers(MAX_STREAM_BUFFERS, stream.buffers);
#endif
TraceLog(LOG_INFO, "[AUD ID %i] Unloaded audio stream data", stream.source);
}
// 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: IsAudioBufferProcessed()
void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
{
#if USE_MINI_AL
AudioBuffer* audioBuffer = (AudioBuffer*)stream.audioBuffer;
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "UpdateAudioStream() : No audio buffer");
return;
}
if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1])
{
mal_uint32 subBufferToUpdate;
if (audioBuffer->isSubBufferProcessed[0] && audioBuffer->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;
audioBuffer->frameCursorPos = 0;
}
else
{
// Just update whichever sub-buffer is processed.
subBufferToUpdate = (audioBuffer->isSubBufferProcessed[0]) ? 0 : 1;
}
mal_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2;
unsigned char *subBuffer = audioBuffer->buffer + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
// 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 >= (mal_uint32)samplesCount)
{
mal_uint32 framesToWrite = subBufferSizeInFrames;
if (framesToWrite > (mal_uint32)samplesCount) framesToWrite = (mal_uint32)samplesCount;
mal_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
memcpy(subBuffer, data, bytesToWrite);
// Any leftover frames should be filled with zeros.
mal_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
if (leftoverFrameCount > 0)
{
memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));
}
audioBuffer->isSubBufferProcessed[subBufferToUpdate] = false;
}
else
{
TraceLog(LOG_ERROR, "UpdateAudioStream() : Attempting to write too many frames to buffer");
return;
}
}
else
{
TraceLog(LOG_ERROR, "Audio buffer not available for updating");
return;
}
#else
ALuint buffer = 0;
alSourceUnqueueBuffers(stream.source, 1, &buffer);
// Check if any buffer was available for unqueue
if (alGetError() != AL_INVALID_VALUE)
{
alBufferData(buffer, stream.format, data, samplesCount*stream.sampleSize/8*stream.channels, stream.sampleRate);
alSourceQueueBuffers(stream.source, 1, &buffer);
}
else TraceLog(LOG_WARNING, "[AUD ID %i] Audio buffer not available for unqueuing", stream.source);
#endif
}
// Check if any audio stream buffers requires refill
bool IsAudioBufferProcessed(AudioStream stream)
{
#if USE_MINI_AL
AudioBuffer *audioBuffer = (AudioBuffer *)stream.audioBuffer;
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "IsAudioBufferProcessed() : No audio buffer");
return false;
}
return audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1];
#else
ALint processed = 0;
// Determine if music stream is ready to be written
alGetSourcei(stream.source, AL_BUFFERS_PROCESSED, &processed);
return (processed > 0);
#endif
}
// Play audio stream
void PlayAudioStream(AudioStream stream)
{
#if USE_MINI_AL
PlayAudioBuffer((AudioBuffer *)stream.audioBuffer);
#else
alSourcePlay(stream.source);
#endif
}
// Play audio stream
void PauseAudioStream(AudioStream stream)
{
#if USE_MINI_AL
PauseAudioBuffer((AudioBuffer *)stream.audioBuffer);
#else
alSourcePause(stream.source);
#endif
}
// Resume audio stream playing
void ResumeAudioStream(AudioStream stream)
{
#if USE_MINI_AL
ResumeAudioBuffer((AudioBuffer *)stream.audioBuffer);
#else
ALenum state;
alGetSourcei(stream.source, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED) alSourcePlay(stream.source);
#endif
}
// Check if audio stream is playing.
bool IsAudioStreamPlaying(AudioStream stream)
{
#if USE_MINI_AL
return IsAudioBufferPlaying((AudioBuffer *)stream.audioBuffer);
#else
bool playing = false;
ALint state;
alGetSourcei(stream.source, AL_SOURCE_STATE, &state);
if (state == AL_PLAYING) playing = true;
return playing;
#endif
}
// Stop audio stream
void StopAudioStream(AudioStream stream)
{
#if USE_MINI_AL
StopAudioBuffer((AudioBuffer *)stream.audioBuffer);
#else
alSourceStop(stream.source);
#endif
}
void SetAudioStreamVolume(AudioStream stream, float volume)
{
#if USE_MINI_AL
SetAudioBufferVolume((AudioBuffer *)stream.audioBuffer, volume);
#else
alSourcef(stream.source, AL_GAIN, volume);
#endif
}
void SetAudioStreamPitch(AudioStream stream, float pitch)
{
#if USE_MINI_AL
SetAudioBufferPitch((AudioBuffer *)stream.audioBuffer, pitch);
#else
alSourcef(stream.source, AL_PITCH, pitch);
#endif
}
//----------------------------------------------------------------------------------
// Module specific Functions Definition
//----------------------------------------------------------------------------------
#if defined(SUPPORT_FILEFORMAT_WAV)
// Load WAV file into Wave structure
static Wave LoadWAV(const char *fileName)
{
// Basic WAV headers structs
typedef struct {
char chunkID[4];
int chunkSize;
char format[4];
} WAVRiffHeader;
typedef struct {
char subChunkID[4];
int subChunkSize;
short audioFormat;
short numChannels;
int sampleRate;
int byteRate;
short blockAlign;
short bitsPerSample;
} WAVFormat;
typedef struct {
char subChunkID[4];
int subChunkSize;
} WAVData;
WAVRiffHeader wavRiffHeader;
WAVFormat wavFormat;
WAVData wavData;
Wave wave = { 0 };
FILE *wavFile;
wavFile = fopen(fileName, "rb");
if (wavFile == NULL)
{
TraceLog(LOG_WARNING, "[%s] WAV file could not be opened", fileName);
wave.data = NULL;
}
else
{
// Read in the first chunk into the struct
fread(&wavRiffHeader, sizeof(WAVRiffHeader), 1, wavFile);
// Check for RIFF and WAVE tags
if (strncmp(wavRiffHeader.chunkID, "RIFF", 4) ||
strncmp(wavRiffHeader.format, "WAVE", 4))
{
TraceLog(LOG_WARNING, "[%s] Invalid RIFF or WAVE Header", fileName);
}
else
{
// Read in the 2nd chunk for the wave info
fread(&wavFormat, sizeof(WAVFormat), 1, wavFile);
// Check for fmt tag
if ((wavFormat.subChunkID[0] != 'f') || (wavFormat.subChunkID[1] != 'm') ||
(wavFormat.subChunkID[2] != 't') || (wavFormat.subChunkID[3] != ' '))
{
TraceLog(LOG_WARNING, "[%s] Invalid Wave format", fileName);
}
else
{
// Check for extra parameters;
if (wavFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR);
// Read in the the last byte of data before the sound file
fread(&wavData, sizeof(WAVData), 1, wavFile);
// Check for data tag
if ((wavData.subChunkID[0] != 'd') || (wavData.subChunkID[1] != 'a') ||
(wavData.subChunkID[2] != 't') || (wavData.subChunkID[3] != 'a'))
{
TraceLog(LOG_WARNING, "[%s] Invalid data header", fileName);
}
else
{
// Allocate memory for data
wave.data = malloc(wavData.subChunkSize);
// Read in the sound data into the soundData variable
fread(wave.data, wavData.subChunkSize, 1, wavFile);
// Store wave parameters
wave.sampleRate = wavFormat.sampleRate;
wave.sampleSize = wavFormat.bitsPerSample;
wave.channels = wavFormat.numChannels;
// NOTE: Only support 8 bit, 16 bit and 32 bit sample sizes
if ((wave.sampleSize != 8) && (wave.sampleSize != 16) && (wave.sampleSize != 32))
{
TraceLog(LOG_WARNING, "[%s] WAV sample size (%ibit) not supported, converted to 16bit", fileName, wave.sampleSize);
WaveFormat(&wave, wave.sampleRate, 16, wave.channels);
}
// NOTE: Only support up to 2 channels (mono, stereo)
if (wave.channels > 2)
{
WaveFormat(&wave, wave.sampleRate, wave.sampleSize, 2);
TraceLog(LOG_WARNING, "[%s] WAV channels number (%i) not supported, converted to 2 channels", fileName, wave.channels);
}
// NOTE: subChunkSize comes in bytes, we need to translate it to number of samples
wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels;
TraceLog(LOG_INFO, "[%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo");
}
}
}
fclose(wavFile);
}
return wave;
}
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
// Load OGG file into Wave structure
// NOTE: Using stb_vorbis library
static Wave LoadOGG(const char *fileName)
{
Wave wave = { 0 };
stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);
if (oggFile == NULL) TraceLog(LOG_WARNING, "[%s] OGG file could not be opened", fileName);
else
{
stb_vorbis_info info = stb_vorbis_get_info(oggFile);
wave.sampleRate = info.sample_rate;
wave.sampleSize = 16; // 16 bit per sample (short)
wave.channels = info.channels;
wave.sampleCount = (int)stb_vorbis_stream_length_in_samples(oggFile); // Independent by channel
float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile);
if (totalSeconds > 10) TraceLog(LOG_WARNING, "[%s] Ogg audio length is larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds);
wave.data = (short *)malloc(wave.sampleCount*wave.channels*sizeof(short));
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, (short *)wave.data, wave.sampleCount*wave.channels);
TraceLog(LOG_DEBUG, "[%s] Samples obtained: %i", fileName, numSamplesOgg);
TraceLog(LOG_INFO, "[%s] OGG file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo");
stb_vorbis_close(oggFile);
}
return wave;
}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
// Load FLAC file into Wave structure
// NOTE: Using dr_flac library
static Wave LoadFLAC(const char *fileName)
{
Wave wave;
// Decode an entire FLAC file in one go
uint64_t totalSampleCount;
wave.data = drflac_open_and_decode_file_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount);
wave.sampleCount = (int)totalSampleCount/wave.channels;
wave.sampleSize = 16;
// NOTE: Only support up to 2 channels (mono, stereo)
if (wave.channels > 2) TraceLog(LOG_WARNING, "[%s] FLAC channels number (%i) not supported", fileName, wave.channels);
if (wave.data == NULL) TraceLog(LOG_WARNING, "[%s] FLAC data could not be loaded", fileName);
else TraceLog(LOG_INFO, "[%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo");
return wave;
}
#endif
// Some required functions for audio standalone module version
#if defined(AUDIO_STANDALONE)
// Check file extension
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;
}
// Show trace log messages (LOG_INFO, LOG_WARNING, LOG_ERROR, LOG_DEBUG)
void TraceLog(int msgType, const char *text, ...)
{
va_list args;
va_start(args, text);
switch (msgType)
{
case LOG_INFO: fprintf(stdout, "INFO: "); break;
case LOG_ERROR: fprintf(stdout, "ERROR: "); break;
case LOG_WARNING: fprintf(stdout, "WARNING: "); break;
case LOG_DEBUG: fprintf(stdout, "DEBUG: "); break;
default: break;
}
vfprintf(stdout, text, args);
fprintf(stdout, "\n");
va_end(args);
if (msgType == LOG_ERROR) exit(1);
}
#endif
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