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raylib/src/raudio.c
jbosh 019434a37e Fix for short non-looping sounds (#1067) 
Short non-looping sounds can sometimes think they need to keep playing and will output their first few frames again. This helps to break out of all the loops instead of just this one.
2020-01-14 09:15:52 +01:00

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/**********************************************************************************************
*
* raudio - A simple and easy-to-use audio library based on miniaudio
*
* 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 RAUDIO_STANDALONE
* Define to use the module as standalone library (independently of raylib).
* Required types and functions are defined in the same module.
*
* #define SUPPORT_FILEFORMAT_WAV
* #define SUPPORT_FILEFORMAT_OGG
* #define SUPPORT_FILEFORMAT_XM
* #define SUPPORT_FILEFORMAT_MOD
* #define SUPPORT_FILEFORMAT_FLAC
* #define SUPPORT_FILEFORMAT_MP3
* Selected desired fileformats to be supported for loading. Some of those formats are
* supported by default, to remove support, just comment unrequired #define in this module
*
* DEPENDENCIES:
* miniaudio.h - Audio device management lib (https://github.com/dr-soft/miniaudio)
* stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
* dr_mp3.h - MP3 audio file loading (https://github.com/mackron/dr_libs)
* dr_flac.h - FLAC audio file loading (https://github.com/mackron/dr_libs)
* jar_xm.h - XM module file loading
* jar_mod.h - MOD audio file loading
*
* CONTRIBUTORS:
* David Reid (github: @mackron) (Nov. 2017):
* - Complete port to miniaudio library
*
* Joshua Reisenauer (github: @kd7tck) (2015)
* - XM audio module support (jar_xm)
* - MOD audio module support (jar_mod)
* - Mixing channels support
* - Raw audio context support
*
*
* LICENSE: zlib/libpng
*
* Copyright (c) 2013-2020 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
#define MA_NO_JACK
#define MINIAUDIO_IMPLEMENTATION
#include "external/miniaudio.h" // miniaudio library
#undef PlaySound // Win32 API: windows.h > mmsystem.h defines PlaySound macro
#include <stdlib.h> // Required for: malloc(), free()
#include <string.h> // Required for: strcmp(), strncmp()
#include <stdio.h> // Required for: FILE, fopen(), fclose(), fread()
#if defined(SUPPORT_FILEFORMAT_OGG)
#define STB_VORBIS_IMPLEMENTATION
#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
#if defined(SUPPORT_FILEFORMAT_MP3)
#define DR_MP3_IMPLEMENTATION
#include "external/dr_mp3.h" // MP3 loading functions
#endif
#if defined(_MSC_VER)
#undef bool
#endif
//----------------------------------------------------------------------------------
// Defines and Macros
//----------------------------------------------------------------------------------
// 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
#define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. 16bit, Mono: 8Kb)
//----------------------------------------------------------------------------------
// 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_WAV = 0,
MUSIC_AUDIO_OGG,
MUSIC_AUDIO_FLAC,
MUSIC_AUDIO_MP3,
MUSIC_MODULE_XM,
MUSIC_MODULE_MOD
} MusicContextType;
#if defined(RAUDIO_STANDALONE)
typedef enum {
LOG_ALL,
LOG_TRACE,
LOG_DEBUG,
LOG_INFO,
LOG_WARNING,
LOG_ERROR,
LOG_FATAL,
LOG_NONE
} TraceLogType;
#endif
//----------------------------------------------------------------------------------
// Global Variables Definition
//----------------------------------------------------------------------------------
// ...
//----------------------------------------------------------------------------------
// Module specific Functions Declaration
//----------------------------------------------------------------------------------
#if defined(SUPPORT_FILEFORMAT_WAV)
static Wave LoadWAV(const char *fileName); // Load WAV file
static int SaveWAV(Wave wave, const char *fileName); // Save wave data as WAV file
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
static Wave LoadOGG(const char *fileName); // Load OGG file
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
static Wave LoadFLAC(const char *fileName); // Load FLAC file
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
static Wave LoadMP3(const char *fileName); // Load MP3 file
#endif
#if defined(RAUDIO_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
//----------------------------------------------------------------------------------
// AudioBuffer Functionality
//----------------------------------------------------------------------------------
#define DEVICE_FORMAT ma_format_f32
#define DEVICE_CHANNELS 2
#define DEVICE_SAMPLE_RATE 44100
#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16
typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioBufferUsage;
// Audio buffer structure
// NOTE: Slightly different logic is used when feeding data to the
// playback device depending on whether or not data is streamed
struct rAudioBuffer {
ma_pcm_converter dsp; // PCM data converter
float volume; // Audio buffer volume
float pitch; // Audio buffer pitch
bool playing; // Audio buffer state: AUDIO_PLAYING
bool paused; // Audio buffer state: AUDIO_PAUSED
bool looping; // Audio buffer looping, always true for AudioStreams
int usage; // Audio buffer usage mode: STATIC or STREAM
bool isSubBufferProcessed[2]; // SubBuffer processed (virtual double buffer)
unsigned int frameCursorPos; // Frame cursor position
unsigned int bufferSizeInFrames; // Total buffer size in frames
unsigned int totalFramesProcessed; // Total frames processed in this buffer (required for play timming)
unsigned char *buffer; // 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 buffers are tracked in a linked list
static AudioBuffer *firstAudioBuffer = NULL; // Pointer to first AudioBuffer in the list
static AudioBuffer *lastAudioBuffer = NULL; // Pointer to last AudioBuffer in the list
// miniaudio global variables
static ma_context context; // miniaudio context data
static ma_device device; // miniaudio device
static ma_mutex audioLock; // miniaudio mutex lock
static bool isAudioInitialized = false; // Check if audio device is initialized
static float masterVolume = 1.0f; // Master volume (multiplied on output mixing)
// Multi channel playback global variables
static AudioBuffer *audioBufferPool[MAX_AUDIO_BUFFER_POOL_CHANNELS] = { 0 }; // Multichannel AudioBuffer pointers pool
static unsigned int audioBufferPoolCounter = 0; // AudioBuffer pointers pool counter
static unsigned int audioBufferPoolChannels[MAX_AUDIO_BUFFER_POOL_CHANNELS] = { 0 }; // AudioBuffer pool channels
// miniaudio 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 ma_uint32 OnAudioBufferDSPRead(ma_pcm_converter *pDSP, void *pFramesOut, ma_uint32 frameCount, void *pUserData);
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume);
// AudioBuffer management functions declaration
// NOTE: Those functions are not exposed by raylib... for the moment
AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, int usage);
void CloseAudioBuffer(AudioBuffer *buffer);
bool IsAudioBufferPlaying(AudioBuffer *buffer);
void PlayAudioBuffer(AudioBuffer *buffer);
void StopAudioBuffer(AudioBuffer *buffer);
void PauseAudioBuffer(AudioBuffer *buffer);
void ResumeAudioBuffer(AudioBuffer *buffer);
void SetAudioBufferVolume(AudioBuffer *buffer, float volume);
void SetAudioBufferPitch(AudioBuffer *buffer, float pitch);
void TrackAudioBuffer(AudioBuffer *buffer);
void UntrackAudioBuffer(AudioBuffer *buffer);
//----------------------------------------------------------------------------------
// miniaudio functions definitions
//----------------------------------------------------------------------------------
// Log callback function
static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message)
{
(void)pContext;
(void)pDevice;
TraceLog(LOG_ERROR, message); // All log messages from miniaudio are errors
}
// Sending audio data to device callback function
// NOTE: All the mixing takes place here
static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount)
{
(void)pDevice;
// Mixing is basically just an accumulation, we need to initialize the output buffer to 0
memset(pFramesOut, 0, frameCount*pDevice->playback.channels*ma_get_bytes_per_sample(pDevice->playback.format));
// Using a mutex here for thread-safety which makes things not real-time
// This is unlikely to be necessary for this project, but may want to consider how you might want to avoid this
ma_mutex_lock(&audioLock);
{
for (AudioBuffer *audioBuffer = firstAudioBuffer; 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)
{
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
ma_uint32 framesToRead = (frameCount - framesRead);
while (framesToRead > 0)
{
float tempBuffer[1024]; // 512 frames for stereo
ma_uint32 framesToReadRightNow = framesToRead;
if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS)
{
framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS;
}
ma_uint32 framesJustRead = (ma_uint32)ma_pcm_converter_read(&audioBuffer->dsp, tempBuffer, framesToReadRightNow);
if (framesJustRead > 0)
{
float *framesOut = (float *)pFramesOut + (framesRead*device.playback.channels);
float *framesIn = tempBuffer;
MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume);
framesToRead -= framesJustRead;
framesRead += framesJustRead;
}
if (!audioBuffer->playing)
{
framesRead = frameCount;
break;
}
// If we weren't able to read all the frames we requested, break
if (framesJustRead < framesToReadRightNow)
{
if (!audioBuffer->looping)
{
StopAudioBuffer(audioBuffer);
break;
}
else
{
// Should never get here, but just for safety,
// move the cursor position back to the start and continue the loop
audioBuffer->frameCursorPos = 0;
continue;
}
}
}
// If for some reason we weren't able to read every frame we'll need to break from the loop
// Not doing this could theoretically put us into an infinite loop
if (framesToRead > 0) break;
}
}
}
ma_mutex_unlock(&audioLock);
}
// DSP read from audio buffer callback function
static ma_uint32 OnAudioBufferDSPRead(ma_pcm_converter *pDSP, void *pFramesOut, ma_uint32 frameCount, void *pUserData)
{
AudioBuffer *audioBuffer = (AudioBuffer *)pUserData;
ma_uint32 subBufferSizeInFrames = (audioBuffer->bufferSizeInFrames > 1)? audioBuffer->bufferSizeInFrames/2 : audioBuffer->bufferSizeInFrames;
ma_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];
ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn)*audioBuffer->dsp.formatConverterIn.config.channels;
// 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->bufferSizeInFrames - 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 *)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
ma_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;
}
// This is the main mixing function. Mixing is pretty simple in this project - it's just an accumulation.
// NOTE: framesOut is both an input and an output. It will be initially filled with zeros outside of this function.
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume)
{
for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
{
for (ma_uint32 iChannel = 0; iChannel < device.playback.channels; ++iChannel)
{
float *frameOut = framesOut + (iFrame*device.playback.channels);
const float *frameIn = framesIn + (iFrame*device.playback.channels);
frameOut[iChannel] += (frameIn[iChannel]*masterVolume*localVolume);
}
}
}
// Initialise the multichannel buffer pool
static void InitAudioBufferPool()
{
// Dummy buffers
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
{
audioBufferPool[i] = InitAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC);
}
}
// Close the audio buffers pool
static void CloseAudioBufferPool()
{
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
{
RL_FREE(audioBufferPool[i]->buffer);
RL_FREE(audioBufferPool[i]);
}
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Device initialization and Closing
//----------------------------------------------------------------------------------
// Initialize audio device
void InitAudioDevice(void)
{
// Init audio context
ma_context_config contextConfig = ma_context_config_init();
contextConfig.logCallback = OnLog;
ma_result result = ma_context_init(NULL, 0, &contextConfig, &context);
if (result != MA_SUCCESS)
{
TraceLog(LOG_ERROR, "Failed to initialize audio 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 device.
config.playback.format = DEVICE_FORMAT;
config.playback.channels = DEVICE_CHANNELS;
config.capture.pDeviceID = NULL; // NULL for the default capture device.
config.capture.format = ma_format_s16;
config.capture.channels = 1;
config.sampleRate = DEVICE_SAMPLE_RATE;
config.dataCallback = OnSendAudioDataToDevice;
config.pUserData = NULL;
result = ma_device_init(&context, &config, &device);
if (result != MA_SUCCESS)
{
TraceLog(LOG_ERROR, "Failed to initialize audio playback device");
ma_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 = ma_device_start(&device);
if (result != MA_SUCCESS)
{
TraceLog(LOG_ERROR, "Failed to start audio playback device");
ma_device_uninit(&device);
ma_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 (ma_mutex_init(&context, &audioLock) != MA_SUCCESS)
{
TraceLog(LOG_ERROR, "Failed to create mutex for audio mixing");
ma_device_uninit(&device);
ma_context_uninit(&context);
return;
}
TraceLog(LOG_INFO, "Audio device initialized successfully");
TraceLog(LOG_INFO, "Audio backend: miniaudio / %s", ma_get_backend_name(context.backend));
TraceLog(LOG_INFO, "Audio format: %s -> %s", ma_get_format_name(device.playback.format), ma_get_format_name(device.playback.internalFormat));
TraceLog(LOG_INFO, "Audio channels: %d -> %d", device.playback.channels, device.playback.internalChannels);
TraceLog(LOG_INFO, "Audio sample rate: %d -> %d", device.sampleRate, device.playback.internalSampleRate);
TraceLog(LOG_INFO, "Audio buffer size: %d", device.playback.internalBufferSizeInFrames);
InitAudioBufferPool();
TraceLog(LOG_INFO, "Audio multichannel pool size: %i", MAX_AUDIO_BUFFER_POOL_CHANNELS);
isAudioInitialized = true;
}
// Close the audio device for all contexts
void CloseAudioDevice(void)
{
if (isAudioInitialized)
{
ma_mutex_uninit(&audioLock);
ma_device_uninit(&device);
ma_context_uninit(&context);
CloseAudioBufferPool();
TraceLog(LOG_INFO, "Audio device closed successfully");
}
else TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized");
}
// Check if device has been initialized successfully
bool IsAudioDeviceReady(void)
{
return isAudioInitialized;
}
// Set master volume (listener)
void SetMasterVolume(float volume)
{
if (volume < 0.0f) volume = 0.0f;
else if (volume > 1.0f) volume = 1.0f;
masterVolume = volume;
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Buffer management
//----------------------------------------------------------------------------------
// Initialize a new audio buffer (filled with silence)
AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, int usage)
{
AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer));
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "InitAudioBuffer() : Failed to allocate memory for audio buffer");
return NULL;
}
audioBuffer->buffer = RL_CALLOC(bufferSizeInFrames*channels*ma_get_bytes_per_sample(format), 1);
// Audio data runs through a format converter
ma_pcm_converter_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;
dspConfig.onRead = OnAudioBufferDSPRead; // Callback on data reading
dspConfig.pUserData = audioBuffer; // Audio data pointer
dspConfig.allowDynamicSampleRate = true; // Required for pitch shifting
ma_result result = ma_pcm_converter_init(&dspConfig, &audioBuffer->dsp);
if (result != MA_SUCCESS)
{
TraceLog(LOG_ERROR, "InitAudioBuffer() : Failed to create data conversion pipeline");
RL_FREE(audioBuffer);
return NULL;
}
// Init audio buffer values
audioBuffer->volume = 1.0f;
audioBuffer->pitch = 1.0f;
audioBuffer->playing = false;
audioBuffer->paused = false;
audioBuffer->looping = false;
audioBuffer->usage = usage;
audioBuffer->frameCursorPos = 0;
audioBuffer->bufferSizeInFrames = bufferSizeInFrames;
// 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 CloseAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL)
{
UntrackAudioBuffer(buffer);
RL_FREE(buffer->buffer);
RL_FREE(buffer);
}
else TraceLog(LOG_ERROR, "CloseAudioBuffer() : No audio buffer");
}
// Check if an audio buffer is playing
bool IsAudioBufferPlaying(AudioBuffer *buffer)
{
bool result = false;
if (buffer != NULL) result = (buffer->playing && !buffer->paused);
else TraceLog(LOG_WARNING, "IsAudioBufferPlaying() : No audio buffer");
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;
}
else TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
}
// Stop an audio buffer
void StopAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL)
{
if (IsAudioBufferPlaying(buffer))
{
buffer->playing = false;
buffer->paused = false;
buffer->frameCursorPos = 0;
buffer->totalFramesProcessed = 0;
buffer->isSubBufferProcessed[0] = true;
buffer->isSubBufferProcessed[1] = true;
}
}
else TraceLog(LOG_ERROR, "StopAudioBuffer() : No audio buffer");
}
// Pause an audio buffer
void PauseAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL) buffer->paused = true;
else TraceLog(LOG_ERROR, "PauseAudioBuffer() : No audio buffer");
}
// Resume an audio buffer
void ResumeAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL) buffer->paused = false;
else TraceLog(LOG_ERROR, "ResumeAudioBuffer() : No audio buffer");
}
// Set volume for an audio buffer
void SetAudioBufferVolume(AudioBuffer *buffer, float volume)
{
if (buffer != NULL) buffer->volume = volume;
else TraceLog(LOG_WARNING, "SetAudioBufferVolume() : No audio buffer");
}
// Set pitch for an audio buffer
void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
{
if (buffer != NULL)
{
float pitchMul = pitch/buffer->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
ma_uint32 newOutputSampleRate = (ma_uint32)((float)buffer->dsp.src.config.sampleRateOut/pitchMul);
buffer->pitch *= (float)buffer->dsp.src.config.sampleRateOut/newOutputSampleRate;
ma_pcm_converter_set_output_sample_rate(&buffer->dsp, newOutputSampleRate);
}
else TraceLog(LOG_WARNING, "SetAudioBufferPitch() : No audio buffer");
}
// Track audio buffer to linked list next position
void TrackAudioBuffer(AudioBuffer *buffer)
{
ma_mutex_lock(&audioLock);
{
if (firstAudioBuffer == NULL) firstAudioBuffer = buffer;
else
{
lastAudioBuffer->next = buffer;
buffer->prev = lastAudioBuffer;
}
lastAudioBuffer = buffer;
}
ma_mutex_unlock(&audioLock);
}
// Untrack audio buffer from linked list
void UntrackAudioBuffer(AudioBuffer *buffer)
{
ma_mutex_lock(&audioLock);
{
if (buffer->prev == NULL) firstAudioBuffer = buffer->next;
else buffer->prev->next = buffer->next;
if (buffer->next == NULL) lastAudioBuffer = buffer->prev;
else buffer->next->prev = buffer->prev;
buffer->prev = NULL;
buffer->next = NULL;
}
ma_mutex_unlock(&audioLock);
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Sounds loading and playing (.WAV)
//----------------------------------------------------------------------------------
// Load wave data from file
Wave LoadWave(const char *fileName)
{
Wave wave = { 0 };
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName);
#endif
#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
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (IsFileExtension(fileName, ".mp3")) wave = LoadMP3(fileName);
#endif
else TraceLog(LOG_WARNING, "[%s] Audio fileformat not supported, it can't be loaded", fileName);
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 device. We can do this two ways:
//
// 1) Convert the whole sound in one go at load time (here).
// 2) Convert the audio data in chunks at mixing time.
//
// First option has been selected, format conversion is done on the loading stage.
// The downside is that it uses more memory if the original sound is u8 or s16.
ma_format formatIn = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32));
ma_uint32 frameCountIn = wave.sampleCount/wave.channels;
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, formatIn, wave.channels, wave.sampleRate, frameCountIn);
if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Failed to get frame count for format conversion");
AudioBuffer *audioBuffer = InitAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC);
if (audioBuffer == NULL) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Failed to create audio buffer");
frameCount = (ma_uint32)ma_convert_frames(audioBuffer->buffer, audioBuffer->dsp.formatConverterIn.config.formatIn, audioBuffer->dsp.formatConverterIn.config.channels, audioBuffer->dsp.src.config.sampleRateIn, wave.data, formatIn, wave.channels, wave.sampleRate, frameCountIn);
if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed");
sound.sampleCount = frameCount*DEVICE_CHANNELS;
sound.stream.sampleRate = DEVICE_SAMPLE_RATE;
sound.stream.sampleSize = 32;
sound.stream.channels = DEVICE_CHANNELS;
sound.stream.buffer = audioBuffer;
}
return sound;
}
// Unload wave data
void UnloadWave(Wave wave)
{
if (wave.data != NULL) RL_FREE(wave.data);
TraceLog(LOG_INFO, "Unloaded wave data from RAM");
}
// Unload sound
void UnloadSound(Sound sound)
{
CloseAudioBuffer(sound.stream.buffer);
TraceLog(LOG_INFO, "Unloaded sound data from RAM");
}
// Update sound buffer with new data
void UpdateSound(Sound sound, const void *data, int samplesCount)
{
AudioBuffer *audioBuffer = sound.stream.buffer;
if (audioBuffer != NULL)
{
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.formatConverterIn.config.channels*ma_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn));
}
else TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer");
}
// Export wave data to file
void ExportWave(Wave wave, const char *fileName)
{
bool success = false;
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName);
#endif
else if (IsFileExtension(fileName, ".raw"))
{
// Export raw sample data (without header)
// NOTE: It's up to the user to track wave parameters
FILE *rawFile = fopen(fileName, "wb");
success = fwrite(wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8, 1, rawFile);
fclose(rawFile);
}
if (success) TraceLog(LOG_INFO, "Wave exported successfully: %s", fileName);
else TraceLog(LOG_WARNING, "Wave could not be exported.");
}
// Export wave sample data to code (.h)
void ExportWaveAsCode(Wave wave, const char *fileName)
{
#define BYTES_TEXT_PER_LINE 20
char varFileName[256] = { 0 };
int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;
FILE *txtFile = fopen(fileName, "wt");
if (txtFile != NULL)
{
fprintf(txtFile, "\n//////////////////////////////////////////////////////////////////////////////////\n");
fprintf(txtFile, "// //\n");
fprintf(txtFile, "// WaveAsCode exporter v1.0 - Wave data exported as an array of bytes //\n");
fprintf(txtFile, "// //\n");
fprintf(txtFile, "// more info and bugs-report: github.com/raysan5/raylib //\n");
fprintf(txtFile, "// feedback and support: ray[at]raylib.com //\n");
fprintf(txtFile, "// //\n");
fprintf(txtFile, "// Copyright (c) 2018 Ramon Santamaria (@raysan5) //\n");
fprintf(txtFile, "// //\n");
fprintf(txtFile, "//////////////////////////////////////////////////////////////////////////////////\n\n");
#if !defined(RAUDIO_STANDALONE)
// Get file name from path and convert variable name to uppercase
strcpy(varFileName, GetFileNameWithoutExt(fileName));
for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; }
#else
strcpy(varFileName, fileName);
#endif
fprintf(txtFile, "// Wave data information\n");
fprintf(txtFile, "#define %s_SAMPLE_COUNT %i\n", varFileName, wave.sampleCount);
fprintf(txtFile, "#define %s_SAMPLE_RATE %i\n", varFileName, wave.sampleRate);
fprintf(txtFile, "#define %s_SAMPLE_SIZE %i\n", varFileName, wave.sampleSize);
fprintf(txtFile, "#define %s_CHANNELS %i\n\n", varFileName, wave.channels);
// Write byte data as hexadecimal text
fprintf(txtFile, "static unsigned char %s_DATA[%i] = { ", varFileName, dataSize);
for (int i = 0; i < dataSize - 1; i++) fprintf(txtFile, ((i%BYTES_TEXT_PER_LINE == 0)? "0x%x,\n" : "0x%x, "), ((unsigned char *)wave.data)[i]);
fprintf(txtFile, "0x%x };\n", ((unsigned char *)wave.data)[dataSize - 1]);
fclose(txtFile);
}
}
// 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 (audioBufferPoolChannels[i] > oldAge)
{
oldAge = audioBufferPoolChannels[i];
oldIndex = i;
}
if (!IsAudioBufferPlaying(audioBufferPool[i]))
{
index = i;
break;
}
}
// If no none playing pool members can be index choose the oldest
if (index == -1)
{
TraceLog(LOG_WARNING,"pool age %i ended a sound early no room in buffer pool", audioBufferPoolCounter);
if (oldIndex == -1)
{
// Shouldn't be able to get here... but just in case something odd happens!
TraceLog(LOG_ERROR,"sound buffer pool couldn't determine oldest buffer not playing sound");
return;
}
index = oldIndex;
// Just in case...
StopAudioBuffer(audioBufferPool[index]);
}
// Experimentally mutex lock doesn't seem to be needed this makes sense
// as audioBufferPool[index] isn't playing and the only stuff we're copying
// shouldn't be changing...
audioBufferPoolChannels[index] = audioBufferPoolCounter;
audioBufferPoolCounter++;
audioBufferPool[index]->volume = sound.stream.buffer->volume;
audioBufferPool[index]->pitch = sound.stream.buffer->pitch;
audioBufferPool[index]->looping = sound.stream.buffer->looping;
audioBufferPool[index]->usage = sound.stream.buffer->usage;
audioBufferPool[index]->isSubBufferProcessed[0] = false;
audioBufferPool[index]->isSubBufferProcessed[1] = false;
audioBufferPool[index]->bufferSizeInFrames = sound.stream.buffer->bufferSizeInFrames;
audioBufferPool[index]->buffer = sound.stream.buffer->buffer;
PlayAudioBuffer(audioBufferPool[index]);
}
// Stop any sound played with PlaySoundMulti()
void StopSoundMulti(void)
{
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) StopAudioBuffer(audioBufferPool[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(audioBufferPool[i])) counter++;
}
return counter;
}
// Pause a sound
void PauseSound(Sound sound)
{
PauseAudioBuffer(sound.stream.buffer);
}
// Resume a paused sound
void ResumeSound(Sound sound)
{
ResumeAudioBuffer(sound.stream.buffer);
}
// Stop reproducing a sound
void StopSound(Sound sound)
{
StopAudioBuffer(sound.stream.buffer);
}
// Check if a sound is playing
bool IsSoundPlaying(Sound sound)
{
return IsAudioBufferPlaying(sound.stream.buffer);
}
// Set volume for a sound
void SetSoundVolume(Sound sound, float volume)
{
SetAudioBufferVolume(sound.stream.buffer, volume);
}
// Set pitch for a sound
void SetSoundPitch(Sound sound, float pitch)
{
SetAudioBufferPitch(sound.stream.buffer, pitch);
}
// Convert wave data to desired format
void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
{
ma_format formatIn = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32));
ma_format formatOut = (( sampleSize == 8)? ma_format_u8 : (( sampleSize == 16)? ma_format_s16 : ma_format_f32));
ma_uint32 frameCountIn = wave->sampleCount; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so.
ma_uint32 frameCount = (ma_uint32)ma_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 = RL_MALLOC(frameCount*channels*(sampleSize/8));
frameCount = (ma_uint32)ma_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;
RL_FREE(wave->data);
wave->data = data;
}
// Copy a wave to a new wave
Wave WaveCopy(Wave wave)
{
Wave newWave = { 0 };
newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8*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) && ((unsigned int)finalSample < wave->sampleCount))
{
int sampleCount = finalSample - initSample;
void *data = RL_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);
RL_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 *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(float));
for (unsigned int i = 0; i < wave.sampleCount; i++)
{
for (unsigned 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 = { 0 };
bool musicLoaded = false;
if (false) { }
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (IsFileExtension(fileName, ".ogg"))
{
// Open ogg audio stream
music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);
if (music.ctxData != NULL)
{
music.ctxType = MUSIC_AUDIO_OGG;
stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData); // Get Ogg file info
// OGG bit rate defaults to 16 bit, it's enough for compressed format
music.stream = InitAudioStream(info.sample_rate, 16, info.channels);
music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels;
music.loopCount = 0; // Infinite loop by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (IsFileExtension(fileName, ".flac"))
{
music.ctxData = drflac_open_file(fileName);
if (music.ctxData != NULL)
{
music.ctxType = MUSIC_AUDIO_FLAC;
drflac *ctxFlac = (drflac *)music.ctxData;
music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music.sampleCount = (unsigned int)ctxFlac->totalSampleCount;
music.loopCount = 0; // Infinite loop by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (IsFileExtension(fileName, ".mp3"))
{
drmp3 *ctxMp3 = RL_MALLOC(sizeof(drmp3));
music.ctxData = ctxMp3;
int result = drmp3_init_file(ctxMp3, fileName, NULL);
if (result > 0)
{
music.ctxType = MUSIC_AUDIO_MP3;
music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music.sampleCount = drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
music.loopCount = 0; // Infinite loop 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, 48000, fileName);
if (result == 0) // XM context created successfully
{
music.ctxType = MUSIC_MODULE_XM;
jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops
// NOTE: Only stereo is supported for XM
music.stream = InitAudioStream(48000, 16, 2);
music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm);
music.loopCount = 0; // Infinite loop by default
jar_xm_reset(ctxXm); // make sure we start at the beginning of the song
musicLoaded = true;
music.ctxData = ctxXm;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
else if (IsFileExtension(fileName, ".mod"))
{
jar_mod_context_t *ctxMod = RL_MALLOC(sizeof(jar_mod_context_t));
music.ctxData = ctxMod;
jar_mod_init(ctxMod);
int result = jar_mod_load_file(ctxMod, fileName);
if (result > 0)
{
music.ctxType = MUSIC_MODULE_MOD;
// NOTE: Only stereo is supported for MOD
music.stream = InitAudioStream(48000, 16, 2);
music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod);
music.loopCount = 0; // Infinite loop by default
musicLoaded = true;
}
}
#endif
if (!musicLoaded)
{
if (false) { }
#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);
#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
TraceLog(LOG_WARNING, "[%s] Music file could not be opened", fileName);
}
else
{
// Show some music stream info
TraceLog(LOG_INFO, "[%s] Music file successfully loaded:", fileName);
TraceLog(LOG_INFO, " Total samples: %i", music.sampleCount);
TraceLog(LOG_INFO, " Sample rate: %i Hz", music.stream.sampleRate);
TraceLog(LOG_INFO, " Sample size: %i bits", music.stream.sampleSize);
TraceLog(LOG_INFO, " Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
}
return music;
}
// Unload music stream
void UnloadMusicStream(Music music)
{
CloseAudioStream(music.stream);
if (false) { }
#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);
#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)
{
AudioBuffer *audioBuffer = music.stream.buffer;
if (audioBuffer != NULL)
{
// For music streams, we need to make sure we maintain the frame cursor position
// This is a hack for this section of code in UpdateMusicStream()
// NOTE: In case window is minimized, music stream is stopped, just make sure to
// play again on window restore: if (IsMusicPlaying(music)) PlayMusicStream(music);
ma_uint32 frameCursorPos = audioBuffer->frameCursorPos;
PlayAudioStream(music.stream); // WARNING: This resets the cursor position.
audioBuffer->frameCursorPos = frameCursorPos;
}
else TraceLog(LOG_ERROR, "PlayMusicStream() : No audio buffer");
}
// 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);
// Restart music context
switch (music.ctxType)
{
#if defined(SUPPORT_FILEFORMAT_OGG)
case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, 0); break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0); break;
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
case MUSIC_MODULE_XM: jar_xm_reset((jar_xm_context_t *)music.ctxData); break;
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
case MUSIC_MODULE_MOD: jar_mod_seek_start((jar_mod_context_t *)music.ctxData); break;
#endif
default: break;
}
}
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(Music music)
{
bool streamEnding = false;
unsigned int subBufferSizeInFrames = music.stream.buffer->bufferSizeInFrames/2;
// NOTE: Using dynamic allocation because it could require more than 16KB
void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1);
int samplesCount = 0; // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts
// TODO: Get the sampleLeft using totalFramesProcessed... but first, get total frames processed correctly...
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
int sampleLeft = music.sampleCount - (music.stream.buffer->totalFramesProcessed*music.stream.channels);
while (IsAudioStreamProcessed(music.stream))
{
if ((sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels;
else samplesCount = sampleLeft;
switch (music.ctxType)
{
#if defined(SUPPORT_FILEFORMAT_OGG)
case MUSIC_AUDIO_OGG:
{
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, samplesCount);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_AUDIO_FLAC:
{
// NOTE: Returns the number of samples to process (not required)
drflac_read_pcm_frames_s16((drflac *)music.ctxData, samplesCount, (short *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3:
{
// NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed
drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
case MUSIC_MODULE_XM:
{
// NOTE: Internally this function considers 2 channels generation, so samplesCount/2
jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, samplesCount/2);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
case MUSIC_MODULE_MOD:
{
// NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2
jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)pcm, samplesCount/2, 0);
} break;
#endif
default: break;
}
UpdateAudioStream(music.stream, pcm, samplesCount);
if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD))
{
if (samplesCount > 1) sampleLeft -= samplesCount/2;
else sampleLeft -= samplesCount;
}
else sampleLeft -= samplesCount;
if (sampleLeft <= 0)
{
streamEnding = true;
break;
}
}
// Free allocated pcm data
RL_FREE(pcm);
// Reset audio stream for looping
if (streamEnding)
{
StopMusicStream(music); // Stop music (and reset)
// Decrease loopCount to stop when required
if (music.loopCount > 1)
{
music.loopCount--; // Decrease loop count
PlayMusicStream(music); // Play again
}
else if (music.loopCount == 0) 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);
}
}
// Check if any music is playing
bool IsMusicPlaying(Music music)
{
return IsAudioStreamPlaying(music.stream);
}
// Set volume for music
void SetMusicVolume(Music music, float volume)
{
SetAudioStreamVolume(music.stream, volume);
}
// Set pitch for music
void SetMusicPitch(Music music, float pitch)
{
SetAudioStreamPitch(music.stream, pitch);
}
// Set music loop count (loop repeats)
// NOTE: If set to 0, means infinite loop
void SetMusicLoopCount(Music music, int count)
{
music.loopCount = count;
}
// Get music time length (in seconds)
float GetMusicTimeLength(Music music)
{
float totalSeconds = 0.0f;
totalSeconds = (float)music.sampleCount/(music.stream.sampleRate*music.stream.channels);
return totalSeconds;
}
// Get current music time played (in seconds)
float GetMusicTimePlayed(Music music)
{
float secondsPlayed = 0.0f;
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
unsigned int samplesPlayed = music.stream.buffer->totalFramesProcessed*music.stream.channels;
secondsPlayed = (float)samplesPlayed/(music.stream.sampleRate*music.stream.channels);
return secondsPlayed;
}
// Init audio stream (to stream audio pcm data)
AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)
{
AudioStream stream = { 0 };
stream.sampleRate = sampleRate;
stream.sampleSize = sampleSize;
stream.channels = channels;
ma_format formatIn = ((stream.sampleSize == 8)? ma_format_u8 : ((stream.sampleSize == 16)? ma_format_s16 : ma_format_f32));
// The size of a streaming buffer must be at least double the size of a period
unsigned int periodSize = device.playback.internalBufferSizeInFrames/device.playback.internalPeriods;
unsigned int subBufferSize = AUDIO_BUFFER_SIZE;
if (subBufferSize < periodSize) subBufferSize = periodSize;
stream.buffer = InitAudioBuffer(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, "Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo");
}
else TraceLog(LOG_ERROR, "InitAudioStream() : Failed to create audio buffer");
return stream;
}
// Close audio stream and free memory
void CloseAudioStream(AudioStream stream)
{
CloseAudioBuffer(stream.buffer);
TraceLog(LOG_INFO, "Unloaded audio stream data");
}
// Update audio stream buffers with data
// NOTE 1: Only updates one buffer of the stream source: unqueue -> update -> queue
// NOTE 2: To unqueue a buffer it needs to be processed: IsAudioStreamProcessed()
void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
{
AudioBuffer *audioBuffer = stream.buffer;
if (audioBuffer != NULL)
{
if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1])
{
ma_uint32 subBufferToUpdate = 0;
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;
}
ma_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2;
unsigned char *subBuffer = audioBuffer->buffer + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
// TODO: Get total frames processed on this buffer... DOES NOT WORK.
audioBuffer->totalFramesProcessed += subBufferSizeInFrames;
// Does this API expect a whole buffer to be updated in one go?
// Assuming so, but if not will need to change this logic.
if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels)
{
ma_uint32 framesToWrite = subBufferSizeInFrames;
if (framesToWrite > ((ma_uint32)samplesCount/stream.channels)) framesToWrite = (ma_uint32)samplesCount/stream.channels;
ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
memcpy(subBuffer, data, bytesToWrite);
// Any leftover frames should be filled with zeros.
ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
if (leftoverFrameCount > 0) memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));
audioBuffer->isSubBufferProcessed[subBufferToUpdate] = false;
}
else TraceLog(LOG_ERROR, "UpdateAudioStream() : Attempting to write too many frames to buffer");
}
else TraceLog(LOG_ERROR, "UpdateAudioStream() : Audio buffer not available for updating");
}
else TraceLog(LOG_ERROR, "UpdateAudioStream() : No audio buffer");
}
// Check if any audio stream buffers requires refill
bool IsAudioStreamProcessed(AudioStream stream)
{
if (stream.buffer == NULL)
{
TraceLog(LOG_ERROR, "IsAudioStreamProcessed() : No audio buffer");
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);
}
void SetAudioStreamVolume(AudioStream stream, float volume)
{
SetAudioBufferVolume(stream.buffer, volume);
}
void SetAudioStreamPitch(AudioStream stream, float pitch)
{
SetAudioBufferPitch(stream.buffer, pitch);
}
//----------------------------------------------------------------------------------
// 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 = RL_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;
}
// Save wave data as WAV file
static int SaveWAV(Wave wave, const char *fileName)
{
int success = 0;
int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;
// Basic WAV headers structs
typedef struct {
char chunkID[4];
int chunkSize;
char format[4];
} RiffHeader;
typedef struct {
char subChunkID[4];
int subChunkSize;
short audioFormat;
short numChannels;
int sampleRate;
int byteRate;
short blockAlign;
short bitsPerSample;
} WaveFormat;
typedef struct {
char subChunkID[4];
int subChunkSize;
} WaveData;
FILE *wavFile = fopen(fileName, "wb");
if (wavFile == NULL) TraceLog(LOG_WARNING, "[%s] WAV audio file could not be created", fileName);
else
{
RiffHeader riffHeader;
WaveFormat waveFormat;
WaveData waveData;
// Fill structs with data
riffHeader.chunkID[0] = 'R';
riffHeader.chunkID[1] = 'I';
riffHeader.chunkID[2] = 'F';
riffHeader.chunkID[3] = 'F';
riffHeader.chunkSize = 44 - 4 + wave.sampleCount*wave.sampleSize/8;
riffHeader.format[0] = 'W';
riffHeader.format[1] = 'A';
riffHeader.format[2] = 'V';
riffHeader.format[3] = 'E';
waveFormat.subChunkID[0] = 'f';
waveFormat.subChunkID[1] = 'm';
waveFormat.subChunkID[2] = 't';
waveFormat.subChunkID[3] = ' ';
waveFormat.subChunkSize = 16;
waveFormat.audioFormat = 1;
waveFormat.numChannels = wave.channels;
waveFormat.sampleRate = wave.sampleRate;
waveFormat.byteRate = wave.sampleRate*wave.sampleSize/8;
waveFormat.blockAlign = wave.sampleSize/8;
waveFormat.bitsPerSample = wave.sampleSize;
waveData.subChunkID[0] = 'd';
waveData.subChunkID[1] = 'a';
waveData.subChunkID[2] = 't';
waveData.subChunkID[3] = 'a';
waveData.subChunkSize = dataSize;
fwrite(&riffHeader, sizeof(RiffHeader), 1, wavFile);
fwrite(&waveFormat, sizeof(WaveFormat), 1, wavFile);
fwrite(&waveData, sizeof(WaveData), 1, wavFile);
success = fwrite(wave.data, dataSize, 1, wavFile);
fclose(wavFile);
}
// If all data has been written correctly to file, success = 1
return success;
}
#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 = (unsigned int)stb_vorbis_stream_length_in_samples(oggFile)*info.channels; // 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 *)RL_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_file_and_read_pcm_frames_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount);
wave.sampleCount = (unsigned int)totalSampleCount;
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
#if defined(SUPPORT_FILEFORMAT_MP3)
// Load MP3 file into Wave structure
// NOTE: Using dr_mp3 library
static Wave LoadMP3(const char *fileName)
{
Wave wave = { 0 };
// Decode an entire MP3 file in one go
uint64_t totalFrameCount = 0;
drmp3_config config = { 0 };
wave.data = drmp3_open_file_and_read_f32(fileName, &config, &totalFrameCount);
wave.channels = config.outputChannels;
wave.sampleRate = config.outputSampleRate;
wave.sampleCount = (int)totalFrameCount*wave.channels;
wave.sampleSize = 32;
// NOTE: Only support up to 2 channels (mono, stereo)
if (wave.channels > 2) TraceLog(LOG_WARNING, "[%s] MP3 channels number (%i) not supported", fileName, wave.channels);
if (wave.data == NULL) TraceLog(LOG_WARNING, "[%s] MP3 data could not be loaded", fileName);
else TraceLog(LOG_INFO, "[%s] MP3 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(RAUDIO_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
#undef AudioBuffer
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