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raylib/src/raudio.c
Ray c001bdb2de Checking issue with sound volume 
It seems individual sound volume level is not set...
2019-03-20 10:57:41 +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) 2014-2019 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
//----------------------------------------------------------------------------------
#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)
//----------------------------------------------------------------------------------
// Types and Structures Definition
//----------------------------------------------------------------------------------
typedef enum {
MUSIC_AUDIO_OGG = 0,
MUSIC_AUDIO_FLAC,
MUSIC_AUDIO_MP3,
MUSIC_MODULE_XM,
MUSIC_MODULE_MOD
} MusicContextType;
// Music type (file streaming from memory)
typedef struct MusicData {
MusicContextType ctxType; // Type of music context
#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_MP3)
drmp3 ctxMp3; // MP3 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(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
//----------------------------------------------------------------------------------
// miniaudio AudioBuffer Functionality
//----------------------------------------------------------------------------------
#define DEVICE_FORMAT ma_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
// NOTE: Slightly different logic is used when feeding data to the playback device depending on whether or not data is streamed
typedef struct rAudioBuffer rAudioBuffer;
struct rAudioBuffer {
ma_pcm_converter dsp; // Required for format conversion
float volume;
float pitch;
bool playing;
bool paused;
bool looping; // Always true for AudioStreams
int usage; // AudioBufferUsage type
bool isSubBufferProcessed[2];
unsigned int frameCursorPos;
unsigned int bufferSizeInFrames;
rAudioBuffer *next;
rAudioBuffer *prev;
unsigned char buffer[1];
};
// HACK: To avoid CoreAudio (macOS) symbol collision
// NOTE: This system should probably be redesigned
#define AudioBuffer rAudioBuffer
// miniaudio global variables
static ma_context context;
static ma_device device;
static ma_mutex audioLock;
static bool isAudioInitialized = MA_FALSE;
static float masterVolume = 1.0f;
// Audio buffers are tracked in a linked list
static AudioBuffer *firstAudioBuffer = NULL;
static AudioBuffer *lastAudioBuffer = NULL;
// 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 *CreateAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, AudioBufferUsage usage);
void DeleteAudioBuffer(AudioBuffer *audioBuffer);
bool IsAudioBufferPlaying(AudioBuffer *audioBuffer);
void PlayAudioBuffer(AudioBuffer *audioBuffer);
void StopAudioBuffer(AudioBuffer *audioBuffer);
void PauseAudioBuffer(AudioBuffer *audioBuffer);
void ResumeAudioBuffer(AudioBuffer *audioBuffer);
void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume);
void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch);
void TrackAudioBuffer(AudioBuffer *audioBuffer);
void UntrackAudioBuffer(AudioBuffer *audioBuffer);
// 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
static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount)
{
// 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->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;
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.
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 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;
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;
}
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);
}
}
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Device initialization and Closing
//----------------------------------------------------------------------------------
// Initialize audio device
void InitAudioDevice(void)
{
// 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;
}
// Device. 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);
isAudioInitialized = MA_TRUE;
}
// Close the audio device for all contexts
void CloseAudioDevice(void)
{
if (!isAudioInitialized)
{
TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized");
return;
}
ma_mutex_uninit(&audioLock);
ma_device_uninit(&device);
ma_context_uninit(&context);
TraceLog(LOG_INFO, "Audio device closed successfully");
}
// 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
//----------------------------------------------------------------------------------
// Create a new audio buffer. Initially filled with silence
AudioBuffer *CreateAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, AudioBufferUsage usage)
{
AudioBuffer *audioBuffer = (AudioBuffer *)calloc(sizeof(*audioBuffer) + (bufferSizeInFrames*channels*ma_get_bytes_per_sample(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.
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;
dspConfig.pUserData = audioBuffer;
dspConfig.allowDynamicSampleRate = MA_TRUE; // <-- Required for pitch shifting.
ma_result result = ma_pcm_converter_init(&dspConfig, &audioBuffer->dsp);
if (result != MA_SUCCESS)
{
TraceLog(LOG_ERROR, "CreateAudioBuffer() : Failed to create data conversion pipeline");
free(audioBuffer);
return NULL;
}
audioBuffer->volume = 1.0f;
audioBuffer->pitch = 1.0f;
audioBuffer->playing = false;
audioBuffer->paused = false;
audioBuffer->looping = false;
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, "DeleteAudioBuffer() : 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, "IsAudioBufferPlaying() : No audio buffer");
return false;
}
return audioBuffer->playing && !audioBuffer->paused;
}
// 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 *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, "StopAudioBuffer() : 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, "PauseAudioBuffer() : No audio buffer");
return;
}
audioBuffer->paused = true;
}
// Resume an audio buffer
void ResumeAudioBuffer(AudioBuffer *audioBuffer)
{
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "ResumeAudioBuffer() : No audio buffer");
return;
}
audioBuffer->paused = false;
}
// Set volume for an audio buffer
void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume)
{
if (audioBuffer == NULL)
{
TraceLog(LOG_WARNING, "SetAudioBufferVolume() : No audio buffer");
return;
}
audioBuffer->volume = volume;
}
// Set pitch for an audio buffer
void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch)
{
if (audioBuffer == NULL)
{
TraceLog(LOG_WARNING, "SetAudioBufferPitch() : No audio buffer");
return;
}
float pitchMul = pitch/audioBuffer->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)audioBuffer->dsp.src.config.sampleRateOut / pitchMul);
audioBuffer->pitch *= (float)audioBuffer->dsp.src.config.sampleRateOut / newOutputSampleRate;
ma_pcm_converter_set_output_sample_rate(&audioBuffer->dsp, newOutputSampleRate);
}
// Track audio buffer to linked list next position
void TrackAudioBuffer(AudioBuffer *audioBuffer)
{
ma_mutex_lock(&audioLock);
{
if (firstAudioBuffer == NULL) firstAudioBuffer = audioBuffer;
else
{
lastAudioBuffer->next = audioBuffer;
audioBuffer->prev = lastAudioBuffer;
}
lastAudioBuffer = audioBuffer;
}
ma_mutex_unlock(&audioLock);
}
// Untrack audio buffer from linked list
void UntrackAudioBuffer(AudioBuffer *audioBuffer)
{
ma_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;
}
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 defined(SUPPORT_FILEFORMAT_WAV)
if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName);
#else
if (false) {}
#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 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)
{
// 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.
//
// 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.
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 = CreateAudioBuffer(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.audioBuffer = audioBuffer;
}
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)
{
DeleteAudioBuffer((AudioBuffer *)sound.audioBuffer);
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)
{
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.formatConverterIn.config.channels*ma_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn));
}
// Export wave data to file
void ExportWave(Wave wave, const char *fileName)
{
bool success = false;
#if defined(SUPPORT_FILEFORMAT_WAV)
if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName);
#else
if (false) {}
#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");
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((AudioBuffer *)sound.audioBuffer);
}
// Pause a sound
void PauseSound(Sound sound)
{
PauseAudioBuffer((AudioBuffer *)sound.audioBuffer);
}
// Resume a paused sound
void ResumeSound(Sound sound)
{
ResumeAudioBuffer((AudioBuffer *)sound.audioBuffer);
}
// Stop reproducing a sound
void StopSound(Sound sound)
{
StopAudioBuffer((AudioBuffer *)sound.audioBuffer);
}
// Check if a sound is playing
bool IsSoundPlaying(Sound sound)
{
return IsAudioBufferPlaying((AudioBuffer *)sound.audioBuffer);
}
// Set volume for a sound
void SetSoundVolume(Sound sound, float volume)
{
SetAudioBufferVolume((AudioBuffer *)sound.audioBuffer, volume);
}
// Set pitch for a sound
void SetSoundPitch(Sound sound, float pitch)
{
SetAudioBufferPitch((AudioBuffer *)sound.audioBuffer, 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 = 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;
free(wave->data);
wave->data = data;
}
// 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) && ((unsigned int)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 (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 = (MusicData *)malloc(sizeof(MusicData));
bool musicLoaded = true;
#if defined(SUPPORT_FILEFORMAT_OGG)
if (IsFileExtension(fileName, ".ogg"))
{
// Open ogg audio stream
music->ctxOgg = stb_vorbis_open_filename(fileName, NULL, NULL);
if (music->ctxOgg == NULL) musicLoaded = false;
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)*info.channels;
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_AUDIO_OGG;
music->loopCount = -1; // Infinite loop by default
TraceLog(LOG_DEBUG, "[%s] OGG 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);
}
}
#else
if (false) {}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (IsFileExtension(fileName, ".flac"))
{
music->ctxFlac = drflac_open_file(fileName);
if (music->ctxFlac == NULL) musicLoaded = false;
else
{
music->stream = InitAudioStream(music->ctxFlac->sampleRate, music->ctxFlac->bitsPerSample, music->ctxFlac->channels);
music->totalSamples = (unsigned int)music->ctxFlac->totalSampleCount;
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_MP3)
else if (IsFileExtension(fileName, ".mp3"))
{
int result = drmp3_init_file(&music->ctxMp3, fileName, NULL);
if (!result) musicLoaded = false;
else
{
TraceLog(LOG_INFO, "[%s] MP3 sample rate: %i", fileName, music->ctxMp3.sampleRate);
TraceLog(LOG_INFO, "[%s] MP3 bits per sample: %i", fileName, 32);
TraceLog(LOG_INFO, "[%s] MP3 channels: %i", fileName, music->ctxMp3.channels);
music->stream = InitAudioStream(music->ctxMp3.sampleRate, 32, music->ctxMp3.channels);
// TODO: There is not an easy way to compute the total number of samples available
// in an MP3, frames size could be variable... we tried with a 60 seconds music... but crashes...
music->totalSamples = drmp3_get_pcm_frame_count(&music->ctxMp3)*music->ctxMp3.channels;
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_AUDIO_MP3;
music->loopCount = -1; // Infinite loop by default
TraceLog(LOG_INFO, "[%s] MP3 total samples: %i", fileName, music->totalSamples);
}
}
#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_INFO, "[%s] XM number of samples: %i", fileName, music->totalSamples);
TraceLog(LOG_INFO, "[%s] XM track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f);
}
else musicLoaded = false;
}
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
else if (IsFileExtension(fileName, ".mod"))
{
jar_mod_init(&music->ctxMod);
if (jar_mod_load_file(&music->ctxMod, fileName))
{
// NOTE: Only stereo is supported for MOD
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_INFO, "[%s] MOD number of samples: %i", fileName, music->samplesLeft);
TraceLog(LOG_INFO, "[%s] MOD track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f);
}
else musicLoaded = false;
}
#endif
else musicLoaded = false;
if (!musicLoaded)
{
#if defined(SUPPORT_FILEFORMAT_OGG)
if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close(music->ctxOgg);
#else
if (false) {}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (music->ctxType == MUSIC_AUDIO_FLAC) drflac_free(music->ctxFlac);
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (music->ctxType == MUSIC_AUDIO_MP3) drmp3_uninit(&music->ctxMp3);
#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);
music = NULL;
TraceLog(LOG_WARNING, "[%s] Music file could not be opened", fileName);
}
return music;
}
// Unload music stream
void UnloadMusicStream(Music music)
{
if (music == NULL) return;
CloseAudioStream(music->stream);
#if defined(SUPPORT_FILEFORMAT_OGG)
if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close(music->ctxOgg);
#else
if (false) {}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (music->ctxType == MUSIC_AUDIO_FLAC) drflac_free(music->ctxFlac);
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (music->ctxType == MUSIC_AUDIO_MP3) drmp3_uninit(&music->ctxMp3);
#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 (music != NULL)
{
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);
ma_uint32 frameCursorPos = audioBuffer->frameCursorPos;
PlayAudioStream(music->stream); // <-- This resets the cursor position.
audioBuffer->frameCursorPos = frameCursorPos;
}
}
// Pause music playing
void PauseMusicStream(Music music)
{
if (music != NULL) PauseAudioStream(music->stream);
}
// Resume music playing
void ResumeMusicStream(Music music)
{
if (music != NULL) ResumeAudioStream(music->stream);
}
// Stop music playing (close stream)
// TODO: To clear a buffer, make sure they have been already processed!
void StopMusicStream(Music music)
{
if (music == NULL) return;
StopAudioStream(music->stream);
// Restart music context
switch (music->ctxType)
{
#if defined(SUPPORT_FILEFORMAT_OGG)
case MUSIC_AUDIO_OGG: stb_vorbis_seek_start(music->ctxOgg); break;
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_AUDIO_FLAC: /* TODO: Restart FLAC context */ break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame(&music->ctxMp3, 0); 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 (music == NULL) return;
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.channels*music->stream.sampleSize/8, 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/music->stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music->stream.channels;
else samplesCount = music->samplesLeft;
// TODO: Really don't like ctxType thingy...
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(music->ctxOgg, 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
unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, 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(&music->ctxMp3, 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(music->ctxXm, (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(&music->ctxMod, (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) music->samplesLeft -= samplesCount/2;
else music->samplesLeft -= samplesCount;
}
else 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);
}
}
// Check if any music is playing
bool IsMusicPlaying(Music music)
{
if (music == NULL) return false;
else return IsAudioStreamPlaying(music->stream);
}
// Set volume for music
void SetMusicVolume(Music music, float volume)
{
if (music != NULL) SetAudioStreamVolume(music->stream, volume);
}
// Set pitch for music
void SetMusicPitch(Music music, float pitch)
{
if (music != NULL) SetAudioStreamPitch(music->stream, pitch);
}
// Set music loop count (loop repeats)
// NOTE: If set to -1, means infinite loop
void SetMusicLoopCount(Music music, int count)
{
if (music != NULL) music->loopCount = count;
}
// Get music time length (in seconds)
float GetMusicTimeLength(Music music)
{
float totalSeconds = 0.0f;
if (music != NULL) totalSeconds = (float)music->totalSamples/(music->stream.sampleRate*music->stream.channels);
return totalSeconds;
}
// Get current music time played (in seconds)
float GetMusicTimePlayed(Music music)
{
float secondsPlayed = 0.0f;
if (music != NULL)
{
unsigned int samplesPlayed = music->totalSamples - music->samplesLeft;
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;
// 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
}
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;
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;
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)
{
DeleteAudioBuffer((AudioBuffer *)stream.audioBuffer);
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)
{
AudioBuffer *audioBuffer = (AudioBuffer *)stream.audioBuffer;
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "UpdateAudioStream() : No audio buffer");
return;
}
if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1])
{
ma_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;
}
ma_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 >= (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");
return;
}
}
else
{
TraceLog(LOG_ERROR, "Audio buffer not available for updating");
return;
}
}
// Check if any audio stream buffers requires refill
bool IsAudioBufferProcessed(AudioStream stream)
{
AudioBuffer *audioBuffer = (AudioBuffer *)stream.audioBuffer;
if (audioBuffer == NULL)
{
TraceLog(LOG_ERROR, "IsAudioBufferProcessed() : No audio buffer");
return false;
}
return audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1];
}
// Play audio stream
void PlayAudioStream(AudioStream stream)
{
PlayAudioBuffer((AudioBuffer *)stream.audioBuffer);
}
// Play audio stream
void PauseAudioStream(AudioStream stream)
{
PauseAudioBuffer((AudioBuffer *)stream.audioBuffer);
}
// Resume audio stream playing
void ResumeAudioStream(AudioStream stream)
{
ResumeAudioBuffer((AudioBuffer *)stream.audioBuffer);
}
// Check if audio stream is playing.
bool IsAudioStreamPlaying(AudioStream stream)
{
return IsAudioBufferPlaying((AudioBuffer *)stream.audioBuffer);
}
// Stop audio stream
void StopAudioStream(AudioStream stream)
{
StopAudioBuffer((AudioBuffer *)stream.audioBuffer);
}
void SetAudioStreamVolume(AudioStream stream, float volume)
{
SetAudioBufferVolume((AudioBuffer *)stream.audioBuffer, volume);
}
void SetAudioStreamPitch(AudioStream stream, float pitch)
{
SetAudioBufferPitch((AudioBuffer *)stream.audioBuffer, 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 = 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;
success = fwrite(&riffHeader, sizeof(RiffHeader), 1, wavFile);
success = fwrite(&waveFormat, sizeof(WaveFormat), 1, wavFile);
success = 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 *)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 = (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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