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raylib/src/audio.c
raysan5 3884716948 Corrected a couple of warnings
2016-06-11 12:01:39 +02:00

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/**********************************************************************************************
*
* raylib.audio
*
* Basic functions to manage Audio: InitAudioDevice, LoadAudioFiles, PlayAudioFiles
*
* Uses external lib:
* OpenAL Soft - Audio device management lib (http://kcat.strangesoft.net/openal.html)
* stb_vorbis - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
*
* Copyright (c) 2014 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.
*
**********************************************************************************************/
//#define AUDIO_STANDALONE // NOTE: To use the audio module as standalone lib, just uncomment this line
#if defined(AUDIO_STANDALONE)
#include "audio.h"
#else
#include "raylib.h"
#endif
#include "AL/al.h" // OpenAL basic header
#include "AL/alc.h" // OpenAL context header (like OpenGL, OpenAL requires a context to work)
#include <stdlib.h> // Required for: malloc(), free()
#include <string.h> // Required for: strcmp(), strncmp()
#include <stdio.h> // Required for: FILE, fopen(), fclose(), fread()
#ifndef AL_FORMAT_MONO_FLOAT32
#define AL_FORMAT_MONO_FLOAT32 0x10010
#endif
#ifndef AL_FORMAT_STEREO_FLOAT32
#define AL_FORMAT_STEREO_FLOAT32 0x10011
#endif
#if defined(AUDIO_STANDALONE)
#include <stdarg.h> // Required for: va_list, va_start(), vfprintf(), va_end()
#else
#include "utils.h" // Required for: DecompressData()
// NOTE: Includes Android fopen() function map
#endif
//#define STB_VORBIS_HEADER_ONLY
#include "external/stb_vorbis.h" // OGG loading functions
#define JAR_XM_IMPLEMENTATION
#include "external/jar_xm.h" // XM loading functions
#define JAR_MOD_IMPLEMENTATION
#include "external/jar_mod.h" // MOD loading functions
//----------------------------------------------------------------------------------
// Defines and Macros
//----------------------------------------------------------------------------------
#define MAX_STREAM_BUFFERS 2 // Number of buffers for each alSource
#define MAX_MIX_CHANNELS 4 // Number of open AL sources
#define MAX_MUSIC_STREAMS 2 // Number of simultanious music sources
#if defined(PLATFORM_RPI) || defined(PLATFORM_ANDROID)
// NOTE: On RPI and Android should be lower to avoid frame-stalls
#define MUSIC_BUFFER_SIZE_SHORT 4096*2 // PCM data buffer (short) - 16Kb (RPI)
#define MUSIC_BUFFER_SIZE_FLOAT 4096 // PCM data buffer (float) - 16Kb (RPI)
#else
// NOTE: On HTML5 (emscripten) this is allocated on heap, by default it's only 16MB!...just take care...
#define MUSIC_BUFFER_SIZE_SHORT 4096*8 // PCM data buffer (short) - 64Kb
#define MUSIC_BUFFER_SIZE_FLOAT 4096*4 // PCM data buffer (float) - 64Kb
#endif
//----------------------------------------------------------------------------------
// Types and Structures Definition
//----------------------------------------------------------------------------------
// Used to create custom audio streams that are not bound to a specific file. There can be
// no more than 4 concurrent mixchannels in use. This is due to each active mixc being tied to
// a dedicated mix channel.
typedef struct MixChannel_t {
unsigned short sampleRate; // default is 48000
unsigned char channels; // 1=mono,2=stereo
unsigned char mixChannel; // 0-3 or mixA-mixD, each mix channel can receive up to one dedicated audio stream
bool floatingPoint; // if false then the short datatype is used instead
bool playing; // false if paused
ALenum alFormat; // OpenAL format specifier
ALuint alSource; // OpenAL source
ALuint alBuffer[MAX_STREAM_BUFFERS]; // OpenAL sample buffer
} MixChannel_t;
// Music type (file streaming from memory)
// NOTE: Anything longer than ~10 seconds should be streamed into a mix channel...
typedef struct Music {
stb_vorbis *stream;
jar_xm_context_t *xmctx; // XM chiptune context
jar_mod_context_t modctx; // MOD chiptune context
MixChannel_t *mixc; // mix channel
unsigned int totalSamplesLeft;
float totalLengthSeconds;
bool loop;
bool chipTune; // chiptune is loaded?
bool enabled;
} Music;
// Audio errors registered
typedef enum {
ERROR_RAW_CONTEXT_CREATION = 1,
ERROR_XM_CONTEXT_CREATION = 2,
ERROR_MOD_CONTEXT_CREATION = 4,
ERROR_MIX_CHANNEL_CREATION = 8,
ERROR_MUSIC_CHANNEL_CREATION = 16,
ERROR_LOADING_XM = 32,
ERROR_LOADING_MOD = 64,
ERROR_LOADING_WAV = 128,
ERROR_LOADING_OGG = 256,
ERROR_OUT_OF_MIX_CHANNELS = 512,
ERROR_EXTENSION_NOT_RECOGNIZED = 1024,
ERROR_UNABLE_TO_OPEN_RRES_FILE = 2048,
ERROR_INVALID_RRES_FILE = 4096,
ERROR_INVALID_RRES_RESOURCE = 8192,
ERROR_UNINITIALIZED_CHANNELS = 16384
} AudioError;
#if defined(AUDIO_STANDALONE)
typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
#endif
//----------------------------------------------------------------------------------
// Global Variables Definition
//----------------------------------------------------------------------------------
static Music musicChannels_g[MAX_MUSIC_STREAMS]; // Current music loaded, up to two can play at the same time
static MixChannel_t *mixChannels_g[MAX_MIX_CHANNELS]; // What mix channels are currently active
static int lastAudioError = 0; // Registers last audio error
//----------------------------------------------------------------------------------
// Module specific Functions Declaration
//----------------------------------------------------------------------------------
static Wave LoadWAV(const char *fileName); // Load WAV file
static Wave LoadOGG(char *fileName); // Load OGG file
static void UnloadWave(Wave wave); // Unload wave data
static bool BufferMusicStream(int index, int numBuffers); // Fill music buffers with data
static void EmptyMusicStream(int index); // Empty music buffers
static MixChannel_t *InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint); // For streaming into mix channels.
static void CloseMixChannel(MixChannel_t *mixc); // Frees mix channel
static int BufferMixChannel(MixChannel_t *mixc, void *data, int numberElements); // Pushes more audio data into mixc mix channel, if NULL is passed it pauses
static int FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer); // Fill buffer with zeros, returns number processed
static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // Pass two arrays of the same legnth in
static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // Pass two arrays of same length in
static int IsMusicStreamReadyForBuffering(int index); // Checks if music buffer is ready to be refilled
#if defined(AUDIO_STANDALONE)
const char *GetExtension(const char *fileName); // Get the extension for a filename
void TraceLog(int msgType, const char *text, ...); // Outputs a trace log message (INFO, ERROR, WARNING)
#endif
//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Device initialization and Closing
//----------------------------------------------------------------------------------
// Initialize audio device and mixc
void InitAudioDevice(void)
{
// Open and initialize a device with default settings
ALCdevice *device = alcOpenDevice(NULL);
if (!device) TraceLog(ERROR, "Audio device could not be opened");
ALCcontext *context = alcCreateContext(device, NULL);
if ((context == NULL) || (alcMakeContextCurrent(context) == ALC_FALSE))
{
if (context != NULL) alcDestroyContext(context);
alcCloseDevice(device);
TraceLog(ERROR, "Could not setup mix channel");
}
TraceLog(INFO, "Audio device and context initialized successfully: %s", alcGetString(device, ALC_DEVICE_SPECIFIER));
// Listener definition (just for 2D)
alListener3f(AL_POSITION, 0, 0, 0);
alListener3f(AL_VELOCITY, 0, 0, 0);
alListener3f(AL_ORIENTATION, 0, 0, -1);
}
// Close the audio device for all contexts
void CloseAudioDevice(void)
{
for (int index=0; index<MAX_MUSIC_STREAMS; index++)
{
if (musicChannels_g[index].mixc) StopMusicStream(index); // Stop music streaming and close current stream
}
ALCdevice *device;
ALCcontext *context = alcGetCurrentContext();
if (context == NULL) TraceLog(WARNING, "Could not get current mix channel for closing");
device = alcGetContextsDevice(context);
alcMakeContextCurrent(NULL);
alcDestroyContext(context);
alcCloseDevice(device);
}
// True if call to InitAudioDevice() was successful and CloseAudioDevice() has not been called yet
bool IsAudioDeviceReady(void)
{
ALCcontext *context = alcGetCurrentContext();
if (context == NULL) return false;
else
{
ALCdevice *device = alcGetContextsDevice(context);
if (device == NULL) return false;
else return true;
}
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Custom audio output
//----------------------------------------------------------------------------------
// For streaming into mix channels.
// The mixChannel is what audio muxing channel you want to operate on, 0-3 are the ones available. Each mix channel can only be used one at a time.
// exmple usage is InitMixChannel(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
static MixChannel_t *InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint)
{
if (mixChannel >= MAX_MIX_CHANNELS) return NULL;
if (!IsAudioDeviceReady()) InitAudioDevice();
if (!mixChannels_g[mixChannel])
{
MixChannel_t *mixc = (MixChannel_t *)malloc(sizeof(MixChannel_t));
mixc->sampleRate = sampleRate;
mixc->channels = channels;
mixc->mixChannel = mixChannel;
mixc->floatingPoint = floatingPoint;
mixChannels_g[mixChannel] = mixc;
// Setup OpenAL format
if (channels == 1)
{
if (floatingPoint) mixc->alFormat = AL_FORMAT_MONO_FLOAT32;
else mixc->alFormat = AL_FORMAT_MONO16;
}
else if (channels == 2)
{
if (floatingPoint) mixc->alFormat = AL_FORMAT_STEREO_FLOAT32;
else mixc->alFormat = AL_FORMAT_STEREO16;
}
// Create an audio source
alGenSources(1, &mixc->alSource);
alSourcef(mixc->alSource, AL_PITCH, 1);
alSourcef(mixc->alSource, AL_GAIN, 1);
alSource3f(mixc->alSource, AL_POSITION, 0, 0, 0);
alSource3f(mixc->alSource, AL_VELOCITY, 0, 0, 0);
// Create Buffer
alGenBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
// Fill buffers
for (int i = 0; i < MAX_STREAM_BUFFERS; i++) FillAlBufferWithSilence(mixc, mixc->alBuffer[i]);
alSourceQueueBuffers(mixc->alSource, MAX_STREAM_BUFFERS, mixc->alBuffer);
mixc->playing = true;
alSourcePlay(mixc->alSource);
return mixc;
}
return NULL;
}
// Frees buffer in mix channel
static void CloseMixChannel(MixChannel_t *mixc)
{
if (mixc)
{
alSourceStop(mixc->alSource);
mixc->playing = false;
// Flush out all queued buffers
ALuint buffer = 0;
int queued = 0;
alGetSourcei(mixc->alSource, AL_BUFFERS_QUEUED, &queued);
while (queued > 0)
{
alSourceUnqueueBuffers(mixc->alSource, 1, &buffer);
queued--;
}
// Delete source and buffers
alDeleteSources(1, &mixc->alSource);
alDeleteBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
mixChannels_g[mixc->mixChannel] = NULL;
free(mixc);
mixc = NULL;
}
}
// Pushes more audio data into mixc mix channel, only one buffer per call
// Call "BufferMixChannel(mixc, NULL, 0)" if you want to pause the audio.
// @Returns number of samples that where processed.
static int BufferMixChannel(MixChannel_t *mixc, void *data, int numberElements)
{
if (!mixc || (mixChannels_g[mixc->mixChannel] != mixc)) return 0; // When there is two channels there must be an even number of samples
if (!data || !numberElements)
{
// Pauses audio until data is given
if (mixc->playing)
{
alSourcePause(mixc->alSource);
mixc->playing = false;
}
return 0;
}
else if (!mixc->playing)
{
// Restart audio otherwise
alSourcePlay(mixc->alSource);
mixc->playing = true;
}
ALuint buffer = 0;
alSourceUnqueueBuffers(mixc->alSource, 1, &buffer);
if (!buffer) return 0;
if (mixc->floatingPoint)
{
// Process float buffers
float *ptr = (float *)data;
alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(float), mixc->sampleRate);
}
else
{
// Process short buffers
short *ptr = (short *)data;
alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(short), mixc->sampleRate);
}
alSourceQueueBuffers(mixc->alSource, 1, &buffer);
return numberElements;
}
// fill buffer with zeros, returns number processed
static int FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer)
{
if (mixc->floatingPoint)
{
float pcm[MUSIC_BUFFER_SIZE_FLOAT] = { 0.0f };
alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), mixc->sampleRate);
return MUSIC_BUFFER_SIZE_FLOAT;
}
else
{
short pcm[MUSIC_BUFFER_SIZE_SHORT] = { 0 };
alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), mixc->sampleRate);
return MUSIC_BUFFER_SIZE_SHORT;
}
}
// example usage:
// short sh[3] = {1,2,3};float fl[3];
// ResampleShortToFloat(sh,fl,3);
static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len)
{
for (int i = 0; i < len; i++)
{
if (shorts[i] < 0) floats[i] = (float)shorts[i]/32766.0f;
else floats[i] = (float)shorts[i]/32767.0f;
}
}
// example usage:
// char ch[3] = {1,2,3};float fl[3];
// ResampleByteToFloat(ch,fl,3);
static void ResampleByteToFloat(char *chars, float *floats, unsigned short len)
{
for (int i = 0; i < len; i++)
{
if (chars[i] < 0) floats[i] = (float)chars[i]/127.0f;
else floats[i] = (float)chars[i]/128.0f;
}
}
// used to output raw audio streams, returns negative numbers on error, + number represents the mix channel index
// if floating point is false the data size is 16bit short, otherwise it is float 32bit
RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint)
{
int mixIndex;
for (mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
{
if (mixChannels_g[mixIndex] == NULL) break;
else if (mixIndex == (MAX_MIX_CHANNELS - 1)) return ERROR_OUT_OF_MIX_CHANNELS; // error
}
if (InitMixChannel(sampleRate, mixIndex, channels, floatingPoint)) return mixIndex;
else return ERROR_RAW_CONTEXT_CREATION; // error
}
void CloseRawAudioContext(RawAudioContext ctx)
{
if (mixChannels_g[ctx]) CloseMixChannel(mixChannels_g[ctx]);
}
// if 0 is returned, the buffers are still full and you need to keep trying with the same data until a + number is returned.
// any + number returned is the number of samples that was processed and passed into buffer.
// data either needs to be array of floats or shorts.
int BufferRawAudioContext(RawAudioContext ctx, void *data, unsigned short numberElements)
{
int numBuffered = 0;
if (ctx >= 0)
{
MixChannel_t* mixc = mixChannels_g[ctx];
numBuffered = BufferMixChannel(mixc, data, numberElements);
}
return numBuffered;
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Sounds loading and playing (.WAV)
//----------------------------------------------------------------------------------
// Load sound to memory
Sound LoadSound(char *fileName)
{
Sound sound = { 0 };
Wave wave = { 0 };
// NOTE: The entire file is loaded to memory to play it all at once (no-streaming)
// Audio file loading
// NOTE: Buffer space is allocated inside function, Wave must be freed
if (strcmp(GetExtension(fileName),"wav") == 0) wave = LoadWAV(fileName);
else if (strcmp(GetExtension(fileName),"ogg") == 0) wave = LoadOGG(fileName);
else
{
TraceLog(WARNING, "[%s] Sound extension not recognized, it can't be loaded", fileName);
// TODO: Find a better way to register errors (similar to glGetError())
lastAudioError = ERROR_EXTENSION_NOT_RECOGNIZED;
}
if (wave.data != NULL)
{
ALenum format = 0;
// The OpenAL format is worked out by looking at the number of channels and the bits per sample
if (wave.channels == 1)
{
if (wave.bitsPerSample == 8 ) format = AL_FORMAT_MONO8;
else if (wave.bitsPerSample == 16) format = AL_FORMAT_MONO16;
}
else if (wave.channels == 2)
{
if (wave.bitsPerSample == 8 ) format = AL_FORMAT_STEREO8;
else if (wave.bitsPerSample == 16) format = AL_FORMAT_STEREO16;
}
// Create an audio source
ALuint source;
alGenSources(1, &source); // Generate pointer to audio source
alSourcef(source, AL_PITCH, 1);
alSourcef(source, AL_GAIN, 1);
alSource3f(source, AL_POSITION, 0, 0, 0);
alSource3f(source, AL_VELOCITY, 0, 0, 0);
alSourcei(source, AL_LOOPING, AL_FALSE);
// Convert loaded data to OpenAL buffer
//----------------------------------------
ALuint buffer;
alGenBuffers(1, &buffer); // Generate pointer to buffer
// Upload sound data to buffer
alBufferData(buffer, format, wave.data, wave.dataSize, wave.sampleRate);
// Attach sound buffer to source
alSourcei(source, AL_BUFFER, buffer);
TraceLog(INFO, "[%s] Sound file loaded successfully (SampleRate: %i, BitRate: %i, Channels: %i)", fileName, wave.sampleRate, wave.bitsPerSample, wave.channels);
// Unallocate WAV data
UnloadWave(wave);
sound.source = source;
sound.buffer = buffer;
}
return sound;
}
// Load sound from wave data
Sound LoadSoundFromWave(Wave wave)
{
Sound sound = { 0 };
if (wave.data != NULL)
{
ALenum format = 0;
// The OpenAL format is worked out by looking at the number of channels and the bits per sample
if (wave.channels == 1)
{
if (wave.bitsPerSample == 8 ) format = AL_FORMAT_MONO8;
else if (wave.bitsPerSample == 16) format = AL_FORMAT_MONO16;
}
else if (wave.channels == 2)
{
if (wave.bitsPerSample == 8 ) format = AL_FORMAT_STEREO8;
else if (wave.bitsPerSample == 16) format = AL_FORMAT_STEREO16;
}
// Create an audio source
ALuint source;
alGenSources(1, &source); // Generate pointer to audio source
alSourcef(source, AL_PITCH, 1);
alSourcef(source, AL_GAIN, 1);
alSource3f(source, AL_POSITION, 0, 0, 0);
alSource3f(source, AL_VELOCITY, 0, 0, 0);
alSourcei(source, AL_LOOPING, AL_FALSE);
// Convert loaded data to OpenAL buffer
//----------------------------------------
ALuint buffer;
alGenBuffers(1, &buffer); // Generate pointer to buffer
// Upload sound data to buffer
alBufferData(buffer, format, wave.data, wave.dataSize, wave.sampleRate);
// Attach sound buffer to source
alSourcei(source, AL_BUFFER, buffer);
// Unallocate WAV data
UnloadWave(wave);
TraceLog(INFO, "[Wave] Sound file loaded successfully (SampleRate: %i, BitRate: %i, Channels: %i)", wave.sampleRate, wave.bitsPerSample, wave.channels);
sound.source = source;
sound.buffer = buffer;
}
return sound;
}
// Load sound to memory from rRES file (raylib Resource)
// TODO: Maybe rresName could be directly a char array with all the data?
Sound LoadSoundFromRES(const char *rresName, int resId)
{
Sound sound = { 0 };
#if defined(AUDIO_STANDALONE)
TraceLog(WARNING, "Sound loading from rRES resource file not supported on standalone mode");
#else
bool found = false;
char id[4]; // rRES file identifier
unsigned char version; // rRES file version and subversion
char useless; // rRES header reserved data
short numRes;
ResInfoHeader infoHeader;
FILE *rresFile = fopen(rresName, "rb");
if (rresFile == NULL)
{
TraceLog(WARNING, "[%s] rRES raylib resource file could not be opened", rresName);
lastAudioError = ERROR_UNABLE_TO_OPEN_RRES_FILE;
}
else
{
// Read rres file (basic file check - id)
fread(&id[0], sizeof(char), 1, rresFile);
fread(&id[1], sizeof(char), 1, rresFile);
fread(&id[2], sizeof(char), 1, rresFile);
fread(&id[3], sizeof(char), 1, rresFile);
fread(&version, sizeof(char), 1, rresFile);
fread(&useless, sizeof(char), 1, rresFile);
if ((id[0] != 'r') && (id[1] != 'R') && (id[2] != 'E') &&(id[3] != 'S'))
{
TraceLog(WARNING, "[%s] This is not a valid raylib resource file", rresName);
lastAudioError = ERROR_INVALID_RRES_FILE;
}
else
{
// Read number of resources embedded
fread(&numRes, sizeof(short), 1, rresFile);
for (int i = 0; i < numRes; i++)
{
fread(&infoHeader, sizeof(ResInfoHeader), 1, rresFile);
if (infoHeader.id == resId)
{
found = true;
// Check data is of valid SOUND type
if (infoHeader.type == 1) // SOUND data type
{
// TODO: Check data compression type
// NOTE: We suppose compression type 2 (DEFLATE - default)
// Reading SOUND parameters
Wave wave;
short sampleRate, bps;
char channels, reserved;
fread(&sampleRate, sizeof(short), 1, rresFile); // Sample rate (frequency)
fread(&bps, sizeof(short), 1, rresFile); // Bits per sample
fread(&channels, 1, 1, rresFile); // Channels (1 - mono, 2 - stereo)
fread(&reserved, 1, 1, rresFile); // <reserved>
wave.sampleRate = sampleRate;
wave.dataSize = infoHeader.srcSize;
wave.bitsPerSample = bps;
wave.channels = (short)channels;
unsigned char *data = malloc(infoHeader.size);
fread(data, infoHeader.size, 1, rresFile);
wave.data = DecompressData(data, infoHeader.size, infoHeader.srcSize);
free(data);
// Convert wave to Sound (OpenAL)
ALenum format = 0;
// The OpenAL format is worked out by looking at the number of channels and the bits per sample
if (wave.channels == 1)
{
if (wave.bitsPerSample == 8 ) format = AL_FORMAT_MONO8;
else if (wave.bitsPerSample == 16) format = AL_FORMAT_MONO16;
}
else if (wave.channels == 2)
{
if (wave.bitsPerSample == 8 ) format = AL_FORMAT_STEREO8;
else if (wave.bitsPerSample == 16) format = AL_FORMAT_STEREO16;
}
// Create an audio source
ALuint source;
alGenSources(1, &source); // Generate pointer to audio source
alSourcef(source, AL_PITCH, 1);
alSourcef(source, AL_GAIN, 1);
alSource3f(source, AL_POSITION, 0, 0, 0);
alSource3f(source, AL_VELOCITY, 0, 0, 0);
alSourcei(source, AL_LOOPING, AL_FALSE);
// Convert loaded data to OpenAL buffer
//----------------------------------------
ALuint buffer;
alGenBuffers(1, &buffer); // Generate pointer to buffer
// Upload sound data to buffer
alBufferData(buffer, format, (void*)wave.data, wave.dataSize, wave.sampleRate);
// Attach sound buffer to source
alSourcei(source, AL_BUFFER, buffer);
TraceLog(INFO, "[%s] Sound loaded successfully from resource (SampleRate: %i, BitRate: %i, Channels: %i)", rresName, wave.sampleRate, wave.bitsPerSample, wave.channels);
// Unallocate WAV data
UnloadWave(wave);
sound.source = source;
sound.buffer = buffer;
}
else
{
TraceLog(WARNING, "[%s] Required resource do not seem to be a valid SOUND resource", rresName);
lastAudioError = ERROR_INVALID_RRES_RESOURCE;
}
}
else
{
// Depending on type, skip the right amount of parameters
switch (infoHeader.type)
{
case 0: fseek(rresFile, 6, SEEK_CUR); break; // IMAGE: Jump 6 bytes of parameters
case 1: fseek(rresFile, 6, SEEK_CUR); break; // SOUND: Jump 6 bytes of parameters
case 2: fseek(rresFile, 5, SEEK_CUR); break; // MODEL: Jump 5 bytes of parameters (TODO: Review)
case 3: break; // TEXT: No parameters
case 4: break; // RAW: No parameters
default: break;
}
// Jump DATA to read next infoHeader
fseek(rresFile, infoHeader.size, SEEK_CUR);
}
}
}
fclose(rresFile);
}
if (!found) TraceLog(WARNING, "[%s] Required resource id [%i] could not be found in the raylib resource file", rresName, resId);
#endif
return sound;
}
// Unload sound
void UnloadSound(Sound sound)
{
alDeleteSources(1, &sound.source);
alDeleteBuffers(1, &sound.buffer);
TraceLog(INFO, "Unloaded sound data");
}
// Play a sound
void PlaySound(Sound sound)
{
alSourcePlay(sound.source); // Play the sound
//TraceLog(INFO, "Playing sound");
// Find the current position of the sound being played
// NOTE: Only work when the entire file is in a single buffer
//int byteOffset;
//alGetSourcei(sound.source, AL_BYTE_OFFSET, &byteOffset);
//
//int sampleRate;
//alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate); // AL_CHANNELS, AL_BITS (bps)
//float seconds = (float)byteOffset / sampleRate; // Number of seconds since the beginning of the sound
//or
//float result;
//alGetSourcef(sound.source, AL_SEC_OFFSET, &result); // AL_SAMPLE_OFFSET
}
// Pause a sound
void PauseSound(Sound sound)
{
alSourcePause(sound.source);
}
// Stop reproducing a sound
void StopSound(Sound sound)
{
alSourceStop(sound.source);
}
// Check if a sound is playing
bool IsSoundPlaying(Sound sound)
{
bool playing = false;
ALint state;
alGetSourcei(sound.source, AL_SOURCE_STATE, &state);
if (state == AL_PLAYING) playing = true;
return playing;
}
// Set volume for a sound
void SetSoundVolume(Sound sound, float volume)
{
alSourcef(sound.source, AL_GAIN, volume);
}
// Set pitch for a sound
void SetSoundPitch(Sound sound, float pitch)
{
alSourcef(sound.source, AL_PITCH, pitch);
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Music loading and stream playing (.OGG)
//----------------------------------------------------------------------------------
// Start music playing (open stream)
// returns 0 on success
int PlayMusicStream(int index, char *fileName)
{
int mixIndex;
if (musicChannels_g[index].stream || musicChannels_g[index].xmctx) return ERROR_UNINITIALIZED_CHANNELS; // error
for (mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
{
if (mixChannels_g[mixIndex] == NULL) break;
else if (mixIndex == (MAX_MIX_CHANNELS - 1)) return ERROR_OUT_OF_MIX_CHANNELS; // error
}
if (strcmp(GetExtension(fileName),"ogg") == 0)
{
// Open audio stream
musicChannels_g[index].stream = stb_vorbis_open_filename(fileName, NULL, NULL);
if (musicChannels_g[index].stream == NULL)
{
TraceLog(WARNING, "[%s] OGG audio file could not be opened", fileName);
return ERROR_LOADING_OGG; // error
}
else
{
// Get file info
stb_vorbis_info info = stb_vorbis_get_info(musicChannels_g[index].stream);
TraceLog(INFO, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
TraceLog(INFO, "[%s] Ogg channels: %i", fileName, info.channels);
TraceLog(DEBUG, "[%s] Temp memory required: %i", fileName, info.temp_memory_required);
musicChannels_g[index].loop = true; // We loop by default
musicChannels_g[index].enabled = true;
musicChannels_g[index].totalSamplesLeft = (unsigned int)stb_vorbis_stream_length_in_samples(musicChannels_g[index].stream) * info.channels;
musicChannels_g[index].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(musicChannels_g[index].stream);
if (info.channels == 2)
{
musicChannels_g[index].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false);
musicChannels_g[index].mixc->playing = true;
}
else
{
musicChannels_g[index].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false);
musicChannels_g[index].mixc->playing = true;
}
if (!musicChannels_g[index].mixc) return ERROR_LOADING_OGG; // error
}
}
else if (strcmp(GetExtension(fileName),"xm") == 0)
{
// only stereo is supported for xm
if (!jar_xm_create_context_from_file(&musicChannels_g[index].xmctx, 48000, fileName))
{
musicChannels_g[index].chipTune = true;
musicChannels_g[index].loop = true;
jar_xm_set_max_loop_count(musicChannels_g[index].xmctx, 0); // infinite number of loops
musicChannels_g[index].totalSamplesLeft = (unsigned int)jar_xm_get_remaining_samples(musicChannels_g[index].xmctx);
musicChannels_g[index].totalLengthSeconds = ((float)musicChannels_g[index].totalSamplesLeft) / 48000.f;
musicChannels_g[index].enabled = true;
TraceLog(INFO, "[%s] XM number of samples: %i", fileName, musicChannels_g[index].totalSamplesLeft);
TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, musicChannels_g[index].totalLengthSeconds);
musicChannels_g[index].mixc = InitMixChannel(48000, mixIndex, 2, true);
if (!musicChannels_g[index].mixc) return ERROR_XM_CONTEXT_CREATION; // error
musicChannels_g[index].mixc->playing = true;
}
else
{
TraceLog(WARNING, "[%s] XM file could not be opened", fileName);
return ERROR_LOADING_XM; // error
}
}
else if (strcmp(GetExtension(fileName),"mod") == 0)
{
jar_mod_init(&musicChannels_g[index].modctx);
if (jar_mod_load_file(&musicChannels_g[index].modctx, fileName))
{
musicChannels_g[index].chipTune = true;
musicChannels_g[index].loop = true;
musicChannels_g[index].totalSamplesLeft = (unsigned int)jar_mod_max_samples(&musicChannels_g[index].modctx);
musicChannels_g[index].totalLengthSeconds = ((float)musicChannels_g[index].totalSamplesLeft) / 48000.f;
musicChannels_g[index].enabled = true;
TraceLog(INFO, "[%s] MOD number of samples: %i", fileName, musicChannels_g[index].totalSamplesLeft);
TraceLog(INFO, "[%s] MOD track length: %11.6f sec", fileName, musicChannels_g[index].totalLengthSeconds);
musicChannels_g[index].mixc = InitMixChannel(48000, mixIndex, 2, false);
if (!musicChannels_g[index].mixc) return ERROR_MOD_CONTEXT_CREATION; // error
musicChannels_g[index].mixc->playing = true;
}
else
{
TraceLog(WARNING, "[%s] MOD file could not be opened", fileName);
return ERROR_LOADING_MOD; // error
}
}
else
{
TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
return ERROR_EXTENSION_NOT_RECOGNIZED; // error
}
return 0; // normal return
}
// Stop music playing for individual music index of musicChannels_g array (close stream)
void StopMusicStream(int index)
{
if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
{
CloseMixChannel(musicChannels_g[index].mixc);
if (musicChannels_g[index].xmctx)
jar_xm_free_context(musicChannels_g[index].xmctx);
else if (musicChannels_g[index].modctx.mod_loaded)
jar_mod_unload(&musicChannels_g[index].modctx);
else
stb_vorbis_close(musicChannels_g[index].stream);
musicChannels_g[index].enabled = false;
if (musicChannels_g[index].stream || musicChannels_g[index].xmctx)
{
musicChannels_g[index].stream = NULL;
musicChannels_g[index].xmctx = NULL;
}
}
}
//get number of music channels active at this time, this does not mean they are playing
int GetMusicStreamCount(void)
{
int musicCount = 0;
// Find empty music slot
for (int musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++)
{
if(musicChannels_g[musicIndex].stream != NULL || musicChannels_g[musicIndex].chipTune) musicCount++;
}
return musicCount;
}
// Pause music playing
void PauseMusicStream(int index)
{
// Pause music stream if music available!
if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc && musicChannels_g[index].enabled)
{
TraceLog(INFO, "Pausing music stream");
alSourcePause(musicChannels_g[index].mixc->alSource);
musicChannels_g[index].mixc->playing = false;
}
}
// Resume music playing
void ResumeMusicStream(int index)
{
// Resume music playing... if music available!
ALenum state;
if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
{
alGetSourcei(musicChannels_g[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED)
{
TraceLog(INFO, "Resuming music stream");
alSourcePlay(musicChannels_g[index].mixc->alSource);
musicChannels_g[index].mixc->playing = true;
}
}
}
// Check if any music is playing
bool IsMusicPlaying(int index)
{
bool playing = false;
ALint state;
if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
{
alGetSourcei(musicChannels_g[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (state == AL_PLAYING) playing = true;
}
return playing;
}
// Set volume for music
void SetMusicVolume(int index, float volume)
{
if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
{
alSourcef(musicChannels_g[index].mixc->alSource, AL_GAIN, volume);
}
}
// Set pitch for music
void SetMusicPitch(int index, float pitch)
{
if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
{
alSourcef(musicChannels_g[index].mixc->alSource, AL_PITCH, pitch);
}
}
// Get music time length (in seconds)
float GetMusicTimeLength(int index)
{
float totalSeconds;
if (musicChannels_g[index].chipTune) totalSeconds = (float)musicChannels_g[index].totalLengthSeconds;
else totalSeconds = stb_vorbis_stream_length_in_seconds(musicChannels_g[index].stream);
return totalSeconds;
}
// Get current music time played (in seconds)
float GetMusicTimePlayed(int index)
{
float secondsPlayed = 0.0f;
if (index < MAX_MUSIC_STREAMS && musicChannels_g[index].mixc)
{
if (musicChannels_g[index].chipTune && musicChannels_g[index].xmctx)
{
uint64_t samples;
jar_xm_get_position(musicChannels_g[index].xmctx, NULL, NULL, NULL, &samples);
secondsPlayed = (float)samples / (48000.f * musicChannels_g[index].mixc->channels); // Not sure if this is the correct value
}
else if(musicChannels_g[index].chipTune && musicChannels_g[index].modctx.mod_loaded)
{
long numsamp = jar_mod_current_samples(&musicChannels_g[index].modctx);
secondsPlayed = (float)numsamp / (48000.f);
}
else
{
int totalSamples = stb_vorbis_stream_length_in_samples(musicChannels_g[index].stream) * musicChannels_g[index].mixc->channels;
int samplesPlayed = totalSamples - musicChannels_g[index].totalSamplesLeft;
secondsPlayed = (float)samplesPlayed / (musicChannels_g[index].mixc->sampleRate * musicChannels_g[index].mixc->channels);
}
}
return secondsPlayed;
}
//----------------------------------------------------------------------------------
// Module specific Functions Definition
//----------------------------------------------------------------------------------
// Fill music buffers with new data from music stream
static bool BufferMusicStream(int index, int numBuffers)
{
short pcm[MUSIC_BUFFER_SIZE_SHORT];
float pcmf[MUSIC_BUFFER_SIZE_FLOAT];
int size = 0; // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts
bool active = true; // We can get more data from stream (not finished)
if (musicChannels_g[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
{
for (int i = 0; i < numBuffers; i++)
{
if (musicChannels_g[index].modctx.mod_loaded)
{
if (musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) size = MUSIC_BUFFER_SIZE_SHORT/2;
else size = musicChannels_g[index].totalSamplesLeft/2;
jar_mod_fillbuffer(&musicChannels_g[index].modctx, pcm, size, 0 );
BufferMixChannel(musicChannels_g[index].mixc, pcm, size*2);
}
else if (musicChannels_g[index].xmctx)
{
if (musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_FLOAT) size = MUSIC_BUFFER_SIZE_FLOAT/2;
else size = musicChannels_g[index].totalSamplesLeft/2;
jar_xm_generate_samples(musicChannels_g[index].xmctx, pcmf, size); // reads 2*readlen shorts and moves them to buffer+size memory location
BufferMixChannel(musicChannels_g[index].mixc, pcmf, size*2);
}
musicChannels_g[index].totalSamplesLeft -= size;
if (musicChannels_g[index].totalSamplesLeft <= 0)
{
active = false;
break;
}
}
}
else
{
if (musicChannels_g[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) size = MUSIC_BUFFER_SIZE_SHORT;
else size = musicChannels_g[index].totalSamplesLeft;
for (int i = 0; i < numBuffers; i++)
{
int streamedBytes = stb_vorbis_get_samples_short_interleaved(musicChannels_g[index].stream, musicChannels_g[index].mixc->channels, pcm, size);
BufferMixChannel(musicChannels_g[index].mixc, pcm, streamedBytes * musicChannels_g[index].mixc->channels);
musicChannels_g[index].totalSamplesLeft -= streamedBytes * musicChannels_g[index].mixc->channels;
if (musicChannels_g[index].totalSamplesLeft <= 0)
{
active = false;
break;
}
}
}
return active;
}
// Empty music buffers
static void EmptyMusicStream(int index)
{
ALuint buffer = 0;
int queued = 0;
alGetSourcei(musicChannels_g[index].mixc->alSource, AL_BUFFERS_QUEUED, &queued);
while (queued > 0)
{
alSourceUnqueueBuffers(musicChannels_g[index].mixc->alSource, 1, &buffer);
queued--;
}
}
// Determine if a music stream is ready to be written
static int IsMusicStreamReadyForBuffering(int index)
{
ALint processed = 0;
alGetSourcei(musicChannels_g[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed);
return processed;
}
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(int index)
{
ALenum state;
bool active = true;
int numBuffers = IsMusicStreamReadyForBuffering(index);
if (musicChannels_g[index].mixc->playing && (index < MAX_MUSIC_STREAMS) && musicChannels_g[index].enabled && musicChannels_g[index].mixc && numBuffers)
{
active = BufferMusicStream(index, numBuffers);
if (!active && musicChannels_g[index].loop)
{
if (musicChannels_g[index].chipTune)
{
if(musicChannels_g[index].modctx.mod_loaded) jar_mod_seek_start(&musicChannels_g[index].modctx);
musicChannels_g[index].totalSamplesLeft = musicChannels_g[index].totalLengthSeconds * 48000.f;
}
else
{
stb_vorbis_seek_start(musicChannels_g[index].stream);
musicChannels_g[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(musicChannels_g[index].stream) * musicChannels_g[index].mixc->channels;
}
active = BufferMusicStream(index, IsMusicStreamReadyForBuffering(index));
}
if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
alGetSourcei(musicChannels_g[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING && active) alSourcePlay(musicChannels_g[index].mixc->alSource);
if (!active) StopMusicStream(index);
}
}
// 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];
} 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;
RiffHeader riffHeader;
WaveFormat waveFormat;
WaveData waveData;
Wave wave = { 0 };
FILE *wavFile;
wavFile = fopen(fileName, "rb");
if (wavFile == NULL)
{
TraceLog(WARNING, "[%s] WAV file could not be opened", fileName);
wave.data = NULL;
}
else
{
// Read in the first chunk into the struct
fread(&riffHeader, sizeof(RiffHeader), 1, wavFile);
// Check for RIFF and WAVE tags
if (strncmp(riffHeader.chunkID, "RIFF", 4) ||
strncmp(riffHeader.format, "WAVE", 4))
{
TraceLog(WARNING, "[%s] Invalid RIFF or WAVE Header", fileName);
}
else
{
// Read in the 2nd chunk for the wave info
fread(&waveFormat, sizeof(WaveFormat), 1, wavFile);
// Check for fmt tag
if ((waveFormat.subChunkID[0] != 'f') || (waveFormat.subChunkID[1] != 'm') ||
(waveFormat.subChunkID[2] != 't') || (waveFormat.subChunkID[3] != ' '))
{
TraceLog(WARNING, "[%s] Invalid Wave format", fileName);
}
else
{
// Check for extra parameters;
if (waveFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR);
// Read in the the last byte of data before the sound file
fread(&waveData, sizeof(WaveData), 1, wavFile);
// Check for data tag
if ((waveData.subChunkID[0] != 'd') || (waveData.subChunkID[1] != 'a') ||
(waveData.subChunkID[2] != 't') || (waveData.subChunkID[3] != 'a'))
{
TraceLog(WARNING, "[%s] Invalid data header", fileName);
}
else
{
// Allocate memory for data
wave.data = (unsigned char *)malloc(sizeof(unsigned char) * waveData.subChunkSize);
// Read in the sound data into the soundData variable
fread(wave.data, waveData.subChunkSize, 1, wavFile);
// Now we set the variables that we need later
wave.dataSize = waveData.subChunkSize;
wave.sampleRate = waveFormat.sampleRate;
wave.channels = waveFormat.numChannels;
wave.bitsPerSample = waveFormat.bitsPerSample;
TraceLog(INFO, "[%s] WAV file loaded successfully (SampleRate: %i, BitRate: %i, Channels: %i)", fileName, wave.sampleRate, wave.bitsPerSample, wave.channels);
}
}
}
fclose(wavFile);
}
return wave;
}
// Load OGG file into Wave structure
// NOTE: Using stb_vorbis library
static Wave LoadOGG(char *fileName)
{
Wave wave;
stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);
if (oggFile == NULL)
{
TraceLog(WARNING, "[%s] OGG file could not be opened", fileName);
wave.data = NULL;
}
else
{
stb_vorbis_info info = stb_vorbis_get_info(oggFile);
wave.sampleRate = info.sample_rate;
wave.bitsPerSample = 16;
wave.channels = info.channels;
TraceLog(DEBUG, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
TraceLog(DEBUG, "[%s] Ogg channels: %i", fileName, info.channels);
int totalSamplesLength = (stb_vorbis_stream_length_in_samples(oggFile) * info.channels);
wave.dataSize = totalSamplesLength*sizeof(short); // Size must be in bytes
TraceLog(DEBUG, "[%s] Samples length: %i", fileName, totalSamplesLength);
float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile);
TraceLog(DEBUG, "[%s] Total seconds: %f", fileName, totalSeconds);
if (totalSeconds > 10) TraceLog(WARNING, "[%s] Ogg audio lenght is larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds);
int totalSamples = totalSeconds*info.sample_rate*info.channels;
TraceLog(DEBUG, "[%s] Total samples calculated: %i", fileName, totalSamples);
wave.data = malloc(sizeof(short)*totalSamplesLength);
int samplesObtained = stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, wave.data, totalSamplesLength);
TraceLog(DEBUG, "[%s] Samples obtained: %i", fileName, samplesObtained);
TraceLog(INFO, "[%s] OGG file loaded successfully (SampleRate: %i, BitRate: %i, Channels: %i)", fileName, wave.sampleRate, wave.bitsPerSample, wave.channels);
stb_vorbis_close(oggFile);
}
return wave;
}
// Unload Wave data
static void UnloadWave(Wave wave)
{
free(wave.data);
TraceLog(INFO, "Unloaded wave data");
}
// Some required functions for audio standalone module version
#if defined(AUDIO_STANDALONE)
// Get the extension for a filename
const char *GetExtension(const char *fileName)
{
const char *dot = strrchr(fileName, '.');
if(!dot || dot == fileName) return "";
return (dot + 1);
}
// Outputs a trace log message (INFO, ERROR, WARNING)
// NOTE: If a file has been init, output log is written there
void TraceLog(int msgType, const char *text, ...)
{
va_list args;
int traceDebugMsgs = 0;
#ifdef DO_NOT_TRACE_DEBUG_MSGS
traceDebugMsgs = 0;
#endif
switch(msgType)
{
case INFO: fprintf(stdout, "INFO: "); break;
case ERROR: fprintf(stdout, "ERROR: "); break;
case WARNING: fprintf(stdout, "WARNING: "); break;
case DEBUG: if (traceDebugMsgs) fprintf(stdout, "DEBUG: "); break;
default: break;
}
if ((msgType != DEBUG) || ((msgType == DEBUG) && (traceDebugMsgs)))
{
va_start(args, text);
vfprintf(stdout, text, args);
va_end(args);
fprintf(stdout, "\n");
}
if (msgType == ERROR) exit(1); // If ERROR message, exit program
}
#endif
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