803 lines
28 KiB
C
803 lines
28 KiB
C
/*********************************************************************************************
|
|
*
|
|
* raylib.audio
|
|
*
|
|
* Basic functions to manage Audio: InitAudioDevice, LoadAudioFiles, PlayAudioFiles
|
|
*
|
|
* Uses external lib:
|
|
* OpenAL - Audio device management lib
|
|
* stb_vorbis - Ogg audio files loading
|
|
*
|
|
* Copyright (c) 2013 Ramon Santamaria (Ray San - raysan@raysanweb.com)
|
|
*
|
|
* 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.
|
|
*
|
|
**********************************************************************************************/
|
|
|
|
#include "raylib.h"
|
|
|
|
#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> // Declares malloc() and free() for memory management
|
|
#include <string.h> // Required for strcmp()
|
|
#include <stdio.h> // Used for .WAV loading
|
|
|
|
#include "utils.h" // rRES data decompression utility function
|
|
|
|
#include "stb_vorbis.h" // OGG loading functions
|
|
|
|
//----------------------------------------------------------------------------------
|
|
// Defines and Macros
|
|
//----------------------------------------------------------------------------------
|
|
#define MUSIC_STREAM_BUFFERS 2
|
|
#define MUSIC_BUFFER_SIZE 4096*8 //4096*32
|
|
|
|
//----------------------------------------------------------------------------------
|
|
// Types and Structures Definition
|
|
//----------------------------------------------------------------------------------
|
|
|
|
// Music type (file streaming from memory)
|
|
// NOTE: Anything longer than ~10 seconds should be streamed...
|
|
typedef struct Music {
|
|
stb_vorbis *stream;
|
|
|
|
ALuint buffers[MUSIC_STREAM_BUFFERS];
|
|
ALuint source;
|
|
ALenum format;
|
|
|
|
int channels;
|
|
int sampleRate;
|
|
int totalSamplesLeft;
|
|
bool loop;
|
|
|
|
} Music;
|
|
|
|
// Wave file data
|
|
typedef struct Wave {
|
|
void *data; // Buffer data pointer
|
|
unsigned int dataSize; // Data size in bytes
|
|
unsigned int sampleRate;
|
|
short bitsPerSample;
|
|
short channels;
|
|
} Wave;
|
|
|
|
//----------------------------------------------------------------------------------
|
|
// Global Variables Definition
|
|
//----------------------------------------------------------------------------------
|
|
bool musicEnabled = false;
|
|
static Music currentMusic; // Current music loaded
|
|
// NOTE: Only one music file playing at a time
|
|
|
|
//----------------------------------------------------------------------------------
|
|
// Module specific Functions Declaration
|
|
//----------------------------------------------------------------------------------
|
|
static Wave LoadWAV(const char *fileName);
|
|
static Wave LoadOGG(char *fileName);
|
|
static void UnloadWave(Wave wave);
|
|
|
|
static bool BufferMusicStream(ALuint buffer); // Fill music buffers with data
|
|
static void EmptyMusicStream(); // Empty music buffers
|
|
extern void UpdateMusicStream(); // Updates buffers (refill) for music streaming
|
|
|
|
//----------------------------------------------------------------------------------
|
|
// Module Functions Definition - Audio Device initialization and Closing
|
|
//----------------------------------------------------------------------------------
|
|
|
|
// Initialize audio device and context
|
|
void InitAudioDevice()
|
|
{
|
|
// Open and initialize a device with default settings
|
|
ALCdevice *device = alcOpenDevice(NULL);
|
|
|
|
if(!device) TraceLog(ERROR, "Could not open audio device");
|
|
|
|
ALCcontext *context = alcCreateContext(device, NULL);
|
|
|
|
if(context == NULL || alcMakeContextCurrent(context) == ALC_FALSE)
|
|
{
|
|
if(context != NULL) alcDestroyContext(context);
|
|
|
|
alcCloseDevice(device);
|
|
|
|
TraceLog(ERROR, "Could not setup audio context");
|
|
}
|
|
|
|
TraceLog(INFO, "Audio device and context initialized: %s\n", 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 the current context, and destroys the context
|
|
void CloseAudioDevice()
|
|
{
|
|
StopMusicStream(); // Stop music streaming and close current stream
|
|
|
|
ALCdevice *device;
|
|
ALCcontext *context = alcGetCurrentContext();
|
|
|
|
if (context == NULL) TraceLog(WARNING, "Could not get current audio context for closing");
|
|
|
|
device = alcGetContextsDevice(context);
|
|
|
|
alcMakeContextCurrent(NULL);
|
|
alcDestroyContext(context);
|
|
alcCloseDevice(device);
|
|
}
|
|
|
|
//----------------------------------------------------------------------------------
|
|
// Module Functions Definition - Sounds loading and playing (.WAV)
|
|
//----------------------------------------------------------------------------------
|
|
|
|
// Load sound to memory
|
|
Sound LoadSound(char *fileName)
|
|
{
|
|
Sound sound;
|
|
Wave wave;
|
|
|
|
// 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);
|
|
|
|
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, "[%s] Sound file loaded successfully", fileName);
|
|
TraceLog(INFO, "[%s] Sample rate: %i - Channels: %i", fileName, wave.sampleRate, wave.channels);
|
|
|
|
sound.source = source;
|
|
sound.buffer = buffer;
|
|
}
|
|
|
|
return sound;
|
|
}
|
|
|
|
// Load sound to memory from rRES file (raylib Resource)
|
|
Sound LoadSoundFromRES(const char *rresName, int resId)
|
|
{
|
|
// NOTE: rresName could be directly a char array with all the data!!! --> TODO
|
|
Sound sound;
|
|
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) TraceLog(WARNING, "[%s] Could not open raylib resource file", rresName);
|
|
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);
|
|
}
|
|
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);
|
|
|
|
// Unallocate WAV data
|
|
UnloadWave(wave);
|
|
|
|
TraceLog(INFO, "[%s] Sound loaded successfully from resource, sample rate: %i", rresName, (int)sampleRate);
|
|
|
|
sound.source = source;
|
|
sound.buffer = buffer;
|
|
}
|
|
else
|
|
{
|
|
TraceLog(WARNING, "[%s] Required resource do not seem to be a valid SOUND resource", rresName);
|
|
}
|
|
}
|
|
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);
|
|
|
|
return sound;
|
|
}
|
|
|
|
// Unload sound
|
|
void UnloadSound(Sound sound)
|
|
{
|
|
alDeleteSources(1, &sound.source);
|
|
alDeleteBuffers(1, &sound.buffer);
|
|
}
|
|
|
|
// 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 SoundIsPlaying(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)
|
|
void PlayMusicStream(char *fileName)
|
|
{
|
|
if (strcmp(GetExtension(fileName),"ogg") == 0)
|
|
{
|
|
// Stop current music, clean buffers, unload current stream
|
|
StopMusicStream();
|
|
|
|
// Open audio stream
|
|
currentMusic.stream = stb_vorbis_open_filename(fileName, NULL, NULL);
|
|
|
|
if (currentMusic.stream == NULL) TraceLog(WARNING, "[%s] Could not open ogg audio file", fileName);
|
|
else
|
|
{
|
|
// Get file info
|
|
stb_vorbis_info info = stb_vorbis_get_info(currentMusic.stream);
|
|
|
|
currentMusic.channels = info.channels;
|
|
currentMusic.sampleRate = info.sample_rate;
|
|
|
|
TraceLog(INFO, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
|
|
TraceLog(INFO, "[%s] Ogg channels: %i", fileName, info.channels);
|
|
TraceLog(INFO, "[%s] Temp memory required: %i", fileName, info.temp_memory_required);
|
|
|
|
if (info.channels == 2) currentMusic.format = AL_FORMAT_STEREO16;
|
|
else currentMusic.format = AL_FORMAT_MONO16;
|
|
|
|
currentMusic.loop = true; // We loop by default
|
|
musicEnabled = true;
|
|
|
|
// Create an audio source
|
|
alGenSources(1, ¤tMusic.source); // Generate pointer to audio source
|
|
|
|
alSourcef(currentMusic.source, AL_PITCH, 1);
|
|
alSourcef(currentMusic.source, AL_GAIN, 1);
|
|
alSource3f(currentMusic.source, AL_POSITION, 0, 0, 0);
|
|
alSource3f(currentMusic.source, AL_VELOCITY, 0, 0, 0);
|
|
//alSourcei(currentMusic.source, AL_LOOPING, AL_TRUE); // ERROR: Buffers do not queue!
|
|
|
|
// Generate two OpenAL buffers
|
|
alGenBuffers(2, currentMusic.buffers);
|
|
|
|
// Fill buffers with music...
|
|
BufferMusicStream(currentMusic.buffers[0]);
|
|
BufferMusicStream(currentMusic.buffers[1]);
|
|
|
|
// Queue buffers and start playing
|
|
alSourceQueueBuffers(currentMusic.source, 2, currentMusic.buffers);
|
|
alSourcePlay(currentMusic.source);
|
|
|
|
// NOTE: Regularly, we must check if a buffer has been processed and refill it: MusicStreamUpdate()
|
|
|
|
currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
|
|
}
|
|
}
|
|
else TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
|
|
}
|
|
|
|
// Stop music playing (close stream)
|
|
void StopMusicStream()
|
|
{
|
|
if (musicEnabled)
|
|
{
|
|
alSourceStop(currentMusic.source);
|
|
|
|
EmptyMusicStream(); // Empty music buffers
|
|
|
|
alDeleteSources(1, ¤tMusic.source);
|
|
alDeleteBuffers(2, currentMusic.buffers);
|
|
|
|
stb_vorbis_close(currentMusic.stream);
|
|
}
|
|
|
|
musicEnabled = false;
|
|
}
|
|
|
|
// Pause music playing
|
|
void PauseMusicStream()
|
|
{
|
|
// TODO: Record music is paused or check if music available!
|
|
alSourcePause(currentMusic.source);
|
|
}
|
|
|
|
// Check if music is playing
|
|
bool MusicIsPlaying()
|
|
{
|
|
ALenum state;
|
|
|
|
alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
|
|
|
|
return (state == AL_PLAYING);
|
|
}
|
|
|
|
// Set volume for music
|
|
void SetMusicVolume(float volume)
|
|
{
|
|
alSourcef(currentMusic.source, AL_GAIN, volume);
|
|
}
|
|
|
|
// Get current music time length (in seconds)
|
|
float GetMusicTimeLength()
|
|
{
|
|
float totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic.stream);
|
|
|
|
return totalSeconds;
|
|
}
|
|
|
|
// Get current music time played (in seconds)
|
|
float GetMusicTimePlayed()
|
|
{
|
|
int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
|
|
|
|
int samplesPlayed = totalSamples - currentMusic.totalSamplesLeft;
|
|
|
|
float secondsPlayed = (float)samplesPlayed / (currentMusic.sampleRate * currentMusic.channels);
|
|
|
|
return secondsPlayed;
|
|
}
|
|
|
|
//----------------------------------------------------------------------------------
|
|
// Module specific Functions Definition
|
|
//----------------------------------------------------------------------------------
|
|
|
|
// Fill music buffers with new data from music stream
|
|
static bool BufferMusicStream(ALuint buffer)
|
|
{
|
|
short pcm[MUSIC_BUFFER_SIZE];
|
|
|
|
int size = 0; // Total size of data steamed (in bytes)
|
|
int streamedBytes = 0; // Bytes of data obtained in one samples get
|
|
|
|
bool active = true; // We can get more data from stream (not finished)
|
|
|
|
if (musicEnabled)
|
|
{
|
|
while (size < MUSIC_BUFFER_SIZE)
|
|
{
|
|
streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic.stream, currentMusic.channels, pcm + size, MUSIC_BUFFER_SIZE - size);
|
|
|
|
if (streamedBytes > 0) size += (streamedBytes*currentMusic.channels);
|
|
else break;
|
|
}
|
|
|
|
TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
|
|
}
|
|
|
|
if (size > 0)
|
|
{
|
|
alBufferData(buffer, currentMusic.format, pcm, size*sizeof(short), currentMusic.sampleRate);
|
|
|
|
currentMusic.totalSamplesLeft -= size;
|
|
}
|
|
else
|
|
{
|
|
active = false;
|
|
TraceLog(WARNING, "No more data obtained from stream");
|
|
}
|
|
|
|
return active;
|
|
}
|
|
|
|
// Empty music buffers
|
|
static void EmptyMusicStream()
|
|
{
|
|
ALuint buffer = 0;
|
|
int queued = 0;
|
|
|
|
alGetSourcei(currentMusic.source, AL_BUFFERS_QUEUED, &queued);
|
|
|
|
while(queued > 0)
|
|
{
|
|
alSourceUnqueueBuffers(currentMusic.source, 1, &buffer);
|
|
|
|
queued--;
|
|
}
|
|
}
|
|
|
|
// Update (re-fill) music buffers if data already processed
|
|
extern void UpdateMusicStream()
|
|
{
|
|
ALuint buffer = 0;
|
|
ALint processed = 0;
|
|
bool active = true;
|
|
|
|
if (musicEnabled)
|
|
{
|
|
// Get the number of already processed buffers (if any)
|
|
alGetSourcei(currentMusic.source, AL_BUFFERS_PROCESSED, &processed);
|
|
|
|
while (processed > 0)
|
|
{
|
|
// Recover processed buffer for refill
|
|
alSourceUnqueueBuffers(currentMusic.source, 1, &buffer);
|
|
|
|
// Refill buffer
|
|
active = BufferMusicStream(buffer);
|
|
|
|
// If no more data to stream, restart music (if loop)
|
|
if ((!active) && (currentMusic.loop))
|
|
{
|
|
if (currentMusic.loop)
|
|
{
|
|
stb_vorbis_seek_start(currentMusic.stream);
|
|
currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
|
|
|
|
active = BufferMusicStream(buffer);
|
|
}
|
|
}
|
|
|
|
// Add refilled buffer to queue again... don't let the music stop!
|
|
alSourceQueueBuffers(currentMusic.source, 1, &buffer);
|
|
|
|
if(alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Ogg playing, error buffering data...");
|
|
|
|
processed--;
|
|
}
|
|
|
|
ALenum state;
|
|
alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
|
|
|
|
if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic.source);
|
|
|
|
if (!active) StopMusicStream();
|
|
}
|
|
}
|
|
|
|
// Load WAV file into Wave structure
|
|
static Wave LoadWAV(const char *fileName)
|
|
{
|
|
// Basic WAV headers structs
|
|
typedef struct {
|
|
char chunkID[4];
|
|
long chunkSize;
|
|
char format[4];
|
|
} RiffHeader;
|
|
|
|
typedef struct {
|
|
char subChunkID[4];
|
|
long subChunkSize;
|
|
short audioFormat;
|
|
short numChannels;
|
|
long sampleRate;
|
|
long byteRate;
|
|
short blockAlign;
|
|
short bitsPerSample;
|
|
} WaveFormat;
|
|
|
|
typedef struct {
|
|
char subChunkID[4];
|
|
long subChunkSize;
|
|
} WaveData;
|
|
|
|
RiffHeader riffHeader;
|
|
WaveFormat waveFormat;
|
|
WaveData waveData;
|
|
|
|
Wave wave;
|
|
FILE *wavFile;
|
|
|
|
wavFile = fopen(fileName, "rb");
|
|
|
|
if (!wavFile)
|
|
{
|
|
TraceLog(WARNING, "[%s] Could not open WAV file", fileName);
|
|
}
|
|
else
|
|
{
|
|
// Read in the first chunk into the struct
|
|
fread(&riffHeader, sizeof(RiffHeader), 1, wavFile);
|
|
|
|
// Check for RIFF and WAVE tags
|
|
if (((riffHeader.chunkID[0] != 'R') || (riffHeader.chunkID[1] != 'I') || (riffHeader.chunkID[2] != 'F') || (riffHeader.chunkID[3] != 'F')) ||
|
|
((riffHeader.format[0] != 'W') || (riffHeader.format[1] != 'A') || (riffHeader.format[2] != 'V') || (riffHeader.format[3] != 'E')))
|
|
{
|
|
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] Wave file loaded successfully", fileName);
|
|
}
|
|
}
|
|
}
|
|
|
|
fclose(wavFile);
|
|
}
|
|
|
|
return wave;
|
|
}
|
|
|
|
// Load OGG file into Wave structure
|
|
static Wave LoadOGG(char *fileName)
|
|
{
|
|
Wave wave;
|
|
|
|
stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);
|
|
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);
|
|
|
|
//short *data
|
|
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);
|
|
|
|
stb_vorbis_close(oggFile);
|
|
|
|
return wave;
|
|
}
|
|
|
|
// Unload Wave data
|
|
static void UnloadWave(Wave wave)
|
|
{
|
|
free(wave.data);
|
|
} |