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5f73850fa6
raylib/src/audio.c
raysan5 5f73850fa6 Renamed functions for consistency
2016-05-03 18:04:21 +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> // Declares malloc() and free() for memory management
#include <string.h> // Required for strcmp()
#include <stdio.h> // Used for .WAV loading
#if defined(AUDIO_STANDALONE)
#include <stdarg.h> // Used for functions with variable number of parameters (TraceLog())
#else
#include "utils.h" // rRES data decompression utility function
// NOTE: Includes Android fopen function map
#endif
//#define STB_VORBIS_HEADER_ONLY
#include "stb_vorbis.h" // OGG loading functions
#define JAR_XM_IMPLEMENTATION
#include "jar_xm.h" // For playing .xm files
//----------------------------------------------------------------------------------
// Defines and Macros
//----------------------------------------------------------------------------------
#define MUSIC_STREAM_BUFFERS 2
#define MAX_AUDIO_CONTEXTS 4
#if defined(PLATFORM_RPI) || defined(PLATFORM_ANDROID)
// NOTE: On RPI and Android should be lower to avoid frame-stalls
#define MUSIC_BUFFER_SIZE 4096*2 // PCM data buffer (short) - 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 4096*8 // PCM data buffer (short) - 64Kb
#endif
//----------------------------------------------------------------------------------
// 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;
jar_xm_context_t *chipctx; // Stores jar_xm context
ALuint buffers[MUSIC_STREAM_BUFFERS];
ALuint source;
ALenum format;
int channels;
int sampleRate;
int totalSamplesLeft;
float totalLengthSeconds;
bool loop;
bool chipTune; // True if chiptune is loaded
} Music;
// Audio Context, used to create custom audio streams that are not bound to a sound file. There can be
// no more than 4 concurrent audio contexts in use. This is due to each active context being tied to
// a dedicated mix channel.
typedef struct AudioContext_t {
unsigned short sampleRate; // default is 48000
unsigned char bitsPerSample; // 16 is default
unsigned char mixChannel; // 0-3 or mixA-mixD, each mix channel can receive up to one dedicated audio stream
unsigned char channels; // 1=mono, 2=stereo
ALenum alFormat; // openAL format specifier
ALuint alSource; // openAL source
ALuint alBuffer[2]; // openAL sample buffer
} AudioContext_t;
#if defined(AUDIO_STANDALONE)
typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
#endif
//----------------------------------------------------------------------------------
// Global Variables Definition
//----------------------------------------------------------------------------------
static AudioContext_t* mixChannelsActive_g[MAX_AUDIO_CONTEXTS]; // What mix channels are currently active
static 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); // Load WAV file
static Wave LoadOGG(char *fileName); // Load OGG file
static void UnloadWave(Wave wave); // Unload wave data
static bool BufferMusicStream(ALuint buffer); // Fill music buffers with data
static void EmptyMusicStream(void); // Empty music buffers
#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 context
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 audio context");
}
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 the current context, and destroys the context
void CloseAudioDevice(void)
{
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);
}
// 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
//----------------------------------------------------------------------------------
// Audio contexts are for outputing custom audio waveforms, This will shut down any other sound sources currently playing
// The mixChannel is what mix 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 InitAudioContext(48000, 16, 0, 2); // stereo, mixchannel 1, 16bit, 48khz
AudioContext InitAudioContext(unsigned short sampleRate, unsigned char bitsPerSample, unsigned char mixChannel, unsigned char channels)
{
if(mixChannel > MAX_AUDIO_CONTEXTS) return NULL;
if(!IsAudioDeviceReady()) InitAudioDevice();
else StopMusicStream();
if(!mixChannelsActive_g[mixChannel]){
AudioContext_t *ac = malloc(sizeof(AudioContext_t));
ac->sampleRate = sampleRate;
ac->bitsPerSample = bitsPerSample;
ac->mixChannel = mixChannel;
ac->channels = channels;
mixChannelsActive_g[mixChannel] = ac;
// setup openAL format
if (channels == 1)
{
if (bitsPerSample == 8 ) ac->alFormat = AL_FORMAT_MONO8;
else if (bitsPerSample == 16) ac->alFormat = AL_FORMAT_MONO16;
}
else if (channels == 2)
{
if (bitsPerSample == 8 ) ac->alFormat = AL_FORMAT_STEREO8;
else if (bitsPerSample == 16) ac->alFormat = AL_FORMAT_STEREO16;
}
// Create an audio source
alGenSources(1, &ac->alSource);
alSourcef(ac->alSource, AL_PITCH, 1);
alSourcef(ac->alSource, AL_GAIN, 1);
alSource3f(ac->alSource, AL_POSITION, 0, 0, 0);
alSource3f(ac->alSource, AL_VELOCITY, 0, 0, 0);
// Create Buffer
alGenBuffers(2, ac->alBuffer);
return ac;
}
return NULL;
}
// Frees buffer in audio context
void CloseAudioContext(AudioContext ctx)
{
AudioContext_t *context = (AudioContext_t*)ctx;
if(context){
alDeleteSources(1, &context->alSource);
alDeleteBuffers(2, context->alBuffer);
mixChannelsActive_g[context->mixChannel] = NULL;
free(context);
ctx = NULL;
}
}
// Pushes more audio data into context mix channel, if none are ever pushed then zeros are fed in
void UpdateAudioContext(AudioContext ctx, void *data, unsigned short *dataLength)
{
AudioContext_t *context = (AudioContext_t*)ctx;
if(!musicEnabled && context && mixChannelsActive_g[context->mixChannel] == context)
{
;
}
}
//----------------------------------------------------------------------------------
// 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);
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);
}
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);
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);
}
}
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)
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] OGG audio file could not be opened", 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(DEBUG, "[%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, &currentMusic.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: UpdateMusicStream()
currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
currentMusic.totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic.stream);
}
}
else if (strcmp(GetExtension(fileName),"xm") == 0)
{
// Stop current music, clean buffers, unload current stream
StopMusicStream();
// new song settings for xm chiptune
currentMusic.chipTune = true;
currentMusic.channels = 2;
currentMusic.sampleRate = 48000;
currentMusic.loop = true;
// only stereo is supported for xm
if(!jar_xm_create_context_from_file(&currentMusic.chipctx, currentMusic.sampleRate, fileName))
{
currentMusic.format = AL_FORMAT_STEREO16;
jar_xm_set_max_loop_count(currentMusic.chipctx, 0); // infinite number of loops
currentMusic.totalSamplesLeft = jar_xm_get_remaining_samples(currentMusic.chipctx);
currentMusic.totalLengthSeconds = ((float)currentMusic.totalSamplesLeft) / ((float)currentMusic.sampleRate);
musicEnabled = true;
TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic.totalSamplesLeft);
TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, currentMusic.totalLengthSeconds);
// Set up OpenAL
alGenSources(1, &currentMusic.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);
alGenBuffers(2, currentMusic.buffers);
BufferMusicStream(currentMusic.buffers[0]);
BufferMusicStream(currentMusic.buffers[1]);
alSourceQueueBuffers(currentMusic.source, 2, currentMusic.buffers);
alSourcePlay(currentMusic.source);
// NOTE: Regularly, we must check if a buffer has been processed and refill it: UpdateMusicStream()
}
else TraceLog(WARNING, "[%s] XM file could not be opened", fileName);
}
else TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
}
// Stop music playing (close stream)
void StopMusicStream(void)
{
if (musicEnabled)
{
alSourceStop(currentMusic.source);
EmptyMusicStream(); // Empty music buffers
alDeleteSources(1, &currentMusic.source);
alDeleteBuffers(2, currentMusic.buffers);
if (currentMusic.chipTune)
{
jar_xm_free_context(currentMusic.chipctx);
}
else
{
stb_vorbis_close(currentMusic.stream);
}
}
musicEnabled = false;
}
// Pause music playing
void PauseMusicStream(void)
{
// Pause music stream if music available!
if (musicEnabled)
{
TraceLog(INFO, "Pausing music stream");
alSourcePause(currentMusic.source);
musicEnabled = false;
}
}
// Resume music playing
void ResumeMusicStream(void)
{
// Resume music playing... if music available!
ALenum state;
alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED)
{
TraceLog(INFO, "Resuming music stream");
alSourcePlay(currentMusic.source);
musicEnabled = true;
}
}
// Check if music is playing
bool IsMusicPlaying(void)
{
bool playing = false;
ALint state;
alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
if (state == AL_PLAYING) playing = true;
return playing;
}
// Set volume for music
void SetMusicVolume(float volume)
{
alSourcef(currentMusic.source, AL_GAIN, volume);
}
// Get current music time length (in seconds)
float GetMusicTimeLength(void)
{
float totalSeconds;
if (currentMusic.chipTune)
{
totalSeconds = currentMusic.totalLengthSeconds;
}
else
{
totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic.stream);
}
return totalSeconds;
}
// Get current music time played (in seconds)
float GetMusicTimePlayed(void)
{
float secondsPlayed;
if (currentMusic.chipTune)
{
uint64_t samples;
jar_xm_get_position(currentMusic.chipctx, NULL, NULL, NULL, &samples);
secondsPlayed = (float)samples / (currentMusic.sampleRate * currentMusic.channels); // Not sure if this is the correct value
}
else
{
int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
int samplesPlayed = totalSamples - currentMusic.totalSamplesLeft;
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; // samples of data obtained, channels are not included in calculation
bool active = true; // We can get more data from stream (not finished)
if (musicEnabled)
{
if (currentMusic.chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
{
int readlen = MUSIC_BUFFER_SIZE / 2;
jar_xm_generate_samples_16bit(currentMusic.chipctx, pcm, readlen); // reads 2*readlen shorts and moves them to buffer+size memory location
size += readlen * currentMusic.channels; // Not sure if this is what it needs
}
else
{
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;
if(currentMusic.totalSamplesLeft <= 0) active = false; // end if no more samples left
}
else
{
active = false;
TraceLog(WARNING, "No more data obtained from stream");
}
return active;
}
// Empty music buffers
static void EmptyMusicStream(void)
{
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
void UpdateMusicStream(void)
{
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.chipTune)
{
currentMusic.totalSamplesLeft = currentMusic.totalLengthSeconds * currentMusic.sampleRate;
}
else
{
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, "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];
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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