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e6b82cb111c3485c5e6131fe29791f938305bce3
raylib/src/audio.c
raysan5 e6b82cb111 Lots of changes, most of them under testing-review 
Added a Tracing/Log system
Added OGG stream music support (DOESN'T WORK)
Added Compressed textures support
* This update is probably very buggy...
2014-04-09 20:25:26 +02:00

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/*********************************************************************************************
*
* raylib.audio
*
* Basic functions to manage Audio: InitAudioDevice, LoadAudioFiles, PlayAudioFiles
*
* Uses external lib:
* OpenAL - Audio device management lib
* TODO: 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> // To use exit() function
#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
//----------------------------------------------------------------------------------
// Nop...
//----------------------------------------------------------------------------------
// Types and Structures Definition
//----------------------------------------------------------------------------------
// Sound source type (all file loaded in memory)
/*
struct Sound {
unsigned int source;
unsigned int buffer;
};
// Music type (file streamming from memory)
// NOTE: Anything longer than ~10 seconds should be Music...
struct Music {
stb_vorbis* stream;
stb_vorbis_info info;
ALuint id;
ALuint buffers[2];
ALuint source;
ALenum format;
int bufferSize;
int totalSamplesLeft;
bool loop;
};
*/
// Wave file data
typedef struct Wave {
unsigned char *data; // Buffer data pointer
unsigned int sampleRate;
unsigned int dataSize;
short bitsPerSample;
short channels;
} Wave;
//----------------------------------------------------------------------------------
// Global Variables Definition
//----------------------------------------------------------------------------------
static bool musicIsPlaying;
static Music *currentMusic;
//----------------------------------------------------------------------------------
// Module specific Functions Declaration
//----------------------------------------------------------------------------------
static Wave LoadWAV(char *fileName);
static void UnloadWAV(Wave wave);
//static Ogg LoadOGG(char *fileName);
static bool MusicStream(Music music, ALuint buffer);
extern bool MusicStreamUpdate();
extern void PlayCurrentMusic();
//----------------------------------------------------------------------------------
// Module Functions Definition - Window and OpenGL Context Functions
//----------------------------------------------------------------------------------
// 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);
musicIsPlaying = false;
}
// Close the audio device for the current context, and destroys the context
void CloseAudioDevice()
{
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);
}
// Load sound to memory
Sound LoadSound(char *fileName)
{
Sound sound;
// NOTE: The entire file is loaded to memory to play it all at once (no-streaming)
// WAV file loading
// NOTE: Buffer space is allocated inside LoadWAV, Wave must be freed
Wave wave = LoadWAV(fileName);
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
UnloadWAV(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
UnloadWAV(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
}
// Play a sound with extended options
// TODO: This function should be reviewed...
void PlaySoundEx(Sound sound, float timePosition, bool loop)
{
// TODO: Review
// Change the current position (e.g. skip some part of the sound)
// NOTE: Only work when the entire file is in a single buffer
//alSourcei(sound.source, AL_BYTE_OFFSET, int(position * sampleRate));
alSourcePlay(sound.source); // Play the sound
if (loop) alSourcei(sound.source, AL_LOOPING, AL_TRUE);
else alSourcei(sound.source, AL_LOOPING, AL_FALSE);
}
// 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;
}
// Check if music is playing
bool MusicIsPlaying(Music music)
{
ALenum state;
alGetSourcei(music.source, AL_SOURCE_STATE, &state);
return (state == AL_PLAYING);
}
// Set volume for a sound
void SetVolume(Sound sound, float volume)
{
alSourcef(sound.source, AL_GAIN, volume);
}
// Set pitch for a sound
void SetPitch(Sound sound, float pitch)
{
alSourcef(sound.source, AL_PITCH, pitch);
}
// Load WAV file into Wave structure
static Wave LoadWAV(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;
}
// Unload WAV file data
static void UnloadWAV(Wave wave)
{
free(wave.data);
}
// TODO: Ogg data loading
Music LoadMusic(char *fileName)
{
Music music;
// Open audio stream
music.stream = stb_vorbis_open_filename(fileName, NULL, NULL);
if (music.stream == NULL) TraceLog(WARNING, "Could not open ogg audio file");
else
{
// Get file info
music.info = stb_vorbis_get_info(music.stream);
printf("Ogg sample rate: %i\n", music.info.sample_rate);
printf("Ogg channels: %i\n", music.info.channels);
printf("Temp memory required: %i\n", music.info.temp_memory_required);
if (music.info.channels == 2) music.format = AL_FORMAT_STEREO16;
else music.format = AL_FORMAT_MONO16;
music.bufferSize = 4096*8;
music.loop = true; // We loop by default
// Create an audio source
alGenSources(1, &music.source); // Generate pointer to audio source
alSourcef(music.source, AL_PITCH, 1);
alSourcef(music.source, AL_GAIN, 1);
alSource3f(music.source, AL_POSITION, 0, 0, 0);
alSource3f(music.source, AL_VELOCITY, 0, 0, 0);
alSourcei(music.source, AL_LOOPING, AL_TRUE); // We loop by default
// Convert loaded data to OpenAL buffers
alGenBuffers(2, music.buffers);
/*
if (!MusicStream(music, music.buffers[0])) exit(1);
if (!MusicStream(music, music.buffers[1])) exit(1);
alSourceQueueBuffers(music.source, 2, music.buffers);
PlayMusic(music);
*/
music.totalSamplesLeft = stb_vorbis_stream_length_in_samples(music.stream) * music.info.channels;
currentMusic = &music;
}
return music;
}
void UnloadMusic(Music music)
{
StopMusic(music);
alDeleteSources(1, &music.source);
alDeleteBuffers(2, music.buffers);
stb_vorbis_close(music.stream);
}
void PlayMusic(Music music)
{
//if (MusicIsPlaying(music)) return true;
if (!MusicStream(music, music.buffers[0])) TraceLog(WARNING, "MusicStream returned 0");
if (!MusicStream(music, music.buffers[1])) TraceLog(WARNING, "MusicStream returned 0");
alSourceQueueBuffers(music.source, 2, music.buffers);
alSourcePlay(music.source);
TraceLog(INFO, "Playing music");
}
extern void PlayCurrentMusic()
{
if (!MusicStream(*currentMusic, currentMusic->buffers[0])) TraceLog(WARNING, "MusicStream returned 0");
if (!MusicStream(*currentMusic, currentMusic->buffers[1])) TraceLog(WARNING, "MusicStream returned 0");
alSourceQueueBuffers(currentMusic->source, 2, currentMusic->buffers);
alSourcePlay(currentMusic->source);
}
// Stop reproducing music
void StopMusic(Music music)
{
alSourceStop(music.source);
musicIsPlaying = false;
}
static bool MusicStream(Music music, ALuint buffer)
{
//Uncomment this to avoid VLAs
//#define BUFFER_SIZE 4096*32
#ifndef BUFFER_SIZE//VLAs ftw
#define BUFFER_SIZE (music.bufferSize)
#endif
ALshort pcm[BUFFER_SIZE];
int size = 0;
int result = 0;
while (size < BUFFER_SIZE)
{
result = stb_vorbis_get_samples_short_interleaved(music.stream, music.info.channels, pcm+size, BUFFER_SIZE-size);
if (result > 0) size += (result*music.info.channels);
else break;
}
if (size == 0) return false;
alBufferData(buffer, music.format, pcm, size*sizeof(ALshort), music.info.sample_rate);
music.totalSamplesLeft -= size;
#undef BUFFER_SIZE
return true;
}
/*
extern bool MusicStreamUpdate()
{
ALint processed = 0;
alGetSourcei(currentMusic->source, AL_BUFFERS_PROCESSED, &processed);
while (processed--)
{
ALuint buffer = 0;
alSourceUnqueueBuffers(currentMusic->source, 1, &buffer);
if (!MusicStream(*currentMusic, buffer))
{
bool shouldExit = true;
if (currentMusic->loop)
{
stb_vorbis_seek_start(currentMusic->stream);
currentMusic->totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic->stream) * currentMusic->info.channels;
shouldExit = !MusicStream(*currentMusic, buffer);
}
if (shouldExit) return false;
}
alSourceQueueBuffers(currentMusic->source, 1, &buffer);
}
return true;
}
*/
extern bool MusicStreamUpdate()
{
int processed;
bool active = true;
alGetSourcei(currentMusic->source, AL_BUFFERS_PROCESSED, &processed);
printf("Data processed: %i\n", processed);
while (processed--)
{
ALuint buffer = 0;
alSourceUnqueueBuffers(currentMusic->source, 1, &buffer);
active = MusicStream(*currentMusic, buffer);
alSourceQueueBuffers(currentMusic->source, 1, &buffer);
}
return active;
}
void MusicStreamEmpty()
{
int queued;
alGetSourcei(currentMusic->source, AL_BUFFERS_QUEUED, &queued);
while(queued--)
{
ALuint buffer;
alSourceUnqueueBuffers(currentMusic->source, 1, &buffer);
}
}
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