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7586031410
raylib/src/audio.c
Ray 7586031410 Support 32bit wav data
2017-01-19 13:18:04 +01:00

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/**********************************************************************************************
*
* raylib.audio
*
* This module provides basic functionality to work with audio:
* Manage audio device (init/close)
* Load and Unload audio files (WAV, OGG, FLAC, XM, MOD)
* Play/Stop/Pause/Resume loaded audio
* Manage mixing channels
* Manage raw audio context
*
* External libs:
* OpenAL Soft - Audio device management (http://kcat.strangesoft.net/openal.html)
* stb_vorbis - OGG audio files loading (http://www.nothings.org/stb_vorbis/)
* jar_xm - XM module file loading
* jar_mod - MOD audio file loading
* dr_flac - FLAC audio file loading
*
* Module Configuration Flags:
* AUDIO_STANDALONE - Use this module as standalone library (independently of raylib)
*
* Some design decisions:
* Support only up to two channels: MONO and STEREO (for additional channels, AL_EXT_MCFORMATS)
* Support only the following sample sizes: 8bit PCM, 16bit PCM, 32-bit float PCM (using AL_EXT_FLOAT32)
*
* Many thanks to Joshua Reisenauer (github: @kd7tck) for the following additions:
* XM audio module support (jar_xm)
* MOD audio module support (jar_mod)
* Mixing channels support
* Raw audio context support
*
*
* Copyright (c) 2014-2016 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"
#include <stdarg.h> // Required for: va_list, va_start(), vfprintf(), va_end()
#else
#include "raylib.h"
#include "utils.h" // Required for: fopen() Android mapping, TraceLog()
#endif
#include "AL/al.h" // OpenAL basic header
#include "AL/alc.h" // OpenAL context header (like OpenGL, OpenAL requires a context to work)
//#include "AL/alext.h" // OpenAL extensions header, required for AL_EXT_FLOAT32 and AL_EXT_MCFORMATS
// OpenAL extension: AL_EXT_FLOAT32 - Support for 32bit float samples
// OpenAL extension: AL_EXT_MCFORMATS - Support for multi-channel formats (Quad, 5.1, 6.1, 7.1)
#include <stdlib.h> // Required for: malloc(), free()
#include <string.h> // Required for: strcmp(), strncmp()
#include <stdio.h> // Required for: FILE, fopen(), fclose(), fread()
//#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
#define DR_FLAC_IMPLEMENTATION
#define DR_FLAC_NO_WIN32_IO
#include "external/dr_flac.h" // FLAC loading functions
#ifdef _MSC_VER
#undef bool
#endif
//----------------------------------------------------------------------------------
// Defines and Macros
//----------------------------------------------------------------------------------
#define MAX_STREAM_BUFFERS 2 // Number of buffers for each audio stream
// NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number
// After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds
// and double-buffering system, I concluded that a 4096 samples buffer should be enough
// In case of music-stalls, just increase this number
#define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. 16bit, Mono: 8Kb)
// Support uncompressed PCM data in 32-bit float IEEE format
// NOTE: This definition is included in "AL/alext.h", but some OpenAL implementations
// could not provide the extensions header (Android), so its defined here
#if !defined(AL_EXT_float32)
#define AL_EXT_float32 1
#define AL_FORMAT_MONO_FLOAT32 0x10010
#define AL_FORMAT_STEREO_FLOAT32 0x10011
#endif
//----------------------------------------------------------------------------------
// Types and Structures Definition
//----------------------------------------------------------------------------------
typedef enum { MUSIC_AUDIO_OGG = 0, MUSIC_AUDIO_FLAC, MUSIC_MODULE_XM, MUSIC_MODULE_MOD } MusicContextType;
// Music type (file streaming from memory)
typedef struct MusicData {
MusicContextType ctxType; // Type of music context (OGG, XM, MOD)
stb_vorbis *ctxOgg; // OGG audio context
drflac *ctxFlac; // FLAC audio context
jar_xm_context_t *ctxXm; // XM chiptune context
jar_mod_context_t ctxMod; // MOD chiptune context
AudioStream stream; // Audio stream (double buffering)
bool loop; // Repeat music after finish (loop)
unsigned int totalSamples; // Total number of samples
unsigned int samplesLeft; // Number of samples left to end
} MusicData;
#if defined(AUDIO_STANDALONE)
typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
#endif
//----------------------------------------------------------------------------------
// Global Variables Definition
//----------------------------------------------------------------------------------
// ...
//----------------------------------------------------------------------------------
// Module specific Functions Declaration
//----------------------------------------------------------------------------------
static Wave LoadWAV(const char *fileName); // Load WAV file
static Wave LoadOGG(const char *fileName); // Load OGG file
static Wave LoadFLAC(const char *fileName); // Load FLAC file
#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
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");
else
{
ALCcontext *context = alcCreateContext(device, NULL);
if ((context == NULL) || (alcMakeContextCurrent(context) == ALC_FALSE))
{
if (context != NULL) alcDestroyContext(context);
alcCloseDevice(device);
TraceLog(ERROR, "Could not initialize audio context");
}
else
{
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)
{
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);
TraceLog(INFO, "Audio device closed successfully");
}
// Check if device has been initialized successfully
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 - Sounds loading and playing (.WAV)
//----------------------------------------------------------------------------------
// Load wave data from file
Wave LoadWave(const char *fileName)
{
Wave wave = { 0 };
if (strcmp(GetExtension(fileName), "wav") == 0) wave = LoadWAV(fileName);
else if (strcmp(GetExtension(fileName), "ogg") == 0) wave = LoadOGG(fileName);
else if (strcmp(GetExtension(fileName), "flac") == 0) wave = LoadFLAC(fileName);
else if (strcmp(GetExtension(fileName),"rres") == 0)
{
RRESData rres = LoadResource(fileName);
// NOTE: Parameters for RRES_WAVE type are: sampleCount, sampleRate, sampleSize, channels
if (rres.type == RRES_WAVE) wave = LoadWaveEx(rres.data, rres.param1, rres.param2, rres.param3, rres.param4);
else TraceLog(WARNING, "[%s] Resource file does not contain wave data", fileName);
UnloadResource(rres);
}
else TraceLog(WARNING, "[%s] File extension not recognized, it can't be loaded", fileName);
return wave;
}
// Load wave data from raw array data
Wave LoadWaveEx(void *data, int sampleCount, int sampleRate, int sampleSize, int channels)
{
Wave wave;
wave.data = data;
wave.sampleCount = sampleCount;
wave.sampleRate = sampleRate;
wave.sampleSize = sampleSize;
wave.channels = channels;
// NOTE: Copy wave data to work with, user is responsible of input data to free
Wave cwave = WaveCopy(wave);
WaveFormat(&cwave, sampleRate, sampleSize, channels);
return cwave;
}
// Load sound from file
// NOTE: The entire file is loaded to memory to be played (no-streaming)
Sound LoadSound(const char *fileName)
{
Wave wave = LoadWave(fileName);
Sound sound = LoadSoundFromWave(wave);
UnloadWave(wave); // Sound is loaded, we can unload wave
return sound;
}
// Load sound from wave data
// NOTE: Wave data must be unallocated manually
Sound LoadSoundFromWave(Wave wave)
{
Sound sound = { 0 };
if (wave.data != NULL)
{
ALenum format = 0;
// The OpenAL format is worked out by looking at the number of channels and the sample size (bits per sample)
if (wave.channels == 1)
{
switch (wave.sampleSize)
{
case 8: format = AL_FORMAT_MONO8; break;
case 16: format = AL_FORMAT_MONO16; break;
case 32: format = AL_FORMAT_MONO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32
default: TraceLog(WARNING, "Wave sample size not supported: %i", wave.sampleSize); break;
}
}
else if (wave.channels == 2)
{
switch (wave.sampleSize)
{
case 8: format = AL_FORMAT_STEREO8; break;
case 16: format = AL_FORMAT_STEREO16; break;
case 32: format = AL_FORMAT_STEREO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32
default: TraceLog(WARNING, "Wave sample size not supported: %i", wave.sampleSize); break;
}
}
else TraceLog(WARNING, "Wave number of channels not supported: %i", wave.channels);
// 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
unsigned int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8; // Size in bytes
// Upload sound data to buffer
alBufferData(buffer, format, wave.data, dataSize, wave.sampleRate);
// Attach sound buffer to source
alSourcei(source, AL_BUFFER, buffer);
TraceLog(INFO, "[SND ID %i][BUFR ID %i] Sound data loaded successfully (%i Hz, %i bit, %s)", source, buffer, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo");
sound.source = source;
sound.buffer = buffer;
sound.format = format;
}
return sound;
}
// Unload wave data
void UnloadWave(Wave wave)
{
if (wave.data != NULL) free(wave.data);
TraceLog(INFO, "Unloaded wave data from RAM");
}
// Unload sound
void UnloadSound(Sound sound)
{
alSourceStop(sound.source);
alDeleteSources(1, &sound.source);
alDeleteBuffers(1, &sound.buffer);
TraceLog(INFO, "[SND ID %i][BUFR ID %i] Unloaded sound data from RAM", sound.source, sound.buffer);
}
// Update sound buffer with new data
// NOTE: data must match sound.format
void UpdateSound(Sound sound, const void *data, int numSamples)
{
ALint sampleRate, sampleSize, channels;
alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate);
alGetBufferi(sound.buffer, AL_BITS, &sampleSize); // It could also be retrieved from sound.format
alGetBufferi(sound.buffer, AL_CHANNELS, &channels); // It could also be retrieved from sound.format
TraceLog(DEBUG, "UpdateSound() : AL_FREQUENCY: %i", sampleRate);
TraceLog(DEBUG, "UpdateSound() : AL_BITS: %i", sampleSize);
TraceLog(DEBUG, "UpdateSound() : AL_CHANNELS: %i", channels);
unsigned int dataSize = numSamples*channels*sampleSize/8; // Size of data in bytes
alSourceStop(sound.source); // Stop sound
alSourcei(sound.source, AL_BUFFER, 0); // Unbind buffer from sound to update
//alDeleteBuffers(1, &sound.buffer); // Delete current buffer data
//alGenBuffers(1, &sound.buffer); // Generate new buffer
// Upload new data to sound buffer
alBufferData(sound.buffer, sound.format, data, dataSize, sampleRate);
// Attach sound buffer to source again
alSourcei(sound.source, AL_BUFFER, 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);
}
// Resume a paused sound
void ResumeSound(Sound sound)
{
ALenum state;
alGetSourcei(sound.source, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED) alSourcePlay(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);
}
// Convert wave data to desired format
void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
{
// Format sample rate
// NOTE: Only supported 22050 <--> 44100
if (wave->sampleRate != sampleRate)
{
// TODO: Resample wave data (upsampling or downsampling)
// NOTE 1: To downsample, you have to drop samples or average them.
// NOTE 2: To upsample, you have to interpolate new samples.
wave->sampleRate = sampleRate;
}
// Format sample size
// NOTE: Only supported 8 bit <--> 16 bit <--> 32 bit
if (wave->sampleSize != sampleSize)
{
void *data = malloc(wave->sampleCount*wave->channels*sampleSize/8);
for (int i = 0; i < wave->sampleCount; i++)
{
for (int j = 0; j < wave->channels; j++)
{
if (sampleSize == 8)
{
if (wave->sampleSize == 16) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float)(((short *)wave->data)[wave->channels*i + j])/32767.0f)*256);
else if (wave->sampleSize == 32) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float *)wave->data)[wave->channels*i + j]*127.0f + 127);
}
else if (sampleSize == 16)
{
if (wave->sampleSize == 8) ((short *)data)[wave->channels*i + j] = (short)(((float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f)*32767);
else if (wave->sampleSize == 32) ((short *)data)[wave->channels*i + j] = (short)((((float *)wave->data)[wave->channels*i + j])*32767);
}
else if (sampleSize == 32)
{
if (wave->sampleSize == 8) ((float *)data)[wave->channels*i + j] = (float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f;
else if (wave->sampleSize == 16) ((float *)data)[wave->channels*i + j] = (float)(((short *)wave->data)[wave->channels*i + j])/32767.0f;
}
}
}
wave->sampleSize = sampleSize;
free(wave->data);
wave->data = data;
}
// Format channels (interlaced mode)
// NOTE: Only supported mono <--> stereo
if (wave->channels != channels)
{
void *data = malloc(wave->sampleCount*channels*wave->sampleSize/8);
if ((wave->channels == 1) && (channels == 2)) // mono ---> stereo (duplicate mono information)
{
for (int i = 0; i < wave->sampleCount; i++)
{
for (int j = 0; j < channels; j++)
{
if (wave->sampleSize == 8) ((unsigned char *)data)[channels*i + j] = ((unsigned char *)wave->data)[i];
else if (wave->sampleSize == 16) ((short *)data)[channels*i + j] = ((short *)wave->data)[i];
else if (wave->sampleSize == 32) ((float *)data)[channels*i + j] = ((float *)wave->data)[i];
}
}
}
else if ((wave->channels == 2) && (channels == 1)) // stereo ---> mono (mix stereo channels)
{
for (int i = 0, j = 0; i < wave->sampleCount; i++, j += 2)
{
if (wave->sampleSize == 8) ((unsigned char *)data)[i] = (((unsigned char *)wave->data)[j] + ((unsigned char *)wave->data)[j + 1])/2;
else if (wave->sampleSize == 16) ((short *)data)[i] = (((short *)wave->data)[j] + ((short *)wave->data)[j + 1])/2;
else if (wave->sampleSize == 32) ((float *)data)[i] = (((float *)wave->data)[j] + ((float *)wave->data)[j + 1])/2.0f;
}
}
// TODO: Add/remove additional interlaced channels
wave->channels = channels;
free(wave->data);
wave->data = data;
}
}
// Copy a wave to a new wave
Wave WaveCopy(Wave wave)
{
Wave newWave = { 0 };
newWave.data = malloc(wave.sampleCount*wave.channels*wave.sampleSize/8);
if (newWave.data != NULL)
{
// NOTE: Size must be provided in bytes
memcpy(newWave.data, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8);
newWave.sampleCount = wave.sampleCount;
newWave.sampleRate = wave.sampleRate;
newWave.sampleSize = wave.sampleSize;
newWave.channels = wave.channels;
}
return newWave;
}
// Crop a wave to defined samples range
// NOTE: Security check in case of out-of-range
void WaveCrop(Wave *wave, int initSample, int finalSample)
{
if ((initSample >= 0) && (initSample < finalSample) &&
(finalSample > 0) && (finalSample < wave->sampleCount))
{
int sampleCount = finalSample - initSample;
void *data = malloc(sampleCount*wave->channels*wave->sampleSize/8);
memcpy(data, wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8);
free(wave->data);
wave->data = data;
}
else TraceLog(WARNING, "Wave crop range out of bounds");
}
// Get samples data from wave as a floats array
// NOTE: Returned sample values are normalized to range [-1..1]
float *GetWaveData(Wave wave)
{
float *samples = (float *)malloc(wave.sampleCount*wave.channels*sizeof(float));
for (int i = 0; i < wave.sampleCount; i++)
{
for (int j = 0; j < wave.channels; j++)
{
if (wave.sampleSize == 8) samples[wave.channels*i + j] = (float)(((unsigned char *)wave.data)[wave.channels*i + j] - 127)/256.0f;
else if (wave.sampleSize == 16) samples[wave.channels*i + j] = (float)((short *)wave.data)[wave.channels*i + j]/32767.0f;
else if (wave.sampleSize == 32) samples[wave.channels*i + j] = ((float *)wave.data)[wave.channels*i + j];
}
}
return samples;
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Music loading and stream playing (.OGG)
//----------------------------------------------------------------------------------
// Load music stream from file
Music LoadMusicStream(const char *fileName)
{
Music music = (MusicData *)malloc(sizeof(MusicData));
if (strcmp(GetExtension(fileName), "ogg") == 0)
{
// Open ogg audio stream
music->ctxOgg = stb_vorbis_open_filename(fileName, NULL, NULL);
if (music->ctxOgg == NULL) TraceLog(WARNING, "[%s] OGG audio file could not be opened", fileName);
else
{
stb_vorbis_info info = stb_vorbis_get_info(music->ctxOgg); // Get Ogg file info
// OGG bit rate defaults to 16 bit, it's enough for compressed format
music->stream = InitAudioStream(info.sample_rate, 16, info.channels);
music->totalSamples = (unsigned int)stb_vorbis_stream_length_in_samples(music->ctxOgg); // Independent by channel
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_AUDIO_OGG;
music->loop = true; // We loop by default
TraceLog(DEBUG, "[%s] FLAC total samples: %i", fileName, music->totalSamples);
TraceLog(DEBUG, "[%s] OGG sample rate: %i", fileName, info.sample_rate);
TraceLog(DEBUG, "[%s] OGG channels: %i", fileName, info.channels);
TraceLog(DEBUG, "[%s] OGG memory required: %i", fileName, info.temp_memory_required);
}
}
else if (strcmp(GetExtension(fileName), "flac") == 0)
{
music->ctxFlac = drflac_open_file(fileName);
if (music->ctxFlac == NULL) TraceLog(WARNING, "[%s] FLAC audio file could not be opened", fileName);
else
{
music->stream = InitAudioStream(music->ctxFlac->sampleRate, music->ctxFlac->bitsPerSample, music->ctxFlac->channels);
music->totalSamples = (unsigned int)music->ctxFlac->totalSampleCount/music->ctxFlac->channels;
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_AUDIO_FLAC;
music->loop = true; // We loop by default
TraceLog(DEBUG, "[%s] FLAC total samples: %i", fileName, music->totalSamples);
TraceLog(DEBUG, "[%s] FLAC sample rate: %i", fileName, music->ctxFlac->sampleRate);
TraceLog(DEBUG, "[%s] FLAC bits per sample: %i", fileName, music->ctxFlac->bitsPerSample);
TraceLog(DEBUG, "[%s] FLAC channels: %i", fileName, music->ctxFlac->channels);
}
}
else if (strcmp(GetExtension(fileName), "xm") == 0)
{
int result = jar_xm_create_context_from_file(&music->ctxXm, 48000, fileName);
if (!result) // XM context created successfully
{
jar_xm_set_max_loop_count(music->ctxXm, 0); // Set infinite number of loops
// NOTE: Only stereo is supported for XM
music->stream = InitAudioStream(48000, 16, 2);
music->totalSamples = (unsigned int)jar_xm_get_remaining_samples(music->ctxXm);
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_MODULE_XM;
music->loop = true;
TraceLog(DEBUG, "[%s] XM number of samples: %i", fileName, music->totalSamples);
TraceLog(DEBUG, "[%s] XM track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f);
}
else TraceLog(WARNING, "[%s] XM file could not be opened", fileName);
}
else if (strcmp(GetExtension(fileName), "mod") == 0)
{
jar_mod_init(&music->ctxMod);
if (jar_mod_load_file(&music->ctxMod, fileName))
{
music->stream = InitAudioStream(48000, 16, 2);
music->totalSamples = (unsigned int)jar_mod_max_samples(&music->ctxMod);
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_MODULE_MOD;
music->loop = true;
TraceLog(DEBUG, "[%s] MOD number of samples: %i", fileName, music->samplesLeft);
TraceLog(DEBUG, "[%s] MOD track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f);
}
else TraceLog(WARNING, "[%s] MOD file could not be opened", fileName);
}
else TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
return music;
}
// Unload music stream
void UnloadMusicStream(Music music)
{
CloseAudioStream(music->stream);
if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close(music->ctxOgg);
else if (music->ctxType == MUSIC_AUDIO_FLAC) drflac_free(music->ctxFlac);
else if (music->ctxType == MUSIC_MODULE_XM) jar_xm_free_context(music->ctxXm);
else if (music->ctxType == MUSIC_MODULE_MOD) jar_mod_unload(&music->ctxMod);
free(music);
}
// Start music playing (open stream)
void PlayMusicStream(Music music)
{
alSourcePlay(music->stream.source);
}
// Pause music playing
void PauseMusicStream(Music music)
{
alSourcePause(music->stream.source);
}
// Resume music playing
void ResumeMusicStream(Music music)
{
ALenum state;
alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED) alSourcePlay(music->stream.source);
}
// Stop music playing (close stream)
void StopMusicStream(Music music)
{
alSourceStop(music->stream.source);
switch (music->ctxType)
{
case MUSIC_AUDIO_OGG: stb_vorbis_seek_start(music->ctxOgg); break;
case MUSIC_MODULE_XM: /* TODO: Restart XM context */ break;
case MUSIC_MODULE_MOD: jar_mod_seek_start(&music->ctxMod); break;
default: break;
}
music->samplesLeft = music->totalSamples;
}
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(Music music)
{
ALenum state;
ALint processed = 0;
alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state); // Get music stream state
alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed); // Get processed buffers
if (processed > 0)
{
bool active = true;
// NOTE: Using dynamic allocation because it could require more than 16KB
void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.channels*music->stream.sampleSize/8, 1);
int numBuffersToProcess = processed;
int numSamples = 0; // Total size of data steamed in L+R samples for xm floats,
// individual L or R for ogg shorts
for (int i = 0; i < numBuffersToProcess; i++)
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE) numSamples = AUDIO_BUFFER_SIZE;
else numSamples = music->samplesLeft;
// TODO: Really don't like ctxType thingy...
switch (music->ctxType)
{
case MUSIC_AUDIO_OGG:
{
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, numSamples*music->stream.channels);
} break;
case MUSIC_AUDIO_FLAC:
{
// NOTE: Returns the number of samples to process
unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, numSamples*music->stream.channels, (short *)pcm);
} break;
case MUSIC_MODULE_XM: jar_xm_generate_samples_16bit(music->ctxXm, pcm, numSamples); break;
case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, numSamples, 0); break;
default: break;
}
UpdateAudioStream(music->stream, pcm, numSamples);
music->samplesLeft -= numSamples;
if (music->samplesLeft <= 0)
{
active = false;
break;
}
}
// This error is registered when UpdateAudioStream() fails
if (alGetError() == AL_INVALID_VALUE) TraceLog(WARNING, "OpenAL: Error buffering data...");
// Reset audio stream for looping
if (!active)
{
StopMusicStream(music); // Stop music (and reset)
if (music->loop) PlayMusicStream(music); // Play again
}
else
{
// NOTE: In case window is minimized, music stream is stopped,
// just make sure to play again on window restore
if (state != AL_PLAYING) PlayMusicStream(music);
}
free(pcm);
}
}
// Check if any music is playing
bool IsMusicPlaying(Music music)
{
bool playing = false;
ALint state;
alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state);
if (state == AL_PLAYING) playing = true;
return playing;
}
// Set volume for music
void SetMusicVolume(Music music, float volume)
{
alSourcef(music->stream.source, AL_GAIN, volume);
}
// Set pitch for music
void SetMusicPitch(Music music, float pitch)
{
alSourcef(music->stream.source, AL_PITCH, pitch);
}
// Get music time length (in seconds)
float GetMusicTimeLength(Music music)
{
float totalSeconds = (float)music->totalSamples/music->stream.sampleRate;
return totalSeconds;
}
// Get current music time played (in seconds)
float GetMusicTimePlayed(Music music)
{
float secondsPlayed = 0.0f;
unsigned int samplesPlayed = music->totalSamples - music->samplesLeft;
secondsPlayed = (float)samplesPlayed/music->stream.sampleRate;
return secondsPlayed;
}
// Init audio stream (to stream audio pcm data)
AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)
{
AudioStream stream = { 0 };
stream.sampleRate = sampleRate;
stream.sampleSize = sampleSize;
// Only mono and stereo channels are supported, more channels require AL_EXT_MCFORMATS extension
if ((channels > 0) && (channels < 3)) stream.channels = channels;
else
{
TraceLog(WARNING, "Init audio stream: Number of channels not supported: %i", channels);
stream.channels = 1; // Fallback to mono channel
}
// Setup OpenAL format
if (stream.channels == 1)
{
switch (sampleSize)
{
case 8: stream.format = AL_FORMAT_MONO8; break;
case 16: stream.format = AL_FORMAT_MONO16; break;
case 32: stream.format = AL_FORMAT_MONO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32
default: TraceLog(WARNING, "Init audio stream: Sample size not supported: %i", sampleSize); break;
}
}
else if (stream.channels == 2)
{
switch (sampleSize)
{
case 8: stream.format = AL_FORMAT_STEREO8; break;
case 16: stream.format = AL_FORMAT_STEREO16; break;
case 32: stream.format = AL_FORMAT_STEREO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32
default: TraceLog(WARNING, "Init audio stream: Sample size not supported: %i", sampleSize); break;
}
}
// Create an audio source
alGenSources(1, &stream.source);
alSourcef(stream.source, AL_PITCH, 1);
alSourcef(stream.source, AL_GAIN, 1);
alSource3f(stream.source, AL_POSITION, 0, 0, 0);
alSource3f(stream.source, AL_VELOCITY, 0, 0, 0);
// Create Buffers (double buffering)
alGenBuffers(MAX_STREAM_BUFFERS, stream.buffers);
// Initialize buffer with zeros by default
// NOTE: Using dynamic allocation because it requires more than 16KB
void *pcm = calloc(AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, 1);
for (int i = 0; i < MAX_STREAM_BUFFERS; i++)
{
alBufferData(stream.buffers[i], stream.format, pcm, AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, stream.sampleRate);
}
free(pcm);
alSourceQueueBuffers(stream.source, MAX_STREAM_BUFFERS, stream.buffers);
TraceLog(INFO, "[AUD ID %i] Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.source, stream.sampleRate, stream.sampleSize, (stream.channels == 1) ? "Mono" : "Stereo");
return stream;
}
// Close audio stream and free memory
void CloseAudioStream(AudioStream stream)
{
// Stop playing channel
alSourceStop(stream.source);
// Flush out all queued buffers
int queued = 0;
alGetSourcei(stream.source, AL_BUFFERS_QUEUED, &queued);
ALuint buffer = 0;
while (queued > 0)
{
alSourceUnqueueBuffers(stream.source, 1, &buffer);
queued--;
}
// Delete source and buffers
alDeleteSources(1, &stream.source);
alDeleteBuffers(MAX_STREAM_BUFFERS, stream.buffers);
TraceLog(INFO, "[AUD ID %i] Unloaded audio stream data", stream.source);
}
// Update audio stream buffers with data
// NOTE: Only updates one buffer per call
void UpdateAudioStream(AudioStream stream, const void *data, int numSamples)
{
ALuint buffer = 0;
alSourceUnqueueBuffers(stream.source, 1, &buffer);
// Check if any buffer was available for unqueue
if (alGetError() != AL_INVALID_VALUE)
{
alBufferData(buffer, stream.format, data, numSamples*stream.channels*stream.sampleSize/8, stream.sampleRate);
alSourceQueueBuffers(stream.source, 1, &buffer);
}
}
// Check if any audio stream buffers requires refill
bool IsAudioBufferProcessed(AudioStream stream)
{
ALint processed = 0;
// Determine if music stream is ready to be written
alGetSourcei(stream.source, AL_BUFFERS_PROCESSED, &processed);
return (processed > 0);
}
// Play audio stream
void PlayAudioStream(AudioStream stream)
{
alSourcePlay(stream.source);
}
// Play audio stream
void PauseAudioStream(AudioStream stream)
{
alSourcePause(stream.source);
}
// Resume audio stream playing
void ResumeAudioStream(AudioStream stream)
{
ALenum state;
alGetSourcei(stream.source, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED) alSourcePlay(stream.source);
}
// Stop audio stream
void StopAudioStream(AudioStream stream)
{
alSourceStop(stream.source);
}
//----------------------------------------------------------------------------------
// Module specific Functions Definition
//----------------------------------------------------------------------------------
// Load WAV file into Wave structure
static Wave LoadWAV(const char *fileName)
{
// Basic WAV headers structs
typedef struct {
char chunkID[4];
int chunkSize;
char format[4];
} WAVRiffHeader;
typedef struct {
char subChunkID[4];
int subChunkSize;
short audioFormat;
short numChannels;
int sampleRate;
int byteRate;
short blockAlign;
short bitsPerSample;
} WAVFormat;
typedef struct {
char subChunkID[4];
int subChunkSize;
} WAVData;
WAVRiffHeader wavRiffHeader;
WAVFormat wavFormat;
WAVData wavData;
Wave wave = { 0 };
FILE *wavFile;
wavFile = fopen(fileName, "rb");
if (wavFile == NULL)
{
TraceLog(WARNING, "[%s] WAV file could not be opened", fileName);
wave.data = NULL;
}
else
{
// Read in the first chunk into the struct
fread(&wavRiffHeader, sizeof(WAVRiffHeader), 1, wavFile);
// Check for RIFF and WAVE tags
if (strncmp(wavRiffHeader.chunkID, "RIFF", 4) ||
strncmp(wavRiffHeader.format, "WAVE", 4))
{
TraceLog(WARNING, "[%s] Invalid RIFF or WAVE Header", fileName);
}
else
{
// Read in the 2nd chunk for the wave info
fread(&wavFormat, sizeof(WAVFormat), 1, wavFile);
// Check for fmt tag
if ((wavFormat.subChunkID[0] != 'f') || (wavFormat.subChunkID[1] != 'm') ||
(wavFormat.subChunkID[2] != 't') || (wavFormat.subChunkID[3] != ' '))
{
TraceLog(WARNING, "[%s] Invalid Wave format", fileName);
}
else
{
// Check for extra parameters;
if (wavFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR);
// Read in the the last byte of data before the sound file
fread(&wavData, sizeof(WAVData), 1, wavFile);
// Check for data tag
if ((wavData.subChunkID[0] != 'd') || (wavData.subChunkID[1] != 'a') ||
(wavData.subChunkID[2] != 't') || (wavData.subChunkID[3] != 'a'))
{
TraceLog(WARNING, "[%s] Invalid data header", fileName);
}
else
{
// Allocate memory for data
wave.data = malloc(wavData.subChunkSize);
// Read in the sound data into the soundData variable
fread(wave.data, 1, wavData.subChunkSize, wavFile);
// Store wave parameters
wave.sampleRate = wavFormat.sampleRate;
wave.sampleSize = wavFormat.bitsPerSample;
wave.channels = wavFormat.numChannels;
// NOTE: Only support 8 bit, 16 bit and 32 bit sample sizes
if ((wave.sampleSize != 8) && (wave.sampleSize != 16) && (wave.sampleSize != 32))
{
TraceLog(WARNING, "[%s] WAV sample size (%ibit) not supported, converted to 16bit", fileName, wave.sampleSize);
WaveFormat(&wave, wave.sampleRate, 16, wave.channels);
}
// NOTE: Only support up to 2 channels (mono, stereo)
if (wave.channels > 2)
{
WaveFormat(&wave, wave.sampleRate, wave.sampleSize, 2);
TraceLog(WARNING, "[%s] WAV channels number (%i) not supported, converted to 2 channels", fileName, wave.channels);
}
// NOTE: subChunkSize comes in bytes, we need to translate it to number of samples
wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels;
TraceLog(INFO, "[%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo");
}
}
}
fclose(wavFile);
}
return wave;
}
// Load OGG file into Wave structure
// NOTE: Using stb_vorbis library
static Wave LoadOGG(const 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.sampleSize = 16; // 16 bit per sample (short)
wave.channels = info.channels;
wave.sampleCount = (int)stb_vorbis_stream_length_in_samples(oggFile);
float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile);
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);
wave.data = (short *)malloc(wave.sampleCount*wave.channels*sizeof(short));
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, (short *)wave.data, wave.sampleCount*wave.channels);
TraceLog(DEBUG, "[%s] Samples obtained: %i", fileName, numSamplesOgg);
TraceLog(INFO, "[%s] OGG file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo");
stb_vorbis_close(oggFile);
}
return wave;
}
// Load FLAC file into Wave structure
// NOTE: Using dr_flac library
static Wave LoadFLAC(const char *fileName)
{
Wave wave;
// Decode an entire FLAC file in one go
uint64_t totalSampleCount;
wave.data = drflac_open_and_decode_file_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount);
wave.sampleCount = (int)totalSampleCount/wave.channels;
wave.sampleSize = 16;
// NOTE: Only support up to 2 channels (mono, stereo)
if (wave.channels > 2) TraceLog(WARNING, "[%s] FLAC channels number (%i) not supported", fileName, wave.channels);
if (wave.data == NULL) TraceLog(WARNING, "[%s] FLAC data could not be loaded", fileName);
else TraceLog(INFO, "[%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo");
return wave;
}
// 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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