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Merge pull request #116 from kd7tck/newaudio 

Redesign audio system to support multiple mix channels
pull/118/head
Ray 8 anos atrás
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8bbbe8cd76 179f2f9e4f
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5 arquivos alterados com 393 adições e 386 exclusões
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  1. +348
    -341
      src/audio.c
  2. +19
    -20
      src/audio.h
  3. +5
    -5
      src/easings.h
  4. +19
    -20
      src/raylib.h
  5. +2
    -0
      src/windows_compile.bat

+ 348
- 341
src/audio.c Ver arquivo

@ -59,8 +59,9 @@
//----------------------------------------------------------------------------------
// Defines and Macros
//----------------------------------------------------------------------------------
#define MAX_STREAM_BUFFERS 2
#define MAX_AUDIO_CONTEXTS 4 // Number of open AL sources
#define MAX_STREAM_BUFFERS 2 // Number of buffers for each alSource
#define MAX_MIX_CHANNELS 4 // Number of open AL sources
#define MAX_MUSIC_STREAMS 2 // Number of simultanious music sources
#if defined(PLATFORM_RPI) || defined(PLATFORM_ANDROID)
// NOTE: On RPI and Android should be lower to avoid frame-stalls
@ -76,37 +77,32 @@
// 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;
jar_xm_context_t *chipctx; // Stores jar_xm context
ALuint buffers[MAX_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. All audio is 32bit floating point in stereo.
typedef struct AudioContext_t {
// Used to create custom audio streams that are not bound to a specific file. There can be
// no more than 4 concurrent mixchannels in use. This is due to each active mixc being tied to
// a dedicated mix channel.
typedef struct MixChannel_t {
unsigned short sampleRate; // default is 48000
unsigned char channels; // 1=mono,2=stereo
unsigned char mixChannel; // 0-3 or mixA-mixD, each mix channel can receive up to one dedicated audio stream
bool floatingPoint; // if false then the short datatype is used instead
bool playing;
bool playing; // false if paused
ALenum alFormat; // openAL format specifier
ALuint alSource; // openAL source
ALuint alBuffer[MAX_STREAM_BUFFERS]; // openAL sample buffer
} AudioContext_t;
} MixChannel_t;
// Music type (file streaming from memory)
// NOTE: Anything longer than ~10 seconds should be streamed into a mix channel...
typedef struct Music {
stb_vorbis *stream;
jar_xm_context_t *chipctx; // Stores jar_xm mixc
MixChannel_t *mixc; // mix channel
int totalSamplesLeft;
float totalLengthSeconds;
bool loop;
bool chipTune; // True if chiptune is loaded
} Music;
#if defined(AUDIO_STANDALONE)
typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
@ -115,23 +111,28 @@ typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
//----------------------------------------------------------------------------------
// 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
static MixChannel_t* mixChannelsActive_g[MAX_MIX_CHANNELS]; // What mix channels are currently active
static bool musicEnabled_g = false;
static Music currentMusic[MAX_MUSIC_STREAMS]; // Current music loaded, up to two can play at the same 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 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(n">ALuint buffer); // Fill music buffers with data
static void EmptyMusicStream(void); // Empty music buffers
static bool BufferMusicStream(kt">int index, int numBuffers); // Fill music buffers with data
static void EmptyMusicStream(int index); // Empty music buffers
static unsigned short FillAlBufferWithSilence(AudioContext_t *context, ALuint buffer);// fill buffer with zeros, returns number processed
static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // pass two arrays of the same legnth in
static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // pass two arrays of same length in
static MixChannel_t* InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint); // For streaming into mix channels.
static void CloseMixChannel(MixChannel_t* mixc); // Frees mix channel
static int BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements); // Pushes more audio data into mixc mix channel, if NULL is passed it pauses
static int FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer); // Fill buffer with zeros, returns number processed
static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // Pass two arrays of the same legnth in
static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // Pass two arrays of same length in
static int IsMusicStreamReadyForBuffering(int index); // Checks if music buffer is ready to be refilled
#if defined(AUDIO_STANDALONE)
const char *GetExtension(const char *fileName); // Get the extension for a filename
@ -142,7 +143,7 @@ void TraceLog(int msgType, const char *text, ...); // Outputs a trace log messa
// Module Functions Definition - Audio Device initialization and Closing
//----------------------------------------------------------------------------------
// Initialize audio device and context
// Initialize audio device and mixc
void InitAudioDevice(void)
{
// Open and initialize a device with default settings
@ -158,7 +159,7 @@ void InitAudioDevice(void)
alcCloseDevice(device);
TraceLog(ERROR, "Could not setup audio context");
TraceLog(ERROR, "Could not setup mix channel");
}
TraceLog(INFO, "Audio device and context initialized successfully: %s", alcGetString(device, ALC_DEVICE_SPECIFIER));
@ -169,15 +170,19 @@ void InitAudioDevice(void)
alListener3f(AL_ORIENTATION, 0, 0, -1);
}
// Close the audio device for the current context, and destroys the context
// Close the audio device for all contexts
void CloseAudioDevice(void)
{
StopMusicStream(); // Stop music streaming and close current stream
for(int index=0; index<MAX_MUSIC_STREAMS; index++)
{
if(currentMusic[index].mixc) StopMusicStream(index); // Stop music streaming and close current stream
}
ALCdevice *device;
ALCcontext *context = alcGetCurrentContext();
if (context == NULL) TraceLog(WARNING, "Could not get current audio context for closing");
if (context == NULL) TraceLog(WARNING, "Could not get current mix channel for closing");
device = alcGetContextsDevice(context);
@ -202,187 +207,141 @@ bool IsAudioDeviceReady(void)
// 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, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
n">AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint)
// For streaming into mix channels.
// The mixChannel is what audio muxing channel you want to operate on, 0-3 are the ones available. Each mix channel can only be used one at a time.
// exmple usage is InitMixChannel(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
k">static MixChannel_t* InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint)
{
if(mixChannel >= MAX_AUDIO_CONTEXTS) return NULL;
if(mixChannel >= MAX_MIX_CHANNELS) return NULL;
if(!IsAudioDeviceReady()) InitAudioDevice();
else StopMusicStream();
if(!mixChannelsActive_g[mixChannel]){
AudioContext_t *ac = (AudioContext_t*)malloc(sizeof(AudioContext_t));
ac->sampleRate = sampleRate;
ac->channels = channels;
ac->mixChannel = mixChannel;
ac->floatingPoint = floatingPoint;
mixChannelsActive_g[mixChannel] = ac;
MixChannel_t *mixc = (MixChannel_t*)malloc(sizeof(MixChannel_t));
mixc->sampleRate = sampleRate;
mixc->channels = channels;
mixc->mixChannel = mixChannel;
mixc->floatingPoint = floatingPoint;
mixChannelsActive_g[mixChannel] = mixc;
// setup openAL format
if(channels == 1)
{
if(floatingPoint)
ac->alFormat = AL_FORMAT_MONO_FLOAT32;
mixc->alFormat = AL_FORMAT_MONO_FLOAT32;
else
ac->alFormat = AL_FORMAT_MONO16;
mixc->alFormat = AL_FORMAT_MONO16;
}
else if(channels == 2)
{
if(floatingPoint)
ac->alFormat = AL_FORMAT_STEREO_FLOAT32;
mixc->alFormat = AL_FORMAT_STEREO_FLOAT32;
else
ac->alFormat = AL_FORMAT_STEREO16;
mixc->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);
alGenSources(1, &mixc->alSource);
alSourcef(mixc->alSource, AL_PITCH, 1);
alSourcef(mixc->alSource, AL_GAIN, 1);
alSource3f(mixc->alSource, AL_POSITION, 0, 0, 0);
alSource3f(mixc->alSource, AL_VELOCITY, 0, 0, 0);
// Create Buffer
alGenBuffers(MAX_STREAM_BUFFERS, ac->alBuffer);
alGenBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
//fill buffers
int x;
for(x=0;x<MAX_STREAM_BUFFERS;x++)
FillAlBufferWithSilence(ac, ac->alBuffer[x]);
FillAlBufferWithSilence(mixc, mixc->alBuffer[x]);
alSourceQueueBuffers(ac->alSource, MAX_STREAM_BUFFERS, ac->alBuffer);
alSourcePlay(ac->alSource);
ac->playing = true;
alSourceQueueBuffers(mixc->alSource, MAX_STREAM_BUFFERS, mixc->alBuffer);
mixc->playing = true;
alSourcePlay(mixc->alSource);
return ac;
return mixc;
}
return NULL;
}
// Frees buffer in audio context
void CloseAudioContext(AudioContext ctx)
// Frees buffer in mix channel
static void CloseMixChannel(MixChannel_t* mixc)
{
AudioContext_t *context = (AudioContext_t*)ctx;
if(context){
alSourceStop(context->alSource);
context->playing = false;
if(mixc){
alSourceStop(mixc->alSource);
mixc->playing = false;
//flush out all queued buffers
ALuint buffer = 0;
int queued = 0;
alGetSourcei(context->alSource, AL_BUFFERS_QUEUED, &queued);
alGetSourcei(mixc->alSource, AL_BUFFERS_QUEUED, &queued);
while (queued > 0)
{
alSourceUnqueueBuffers(context->alSource, 1, &buffer);
alSourceUnqueueBuffers(mixc->alSource, 1, &buffer);
queued--;
}
//delete source and buffers
alDeleteSources(1, &context->alSource);
alDeleteBuffers(MAX_STREAM_BUFFERS, context->alBuffer);
mixChannelsActive_g[context->mixChannel] = NULL;
free(context);
ctx = NULL;
alDeleteSources(1, &mixc->alSource);
alDeleteBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
mixChannelsActive_g[mixc->mixChannel] = NULL;
free(mixc);
mixc = NULL;
}
}
// Pushes more audio data into context mix channel, k">if none are ever pushed then zeros are fed in.
// Call "UpdateAudioContext(ctx, NULL, 0)" if you want to pause the audio.
// Pushes more audio data into mixc mix channel, n">only one buffer per call
// Call "BufferMixChannel(mixc, NULL, 0)" if you want to pause the audio.
// @Returns number of samples that where processed.
t">unsigned short UpdateAudioContext(AudioContext ctx, void *data, unsigned short numberElements)
">static int BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements)
{
AudioContext_t *context = (AudioContext_t*)ctx;
if(!context || (context->channels == 2 && numberElements % 2 != 0)) return 0; // when there is two channels there must be an even number of samples
if(!mixc || mixChannelsActive_g[mixc->mixChannel] != mixc) return 0; // when there is two channels there must be an even number of samples
if (!data || !numberElements)
{ // pauses audio until data is given
alSourcePause(context->alSource);
context->playing = false;
if(mixc->playing){
alSourcePause(mixc->alSource);
mixc->playing = false;
}
return 0;
}
else
else if(!mixc->playing)
{ // restart audio otherwise
ALint state;
alGetSourcei(context->alSource, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING){
alSourcePlay(context->alSource);
context->playing = true;
}
alSourcePlay(mixc->alSource);
mixc->playing = true;
}
if (context && context->playing && mixChannelsActive_g[context->mixChannel] == context)
ALuint buffer = 0;
alSourceUnqueueBuffers(mixc->alSource, 1, &buffer);
if(!buffer) return 0;
if(mixc->floatingPoint) // process float buffers
{
ALint processed = 0;
ALuint buffer = 0;
unsigned short numberProcessed = 0;
unsigned short numberRemaining = numberElements;
alGetSourcei(context->alSource, AL_BUFFERS_PROCESSED, &processed); // Get the number of already processed buffers (if any)
if(!processed) return 0; // nothing to process, queue is still full
while (processed > 0)
{
if(context->floatingPoint) // process float buffers
{
float *ptr = (float*)data;
alSourceUnqueueBuffers(context->alSource, 1, &buffer);
if(numberRemaining >= MUSIC_BUFFER_SIZE_FLOAT)
{
alBufferData(buffer, context->alFormat, &ptr[numberProcessed], MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), context->sampleRate);
numberProcessed+=MUSIC_BUFFER_SIZE_FLOAT;
numberRemaining-=MUSIC_BUFFER_SIZE_FLOAT;
}
else
{
alBufferData(buffer, context->alFormat, &ptr[numberProcessed], numberRemaining*sizeof(float), context->sampleRate);
numberProcessed+=numberRemaining;
numberRemaining=0;
}
alSourceQueueBuffers(context->alSource, 1, &buffer);
processed--;
}
else if(!context->floatingPoint) // process short buffers
{
short *ptr = (short*)data;
alSourceUnqueueBuffers(context->alSource, 1, &buffer);
if(numberRemaining >= MUSIC_BUFFER_SIZE_SHORT)
{
alBufferData(buffer, context->alFormat, &ptr[numberProcessed], MUSIC_BUFFER_SIZE_FLOAT*sizeof(short), context->sampleRate);
numberProcessed+=MUSIC_BUFFER_SIZE_SHORT;
numberRemaining-=MUSIC_BUFFER_SIZE_SHORT;
}
else
{
alBufferData(buffer, context->alFormat, &ptr[numberProcessed], numberRemaining*sizeof(short), context->sampleRate);
numberProcessed+=numberRemaining;
numberRemaining=0;
}
alSourceQueueBuffers(context->alSource, 1, &buffer);
processed--;
}
else
break;
}
return numberProcessed;
float *ptr = (float*)data;
alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(float), mixc->sampleRate);
}
else // process short buffers
{
short *ptr = (short*)data;
alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(short), mixc->sampleRate);
}
return 0;
alSourceQueueBuffers(mixc->alSource, 1, &buffer);
return numberElements;
}
// fill buffer with zeros, returns number processed
static unsigned short FillAlBufferWithSilence(AudioContext_t *context, ALuint buffer)
static int FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer)
{
if(context->floatingPoint){
if(mixc->floatingPoint){
float pcm[MUSIC_BUFFER_SIZE_FLOAT] = {0.f};
alBufferData(buffer, context->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), context->sampleRate);
alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), mixc->sampleRate);
return MUSIC_BUFFER_SIZE_FLOAT;
}
else
{
short pcm[MUSIC_BUFFER_SIZE_SHORT] = {0};
alBufferData(buffer, context->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), context->sampleRate);
alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), mixc->sampleRate);
return MUSIC_BUFFER_SIZE_SHORT;
}
}
@ -417,6 +376,42 @@ static void ResampleByteToFloat(char *chars, float *floats, unsigned short len)
}
}
// used to output raw audio streams, returns negative numbers on error
// if floating point is false the data size is 16bit short, otherwise it is float 32bit
RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint)
{
int mixIndex;
for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
{
if(mixChannelsActive_g[mixIndex] == NULL) break;
else if(mixIndex = MAX_MIX_CHANNELS - 1) return -1; // error
}
if(InitMixChannel(sampleRate, mixIndex, channels, floatingPoint))
return mixIndex;
else
return -2; // error
}
void CloseRawAudioContext(RawAudioContext ctx)
{
if(mixChannelsActive_g[ctx])
CloseMixChannel(mixChannelsActive_g[ctx]);
}
int BufferRawAudioContext(RawAudioContext ctx, void *data, int numberElements)
{
int numBuffered = 0;
if(ctx >= 0)
{
MixChannel_t* mixc = mixChannelsActive_g[ctx];
numBuffered = BufferMixChannel(mixc, data, numberElements);
}
return numBuffered;
}
//----------------------------------------------------------------------------------
@ -767,205 +762,215 @@ void SetSoundPitch(Sound sound, float pitch)
//----------------------------------------------------------------------------------
// Start music playing (open stream)
void PlayMusicStream(char *fileName)
// returns 0 on success
int PlayMusicStream(int musicIndex, char *fileName)
{
int mixIndex;
if(currentMusic[musicIndex].stream || currentMusic[musicIndex].chipctx) return 1; // error
for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot
{
if(mixChannelsActive_g[mixIndex] == NULL) break;
else if(mixIndex = MAX_MIX_CHANNELS - 1) return 2; // error
}
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);
currentMusic[musicIndex].stream = stb_vorbis_open_filename(fileName, NULL, NULL);
if (currentMusic.stream == NULL)
if (currentMusic[musicIndex].stream == NULL)
{
TraceLog(WARNING, "[%s] OGG audio file could not be opened", fileName);
return 3; // error
}
else
{
// Get file info
stb_vorbis_info info = stb_vorbis_get_info(currentMusic.stream);
currentMusic.channels = info.channels;
currentMusic.sampleRate = info.sample_rate;
stb_vorbis_info info = stb_vorbis_get_info(currentMusic[musicIndex].stream);
TraceLog(INFO, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
TraceLog(INFO, "[%s] Ogg channels: %i", fileName, info.channels);
TraceLog(DEBUG, "[%s] Temp memory required: %i", fileName, info.temp_memory_required);
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[musicIndex].loop = true; // We loop by default
musicEnabled_g = true;
currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
currentMusic.totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic.stream);
currentMusic[musicIndex].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[musicIndex].stream) * info.channels;
currentMusic[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[musicIndex].stream);
if (info.channels == 2){
currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false);
currentMusic[musicIndex].mixc->playing = true;
}
else{
currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false);
currentMusic[musicIndex].mixc->playing = true;
}
if(!currentMusic[musicIndex].mixc) return 4; // error
}
}
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))
if(!jar_xm_create_context_from_file(&currentMusic[musicIndex].chipctx, 48000, 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;
currentMusic[musicIndex].chipTune = true;
currentMusic[musicIndex].loop = true;
jar_xm_set_max_loop_count(currentMusic[musicIndex].chipctx, 0); // infinite number of loops
currentMusic[musicIndex].totalSamplesLeft = jar_xm_get_remaining_samples(currentMusic[musicIndex].chipctx);
currentMusic[musicIndex].totalLengthSeconds = ((float)currentMusic[musicIndex].totalSamplesLeft) / 48000.f;
musicEnabled_g = true;
TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic.totalSamplesLeft);
TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, currentMusic.totalLengthSeconds);
TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic[musicIndex].totalSamplesLeft);
TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, currentMusic[musicIndex].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()
currentMusic[musicIndex].mixc = InitMixChannel(48000, mixIndex, 2, false);
if(!currentMusic[musicIndex].mixc) return 5; // error
currentMusic[musicIndex].mixc->playing = true;
}
else TraceLog(WARNING, "[%s] XM file could not be opened", fileName);
else
{
TraceLog(WARNING, "[%s] XM file could not be opened", fileName);
return 6; // error
}
}
else
{
TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
return 7; // error
}
else TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
return 0; // normal return
}
// Stop music playing (close stream)
void StopMusicStream(void)
// Stop music playing k">for individual music index of currentMusic array (close stream)
void StopMusicStream(int index)
{
if (musicEnabled)
if (index < MAX_MUSIC_STREAMS && currentMusic[index].mixc)
{
alSourceStop(currentMusic.source);
EmptyMusicStream(); // Empty music buffers
alDeleteSources(1, &currentMusic.source);
alDeleteBuffers(2, currentMusic.buffers);
CloseMixChannel(currentMusic[index].mixc);
if (currentMusic.chipTune)
if (currentMusic[index].chipTune)
{
jar_xm_free_context(currentMusic.chipctx);
jar_xm_free_context(currentMusic[index].chipctx);
}
else
{
stb_vorbis_close(currentMusic.stream);
stb_vorbis_close(currentMusic[index].stream);
}
if(!getMusicStreamCount()) musicEnabled_g = false;
if(currentMusic[index].stream || currentMusic[index].chipctx)
{
currentMusic[index].stream = NULL;
currentMusic[index].chipctx = NULL;
}
}
}
musicEnabled = false;
//get number of music channels active at this time, this does not mean they are playing
int getMusicStreamCount(void)
{
int musicCount = 0;
for(int musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++) // find empty music slot
if(currentMusic[musicIndex].stream != NULL || currentMusic[musicIndex].chipTune) musicCount++;
return musicCount;
}
// Pause music playing
void PauseMusicStream(void)
void PauseMusicStream(int index)
{
// Pause music stream if music available!
if (musicEnabled)
if (index < MAX_MUSIC_STREAMS && currentMusic[index].mixc && musicEnabled_g)
{
TraceLog(INFO, "Pausing music stream");
alSourcePause(currentMusic.source);
musicEnabled = false;
alSourcePause(currentMusic[index].mixcspan>->alSource);
currentMusic[index].mixc->playing = false;
}
}
// Resume music playing
void ResumeMusicStream(void)
void ResumeMusicStream(int index)
{
// 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;
if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){
alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED)
{
TraceLog(INFO, "Resuming music stream");
alSourcePlay(currentMusic[index].mixc->alSource);
currentMusic[index].mixc->playing = true;
}
}
}
// Check if music is playing
bool IsMusicPlaying(void)
// Check if any music is playing
bool IsMusicPlaying(int index)
{
bool playing = false;
ALint state;
alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
if (state == AL_PLAYING) playing = true;
if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){
alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (state == AL_PLAYING) playing = true;
}
return playing;
}
// Set volume for music
void SetMusicVolume(float volume)
void SetMusicVolume(int index, float volume)
{
alSourcef(currentMusic.source, AL_GAIN, volume);
if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){
alSourcef(currentMusic[index].mixc->alSource, AL_GAIN, volume);
}
}
void SetMusicPitch(int index, float pitch)
{
if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){
alSourcef(currentMusic[index].mixc->alSource, AL_PITCH, pitch);
}
}
// Get current music time length (in seconds)
float GetMusicTimeLength(void)
float GetMusicTimeLength(int index)
{
float totalSeconds;
if (currentMusic.chipTune)
if (currentMusic[index].chipTune)
{
totalSeconds = currentMusic.totalLengthSeconds;
totalSeconds = currentMusic[index].totalLengthSeconds;
}
else
{
totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic.stream);
totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[index].stream);
}
return totalSeconds;
}
// Get current music time played (in seconds)
float GetMusicTimePlayed(void)
float GetMusicTimePlayed(int index)
{
float secondsPlayed;
if (currentMusic.chipTune)
if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc)
{
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);
if (currentMusic[index].chipTune)
{
uint64_t samples;
jar_xm_get_position(currentMusic[index].chipctx, NULL, NULL, NULL, &samples);
secondsPlayed = (float)samples / (48000 * currentMusic[index].mixc->channels); // Not sure if this is the correct value
}
else
{
int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].mixc->channels;
int samplesPlayed = totalSamples - currentMusic[index].totalSamplesLeft;
secondsPlayed = (float)samplesPlayed / (currentMusic[index].mixc->sampleRate * currentMusic[index].mixc->channels);
}
}
@ -977,116 +982,118 @@ float GetMusicTimePlayed(void)
//----------------------------------------------------------------------------------
// Fill music buffers with new data from music stream
static bool BufferMusicStream(n">ALuint buffer)
static bool BufferMusicStream(kt">int index, int numBuffers)
{
short pcm[MUSIC_BUFFER_SIZE_SHORT];
float pcmf[MUSIC_BUFFER_SIZE_FLOAT];
int size = 0; // Total size of data steamed (in bytes)
int streamedBytes = 0; // samples of data obtained, channels are not included in calculation
int size = 0; // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts
bool active = true; // We can get more data from stream (not finished)
if (musicEnabled)
if (currentMusic[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
{
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_SHORT / 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
}
if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT)
size = MUSIC_BUFFER_SIZE_SHORT / 2;
else
size = currentMusic[index].totalSamplesLeft / 2;
for(int x=0; x<numBuffers; x++)
{
while (size < MUSIC_BUFFER_SIZE_SHORT)
jar_xm_generate_samples_16bit(currentMusic[index].chipctx, pcm, size); // reads 2*readlen shorts and moves them to buffer+size memory location
BufferMixChannel(currentMusic[index].mixc, pcm, size * 2);
currentMusic[index].totalSamplesLeft -= size * 2;
if(currentMusic[index].totalSamplesLeft <= 0)
{
streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic.stream, currentMusic.channels, pcm + size, MUSIC_BUFFER_SIZE_SHORT - size);
if (streamedBytes > 0) size += (streamedBytes*currentMusic.channels);
else break;
active = false;
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");
if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT)
size = MUSIC_BUFFER_SIZE_SHORT;
else
size = currentMusic[index].totalSamplesLeft;
for(int x=0; x<numBuffers; x++)
{
int streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].mixc->channels, pcm, size);
BufferMixChannel(currentMusic[index].mixc, pcm, streamedBytes * currentMusic[index].mixc->channels);
currentMusic[index].totalSamplesLeft -= streamedBytes * currentMusic[index].mixc->channels;
if(currentMusic[index].totalSamplesLeft <= 0)
{
active = false;
break;
}
}
}
return active;
}
// Empty music buffers
static void EmptyMusicStream(void)
static void EmptyMusicStream(int index)
{
ALuint buffer = 0;
int queued = 0;
alGetSourcei(currentMusic.source, AL_BUFFERS_QUEUED, &queued);
alGetSourcei(currentMusic[index].mixcspan>->alSource, AL_BUFFERS_QUEUED, &queued);
while (queued > 0)
{
alSourceUnqueueBuffers(currentMusic.source, 1, &buffer);
alSourceUnqueueBuffers(currentMusic[index].mixcspan>->alSource, 1, &buffer);
queued--;
}
}
// Update (re-fill) music buffers if data already processed
t">void UpdateMusicStream(void)
//determine if a music stream is ready to be written to
">static int IsMusicStreamReadyForBuffering(int index)
{
ALuint buffer = 0;
ALint processed = 0;
bool active = true;
alGetSourcei(currentMusic[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed);
return processed;
}
if (musicEnabled)
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(int index)
{
ALenum state;
bool active = true;
int numBuffers = IsMusicStreamReadyForBuffering(index);
if (currentMusic[index].mixc->playing && index < MAX_MUSIC_STREAMS && musicEnabled_g && currentMusic[index].mixc && numBuffers)
{
// Get the number of already processed buffers (if any)
alGetSourcei(currentMusic.source, AL_BUFFERS_PROCESSED, &processed);
while (processed > 0)
active = BufferMusicStream(index, numBuffers);
if (!active && currentMusic[index].loop)
{
// 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[index].chipTune)
{
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);
currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * 48000;
}
// 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--;
else
{
stb_vorbis_seek_start(currentMusic[index].stream);
currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].mixc->channels;
}
active = true;
}
ALenum state;
alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (p">(state != AL_PLAYING) && active) alSourcePlay(currentMusic.source);
if (state != AL_PLAYING && active) alSourcePlay(currentMusic[index].mixcspan>->alSource);
if (!active) StopMusicStream();
if (!active) StopMusicStream(index);
}
else
return;
}
// Load WAV file into Wave structure

+ 19
- 20
src/audio.h Ver arquivo

@ -61,10 +61,7 @@ typedef struct Wave {
short channels;
} Wave;
// 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 void* AudioContext;
typedef int RawAudioContext;
#ifdef __cplusplus
extern "C" { // Prevents name mangling of functions
@ -82,13 +79,6 @@ void InitAudioDevice(void); // Initialize au
void CloseAudioDevice(void); // Close the audio device and context (and music stream)
bool IsAudioDeviceReady(void); // True if call to InitAudioDevice() was successful and CloseAudioDevice() has not been called yet
// 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, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint);
void CloseAudioContext(AudioContext ctx); // Frees audio context
unsigned short UpdateAudioContext(AudioContext ctx, void *data, unsigned short numberElements); // Pushes more audio data into context mix channel, if NULL is passed to data then zeros are played
Sound LoadSound(char *fileName); // Load sound to memory
Sound LoadSoundFromWave(Wave wave); // Load sound to memory from wave data
Sound LoadSoundFromRES(const char *rresName, int resId); // Load sound to memory from rRES file (raylib Resource)
@ -100,15 +90,24 @@ bool IsSoundPlaying(Sound sound); // Check if a so
void SetSoundVolume(Sound sound, float volume); // Set volume for a sound (1.0 is max level)
void SetSoundPitch(Sound sound, float pitch); // Set pitch for a sound (1.0 is base level)
void PlayMusicStream(char *fileName); // Start music playing (open stream)
void UpdateMusicStream(void); // Updates buffers for music streaming
void StopMusicStream(void); // Stop music playing (close stream)
void PauseMusicStream(void); // Pause music playing
void ResumeMusicStream(void); // Resume playing paused music
bool IsMusicPlaying(void); // Check if music is playing
void SetMusicVolume(float volume); // Set volume for music (1.0 is max level)
float GetMusicTimeLength(void); // Get music time length (in seconds)
float GetMusicTimePlayed(void); // Get current music time played (in seconds)
int PlayMusicStream(int musicIndex, char *fileName); // Start music playing (open stream)
void UpdateMusicStream(int index); // Updates buffers for music streaming
void StopMusicStream(int index); // Stop music playing (close stream)
void PauseMusicStream(int index); // Pause music playing
void ResumeMusicStream(int index); // Resume playing paused music
bool IsMusicPlaying(int index); // Check if music is playing
void SetMusicVolume(int index, float volume); // Set volume for music (1.0 is max level)
float GetMusicTimeLength(int index); // Get music time length (in seconds)
float GetMusicTimePlayed(int index); // Get current music time played (in seconds)
int getMusicStreamCount(void);
void SetMusicPitch(int index, float pitch);
// used to output raw audio streams, returns negative numbers on error
// if floating point is false the data size is 16bit short, otherwise it is float 32bit
RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint);
void CloseRawAudioContext(RawAudioContext ctx);
int BufferRawAudioContext(RawAudioContext ctx, void *data, int numberElements); // returns number of elements buffered
#ifdef __cplusplus
}

+ 5
- 5
src/easings.h Ver arquivo

@ -18,11 +18,11 @@
* float speed = 1.f;
* float currentTime = 0.f;
* float currentPos[2] = {0,0};
* float newPos[2] = {1,1};
* float tempPosition[2] = currentPos;//x,y positions
* while(currentPos[0] < newPos[0])
* currentPos[0] = EaseSineIn(currentTime, tempPosition[0], tempPosition[0]-newPos[0], speed);
* currentPos[1] = EaseSineIn(currentTime, tempPosition[1], tempPosition[1]-newPos[0], speed);
* float finalPos[2] = {1,1};
* float startPosition[2] = currentPos;//x,y positions
* while(currentPos[0] < finalPos[0])
* currentPos[0] = EaseSineIn(currentTime, startPosition[0], startPosition[0]-finalPos[0], speed);
* currentPos[1] = EaseSineIn(currentTime, startPosition[1], startPosition[1]-finalPos[0], speed);
* currentTime += diffTime();
*
* A port of Robert Penner's easing equations to C (http://robertpenner.com/easing/)

+ 19
- 20
src/raylib.h Ver arquivo

@ -451,10 +451,7 @@ typedef struct Wave {
short channels;
} Wave;
// 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 void* AudioContext;
typedef int RawAudioContext;
// Texture formats
// NOTE: Support depends on OpenGL version and platform
@ -876,13 +873,6 @@ void InitAudioDevice(void); // Initialize au
void CloseAudioDevice(void); // Close the audio device and context (and music stream)
bool IsAudioDeviceReady(void); // True if call to InitAudioDevice() was successful and CloseAudioDevice() has not been called yet
// 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, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint);
void CloseAudioContext(AudioContext ctx); // Frees audio context
unsigned short UpdateAudioContext(AudioContext ctx, void *data, unsigned short numberElements); // Pushes more audio data into context mix channel, if NULL is passed to data then zeros are played
Sound LoadSound(char *fileName); // Load sound to memory
Sound LoadSoundFromWave(Wave wave); // Load sound to memory from wave data
Sound LoadSoundFromRES(const char *rresName, int resId); // Load sound to memory from rRES file (raylib Resource)
@ -894,15 +884,24 @@ bool IsSoundPlaying(Sound sound); // Check if a so
void SetSoundVolume(Sound sound, float volume); // Set volume for a sound (1.0 is max level)
void SetSoundPitch(Sound sound, float pitch); // Set pitch for a sound (1.0 is base level)
void PlayMusicStream(char *fileName); // Start music playing (open stream)
void UpdateMusicStream(void); // Updates buffers for music streaming
void StopMusicStream(void); // Stop music playing (close stream)
void PauseMusicStream(void); // Pause music playing
void ResumeMusicStream(void); // Resume playing paused music
bool IsMusicPlaying(void); // Check if music is playing
void SetMusicVolume(float volume); // Set volume for music (1.0 is max level)
float GetMusicTimeLength(void); // Get current music time length (in seconds)
float GetMusicTimePlayed(void); // Get current music time played (in seconds)
int PlayMusicStream(int musicIndex, char *fileName); // Start music playing (open stream)
void UpdateMusicStream(int index); // Updates buffers for music streaming
void StopMusicStream(int index); // Stop music playing (close stream)
void PauseMusicStream(int index); // Pause music playing
void ResumeMusicStream(int index); // Resume playing paused music
bool IsMusicPlaying(int index); // Check if music is playing
void SetMusicVolume(int index, float volume); // Set volume for music (1.0 is max level)
float GetMusicTimeLength(int index); // Get current music time length (in seconds)
float GetMusicTimePlayed(int index); // Get current music time played (in seconds)
int getMusicStreamCount(void);
void SetMusicPitch(int index, float pitch);
// used to output raw audio streams, returns negative numbers on error
// if floating point is false the data size is 16bit short, otherwise it is float 32bit
RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint);
void CloseRawAudioContext(RawAudioContext ctx);
int BufferRawAudioContext(RawAudioContext ctx, void *data, int numberElements); // returns number of elements buffered
#ifdef __cplusplus
}

+ 2
- 0
src/windows_compile.bat Ver arquivo

@ -0,0 +1,2 @@
set PATH=C:\raylib\MinGW\bin;%PATH%
mingw32-make

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