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renamed everything so it is obvious what it does

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This commit is contained in:
Joshua Reisenauer 2016-05-15 19:37:15 -07:00
parent 86fbf4fd8f
commit 76ff4d220e
3 changed files with 190 additions and 220 deletions
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380
src/audio.c
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@ -77,10 +77,10 @@
// Types and Structures Definition // Types and Structures Definition
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
// Audio Context, used to create custom audio streams that are not bound to a sound file. There can be // Used to create custom audio streams that are not bound to a specific file. There can be
// no more than 4 concurrent audio contexts in use. This is due to each active context being tied to // no more than 4 concurrent mixchannels in use. This is due to each active mixc being tied to
// a dedicated mix channel. All audio is 32bit floating point in stereo. // a dedicated mix channel.
typedef struct AudioContext_t { typedef struct MixChannel_t {
unsigned short sampleRate; // default is 48000 unsigned short sampleRate; // default is 48000
unsigned char channels; // 1=mono,2=stereo 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 unsigned char mixChannel; // 0-3 or mixA-mixD, each mix channel can receive up to one dedicated audio stream
@ -89,14 +89,14 @@ typedef struct AudioContext_t {
ALenum alFormat; // openAL format specifier ALenum alFormat; // openAL format specifier
ALuint alSource; // openAL source ALuint alSource; // openAL source
ALuint alBuffer[MAX_STREAM_BUFFERS]; // openAL sample buffer ALuint alBuffer[MAX_STREAM_BUFFERS]; // openAL sample buffer
} AudioContext_t; } MixChannel_t;
// Music type (file streaming from memory) // Music type (file streaming from memory)
// NOTE: Anything longer than ~10 seconds should be streamed... // NOTE: Anything longer than ~10 seconds should be streamed into a mix channel...
typedef struct Music { typedef struct Music {
stb_vorbis *stream; stb_vorbis *stream;
jar_xm_context_t *chipctx; // Stores jar_xm context jar_xm_context_t *chipctx; // Stores jar_xm mixc
AudioContext_t *ctx; // audio context MixChannel_t *mixc; // mix channel
int totalSamplesLeft; int totalSamplesLeft;
float totalLengthSeconds; float totalLengthSeconds;
@ -111,9 +111,9 @@ typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType;
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
// Global Variables Definition // Global Variables Definition
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
static AudioContext_t* mixChannelsActive_g[MAX_AUDIO_CONTEXTS]; // What mix channels are currently active static MixChannel_t* mixChannelsActive_g[MAX_AUDIO_CONTEXTS]; // What mix channels are currently active
static bool musicEnabled_g = false; static bool musicEnabled_g = false;
static Music currentMusic[MAX_MUSIC_STREAMS]; // Current music loaded, up to two can play at the same time static Music currentMusic[MAX_MUSIC_STREAMS]; // Current music loaded, up to two can play at the same time
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
// Module specific Functions Declaration // Module specific Functions Declaration
@ -122,13 +122,17 @@ static Wave LoadWAV(const char *fileName); // Load WAV file
static Wave LoadOGG(char *fileName); // Load OGG file static Wave LoadOGG(char *fileName); // Load OGG file
static void UnloadWave(Wave wave); // Unload wave data static void UnloadWave(Wave wave); // Unload wave data
static bool BufferMusicStream(int index); // Fill music buffers with data static bool BufferMusicStream(int index, int numBuffers); // Fill music buffers with data
static void EmptyMusicStream(int index); // Empty music buffers 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 MixChannel_t* InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint); // For streaming into mix channels.
static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // pass two arrays of same length in static void CloseMixChannel(MixChannel_t* mixc); // Frees mix channel
static bool isMusicStreamReady(int index); // Checks if music buffer is ready to be refilled static unsigned short BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements); // Pushes more audio data into mixc mix channel, if NULL is passed it pauses
static unsigned short 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) #if defined(AUDIO_STANDALONE)
const char *GetExtension(const char *fileName); // Get the extension for a filename const char *GetExtension(const char *fileName); // Get the extension for a filename
@ -139,7 +143,7 @@ void TraceLog(int msgType, const char *text, ...); // Outputs a trace log messa
// Module Functions Definition - Audio Device initialization and Closing // Module Functions Definition - Audio Device initialization and Closing
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
// Initialize audio device and context // Initialize audio device and mixc
void InitAudioDevice(void) void InitAudioDevice(void)
{ {
// Open and initialize a device with default settings // Open and initialize a device with default settings
@ -155,7 +159,7 @@ void InitAudioDevice(void)
alcCloseDevice(device); 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)); TraceLog(INFO, "Audio device and context initialized successfully: %s", alcGetString(device, ALC_DEVICE_SPECIFIER));
@ -171,14 +175,14 @@ void CloseAudioDevice(void)
{ {
for(int index=0; index<MAX_MUSIC_STREAMS; index++) for(int index=0; index<MAX_MUSIC_STREAMS; index++)
{ {
if(currentMusic[index].ctx) StopMusicStream(index); // Stop music streaming and close current stream if(currentMusic[index].mixc) StopMusicStream(index); // Stop music streaming and close current stream
} }
ALCdevice *device; ALCdevice *device;
ALCcontext *context = alcGetCurrentContext(); 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); device = alcGetContextsDevice(context);
@ -203,186 +207,141 @@ bool IsAudioDeviceReady(void)
// Module Functions Definition - Custom audio output // Module Functions Definition - Custom audio output
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
// Audio contexts are for outputing custom audio waveforms, This will shut down any other sound sources currently playing // For streaming into mix channels.
// 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. // 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 InitAudioContext(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point // exmple usage is InitMixChannel(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint) 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_AUDIO_CONTEXTS) return NULL;
if(!IsAudioDeviceReady()) InitAudioDevice(); if(!IsAudioDeviceReady()) InitAudioDevice();
if(!mixChannelsActive_g[mixChannel]){ if(!mixChannelsActive_g[mixChannel]){
AudioContext_t *ac = (AudioContext_t*)malloc(sizeof(AudioContext_t)); MixChannel_t *mixc = (MixChannel_t*)malloc(sizeof(MixChannel_t));
ac->sampleRate = sampleRate; mixc->sampleRate = sampleRate;
ac->channels = channels; mixc->channels = channels;
ac->mixChannel = mixChannel; mixc->mixChannel = mixChannel;
ac->floatingPoint = floatingPoint; mixc->floatingPoint = floatingPoint;
mixChannelsActive_g[mixChannel] = ac; mixChannelsActive_g[mixChannel] = mixc;
// setup openAL format // setup openAL format
if(channels == 1) if(channels == 1)
{ {
if(floatingPoint) if(floatingPoint)
ac->alFormat = AL_FORMAT_MONO_FLOAT32; mixc->alFormat = AL_FORMAT_MONO_FLOAT32;
else else
ac->alFormat = AL_FORMAT_MONO16; mixc->alFormat = AL_FORMAT_MONO16;
} }
else if(channels == 2) else if(channels == 2)
{ {
if(floatingPoint) if(floatingPoint)
ac->alFormat = AL_FORMAT_STEREO_FLOAT32; mixc->alFormat = AL_FORMAT_STEREO_FLOAT32;
else else
ac->alFormat = AL_FORMAT_STEREO16; mixc->alFormat = AL_FORMAT_STEREO16;
} }
// Create an audio source // Create an audio source
alGenSources(1, &ac->alSource); alGenSources(1, &mixc->alSource);
alSourcef(ac->alSource, AL_PITCH, 1); alSourcef(mixc->alSource, AL_PITCH, 1);
alSourcef(ac->alSource, AL_GAIN, 1); alSourcef(mixc->alSource, AL_GAIN, 1);
alSource3f(ac->alSource, AL_POSITION, 0, 0, 0); alSource3f(mixc->alSource, AL_POSITION, 0, 0, 0);
alSource3f(ac->alSource, AL_VELOCITY, 0, 0, 0); alSource3f(mixc->alSource, AL_VELOCITY, 0, 0, 0);
// Create Buffer // Create Buffer
alGenBuffers(MAX_STREAM_BUFFERS, ac->alBuffer); alGenBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
//fill buffers //fill buffers
int x; int x;
for(x=0;x<MAX_STREAM_BUFFERS;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); alSourceQueueBuffers(mixc->alSource, MAX_STREAM_BUFFERS, mixc->alBuffer);
alSourcePlay(ac->alSource); mixc->playing = true;
ac->playing = true; alSourcePlay(mixc->alSource);
return ac; return mixc;
} }
return NULL; return NULL;
} }
// Frees buffer in audio context // Frees buffer in mix channel
void CloseAudioContext(AudioContext ctx) static void CloseMixChannel(MixChannel_t* mixc)
{ {
AudioContext_t *context = (AudioContext_t*)ctx; if(mixc){
if(context){ alSourceStop(mixc->alSource);
alSourceStop(context->alSource); mixc->playing = false;
context->playing = false;
//flush out all queued buffers //flush out all queued buffers
ALuint buffer = 0; ALuint buffer = 0;
int queued = 0; int queued = 0;
alGetSourcei(context->alSource, AL_BUFFERS_QUEUED, &queued); alGetSourcei(mixc->alSource, AL_BUFFERS_QUEUED, &queued);
while (queued > 0) while (queued > 0)
{ {
alSourceUnqueueBuffers(context->alSource, 1, &buffer); alSourceUnqueueBuffers(mixc->alSource, 1, &buffer);
queued--; queued--;
} }
//delete source and buffers //delete source and buffers
alDeleteSources(1, &context->alSource); alDeleteSources(1, &mixc->alSource);
alDeleteBuffers(MAX_STREAM_BUFFERS, context->alBuffer); alDeleteBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer);
mixChannelsActive_g[context->mixChannel] = NULL; mixChannelsActive_g[mixc->mixChannel] = NULL;
free(context); free(mixc);
ctx = NULL; mixc = NULL;
} }
} }
// Pushes more audio data into context mix channel, if none are ever pushed then zeros are fed in. // Pushes more audio data into mixc mix channel, only one buffer per call
// Call "UpdateAudioContext(ctx, NULL, 0)" if you want to pause the audio. // Call "BufferMixChannel(mixc, NULL, 0)" if you want to pause the audio.
// @Returns number of samples that where processed. // @Returns number of samples that where processed.
unsigned short UpdateAudioContext(AudioContext ctx, void *data, unsigned short numberElements) static unsigned short BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements)
{ {
AudioContext_t *context = (AudioContext_t*)ctx; if(!mixc || mixChannelsActive_g[mixc->mixChannel] != mixc) return 0; // when there is two channels there must be an even number of samples
if(!context || (context->channels == 2 && numberElements % 2 != 0)) return 0; // when there is two channels there must be an even number of samples
if (!data || !numberElements) if (!data || !numberElements)
{ // pauses audio until data is given { // pauses audio until data is given
alSourcePause(context->alSource); if(mixc->playing){
context->playing = false; alSourcePause(mixc->alSource);
mixc->playing = false;
}
return 0; return 0;
} }
else else if(!mixc->playing)
{ // restart audio otherwise { // restart audio otherwise
ALint state; alSourcePlay(mixc->alSource);
alGetSourcei(context->alSource, AL_SOURCE_STATE, &state); mixc->playing = true;
if (state != AL_PLAYING){
alSourcePlay(context->alSource);
context->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; float *ptr = (float*)data;
ALuint buffer = 0; alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(float), mixc->sampleRate);
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;
} }
return 0; else // process short buffers
{
short *ptr = (short*)data;
alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(short), mixc->sampleRate);
}
alSourceQueueBuffers(mixc->alSource, 1, &buffer);
return numberElements;
} }
// fill buffer with zeros, returns number processed // fill buffer with zeros, returns number processed
static unsigned short FillAlBufferWithSilence(AudioContext_t *context, ALuint buffer) static unsigned short FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer)
{ {
if(context->floatingPoint){ if(mixc->floatingPoint){
float pcm[MUSIC_BUFFER_SIZE_FLOAT] = {0.f}; 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; return MUSIC_BUFFER_SIZE_FLOAT;
} }
else else
{ {
short pcm[MUSIC_BUFFER_SIZE_SHORT] = {0}; 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; return MUSIC_BUFFER_SIZE_SHORT;
} }
} }
@ -417,6 +376,28 @@ static void ResampleByteToFloat(char *chars, float *floats, unsigned short len)
} }
} }
// used to output raw audio streams, returns negative numbers on error
RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint)
{
int mixIndex;
for(mixIndex = 0; mixIndex < MAX_AUDIO_CONTEXTS; mixIndex++) // find empty mix channel slot
{
if(mixChannelsActive_g[mixIndex] == NULL) break;
else if(mixIndex = MAX_AUDIO_CONTEXTS - 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]);
}
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
@ -807,14 +788,14 @@ int PlayMusicStream(int musicIndex, char *fileName)
currentMusic[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[musicIndex].stream); currentMusic[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[musicIndex].stream);
if (info.channels == 2){ if (info.channels == 2){
currentMusic[musicIndex].ctx = InitAudioContext(info.sample_rate, mixIndex, 2, false); currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false);
currentMusic[musicIndex].ctx->playing = true; currentMusic[musicIndex].mixc->playing = true;
} }
else{ else{
currentMusic[musicIndex].ctx = InitAudioContext(info.sample_rate, mixIndex, 1, false); currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false);
currentMusic[musicIndex].ctx->playing = true; currentMusic[musicIndex].mixc->playing = true;
} }
if(!currentMusic[musicIndex].ctx) return 4; // error if(!currentMusic[musicIndex].mixc) return 4; // error
} }
} }
else if (strcmp(GetExtension(fileName),"xm") == 0) else if (strcmp(GetExtension(fileName),"xm") == 0)
@ -832,9 +813,9 @@ int PlayMusicStream(int musicIndex, char *fileName)
TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic[musicIndex].totalSamplesLeft); 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); TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, currentMusic[musicIndex].totalLengthSeconds);
currentMusic[musicIndex].ctx = InitAudioContext(48000, mixIndex, 2, true); currentMusic[musicIndex].mixc = InitMixChannel(48000, mixIndex, 2, false);
if(!currentMusic[musicIndex].ctx) return 5; // error if(!currentMusic[musicIndex].mixc) return 5; // error
currentMusic[musicIndex].ctx->playing = true; currentMusic[musicIndex].mixc->playing = true;
} }
else else
{ {
@ -853,9 +834,9 @@ int PlayMusicStream(int musicIndex, char *fileName)
// Stop music playing for individual music index of currentMusic array (close stream) // Stop music playing for individual music index of currentMusic array (close stream)
void StopMusicStream(int index) void StopMusicStream(int index)
{ {
if (index < MAX_MUSIC_STREAMS && currentMusic[index].ctx) if (index < MAX_MUSIC_STREAMS && currentMusic[index].mixc)
{ {
CloseAudioContext(currentMusic[index].ctx); CloseMixChannel(currentMusic[index].mixc);
if (currentMusic[index].chipTune) if (currentMusic[index].chipTune)
{ {
@ -889,11 +870,11 @@ int getMusicStreamCount(void)
void PauseMusicStream(int index) void PauseMusicStream(int index)
{ {
// Pause music stream if music available! // Pause music stream if music available!
if (index < MAX_MUSIC_STREAMS && currentMusic[index].ctx && musicEnabled_g) if (index < MAX_MUSIC_STREAMS && currentMusic[index].mixc && musicEnabled_g)
{ {
TraceLog(INFO, "Pausing music stream"); TraceLog(INFO, "Pausing music stream");
alSourcePause(currentMusic[index].ctx->alSource); alSourcePause(currentMusic[index].mixc->alSource);
currentMusic[index].ctx->playing = false; currentMusic[index].mixc->playing = false;
} }
} }
@ -902,13 +883,13 @@ void ResumeMusicStream(int index)
{ {
// Resume music playing... if music available! // Resume music playing... if music available!
ALenum state; ALenum state;
if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){ if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){
alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state); alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED) if (state == AL_PAUSED)
{ {
TraceLog(INFO, "Resuming music stream"); TraceLog(INFO, "Resuming music stream");
alSourcePlay(currentMusic[index].ctx->alSource); alSourcePlay(currentMusic[index].mixc->alSource);
currentMusic[index].ctx->playing = true; currentMusic[index].mixc->playing = true;
} }
} }
} }
@ -919,8 +900,8 @@ bool IsMusicPlaying(int index)
bool playing = false; bool playing = false;
ALint state; ALint state;
if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){ if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){
alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state); alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (state == AL_PLAYING) playing = true; if (state == AL_PLAYING) playing = true;
} }
@ -930,15 +911,15 @@ bool IsMusicPlaying(int index)
// Set volume for music // Set volume for music
void SetMusicVolume(int index, float volume) void SetMusicVolume(int index, float volume)
{ {
if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){ if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){
alSourcef(currentMusic[index].ctx->alSource, AL_GAIN, volume); alSourcef(currentMusic[index].mixc->alSource, AL_GAIN, volume);
} }
} }
void SetMusicPitch(int index, float pitch) void SetMusicPitch(int index, float pitch)
{ {
if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){ if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){
alSourcef(currentMusic[index].ctx->alSource, AL_PITCH, pitch); alSourcef(currentMusic[index].mixc->alSource, AL_PITCH, pitch);
} }
} }
@ -962,19 +943,19 @@ float GetMusicTimeLength(int index)
float GetMusicTimePlayed(int index) float GetMusicTimePlayed(int index)
{ {
float secondsPlayed; float secondsPlayed;
if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx) if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc)
{ {
if (currentMusic[index].chipTune) if (currentMusic[index].chipTune)
{ {
uint64_t samples; uint64_t samples;
jar_xm_get_position(currentMusic[index].chipctx, NULL, NULL, NULL, &samples); jar_xm_get_position(currentMusic[index].chipctx, NULL, NULL, NULL, &samples);
secondsPlayed = (float)samples / (48000 * currentMusic[index].ctx->channels); // Not sure if this is the correct value secondsPlayed = (float)samples / (48000 * currentMusic[index].mixc->channels); // Not sure if this is the correct value
} }
else else
{ {
int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].ctx->channels; int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].mixc->channels;
int samplesPlayed = totalSamples - currentMusic[index].totalSamplesLeft; int samplesPlayed = totalSamples - currentMusic[index].totalSamplesLeft;
secondsPlayed = (float)samplesPlayed / (currentMusic[index].ctx->sampleRate * currentMusic[index].ctx->channels); secondsPlayed = (float)samplesPlayed / (currentMusic[index].mixc->sampleRate * currentMusic[index].mixc->channels);
} }
} }
@ -987,32 +968,32 @@ float GetMusicTimePlayed(int index)
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
// Fill music buffers with new data from music stream // Fill music buffers with new data from music stream
static bool BufferMusicStream(int index) static bool BufferMusicStream(int index, int numBuffers)
{ {
short pcm[MUSIC_BUFFER_SIZE_SHORT]; short pcm[MUSIC_BUFFER_SIZE_SHORT];
float pcmf[MUSIC_BUFFER_SIZE_FLOAT]; float pcmf[MUSIC_BUFFER_SIZE_FLOAT];
int size = 0; // Total size of data steamed in L+R samples 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) bool active = true; // We can get more data from stream (not finished)
if (!currentMusic[index].ctx->playing && currentMusic[index].totalSamplesLeft > 0)
{
UpdateAudioContext(currentMusic[index].ctx, NULL, 0);
return true; // it is still active but it is paused
}
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[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
{ {
if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_FLOAT / 2) if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT)
size = MUSIC_BUFFER_SIZE_FLOAT / 2; size = MUSIC_BUFFER_SIZE_SHORT / 2;
else else
size = currentMusic[index].totalSamplesLeft / 2; size = currentMusic[index].totalSamplesLeft / 2;
jar_xm_generate_samples(currentMusic[index].chipctx, pcmf, size); // reads 2*readlen shorts and moves them to buffer+size memory location for(int x=0; x<numBuffers; x++)
UpdateAudioContext(currentMusic[index].ctx, pcmf, size * 2); {
currentMusic[index].totalSamplesLeft -= size * 2; 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)
{
active = false;
break;
}
}
} }
else else
{ {
@ -1021,13 +1002,18 @@ static bool BufferMusicStream(int index)
else else
size = currentMusic[index].totalSamplesLeft; size = currentMusic[index].totalSamplesLeft;
int streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].ctx->channels, pcm, size); for(int x=0; x<numBuffers; x++)
UpdateAudioContext(currentMusic[index].ctx, pcm, streamedBytes * currentMusic[index].ctx->channels); {
currentMusic[index].totalSamplesLeft -= streamedBytes * currentMusic[index].ctx->channels; 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;
}
}
} }
TraceLog(DEBUG, "Buffering index:%i, chiptune:%i", index, (int)currentMusic[index].chipTune);
if(currentMusic[index].totalSamplesLeft <= 0) active = false;
return active; return active;
} }
@ -1038,25 +1024,22 @@ static void EmptyMusicStream(int index)
ALuint buffer = 0; ALuint buffer = 0;
int queued = 0; int queued = 0;
alGetSourcei(currentMusic[index].ctx->alSource, AL_BUFFERS_QUEUED, &queued); alGetSourcei(currentMusic[index].mixc->alSource, AL_BUFFERS_QUEUED, &queued);
while (queued > 0) while (queued > 0)
{ {
alSourceUnqueueBuffers(currentMusic[index].ctx->alSource, 1, &buffer); alSourceUnqueueBuffers(currentMusic[index].mixc->alSource, 1, &buffer);
queued--; queued--;
} }
} }
//determine if a music stream is ready to be written to //determine if a music stream is ready to be written to
static bool isMusicStreamReady(int index) static int IsMusicStreamReadyForBuffering(int index)
{ {
ALint processed = 0; ALint processed = 0;
alGetSourcei(currentMusic[index].ctx->alSource, AL_BUFFERS_PROCESSED, &processed); alGetSourcei(currentMusic[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed);
return processed;
if(processed) return true;
return false;
} }
// Update (re-fill) music buffers if data already processed // Update (re-fill) music buffers if data already processed
@ -1064,21 +1047,22 @@ void UpdateMusicStream(int index)
{ {
ALenum state; ALenum state;
bool active = true; bool active = true;
int numBuffers = IsMusicStreamReadyForBuffering(index);
if (index < MAX_MUSIC_STREAMS && musicEnabled_g && currentMusic[index].ctx && isMusicStreamReady(index))
if (currentMusic[index].mixc->playing && index < MAX_MUSIC_STREAMS && musicEnabled_g && currentMusic[index].mixc && numBuffers)
{ {
active = BufferMusicStream(index); active = BufferMusicStream(index, numBuffers);
if (!active && currentMusic[index].loop && currentMusic[index].ctx->playing) if (!active && currentMusic[index].loop)
{ {
if (currentMusic[index].chipTune) if (currentMusic[index].chipTune)
{ {
currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * currentMusic[index].ctx->sampleRate; currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * 48000;
} }
else else
{ {
stb_vorbis_seek_start(currentMusic[index].stream); stb_vorbis_seek_start(currentMusic[index].stream);
currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].ctx->channels; currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].mixc->channels;
} }
active = true; active = true;
} }
@ -1086,9 +1070,9 @@ void UpdateMusicStream(int index)
if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data..."); if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state); alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING && active && currentMusic[index].ctx->playing) alSourcePlay(currentMusic[index].ctx->alSource); if (state != AL_PLAYING && active) alSourcePlay(currentMusic[index].mixc->alSource);
if (!active) StopMusicStream(index); if (!active) StopMusicStream(index);

15
src/audio.h
View file

@ -61,10 +61,7 @@ typedef struct Wave {
short channels; short channels;
} Wave; } Wave;
// Audio Context, used to create custom audio streams that are not bound to a sound file. There can be typedef int RawAudioContext;
// 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;
#ifdef __cplusplus #ifdef __cplusplus
extern "C" { // Prevents name mangling of functions 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) 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 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 LoadSound(char *fileName); // Load sound to memory
Sound LoadSoundFromWave(Wave wave); // Load sound to memory from wave data 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) Sound LoadSoundFromRES(const char *rresName, int resId); // Load sound to memory from rRES file (raylib Resource)
@ -112,6 +102,9 @@ float GetMusicTimePlayed(int index); // Get current m
int getMusicStreamCount(void); int getMusicStreamCount(void);
void SetMusicPitch(int index, float pitch); void SetMusicPitch(int index, float pitch);
RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint);
void CloseRawAudioContext(RawAudioContext ctx);
#ifdef __cplusplus #ifdef __cplusplus
} }
#endif #endif

15
src/raylib.h
View file

@ -451,10 +451,7 @@ typedef struct Wave {
short channels; short channels;
} Wave; } Wave;
// Audio Context, used to create custom audio streams that are not bound to a sound file. There can be typedef int RawAudioContext;
// 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;
// Texture formats // Texture formats
// NOTE: Support depends on OpenGL version and platform // 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) 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 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 LoadSound(char *fileName); // Load sound to memory
Sound LoadSoundFromWave(Wave wave); // Load sound to memory from wave data 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) Sound LoadSoundFromRES(const char *rresName, int resId); // Load sound to memory from rRES file (raylib Resource)
@ -906,6 +896,9 @@ float GetMusicTimePlayed(int index); // Get current m
int getMusicStreamCount(void); int getMusicStreamCount(void);
void SetMusicPitch(int index, float pitch); void SetMusicPitch(int index, float pitch);
RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint); // used to output raw audio streams, returns negative numbers on error
void CloseRawAudioContext(RawAudioContext ctx);
#ifdef __cplusplus #ifdef __cplusplus
} }
#endif #endif