new boolean floatingPoint option
Now floating point is either on or off. This simplifies the use of 16bit vs float.
This commit is contained in:
parent
c3208c5cd6
commit
9d09ada33b
3 changed files with 88 additions and 58 deletions
132
src/audio.c
132
src/audio.c
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@ -59,15 +59,17 @@
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//----------------------------------------------------------------------------------
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//----------------------------------------------------------------------------------
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// Defines and Macros
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// Defines and Macros
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//----------------------------------------------------------------------------------
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//----------------------------------------------------------------------------------
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#define MUSIC_STREAM_BUFFERS 2
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#define MAX_STREAM_BUFFERS 2
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#define MAX_AUDIO_CONTEXTS 4
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#define MAX_AUDIO_CONTEXTS 4 // Number of open AL sources
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#if defined(PLATFORM_RPI) || defined(PLATFORM_ANDROID)
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#if defined(PLATFORM_RPI) || defined(PLATFORM_ANDROID)
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// NOTE: On RPI and Android should be lower to avoid frame-stalls
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// NOTE: On RPI and Android should be lower to avoid frame-stalls
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#define MUSIC_BUFFER_SIZE 4096*2 // PCM data buffer (short) - 16Kb (RPI)
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#define MUSIC_BUFFER_SIZE_SHORT 4096*2 // PCM data buffer (short) - 16Kb (RPI)
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#define MUSIC_BUFFER_SIZE_FLOAT 4096 // PCM data buffer (float) - 16Kb (RPI)
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#else
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#else
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// NOTE: On HTML5 (emscripten) this is allocated on heap, by default it's only 16MB!...just take care...
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// NOTE: On HTML5 (emscripten) this is allocated on heap, by default it's only 16MB!...just take care...
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#define MUSIC_BUFFER_SIZE 4096*8 // PCM data buffer (short) - 64Kb
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#define MUSIC_BUFFER_SIZE_SHORT 4096*8 // PCM data buffer (short) - 64Kb
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#define MUSIC_BUFFER_SIZE_FLOAT 4096*4 // PCM data buffer (float) - 64Kb
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#endif
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#endif
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//----------------------------------------------------------------------------------
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//----------------------------------------------------------------------------------
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@ -80,7 +82,7 @@ typedef struct Music {
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stb_vorbis *stream;
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stb_vorbis *stream;
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jar_xm_context_t *chipctx; // Stores jar_xm context
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jar_xm_context_t *chipctx; // Stores jar_xm context
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ALuint buffers[MUSIC_STREAM_BUFFERS];
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ALuint buffers[MAX_STREAM_BUFFERS];
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ALuint source;
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ALuint source;
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ALenum format;
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ALenum format;
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@ -96,12 +98,13 @@ typedef struct Music {
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// no more than 4 concurrent audio contexts in use. This is due to each active context being tied to
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// no more than 4 concurrent audio contexts in use. This is due to each active context being tied to
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// a dedicated mix channel. All audio is 32bit floating point in stereo.
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// a dedicated mix channel. All audio is 32bit floating point in stereo.
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typedef struct AudioContext_t {
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typedef struct AudioContext_t {
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unsigned short sampleRate; // default is 48000
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unsigned short sampleRate; // default is 48000
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unsigned char channels; // 1=mono,2=stereo
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unsigned char channels; // 1=mono,2=stereo
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unsigned char mixChannel; // 0-3 or mixA-mixD, each mix channel can receive up to one dedicated audio stream
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unsigned char mixChannel; // 0-3 or mixA-mixD, each mix channel can receive up to one dedicated audio stream
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ALenum alFormat; // openAL format specifier
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bool floatingPoint; // if false then the short datatype is used instead
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ALuint alSource; // openAL source
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ALenum alFormat; // openAL format specifier
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ALuint alBuffer[2]; // openAL sample buffer
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ALuint alSource; // openAL source
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ALuint alBuffer[MAX_STREAM_BUFFERS]; // openAL sample buffer
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} AudioContext_t;
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} AudioContext_t;
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#if defined(AUDIO_STANDALONE)
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#if defined(AUDIO_STANDALONE)
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@ -126,7 +129,7 @@ static void UnloadWave(Wave wave); // Unload wave data
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static bool BufferMusicStream(ALuint buffer); // Fill music buffers with data
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static bool BufferMusicStream(ALuint buffer); // Fill music buffers with data
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static void EmptyMusicStream(void); // Empty music buffers
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static void EmptyMusicStream(void); // Empty music buffers
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static void FillAlBufferWithSilence(AudioContext_t *ac, ALuint buffer);// fill buffer with zeros
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static unsigned short FillAlBufferWithSilence(AudioContext_t *context, ALuint buffer);// fill buffer with zeros, returns number processed
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static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // pass two arrays of the same legnth in
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static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // pass two arrays of the same legnth in
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static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // pass two arrays of same length in
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static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // pass two arrays of same length in
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@ -201,11 +204,10 @@ bool IsAudioDeviceReady(void)
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// Audio contexts are for outputing custom audio waveforms, This will shut down any other sound sources currently playing
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// Audio contexts are for outputing custom audio waveforms, This will shut down any other sound sources currently playing
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// 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.
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// 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.
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// exmple usage is InitAudioContext(48000, 0, 2); // mixchannel 1, 48khz, stereo
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// exmple usage is InitAudioContext(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
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// all samples are floating point by default
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AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint)
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AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels)
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{
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{
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if(mixChannel > MAX_AUDIO_CONTEXTS) return NULL;
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if(mixChannel >= MAX_AUDIO_CONTEXTS) return NULL;
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if(!IsAudioDeviceReady()) InitAudioDevice();
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if(!IsAudioDeviceReady()) InitAudioDevice();
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else StopMusicStream();
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else StopMusicStream();
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@ -214,13 +216,24 @@ AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChanne
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ac->sampleRate = sampleRate;
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ac->sampleRate = sampleRate;
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ac->channels = channels;
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ac->channels = channels;
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ac->mixChannel = mixChannel;
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ac->mixChannel = mixChannel;
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ac->floatingPoint = floatingPoint;
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mixChannelsActive_g[mixChannel] = ac;
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mixChannelsActive_g[mixChannel] = ac;
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// setup openAL format
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// setup openAL format
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if(channels == 1)
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if(channels == 1)
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ac->alFormat = AL_FORMAT_MONO_FLOAT32;
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{
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else
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if(floatingPoint)
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ac->alFormat = AL_FORMAT_STEREO_FLOAT32;
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ac->alFormat = AL_FORMAT_MONO_FLOAT32;
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else
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ac->alFormat = AL_FORMAT_MONO16;
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}
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else if(channels == 2)
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{
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if(floatingPoint)
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ac->alFormat = AL_FORMAT_STEREO_FLOAT32;
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else
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ac->alFormat = AL_FORMAT_STEREO16;
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}
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// Create an audio source
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// Create an audio source
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alGenSources(1, &ac->alSource);
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alGenSources(1, &ac->alSource);
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@ -230,16 +243,17 @@ AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChanne
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alSource3f(ac->alSource, AL_VELOCITY, 0, 0, 0);
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alSource3f(ac->alSource, AL_VELOCITY, 0, 0, 0);
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// Create Buffer
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// Create Buffer
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alGenBuffers(2, ac->alBuffer);
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alGenBuffers(MAX_STREAM_BUFFERS, ac->alBuffer);
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//fill buffers
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//fill buffers
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FillAlBufferWithSilence(ac, ac->alBuffer[0]);
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int x;
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FillAlBufferWithSilence(ac, ac->alBuffer[1]);
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for(x=0;x<MAX_STREAM_BUFFERS;x++)
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alSourceQueueBuffers(ac->alSource, 2, ac->alBuffer);
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FillAlBufferWithSilence(ac, ac->alBuffer[x]);
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alSourceQueueBuffers(ac->alSource, MAX_STREAM_BUFFERS, ac->alBuffer);
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alSourcei(ac->alSource, AL_LOOPING, AL_FALSE); // this could cause errors
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alSourcei(ac->alSource, AL_LOOPING, AL_FALSE); // this could cause errors
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alSourcePlay(ac->alSource);
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alSourcePlay(ac->alSource);
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return ac;
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return ac;
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}
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}
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return NULL;
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return NULL;
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@ -264,20 +278,22 @@ void CloseAudioContext(AudioContext ctx)
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//delete source and buffers
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//delete source and buffers
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alDeleteSources(1, &context->alSource);
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alDeleteSources(1, &context->alSource);
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alDeleteBuffers(2, context->alBuffer);
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alDeleteBuffers(MAX_STREAM_BUFFERS, context->alBuffer);
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mixChannelsActive_g[context->mixChannel] = NULL;
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mixChannelsActive_g[context->mixChannel] = NULL;
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free(context);
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free(context);
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ctx = NULL;
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ctx = NULL;
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}
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}
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}
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}
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// Pushes more audio data into context mix channel, if none are ever pushed then zeros are fed in
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// Pushes more audio data into context mix channel, if none are ever pushed then zeros are fed in.
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// Call "UpdateAudioContext(ctx, NULL, 0)" every game tick if you want to pause the audio
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// Call "UpdateAudioContext(ctx, NULL, 0)" every game tick if you want to pause the audio.
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// Returns number of floats that where processed
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// @Returns number of samples that where processed.
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unsigned short UpdateAudioContext(AudioContext ctx, float *data, unsigned short dataLength)
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// All data streams should be of a length that is evenly divisible by MUSIC_BUFFER_SIZE,
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// otherwise the remaining data will not be pushed.
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unsigned short UpdateAudioContext(AudioContext ctx, void *data, unsigned short numberElements)
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{
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{
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unsigned short numberProcessed = 0;
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unsigned short numberProcessed = 0;
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unsigned short numberRemaining = dataLength;
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unsigned short numberRemaining = numberElements;
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AudioContext_t *context = (AudioContext_t*)ctx;
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AudioContext_t *context = (AudioContext_t*)ctx;
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if (context && mixChannelsActive_g[context->mixChannel] == context)
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if (context && mixChannelsActive_g[context->mixChannel] == context)
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@ -288,44 +304,60 @@ unsigned short UpdateAudioContext(AudioContext ctx, float *data, unsigned short
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if(!processed) return 0;//nothing to process, queue is still full
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if(!processed) return 0;//nothing to process, queue is still full
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if (!data || !dataLength)// play silence
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if (!data || !numberElements)// play silence
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{
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while (processed > 0)
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while (processed > 0)
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{
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{
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alSourceUnqueueBuffers(context->alSource, 1, &buffer);
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alSourceUnqueueBuffers(context->alSource, 1, &buffer);
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FillAlBufferWithSilence(context, buffer);
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numberProcessed += FillAlBufferWithSilence(context, buffer);
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alSourceQueueBuffers(context->alSource, 1, &buffer);
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alSourceQueueBuffers(context->alSource, 1, &buffer);
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processed--;
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processed--;
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numberProcessed+=MUSIC_BUFFER_SIZE;
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}
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}
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}
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if(numberRemaining)// buffer data stream in increments of MUSIC_BUFFER_SIZE
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if(numberRemaining)// buffer data stream in increments of MUSIC_BUFFER_SIZE
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{
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while (processed > 0)
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while (processed > 0)
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{
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{
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alSourceUnqueueBuffers(context->alSource, 1, &buffer);
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if(context->floatingPoint && numberRemaining >= MUSIC_BUFFER_SIZE_FLOAT) // process float buffers
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if(numberRemaining >= MUSIC_BUFFER_SIZE)
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{
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{
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float pcm[MUSIC_BUFFER_SIZE];
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float *ptr = (float*)data;
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memcpy(pcm, &data[numberProcessed], MUSIC_BUFFER_SIZE);
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alSourceUnqueueBuffers(context->alSource, 1, &buffer);
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alBufferData(buffer, context->alFormat, pcm, MUSIC_BUFFER_SIZE*sizeof(float), context->sampleRate);
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alBufferData(buffer, context->alFormat, &ptr[numberProcessed], MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), context->sampleRate);
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alSourceQueueBuffers(context->alSource, 1, &buffer);
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alSourceQueueBuffers(context->alSource, 1, &buffer);
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numberProcessed+=MUSIC_BUFFER_SIZE;
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numberProcessed+=MUSIC_BUFFER_SIZE_FLOAT;
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numberRemaining-=MUSIC_BUFFER_SIZE;
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numberRemaining-=MUSIC_BUFFER_SIZE_FLOAT;
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}
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}
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else // less than MUSIC_BUFFER_SIZE number of samples left to buffer, the remaining data is padded with zeros
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else if(!context->floatingPoint && numberRemaining >= MUSIC_BUFFER_SIZE_SHORT) // process short buffers
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{
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{
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float pcm[MUSIC_BUFFER_SIZE] = {0.f}; // pad with zeros
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short *ptr = (short*)data;
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alSourceUnqueueBuffers(context->alSource, 1, &buffer);
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alBufferData(buffer, context->alFormat, &ptr[numberProcessed], MUSIC_BUFFER_SIZE_SHORT*sizeof(short), context->sampleRate);
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alSourceQueueBuffers(context->alSource, 1, &buffer);
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numberProcessed+=MUSIC_BUFFER_SIZE_SHORT;
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numberRemaining-=MUSIC_BUFFER_SIZE_SHORT;
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}
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}
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processed--;
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processed--;
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}
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}
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}
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}
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}
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return numberProcessed;
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return numberProcessed;
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}
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}
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// fill buffer with zeros
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// fill buffer with zeros, returns number processed
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static void FillAlBufferWithSilence(AudioContext_t *context, ALuint buffer)
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static unsigned short FillAlBufferWithSilence(AudioContext_t *context, ALuint buffer)
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{
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{
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float pcm[MUSIC_BUFFER_SIZE] = {0.f};
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if(context->floatingPoint){
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alBufferData(buffer, context->alFormat, pcm, MUSIC_BUFFER_SIZE*sizeof(float), context->sampleRate);
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float pcm[MUSIC_BUFFER_SIZE_FLOAT] = {0.f};
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alBufferData(buffer, context->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), context->sampleRate);
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return MUSIC_BUFFER_SIZE_FLOAT;
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}
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else
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{
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short pcm[MUSIC_BUFFER_SIZE_SHORT] = {0};
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alBufferData(buffer, context->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), context->sampleRate);
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return MUSIC_BUFFER_SIZE_SHORT;
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}
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}
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}
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// example usage:
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// example usage:
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// Fill music buffers with new data from music stream
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// Fill music buffers with new data from music stream
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static bool BufferMusicStream(ALuint buffer)
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static bool BufferMusicStream(ALuint buffer)
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{
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{
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short pcm[MUSIC_BUFFER_SIZE];
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short pcm[MUSIC_BUFFER_SIZE_SHORT];
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int size = 0; // Total size of data steamed (in bytes)
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int size = 0; // Total size of data steamed (in bytes)
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int streamedBytes = 0; // samples of data obtained, channels are not included in calculation
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int streamedBytes = 0; // samples of data obtained, channels are not included in calculation
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{
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{
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if (currentMusic.chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
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if (currentMusic.chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
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{
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{
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int readlen = MUSIC_BUFFER_SIZE / 2;
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int readlen = MUSIC_BUFFER_SIZE_SHORT / 2;
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jar_xm_generate_samples_16bit(currentMusic.chipctx, pcm, readlen); // reads 2*readlen shorts and moves them to buffer+size memory location
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jar_xm_generate_samples_16bit(currentMusic.chipctx, pcm, readlen); // reads 2*readlen shorts and moves them to buffer+size memory location
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size += readlen * currentMusic.channels; // Not sure if this is what it needs
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size += readlen * currentMusic.channels; // Not sure if this is what it needs
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}
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}
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else
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else
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{
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{
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while (size < MUSIC_BUFFER_SIZE)
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while (size < MUSIC_BUFFER_SIZE_SHORT)
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{
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{
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streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic.stream, currentMusic.channels, pcm + size, MUSIC_BUFFER_SIZE - size);
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streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic.stream, currentMusic.channels, pcm + size, MUSIC_BUFFER_SIZE_SHORT - size);
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if (streamedBytes > 0) size += (streamedBytes*currentMusic.channels);
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if (streamedBytes > 0) size += (streamedBytes*currentMusic.channels);
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else break;
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else break;
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}
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}
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@ -84,11 +84,10 @@ bool IsAudioDeviceReady(void); // True if call
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// Audio contexts are for outputing custom audio waveforms, This will shut down any other sound sources currently playing
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// Audio contexts are for outputing custom audio waveforms, This will shut down any other sound sources currently playing
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// 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.
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// 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.
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// exmple usage is InitAudioContext(48000, 0, 2); // mixchannel 1, 48khz, stereo
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// exmple usage is InitAudioContext(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
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// all samples are floating point by default
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AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint);
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AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels);
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void CloseAudioContext(AudioContext ctx); // Frees audio context
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void CloseAudioContext(AudioContext ctx); // Frees audio context
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unsigned short UpdateAudioContext(AudioContext ctx, float *data, unsigned short dataLength); // Pushes more audio data into context mix channel, if NULL is passed to data then zeros are played
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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
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Sound LoadSound(char *fileName); // Load sound to memory
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Sound LoadSound(char *fileName); // Load sound to memory
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Sound LoadSoundFromWave(Wave wave); // Load sound to memory from wave data
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Sound LoadSoundFromWave(Wave wave); // Load sound to memory from wave data
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@ -870,11 +870,10 @@ bool IsAudioDeviceReady(void); // True if call
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// Audio contexts are for outputing custom audio waveforms, This will shut down any other sound sources currently playing
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// Audio contexts are for outputing custom audio waveforms, This will shut down any other sound sources currently playing
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// 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.
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// 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.
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// exmple usage is InitAudioContext(48000, 0, 2); // mixchannel 1, 48khz, stereo
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// exmple usage is InitAudioContext(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point
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// all samples are floating point by default
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AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint);
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AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels);
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void CloseAudioContext(AudioContext ctx); // Frees audio context
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void CloseAudioContext(AudioContext ctx); // Frees audio context
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unsigned short UpdateAudioContext(AudioContext ctx, float *data, unsigned short dataLength); // Pushes more audio data into context mix channel, if NULL is passed to data then zeros are played
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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
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Sound LoadSound(char *fileName); // Load sound to memory
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Sound LoadSound(char *fileName); // Load sound to memory
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Sound LoadSoundFromWave(Wave wave); // Load sound to memory from wave data
|
Sound LoadSoundFromWave(Wave wave); // Load sound to memory from wave data
|
||||||
|
|
Loading…
Add table
Add a link
Reference in a new issue