mirror of
https://github.com/raysan5/raylib.git
synced 2025-12-25 10:22:33 -05:00
Lots of changes, most of them under testing-review
Added a Tracing/Log system Added OGG stream music support (DOESN'T WORK) Added Compressed textures support * This update is probably very buggy...
This commit is contained in:
raysan5
613
src/audio.c
613
src/audio.c
@ -37,7 +37,7 @@
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#include "utils.h" // rRES data decompression utility function
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//#include "stb_vorbis.h" // TODO: OGG loading functions
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//#include "stb_vorbis.h" // OGG loading functions
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//----------------------------------------------------------------------------------
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// Defines and Macros
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@ -47,6 +47,29 @@
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//----------------------------------------------------------------------------------
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// Types and Structures Definition
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//----------------------------------------------------------------------------------
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// Sound source type (all file loaded in memory)
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/*
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struct Sound {
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unsigned int source;
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unsigned int buffer;
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};
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// Music type (file streamming from memory)
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// NOTE: Anything longer than ~10 seconds should be Music...
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struct Music {
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stb_vorbis* stream;
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stb_vorbis_info info;
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ALuint id;
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ALuint buffers[2];
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ALuint source;
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ALenum format;
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int bufferSize;
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int totalSamplesLeft;
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bool loop;
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};
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*/
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// Wave file data
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typedef struct Wave {
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@ -60,7 +83,8 @@ typedef struct Wave {
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//----------------------------------------------------------------------------------
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// Global Variables Definition
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//----------------------------------------------------------------------------------
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// Nop...
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static bool musicIsPlaying;
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static Music *currentMusic;
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//----------------------------------------------------------------------------------
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// Module specific Functions Declaration
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@ -68,6 +92,10 @@ typedef struct Wave {
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static Wave LoadWAV(char *fileName);
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static void UnloadWAV(Wave wave);
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//static Ogg LoadOGG(char *fileName);
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static bool MusicStream(Music music, ALuint buffer);
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extern bool MusicStreamUpdate();
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extern void PlayCurrentMusic();
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//----------------------------------------------------------------------------------
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// Module Functions Definition - Window and OpenGL Context Functions
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@ -79,30 +107,27 @@ void InitAudioDevice()
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// Open and initialize a device with default settings
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ALCdevice *device = alcOpenDevice(NULL);
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if(!device)
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{
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fprintf(stderr, "Could not open a device!\n");
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exit(1);
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}
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if(!device) TraceLog(ERROR, "Could not open audio device");
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ALCcontext *context = alcCreateContext(device, NULL);
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if(context == NULL || alcMakeContextCurrent(context) == ALC_FALSE)
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{
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if(context != NULL) alcDestroyContext(context);
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if(context != NULL) alcDestroyContext(context);
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alcCloseDevice(device);
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fprintf(stderr, "Could not set a context!\n");
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exit(1);
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TraceLog(ERROR, "Could not setup audio context");
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}
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printf("Opened \"%s\"\n", alcGetString(device, ALC_DEVICE_SPECIFIER));
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TraceLog(INFO, "Audio device and context initialized: %s\n", alcGetString(device, ALC_DEVICE_SPECIFIER));
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// Listener definition (just for 2D)
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alListener3f(AL_POSITION, 0, 0, 0);
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alListener3f(AL_VELOCITY, 0, 0, 0);
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alListener3f(AL_ORIENTATION, 0, 0, -1);
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musicIsPlaying = false;
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}
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// Close the audio device for the current context, and destroys the context
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@ -111,7 +136,7 @@ void CloseAudioDevice()
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ALCdevice *device;
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ALCcontext *context = alcGetCurrentContext();
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if (context == NULL) return;
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if (context == NULL) TraceLog(WARNING, "Could not get current audio context for closing");
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device = alcGetContextsDevice(context);
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@ -169,10 +194,8 @@ Sound LoadSound(char *fileName)
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// Unallocate WAV data
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UnloadWAV(wave);
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printf("Sample rate: %i\n", wave.sampleRate);
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printf("Channels: %i\n", wave.channels);
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printf("Audio file loaded...!\n");
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TraceLog(INFO, "[%s] Sound file loaded successfully", fileName);
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TraceLog(INFO, "[%s] Sample rate: %i - Channels: %i", fileName, wave.sampleRate, wave.channels);
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sound.source = source;
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sound.buffer = buffer;
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@ -196,139 +219,136 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
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FILE *rresFile = fopen(rresName, "rb");
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if (!rresFile) printf("Error opening raylib Resource file\n");
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// Read rres file (basic file check - id)
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fread(&id[0], sizeof(char), 1, rresFile);
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fread(&id[1], sizeof(char), 1, rresFile);
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fread(&id[2], sizeof(char), 1, rresFile);
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fread(&id[3], sizeof(char), 1, rresFile);
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fread(&version, sizeof(char), 1, rresFile);
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fread(&useless, sizeof(char), 1, rresFile);
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if ((id[0] != 'r') && (id[1] != 'R') && (id[2] != 'E') &&(id[3] != 'S'))
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if (!rresFile) TraceLog(WARNING, "[%s] Could not open raylib resource file", rresName);
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else
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{
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printf("This is not a valid raylib Resource file!\n");
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exit(1);
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}
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// Read number of resources embedded
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fread(&numRes, sizeof(short), 1, rresFile);
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for (int i = 0; i < numRes; i++)
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{
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fread(&infoHeader, sizeof(ResInfoHeader), 1, rresFile);
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// Read rres file (basic file check - id)
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fread(&id[0], sizeof(char), 1, rresFile);
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fread(&id[1], sizeof(char), 1, rresFile);
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fread(&id[2], sizeof(char), 1, rresFile);
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fread(&id[3], sizeof(char), 1, rresFile);
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fread(&version, sizeof(char), 1, rresFile);
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fread(&useless, sizeof(char), 1, rresFile);
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if (infoHeader.id == resId)
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if ((id[0] != 'r') && (id[1] != 'R') && (id[2] != 'E') &&(id[3] != 'S'))
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{
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found = true;
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// Check data is of valid SOUND type
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if (infoHeader.type == 1) // SOUND data type
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{
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// TODO: Check data compression type
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// NOTE: We suppose compression type 2 (DEFLATE - default)
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// Reading SOUND parameters
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Wave wave;
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short sampleRate, bps;
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char channels, reserved;
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fread(&sampleRate, sizeof(short), 1, rresFile); // Sample rate (frequency)
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fread(&bps, sizeof(short), 1, rresFile); // Bits per sample
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fread(&channels, 1, 1, rresFile); // Channels (1 - mono, 2 - stereo)
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fread(&reserved, 1, 1, rresFile); // <reserved>
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printf("Sample rate: %i\n", (int)sampleRate);
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printf("Bits per sample: %i\n", (int)bps);
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printf("Channels: %i\n", (int)channels);
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wave.sampleRate = sampleRate;
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wave.dataSize = infoHeader.srcSize;
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wave.bitsPerSample = bps;
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wave.channels = (short)channels;
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unsigned char *data = malloc(infoHeader.size);
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fread(data, infoHeader.size, 1, rresFile);
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wave.data = DecompressData(data, infoHeader.size, infoHeader.srcSize);
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free(data);
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// Convert wave to Sound (OpenAL)
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ALenum format = 0;
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// The OpenAL format is worked out by looking at the number of channels and the bits per sample
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if (wave.channels == 1)
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{
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if (wave.bitsPerSample == 8 ) format = AL_FORMAT_MONO8;
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else if (wave.bitsPerSample == 16) format = AL_FORMAT_MONO16;
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}
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else if (wave.channels == 2)
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{
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if (wave.bitsPerSample == 8 ) format = AL_FORMAT_STEREO8;
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else if (wave.bitsPerSample == 16) format = AL_FORMAT_STEREO16;
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}
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// Create an audio source
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ALuint source;
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alGenSources(1, &source); // Generate pointer to audio source
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alSourcef(source, AL_PITCH, 1);
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alSourcef(source, AL_GAIN, 1);
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alSource3f(source, AL_POSITION, 0, 0, 0);
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alSource3f(source, AL_VELOCITY, 0, 0, 0);
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alSourcei(source, AL_LOOPING, AL_FALSE);
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// Convert loaded data to OpenAL buffer
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//----------------------------------------
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ALuint buffer;
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alGenBuffers(1, &buffer); // Generate pointer to buffer
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// Upload sound data to buffer
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alBufferData(buffer, format, (void*)wave.data, wave.dataSize, wave.sampleRate);
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// Attach sound buffer to source
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alSourcei(source, AL_BUFFER, buffer);
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// Unallocate WAV data
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UnloadWAV(wave);
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printf("Audio file loaded...!\n");
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sound.source = source;
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sound.buffer = buffer;
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}
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else
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{
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printf("Required resource do not seem to be a valid IMAGE resource\n");
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exit(2);
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}
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TraceLog(WARNING, "[%s] This is not a valid raylib resource file", rresName);
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}
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else
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{
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// Depending on type, skip the right amount of parameters
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switch (infoHeader.type)
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{
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case 0: fseek(rresFile, 6, SEEK_CUR); break; // IMAGE: Jump 6 bytes of parameters
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case 1: fseek(rresFile, 6, SEEK_CUR); break; // SOUND: Jump 6 bytes of parameters
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case 2: fseek(rresFile, 5, SEEK_CUR); break; // MODEL: Jump 5 bytes of parameters (TODO: Review)
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case 3: break; // TEXT: No parameters
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case 4: break; // RAW: No parameters
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default: break;
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}
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// Read number of resources embedded
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fread(&numRes, sizeof(short), 1, rresFile);
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// Jump DATA to read next infoHeader
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fseek(rresFile, infoHeader.size, SEEK_CUR);
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}
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for (int i = 0; i < numRes; i++)
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{
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fread(&infoHeader, sizeof(ResInfoHeader), 1, rresFile);
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if (infoHeader.id == resId)
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{
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found = true;
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// Check data is of valid SOUND type
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if (infoHeader.type == 1) // SOUND data type
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{
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// TODO: Check data compression type
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// NOTE: We suppose compression type 2 (DEFLATE - default)
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// Reading SOUND parameters
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Wave wave;
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short sampleRate, bps;
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char channels, reserved;
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fread(&sampleRate, sizeof(short), 1, rresFile); // Sample rate (frequency)
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fread(&bps, sizeof(short), 1, rresFile); // Bits per sample
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fread(&channels, 1, 1, rresFile); // Channels (1 - mono, 2 - stereo)
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fread(&reserved, 1, 1, rresFile); // <reserved>
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wave.sampleRate = sampleRate;
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wave.dataSize = infoHeader.srcSize;
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wave.bitsPerSample = bps;
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wave.channels = (short)channels;
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unsigned char *data = malloc(infoHeader.size);
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fread(data, infoHeader.size, 1, rresFile);
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wave.data = DecompressData(data, infoHeader.size, infoHeader.srcSize);
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free(data);
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// Convert wave to Sound (OpenAL)
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ALenum format = 0;
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// The OpenAL format is worked out by looking at the number of channels and the bits per sample
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if (wave.channels == 1)
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{
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if (wave.bitsPerSample == 8 ) format = AL_FORMAT_MONO8;
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else if (wave.bitsPerSample == 16) format = AL_FORMAT_MONO16;
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}
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else if (wave.channels == 2)
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{
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if (wave.bitsPerSample == 8 ) format = AL_FORMAT_STEREO8;
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else if (wave.bitsPerSample == 16) format = AL_FORMAT_STEREO16;
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}
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// Create an audio source
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ALuint source;
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alGenSources(1, &source); // Generate pointer to audio source
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alSourcef(source, AL_PITCH, 1);
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alSourcef(source, AL_GAIN, 1);
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alSource3f(source, AL_POSITION, 0, 0, 0);
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alSource3f(source, AL_VELOCITY, 0, 0, 0);
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alSourcei(source, AL_LOOPING, AL_FALSE);
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// Convert loaded data to OpenAL buffer
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//----------------------------------------
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ALuint buffer;
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alGenBuffers(1, &buffer); // Generate pointer to buffer
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// Upload sound data to buffer
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alBufferData(buffer, format, (void*)wave.data, wave.dataSize, wave.sampleRate);
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// Attach sound buffer to source
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alSourcei(source, AL_BUFFER, buffer);
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// Unallocate WAV data
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UnloadWAV(wave);
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TraceLog(INFO, "[%s] Sound loaded successfully from resource, sample rate: %i", rresName, (int)sampleRate);
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sound.source = source;
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sound.buffer = buffer;
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}
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else
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{
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TraceLog(WARNING, "[%s] Required resource do not seem to be a valid SOUND resource", rresName);
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}
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}
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else
|
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{
|
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// Depending on type, skip the right amount of parameters
|
||||
switch (infoHeader.type)
|
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{
|
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case 0: fseek(rresFile, 6, SEEK_CUR); break; // IMAGE: Jump 6 bytes of parameters
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case 1: fseek(rresFile, 6, SEEK_CUR); break; // SOUND: Jump 6 bytes of parameters
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case 2: fseek(rresFile, 5, SEEK_CUR); break; // MODEL: Jump 5 bytes of parameters (TODO: Review)
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case 3: break; // TEXT: No parameters
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case 4: break; // RAW: No parameters
|
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default: break;
|
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}
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// Jump DATA to read next infoHeader
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fseek(rresFile, infoHeader.size, SEEK_CUR);
|
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}
|
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}
|
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}
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fclose(rresFile);
|
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}
|
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|
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fclose(rresFile);
|
||||
|
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if (!found) printf("Required resource id could not be found in the raylib Resource file!\n");
|
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if (!found) TraceLog(WARNING, "[%s] Required resource id [%i] could not be found in the raylib resource file", rresName, resId);
|
||||
|
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return sound;
|
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}
|
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@ -345,7 +365,7 @@ void PlaySound(Sound sound)
|
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{
|
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alSourcePlay(sound.source); // Play the sound
|
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|
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printf("Playing sound!\n");
|
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TraceLog(INFO, "Playing sound");
|
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|
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// Find the current position of the sound being played
|
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// NOTE: Only work when the entire file is in a single buffer
|
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@ -390,7 +410,7 @@ void StopSound(Sound sound)
|
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}
|
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|
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// Check if a sound is playing
|
||||
bool IsPlaying(Sound sound)
|
||||
bool SoundIsPlaying(Sound sound)
|
||||
{
|
||||
bool playing = false;
|
||||
ALint state;
|
||||
@ -401,6 +421,16 @@ bool IsPlaying(Sound sound)
|
||||
return playing;
|
||||
}
|
||||
|
||||
// Check if music is playing
|
||||
bool MusicIsPlaying(Music music)
|
||||
{
|
||||
ALenum state;
|
||||
|
||||
alGetSourcei(music.source, AL_SOURCE_STATE, &state);
|
||||
|
||||
return (state == AL_PLAYING);
|
||||
}
|
||||
|
||||
// Set volume for a sound
|
||||
void SetVolume(Sound sound, float volume)
|
||||
{
|
||||
@ -450,61 +480,65 @@ static Wave LoadWAV(char *fileName)
|
||||
|
||||
if (!wavFile)
|
||||
{
|
||||
printf("Could not open WAV file.\n");
|
||||
exit(1);
|
||||
TraceLog(WARNING, "[%s] Could not open WAV file", fileName);
|
||||
}
|
||||
|
||||
// Read in the first chunk into the struct
|
||||
fread(&riffHeader, sizeof(RiffHeader), 1, wavFile);
|
||||
|
||||
// Check for RIFF and WAVE tags
|
||||
if ((riffHeader.chunkID[0] != 'R' ||
|
||||
riffHeader.chunkID[1] != 'I' ||
|
||||
riffHeader.chunkID[2] != 'F' ||
|
||||
riffHeader.chunkID[3] != 'F') ||
|
||||
(riffHeader.format[0] != 'W' ||
|
||||
riffHeader.format[1] != 'A' ||
|
||||
riffHeader.format[2] != 'V' ||
|
||||
riffHeader.format[3] != 'E'))
|
||||
printf("Invalid RIFF or WAVE Header");
|
||||
|
||||
// Read in the 2nd chunk for the wave info
|
||||
fread(&waveFormat, sizeof(WaveFormat), 1, wavFile);
|
||||
|
||||
// Check for fmt tag
|
||||
if (waveFormat.subChunkID[0] != 'f' ||
|
||||
waveFormat.subChunkID[1] != 'm' ||
|
||||
waveFormat.subChunkID[2] != 't' ||
|
||||
waveFormat.subChunkID[3] != ' ')
|
||||
printf("Invalid Wave Format");
|
||||
|
||||
// Check for extra parameters;
|
||||
if (waveFormat.subChunkSize > 16)
|
||||
fseek(wavFile, sizeof(short), SEEK_CUR);
|
||||
|
||||
// Read in the the last byte of data before the sound file
|
||||
fread(&waveData, sizeof(WaveData), 1, wavFile);
|
||||
|
||||
// Check for data tag
|
||||
if (waveData.subChunkID[0] != 'd' ||
|
||||
waveData.subChunkID[1] != 'a' ||
|
||||
waveData.subChunkID[2] != 't' ||
|
||||
waveData.subChunkID[3] != 'a')
|
||||
printf("Invalid data header");
|
||||
|
||||
// Allocate memory for data
|
||||
wave.data = (unsigned char *)malloc(sizeof(unsigned char) * waveData.subChunkSize);
|
||||
|
||||
// Read in the sound data into the soundData variable
|
||||
fread(wave.data, waveData.subChunkSize, 1, wavFile);
|
||||
|
||||
// Now we set the variables that we need later
|
||||
wave.dataSize = waveData.subChunkSize;
|
||||
wave.sampleRate = waveFormat.sampleRate;
|
||||
wave.channels = waveFormat.numChannels;
|
||||
wave.bitsPerSample = waveFormat.bitsPerSample;
|
||||
else
|
||||
{
|
||||
// Read in the first chunk into the struct
|
||||
fread(&riffHeader, sizeof(RiffHeader), 1, wavFile);
|
||||
|
||||
// Check for RIFF and WAVE tags
|
||||
if (((riffHeader.chunkID[0] != 'R') || (riffHeader.chunkID[1] != 'I') || (riffHeader.chunkID[2] != 'F') || (riffHeader.chunkID[3] != 'F')) ||
|
||||
((riffHeader.format[0] != 'W') || (riffHeader.format[1] != 'A') || (riffHeader.format[2] != 'V') || (riffHeader.format[3] != 'E')))
|
||||
{
|
||||
TraceLog(WARNING, "[%s] Invalid RIFF or WAVE Header", fileName);
|
||||
}
|
||||
else
|
||||
{
|
||||
// Read in the 2nd chunk for the wave info
|
||||
fread(&waveFormat, sizeof(WaveFormat), 1, wavFile);
|
||||
|
||||
// Check for fmt tag
|
||||
if ((waveFormat.subChunkID[0] != 'f') || (waveFormat.subChunkID[1] != 'm') ||
|
||||
(waveFormat.subChunkID[2] != 't') || (waveFormat.subChunkID[3] != ' '))
|
||||
{
|
||||
TraceLog(WARNING, "[%s] Invalid Wave format", fileName);
|
||||
}
|
||||
else
|
||||
{
|
||||
// Check for extra parameters;
|
||||
if (waveFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR);
|
||||
|
||||
// Read in the the last byte of data before the sound file
|
||||
fread(&waveData, sizeof(WaveData), 1, wavFile);
|
||||
|
||||
// Check for data tag
|
||||
if ((waveData.subChunkID[0] != 'd') || (waveData.subChunkID[1] != 'a') ||
|
||||
(waveData.subChunkID[2] != 't') || (waveData.subChunkID[3] != 'a'))
|
||||
{
|
||||
TraceLog(WARNING, "[%s] Invalid data header", fileName);
|
||||
}
|
||||
else
|
||||
{
|
||||
// Allocate memory for data
|
||||
wave.data = (unsigned char *)malloc(sizeof(unsigned char) * waveData.subChunkSize);
|
||||
|
||||
// Read in the sound data into the soundData variable
|
||||
fread(wave.data, waveData.subChunkSize, 1, wavFile);
|
||||
|
||||
// Now we set the variables that we need later
|
||||
wave.dataSize = waveData.subChunkSize;
|
||||
wave.sampleRate = waveFormat.sampleRate;
|
||||
wave.channels = waveFormat.numChannels;
|
||||
wave.bitsPerSample = waveFormat.bitsPerSample;
|
||||
|
||||
TraceLog(INFO, "[%s] Wave file loaded successfully", fileName);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
fclose(wavFile);
|
||||
fclose(wavFile);
|
||||
}
|
||||
|
||||
return wave;
|
||||
}
|
||||
@ -516,5 +550,192 @@ static void UnloadWAV(Wave wave)
|
||||
}
|
||||
|
||||
// TODO: Ogg data loading
|
||||
//static Ogg LoadOGG(char *fileName) { }
|
||||
Music LoadMusic(char *fileName)
|
||||
{
|
||||
Music music;
|
||||
|
||||
// Open audio stream
|
||||
music.stream = stb_vorbis_open_filename(fileName, NULL, NULL);
|
||||
|
||||
if (music.stream == NULL) TraceLog(WARNING, "Could not open ogg audio file");
|
||||
else
|
||||
{
|
||||
// Get file info
|
||||
music.info = stb_vorbis_get_info(music.stream);
|
||||
|
||||
printf("Ogg sample rate: %i\n", music.info.sample_rate);
|
||||
printf("Ogg channels: %i\n", music.info.channels);
|
||||
printf("Temp memory required: %i\n", music.info.temp_memory_required);
|
||||
|
||||
if (music.info.channels == 2) music.format = AL_FORMAT_STEREO16;
|
||||
else music.format = AL_FORMAT_MONO16;
|
||||
|
||||
music.bufferSize = 4096*8;
|
||||
music.loop = true; // We loop by default
|
||||
|
||||
// Create an audio source
|
||||
alGenSources(1, &music.source); // Generate pointer to audio source
|
||||
|
||||
alSourcef(music.source, AL_PITCH, 1);
|
||||
alSourcef(music.source, AL_GAIN, 1);
|
||||
alSource3f(music.source, AL_POSITION, 0, 0, 0);
|
||||
alSource3f(music.source, AL_VELOCITY, 0, 0, 0);
|
||||
alSourcei(music.source, AL_LOOPING, AL_TRUE); // We loop by default
|
||||
|
||||
// Convert loaded data to OpenAL buffers
|
||||
alGenBuffers(2, music.buffers);
|
||||
/*
|
||||
if (!MusicStream(music, music.buffers[0])) exit(1);
|
||||
if (!MusicStream(music, music.buffers[1])) exit(1);
|
||||
|
||||
alSourceQueueBuffers(music.source, 2, music.buffers);
|
||||
|
||||
PlayMusic(music);
|
||||
*/
|
||||
music.totalSamplesLeft = stb_vorbis_stream_length_in_samples(music.stream) * music.info.channels;
|
||||
|
||||
currentMusic = &music;
|
||||
}
|
||||
|
||||
return music;
|
||||
}
|
||||
|
||||
void UnloadMusic(Music music)
|
||||
{
|
||||
StopMusic(music);
|
||||
|
||||
alDeleteSources(1, &music.source);
|
||||
alDeleteBuffers(2, music.buffers);
|
||||
|
||||
stb_vorbis_close(music.stream);
|
||||
}
|
||||
|
||||
void PlayMusic(Music music)
|
||||
{
|
||||
//if (MusicIsPlaying(music)) return true;
|
||||
|
||||
if (!MusicStream(music, music.buffers[0])) TraceLog(WARNING, "MusicStream returned 0");
|
||||
if (!MusicStream(music, music.buffers[1])) TraceLog(WARNING, "MusicStream returned 0");
|
||||
|
||||
alSourceQueueBuffers(music.source, 2, music.buffers);
|
||||
alSourcePlay(music.source);
|
||||
|
||||
TraceLog(INFO, "Playing music");
|
||||
}
|
||||
|
||||
extern void PlayCurrentMusic()
|
||||
{
|
||||
if (!MusicStream(*currentMusic, currentMusic->buffers[0])) TraceLog(WARNING, "MusicStream returned 0");
|
||||
if (!MusicStream(*currentMusic, currentMusic->buffers[1])) TraceLog(WARNING, "MusicStream returned 0");
|
||||
|
||||
alSourceQueueBuffers(currentMusic->source, 2, currentMusic->buffers);
|
||||
alSourcePlay(currentMusic->source);
|
||||
}
|
||||
|
||||
// Stop reproducing music
|
||||
void StopMusic(Music music)
|
||||
{
|
||||
alSourceStop(music.source);
|
||||
|
||||
musicIsPlaying = false;
|
||||
}
|
||||
|
||||
static bool MusicStream(Music music, ALuint buffer)
|
||||
{
|
||||
//Uncomment this to avoid VLAs
|
||||
//#define BUFFER_SIZE 4096*32
|
||||
#ifndef BUFFER_SIZE//VLAs ftw
|
||||
#define BUFFER_SIZE (music.bufferSize)
|
||||
#endif
|
||||
ALshort pcm[BUFFER_SIZE];
|
||||
|
||||
int size = 0;
|
||||
int result = 0;
|
||||
|
||||
while (size < BUFFER_SIZE)
|
||||
{
|
||||
result = stb_vorbis_get_samples_short_interleaved(music.stream, music.info.channels, pcm+size, BUFFER_SIZE-size);
|
||||
|
||||
if (result > 0) size += (result*music.info.channels);
|
||||
else break;
|
||||
}
|
||||
|
||||
if (size == 0) return false;
|
||||
|
||||
alBufferData(buffer, music.format, pcm, size*sizeof(ALshort), music.info.sample_rate);
|
||||
|
||||
music.totalSamplesLeft -= size;
|
||||
|
||||
#undef BUFFER_SIZE
|
||||
|
||||
return true;
|
||||
}
|
||||
/*
|
||||
extern bool MusicStreamUpdate()
|
||||
{
|
||||
ALint processed = 0;
|
||||
|
||||
alGetSourcei(currentMusic->source, AL_BUFFERS_PROCESSED, &processed);
|
||||
|
||||
while (processed--)
|
||||
{
|
||||
ALuint buffer = 0;
|
||||
|
||||
alSourceUnqueueBuffers(currentMusic->source, 1, &buffer);
|
||||
|
||||
if (!MusicStream(*currentMusic, buffer))
|
||||
{
|
||||
bool shouldExit = true;
|
||||
|
||||
if (currentMusic->loop)
|
||||
{
|
||||
stb_vorbis_seek_start(currentMusic->stream);
|
||||
currentMusic->totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic->stream) * currentMusic->info.channels;
|
||||
|
||||
shouldExit = !MusicStream(*currentMusic, buffer);
|
||||
}
|
||||
|
||||
if (shouldExit) return false;
|
||||
}
|
||||
|
||||
alSourceQueueBuffers(currentMusic->source, 1, &buffer);
|
||||
}
|
||||
|
||||
return true;
|
||||
}
|
||||
*/
|
||||
extern bool MusicStreamUpdate()
|
||||
{
|
||||
int processed;
|
||||
bool active = true;
|
||||
|
||||
alGetSourcei(currentMusic->source, AL_BUFFERS_PROCESSED, &processed);
|
||||
|
||||
printf("Data processed: %i\n", processed);
|
||||
|
||||
while (processed--)
|
||||
{
|
||||
ALuint buffer = 0;
|
||||
|
||||
alSourceUnqueueBuffers(currentMusic->source, 1, &buffer);
|
||||
|
||||
active = MusicStream(*currentMusic, buffer);
|
||||
|
||||
alSourceQueueBuffers(currentMusic->source, 1, &buffer);
|
||||
}
|
||||
|
||||
return active;
|
||||
}
|
||||
|
||||
void MusicStreamEmpty()
|
||||
{
|
||||
int queued;
|
||||
|
||||
alGetSourcei(currentMusic->source, AL_BUFFERS_QUEUED, &queued);
|
||||
|
||||
while(queued--)
|
||||
{
|
||||
ALuint buffer;
|
||||
alSourceUnqueueBuffers(currentMusic->source, 1, &buffer);
|
||||
}
|
||||
}
|
||||
Reference in New Issue
Block a user