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raylib 1.1
View CHANGELOG for a detailed list of changes
This commit is contained in:
raysan5
662
src/audio.c
662
src/audio.c
@ -6,7 +6,7 @@
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*
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* Uses external lib:
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* OpenAL - Audio device management lib
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* TODO: stb_vorbis - Ogg audio files loading
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* stb_vorbis - Ogg audio files loading
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*
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* Copyright (c) 2013 Ramon Santamaria (Ray San - raysan@raysanweb.com)
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*
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@ -32,50 +32,45 @@
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#include <AL/al.h> // OpenAL basic header
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#include <AL/alc.h> // OpenAL context header (like OpenGL, OpenAL requires a context to work)
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#include <stdlib.h> // To use exit() function
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#include <stdlib.h> // Declares malloc() and free() for memory management
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#include <string.h> // Required for strcmp()
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#include <stdio.h> // Used for .WAV loading
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#include "utils.h" // rRES data decompression utility function
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//#include "stb_vorbis.h" // 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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//----------------------------------------------------------------------------------
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// Nop...
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#define MUSIC_STREAM_BUFFERS 2
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#define MUSIC_BUFFER_SIZE 4096*8 //4096*32
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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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// Music type (file streaming from memory)
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// NOTE: Anything longer than ~10 seconds should be streamed...
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typedef struct Music {
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stb_vorbis *stream;
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ALuint id;
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ALuint buffers[2];
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ALuint buffers[MUSIC_STREAM_BUFFERS];
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ALuint source;
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ALenum format;
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int bufferSize;
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int channels;
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int sampleRate;
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int totalSamplesLeft;
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bool loop;
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};
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*/
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} Music;
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// Wave file data
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typedef struct Wave {
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unsigned char *data; // Buffer data pointer
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void *data; // Buffer data pointer
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unsigned int dataSize; // Data size in bytes
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unsigned int sampleRate;
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unsigned int dataSize;
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short bitsPerSample;
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short channels;
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} Wave;
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@ -83,22 +78,23 @@ typedef struct Wave {
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//----------------------------------------------------------------------------------
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// Global Variables Definition
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//----------------------------------------------------------------------------------
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static bool musicIsPlaying;
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static Music *currentMusic;
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bool musicEnabled = false;
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static Music currentMusic; // Current music loaded
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// NOTE: Only one music file playing at a time
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//----------------------------------------------------------------------------------
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// Module specific Functions Declaration
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//----------------------------------------------------------------------------------
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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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static Wave LoadWAV(const char *fileName);
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static Wave LoadOGG(char *fileName);
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static void UnloadWave(Wave wave);
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extern bool MusicStreamUpdate();
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extern void PlayCurrentMusic();
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static bool BufferMusicStream(ALuint buffer); // Fill music buffers with data
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static void EmptyMusicStream(); // Empty music buffers
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extern void UpdateMusicStream(); // Updates buffers (refill) for music streaming
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//----------------------------------------------------------------------------------
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// Module Functions Definition - Window and OpenGL Context Functions
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// Module Functions Definition - Audio Device initialization and Closing
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//----------------------------------------------------------------------------------
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// Initialize audio device and context
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@ -126,13 +122,13 @@ void InitAudioDevice()
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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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void CloseAudioDevice()
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{
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StopMusicStream(); // Stop music streaming and close current stream
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ALCdevice *device;
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ALCcontext *context = alcGetCurrentContext();
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@ -145,61 +141,71 @@ void CloseAudioDevice()
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alcCloseDevice(device);
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}
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//----------------------------------------------------------------------------------
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// Module Functions Definition - Sounds loading and playing (.WAV)
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//----------------------------------------------------------------------------------
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// Load sound to memory
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Sound LoadSound(char *fileName)
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{
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Sound sound;
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Wave wave;
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// NOTE: The entire file is loaded to memory to play it all at once (no-streaming)
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// WAV file loading
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// NOTE: Buffer space is allocated inside LoadWAV, Wave must be freed
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Wave wave = LoadWAV(fileName);
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// Audio file loading
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// NOTE: Buffer space is allocated inside function, Wave must be freed
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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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if (strcmp(GetExtension(fileName),"wav") == 0) wave = LoadWAV(fileName);
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else if (strcmp(GetExtension(fileName),"ogg") == 0) wave = LoadOGG(fileName);
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else TraceLog(WARNING, "[%s] Sound extension not recognized, it can't be loaded", fileName);
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if (wave.data != NULL)
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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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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, 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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UnloadWave(wave);
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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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}
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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 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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return sound;
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}
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@ -314,7 +320,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
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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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UnloadWave(wave);
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TraceLog(INFO, "[%s] Sound loaded successfully from resource, sample rate: %i", rresName, (int)sampleRate);
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@ -381,22 +387,6 @@ void PlaySound(Sound sound)
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//alGetSourcef(sound.source, AL_SEC_OFFSET, &result); // AL_SAMPLE_OFFSET
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}
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// Play a sound with extended options
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// TODO: This function should be reviewed...
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void PlaySoundEx(Sound sound, float timePosition, bool loop)
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{
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// TODO: Review
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// Change the current position (e.g. skip some part of the sound)
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// NOTE: Only work when the entire file is in a single buffer
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//alSourcei(sound.source, AL_BYTE_OFFSET, int(position * sampleRate));
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alSourcePlay(sound.source); // Play the sound
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if (loop) alSourcei(sound.source, AL_LOOPING, AL_TRUE);
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else alSourcei(sound.source, AL_LOOPING, AL_FALSE);
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}
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// Pause a sound
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void PauseSound(Sound sound)
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{
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@ -421,30 +411,250 @@ bool SoundIsPlaying(Sound sound)
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return playing;
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}
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// Check if music is playing
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bool MusicIsPlaying(Music music)
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{
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ALenum state;
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alGetSourcei(music.source, AL_SOURCE_STATE, &state);
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return (state == AL_PLAYING);
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}
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// Set volume for a sound
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void SetVolume(Sound sound, float volume)
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void SetSoundVolume(Sound sound, float volume)
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{
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alSourcef(sound.source, AL_GAIN, volume);
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}
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// Set pitch for a sound
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void SetPitch(Sound sound, float pitch)
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void SetSoundPitch(Sound sound, float pitch)
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{
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alSourcef(sound.source, AL_PITCH, pitch);
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}
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//----------------------------------------------------------------------------------
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// Module Functions Definition - Music loading and stream playing (.OGG)
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//----------------------------------------------------------------------------------
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// Start music playing (open stream)
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void PlayMusicStream(char *fileName)
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{
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if (strcmp(GetExtension(fileName),"ogg") == 0)
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{
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// Stop current music, clean buffers, unload current stream
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StopMusicStream();
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// Open audio stream
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currentMusic.stream = stb_vorbis_open_filename(fileName, NULL, NULL);
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if (currentMusic.stream == NULL) TraceLog(WARNING, "[%s] Could not open ogg audio file", fileName);
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else
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{
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// Get file info
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stb_vorbis_info info = stb_vorbis_get_info(currentMusic.stream);
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currentMusic.channels = info.channels;
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currentMusic.sampleRate = info.sample_rate;
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TraceLog(INFO, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
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TraceLog(INFO, "[%s] Ogg channels: %i", fileName, info.channels);
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TraceLog(INFO, "[%s] Temp memory required: %i", fileName, info.temp_memory_required);
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if (info.channels == 2) currentMusic.format = AL_FORMAT_STEREO16;
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else currentMusic.format = AL_FORMAT_MONO16;
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currentMusic.loop = true; // We loop by default
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musicEnabled = true;
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// Create an audio source
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alGenSources(1, ¤tMusic.source); // Generate pointer to audio source
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alSourcef(currentMusic.source, AL_PITCH, 1);
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alSourcef(currentMusic.source, AL_GAIN, 1);
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alSource3f(currentMusic.source, AL_POSITION, 0, 0, 0);
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alSource3f(currentMusic.source, AL_VELOCITY, 0, 0, 0);
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//alSourcei(currentMusic.source, AL_LOOPING, AL_TRUE); // ERROR: Buffers do not queue!
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// Generate two OpenAL buffers
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alGenBuffers(2, currentMusic.buffers);
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// Fill buffers with music...
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BufferMusicStream(currentMusic.buffers[0]);
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BufferMusicStream(currentMusic.buffers[1]);
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// Queue buffers and start playing
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alSourceQueueBuffers(currentMusic.source, 2, currentMusic.buffers);
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alSourcePlay(currentMusic.source);
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// NOTE: Regularly, we must check if a buffer has been processed and refill it: MusicStreamUpdate()
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|
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currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
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}
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}
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else TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName);
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}
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|
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// Stop music playing (close stream)
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void StopMusicStream()
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{
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if (musicEnabled)
|
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{
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alSourceStop(currentMusic.source);
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EmptyMusicStream(); // Empty music buffers
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alDeleteSources(1, ¤tMusic.source);
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alDeleteBuffers(2, currentMusic.buffers);
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|
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stb_vorbis_close(currentMusic.stream);
|
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}
|
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|
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musicEnabled = false;
|
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}
|
||||
|
||||
// Pause music playing
|
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void PauseMusicStream()
|
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{
|
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// TODO: Record music is paused or check if music available!
|
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alSourcePause(currentMusic.source);
|
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}
|
||||
|
||||
// Check if music is playing
|
||||
bool MusicIsPlaying()
|
||||
{
|
||||
ALenum state;
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|
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alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
|
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|
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return (state == AL_PLAYING);
|
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}
|
||||
|
||||
// Set volume for music
|
||||
void SetMusicVolume(float volume)
|
||||
{
|
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alSourcef(currentMusic.source, AL_GAIN, volume);
|
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}
|
||||
|
||||
// Get current music time length (in seconds)
|
||||
float GetMusicTimeLength()
|
||||
{
|
||||
float totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic.stream);
|
||||
|
||||
return totalSeconds;
|
||||
}
|
||||
|
||||
// Get current music time played (in seconds)
|
||||
float GetMusicTimePlayed()
|
||||
{
|
||||
int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
|
||||
|
||||
int samplesPlayed = totalSamples - currentMusic.totalSamplesLeft;
|
||||
|
||||
float secondsPlayed = (float)samplesPlayed / (currentMusic.sampleRate * currentMusic.channels);
|
||||
|
||||
return secondsPlayed;
|
||||
}
|
||||
|
||||
//----------------------------------------------------------------------------------
|
||||
// Module specific Functions Definition
|
||||
//----------------------------------------------------------------------------------
|
||||
|
||||
// Fill music buffers with new data from music stream
|
||||
static bool BufferMusicStream(ALuint buffer)
|
||||
{
|
||||
short pcm[MUSIC_BUFFER_SIZE];
|
||||
|
||||
int size = 0; // Total size of data steamed (in bytes)
|
||||
int streamedBytes = 0; // Bytes of data obtained in one samples get
|
||||
|
||||
bool active = true; // We can get more data from stream (not finished)
|
||||
|
||||
if (musicEnabled)
|
||||
{
|
||||
while (size < MUSIC_BUFFER_SIZE)
|
||||
{
|
||||
streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic.stream, currentMusic.channels, pcm + size, MUSIC_BUFFER_SIZE - size);
|
||||
|
||||
if (streamedBytes > 0) size += (streamedBytes*currentMusic.channels);
|
||||
else break;
|
||||
}
|
||||
|
||||
TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
|
||||
}
|
||||
|
||||
if (size > 0)
|
||||
{
|
||||
alBufferData(buffer, currentMusic.format, pcm, size*sizeof(short), currentMusic.sampleRate);
|
||||
|
||||
currentMusic.totalSamplesLeft -= size;
|
||||
}
|
||||
else
|
||||
{
|
||||
active = false;
|
||||
TraceLog(WARNING, "No more data obtained from stream");
|
||||
}
|
||||
|
||||
return active;
|
||||
}
|
||||
|
||||
// Empty music buffers
|
||||
static void EmptyMusicStream()
|
||||
{
|
||||
ALuint buffer = 0;
|
||||
int queued = 0;
|
||||
|
||||
alGetSourcei(currentMusic.source, AL_BUFFERS_QUEUED, &queued);
|
||||
|
||||
while(queued > 0)
|
||||
{
|
||||
alSourceUnqueueBuffers(currentMusic.source, 1, &buffer);
|
||||
|
||||
queued--;
|
||||
}
|
||||
}
|
||||
|
||||
// Update (re-fill) music buffers if data already processed
|
||||
extern void UpdateMusicStream()
|
||||
{
|
||||
ALuint buffer = 0;
|
||||
ALint processed = 0;
|
||||
bool active = true;
|
||||
|
||||
if (musicEnabled)
|
||||
{
|
||||
// Get the number of already processed buffers (if any)
|
||||
alGetSourcei(currentMusic.source, AL_BUFFERS_PROCESSED, &processed);
|
||||
|
||||
while (processed > 0)
|
||||
{
|
||||
// Recover processed buffer for refill
|
||||
alSourceUnqueueBuffers(currentMusic.source, 1, &buffer);
|
||||
|
||||
// Refill buffer
|
||||
active = BufferMusicStream(buffer);
|
||||
|
||||
// If no more data to stream, restart music (if loop)
|
||||
if ((!active) && (currentMusic.loop))
|
||||
{
|
||||
if (currentMusic.loop)
|
||||
{
|
||||
stb_vorbis_seek_start(currentMusic.stream);
|
||||
currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
|
||||
|
||||
active = BufferMusicStream(buffer);
|
||||
}
|
||||
}
|
||||
|
||||
// Add refilled buffer to queue again... don't let the music stop!
|
||||
alSourceQueueBuffers(currentMusic.source, 1, &buffer);
|
||||
|
||||
if(alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Ogg playing, error buffering data...");
|
||||
|
||||
processed--;
|
||||
}
|
||||
|
||||
ALenum state;
|
||||
alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
|
||||
|
||||
if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic.source);
|
||||
|
||||
if (!active) StopMusicStream();
|
||||
}
|
||||
}
|
||||
|
||||
// Load WAV file into Wave structure
|
||||
static Wave LoadWAV(char *fileName)
|
||||
static Wave LoadWAV(const char *fileName)
|
||||
{
|
||||
// Basic WAV headers structs
|
||||
typedef struct {
|
||||
@ -543,199 +753,51 @@ static Wave LoadWAV(char *fileName)
|
||||
return wave;
|
||||
}
|
||||
|
||||
// Unload WAV file data
|
||||
static void UnloadWAV(Wave wave)
|
||||
// Load OGG file into Wave structure
|
||||
static Wave LoadOGG(char *fileName)
|
||||
{
|
||||
Wave wave;
|
||||
|
||||
stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);
|
||||
stb_vorbis_info info = stb_vorbis_get_info(oggFile);
|
||||
|
||||
wave.sampleRate = info.sample_rate;
|
||||
wave.bitsPerSample = 16;
|
||||
wave.channels = info.channels;
|
||||
|
||||
TraceLog(DEBUG, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
|
||||
TraceLog(DEBUG, "[%s] Ogg channels: %i", fileName, info.channels);
|
||||
|
||||
int totalSamplesLength = (stb_vorbis_stream_length_in_samples(oggFile) * info.channels);
|
||||
|
||||
wave.dataSize = totalSamplesLength*sizeof(short); // Size must be in bytes
|
||||
|
||||
TraceLog(DEBUG, "[%s] Samples length: %i", fileName, totalSamplesLength);
|
||||
|
||||
float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile);
|
||||
|
||||
TraceLog(DEBUG, "[%s] Total seconds: %f", fileName, totalSeconds);
|
||||
|
||||
if (totalSeconds > 10) TraceLog(WARNING, "[%s] Ogg audio lenght is larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds);
|
||||
|
||||
int totalSamples = totalSeconds*info.sample_rate*info.channels;
|
||||
|
||||
TraceLog(DEBUG, "[%s] Total samples calculated: %i", fileName, totalSamples);
|
||||
|
||||
//short *data
|
||||
wave.data = malloc(sizeof(short)*totalSamplesLength);
|
||||
|
||||
int samplesObtained = stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, wave.data, totalSamplesLength);
|
||||
|
||||
TraceLog(DEBUG, "[%s] Samples obtained: %i", fileName, samplesObtained);
|
||||
|
||||
stb_vorbis_close(oggFile);
|
||||
|
||||
return wave;
|
||||
}
|
||||
|
||||
// Unload Wave data
|
||||
static void UnloadWave(Wave wave)
|
||||
{
|
||||
free(wave.data);
|
||||
}
|
||||
|
||||
// TODO: Ogg data loading
|
||||
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