所以我想知道什么是最短的(就有效线而言)打开AL代码从默认麦克风读取数据并输出到默认扬声器?
我在Visual Studio 2008下的Windows 7上开发
答案 0 :(得分:20)
一个古老的问题,但这是一个答案。如果我们真的想要简洁,那肯定会被削减,但这有点不到100条有效线:
#include <AL/al.h> // OpenAL header files
#include <AL/alc.h>
#include <list>
using std::list;
#define FREQ 22050 // Sample rate
#define CAP_SIZE 2048 // How much to capture at a time (affects latency)
int main(int argC,char* argV[])
{
list<ALuint> bufferQueue; // A quick and dirty queue of buffer objects
ALenum errorCode=0;
ALuint helloBuffer[16], helloSource[1];
ALCdevice* audioDevice = alcOpenDevice(NULL); // Request default audio device
errorCode = alcGetError(audioDevice);
ALCcontext* audioContext = alcCreateContext(audioDevice,NULL); // Create the audio context
alcMakeContextCurrent(audioContext);
errorCode = alcGetError(audioDevice);
// Request the default capture device with a half-second buffer
ALCdevice* inputDevice = alcCaptureOpenDevice(NULL,FREQ,AL_FORMAT_MONO16,FREQ/2);
errorCode = alcGetError(inputDevice);
alcCaptureStart(inputDevice); // Begin capturing
errorCode = alcGetError(inputDevice);
alGenBuffers(16,&helloBuffer[0]); // Create some buffer-objects
errorCode = alGetError();
// Queue our buffers onto an STL list
for (int ii=0;ii<16;++ii) {
bufferQueue.push_back(helloBuffer[ii]);
}
alGenSources (1, &helloSource[0]); // Create a sound source
errorCode = alGetError();
short buffer[FREQ*2]; // A buffer to hold captured audio
ALCint samplesIn=0; // How many samples are captured
ALint availBuffers=0; // Buffers to be recovered
ALuint myBuff; // The buffer we're using
ALuint buffHolder[16]; // An array to hold catch the unqueued buffers
bool done = false;
while (!done) { // Main loop
// Poll for recoverable buffers
alGetSourcei(helloSource[0],AL_BUFFERS_PROCESSED,&availBuffers);
if (availBuffers>0) {
alSourceUnqueueBuffers(helloSource[0],availBuffers,buffHolder);
for (int ii=0;ii<availBuffers;++ii) {
// Push the recovered buffers back on the queue
bufferQueue.push_back(buffHolder[ii]);
}
}
// Poll for captured audio
alcGetIntegerv(inputDevice,ALC_CAPTURE_SAMPLES,1,&samplesIn);
if (samplesIn>CAP_SIZE) {
// Grab the sound
alcCaptureSamples(inputDevice,buffer,CAP_SIZE);
//***** Process/filter captured data here *****//
//for (int ii=0;ii<CAP_SIZE;++ii) {
// buffer[ii]*=0.1; // Make it quieter
//}
// Stuff the captured data in a buffer-object
if (!bufferQueue.empty()) { // We just drop the data if no buffers are available
myBuff = bufferQueue.front(); bufferQueue.pop_front();
alBufferData(myBuff,AL_FORMAT_MONO16,buffer,CAP_SIZE*sizeof(short),FREQ);
// Queue the buffer
alSourceQueueBuffers(helloSource[0],1,&myBuff);
// Restart the source if needed
// (if we take too long and the queue dries up,
// the source stops playing).
ALint sState=0;
alGetSourcei(helloSource[0],AL_SOURCE_STATE,&sState);
if (sState!=AL_PLAYING) {
alSourcePlay(helloSource[0]);
}
}
}
}
// Stop capture
alcCaptureStop(inputDevice);
alcCaptureCloseDevice(inputDevice);
// Stop the sources
alSourceStopv(1,&helloSource[0]);
for (int ii=0;ii<1;++ii) {
alSourcei(helloSource[ii],AL_BUFFER,0);
}
// Clean-up
alDeleteSources(1,&helloSource[0]);
alDeleteBuffers(16,&helloBuffer[0]);
errorCode = alGetError();
alcMakeContextCurrent(NULL);
errorCode = alGetError();
alcDestroyContext(audioContext);
alcCloseDevice(audioDevice);
return 0;
}