OpenAL实现实时录音并保存到本地

我现在正在做用openal的实时录音功能,并保存在本地。获取音频并播放功能已经实现,并且获取到的数据已经保存在buffer中。但是保存音频文件的相关接口或方法我并没有在openal中找到。初步设想是保存文件是alutLoadWAVFile()的逆过程,即加入WAV的文件头后再写入数据。请问大神们,用openal能不能实现呢?

1个回答

以下是照搬别人的代码。。具体链接找不到了。可以实现录音并回放

#include "stdafx.h"
#include
#include //添加openal的 header files(头文件)
#include
#include

using std::list;
using namespace std;

#define FREQ 22050 //定义采样率 Sample Rate
#define CAP_SIZE 2048 //一次捕获音频的长度

int main(int argC,char* argV[])
{

//initial
list bufferQueue; // A quick and dirty queue of buffer objects
ALenum errorCode = 0;
ALuint helloBuffer[16], helloSource[1];

ALCdevice* audioDevice = alcOpenDevice(NULL); //请求默认的音频设备
errorCode = alcGetError(audioDevice);
ALCcontext* audioContext = alcCreateContext(audioDevice, NULL); //创建上下文
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 to where?
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]); //将buffer放入队尾
}

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 -- 这里保存
ALuint myBuff; // The buffer we're using
ALuint buffHolder[16]; // An array to hold catch the unqueued buffers
bool done = false;

//play or cap?
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); //获取可用的样本数
    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;

}

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