简单的AudioQueue正弦波-为什么失真

Question

作为学习练习，我使用AudioQueue生成并播放300 Hz正弦波。(我知道有各种各样的工具来生成和播放音频，但是是的，这只是为了建立我的核心音频，这个任务都是关于AudioQueue的。)

波播放，但有失真。记录和绘制声音表明，在缓冲区之间的边界有一些失真(每半秒)，除了这里和那里的其他短暂的失真爆发。我在下面包含了我的代码。如果有人能对这个问题提出一些建议，那就太棒了——感谢阅读!

编辑:找到问题了。应该是bufferByteSize=numPacketsForTime* asbed . mbytesperpacket;

static void MyAQOutputCallback(void *inUserData,
                               AudioQueueRef inAQ,
                               AudioQueueBufferRef inCompleteAQBuffer){
    int i;
    MyWave *inData=(MyWave*)inUserData;
    // synth params
    int phaseL =inData->sampleCount;
    float FL = (2.0 * 3.14159265 * 300.0) / 44100.0;
    float amp = 0.5;
    int frameCount=22050;
    // Get the info struct and a pointer to our output data
    short *coreAudioBuffer = (short*) inCompleteAQBuffer->mAudioData;
    // Need to set this
    inCompleteAQBuffer->mAudioDataByteSize = 2*frameCount; // two shorts per frame, one frame per packet
    // For each frame/packet (the same in our example)
    for(i=0;i<frameCount;i++) {
        // Render the sine waves - signed interleaved shorts (-32767 -> 32767), 16 bit stereo
        float sampleL = (amp * sin(FL * (float)phaseL));
        short sampleIL = (int)(sampleL * 32767.0);
        coreAudioBuffer[i ] =   sampleIL;
        phaseL++;
    }
    // "Enqueue" the buffer
    AudioQueueEnqueueBuffer(inAQ, inCompleteAQBuffer, 0, NULL);
    inData->sampleCount=phaseL;

}
int main(int argc, const char * argv[])
{
    // Open an audio file
    MyWave thisWave={0};
    // Set up format
    AudioStreamBasicDescription asbd;
    memset(&asbd,0,sizeof(asbd));
    asbd.mSampleRate=SAMPLE_RATE;
    asbd.mFormatID=kAudioFormatLinearPCM;
    asbd.mFormatFlags=kLinearPCMFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
    asbd.mBitsPerChannel=16;
    asbd.mChannelsPerFrame=1;
    asbd.mFramesPerPacket=1;
    asbd.mBytesPerFrame=2;
    asbd.mBytesPerPacket=2;
    // Set up queue
    AudioQueueRef queue;
    CheckError(AudioQueueNewOutput(&asbd,
                                   MyAQOutputCallback,
                                   &thisWave,
                                   NULL,
                                   NULL,
                                   0,
                                   &queue),
               "AudioQueueNewOutput failed");
    UInt32 bufferByteSize;
    Float64 numPacketsForTime=asbd.mSampleRate/asbd.mFramesPerPacket*0.5;
    bufferByteSize=numPacketsForTime;
    AudioQueueBufferRef buffers[kNumberPlaybackBuffers];
    int i;
    for (i=0;i<kNumberPlaybackBuffers;++i){
        CheckError(AudioQueueAllocateBuffer(queue,
                                            bufferByteSize,
                                            &buffers[i]),
                   "AudioQueueAllocateBuffer failed");
        MyAQOutputCallback(&thisWave, queue, buffers[i]);
    }

    // Start queue
    CheckError(AudioQueueStart(queue,
                               NULL),
               "AudioQueueStart failed");
    printf("Playing...n");
    do
    {
        CFRunLoopRunInMode(kCFRunLoopDefaultMode,
                           0.25,
                           false);
    }while (1==1);
    CFRunLoopRunInMode(kCFRunLoopDefaultMode, 2, false);
    // Clean up queue when finished
    CheckError(AudioQueueStop(queue,
                              TRUE),
               "AudioQueueStop failed");
    AudioQueueDispose(queue, TRUE);
    return 0;
}

Answer 1

发现问题，应该是:

 bufferByteSize = numPacketsForTime*asbd.mBytesPerPacket;

我将把它留在这里，因为有人可能会发现代码很有用!