Question

作为一项学习练习，我使用AudioQueue生成并播放300 Hz正弦波。（我知道有很多工具可以生成和播放音频，但是，这只是为了构建我的Core Audio chops而且这个任务完全与AudioQueue有关。）

波浪播放，但失真。记录和绘制声音表明缓冲区之间的边界有一些失真（每半秒），此外还有其他短暂的失真突发。我在下面提供了我的代码。如果有人能够对这个问题有所了解，那将是惊人的 - 感谢阅读！

编辑：发现问题。它应该读取bufferByteSize = numPacketsForTime * asbd.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;

我将此留在这里，因为有人可能会发现代码有用！

简单的AudioQueue正弦波 - 为什么失真？

1 个答案: