SoundTouch音频处理库源码分析及算法提取(8)

  经过前面几个小节的介绍，各位想必已经对ST有个比较初步的认识。在往后的这些章节里，我将提出ST某些重要的算法以及实现。首先谈谈数字滤波器的实现。纵观整个ST的处理流程，仅仅在对声音样本进行重采样的时候可以通过bool bUseAAFilter的值来判断是否采用数字滤波器，具体在RateTransposer类成员函数processSamples中实现。
// Transposes sample rate by applying anti-alias filter to prevent folding.
// Returns amount of samples returned in the "dest" buffer.
// The maximum amount of samples that can be returned at a time is set by
// the 'set_returnBuffer_size' function.
void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
{
    uint count;
    uint sizeReq;

    if (nSamples == 0) return;
    assert(pAAFilter);

    // If anti-alias filter is turned off, simply transpose without applying
    // the filter
    if (bUseAAFilter == FALSE)
    {
        sizeReq = (uint)((float)nSamples / fRate + 1.0f);
        count = transpose(outputBuffer.ptrEnd(sizeReq), src, nSamples);
        outputBuffer.putSamples(count);
        return;
    }

    // Transpose with anti-alias filter
    if (fRate < 1.0f)
    {
        upsample(src, nSamples);
    }
    else 
    {
        downsample(src, nSamples);
    }
}
下面是RateTransposer类成员函数upsample和downsample的实现：
// Transposes up the sample rate, causing the observed playback 'rate' of the
// sound to decrease
void RateTransposer::upsample(const SAMPLETYPE *src, uint nSamples)
{
    uint count, sizeTemp, num;
    sizeTemp = (uint)((float)nSamples / fRate + 16.0f);

    count = transpose(storeBuffer.ptrEnd(sizeTemp), src, nSamples);
    storeBuffer.putSamples(count);

    num = storeBuffer.numSamples();
    count = pAAFilter->evaluate(outputBuffer.ptrEnd(num),
        storeBuff