Function ReadImpulseResponseToFloat

libjamesdsp/JdspImpResToolbox.c:349–466

}

float* ReadImpulseResponseToFloat
(const char* mIRFileName, int targetSampleRate, int* jImpInfo, int convMode, int* javaAdvSetPtr)
{
    if (strlen(mIRFileName) <= 0) return 0;
    unsigned int channels;
    drwav_uint64 frameCount;
    float *pFrameBuffer = loadAudioFile(mIRFileName, targetSampleRate, &channels, &frameCount, 1);
    if (channels == 0 || channels == 3 || channels > 4)
    {
        free(pFrameBuffer);
        return 0;
    }

    int i;
    float *splittedBuffer[4];
    int alloc = frameCount;
    if (alloc < 8)
        alloc = 8;
    for (i = 0; i < channels; i++)
    {
        if (convMode == 2)
        {
            splittedBuffer[i] = (float*)malloc(alloc * 2 * sizeof(float));
            memset(splittedBuffer[i], 0, alloc * 2 * sizeof(float));
        }
        else
            splittedBuffer[i] = (float*)malloc(frameCount * sizeof(float));
    }
    channel_splitFloat(pFrameBuffer, frameCount, splittedBuffer, channels);
    if (convMode > 0)
    {
        free(pFrameBuffer);
        int range[2];
        float startCutdB = javaAdvSetPtr[0];
        float endCutdB = javaAdvSetPtr[1];
        if (convMode == 1)
            checkStartEnd(splittedBuffer, channels, frameCount, startCutdB, endCutdB, range);
        else
        {
            range[0] = 0;
            range[1] = alloc * 2;
        }
        float *outPtr[4];
        int xLen = range[1] - range[0];
        if (convMode == 1)
        {
            checkStartEnd(splittedBuffer, channels, frameCount, startCutdB, endCutdB, range);
            for (i = 0; i < channels; i++)
            {
                outPtr[i] = &splittedBuffer[i][range[0]];
                circshift(outPtr[i], xLen, javaAdvSetPtr[i + 2]);
                for (int j = 0; j < javaAdvSetPtr[i + 2] - 1; j++)
                    outPtr[i][j] = 0.0f;
            }
        }
        else
        {
            fftData fd;