How to delay a channel on runtime?



I’m creating a VST Plugin with Juce and I need to delay each channel on runtime. In my desktop application I used fmod for processing the sound. But now I need to process the sound as VST Plugin. I was searching for a solution and found the JuceDemoPlugin project. But if I change the samples manually it produces noise (same code like in the JuceDemoPlugin). Is there a function for delaying each channel like a DSP filter in fmod? Or can I use the fmod framework for my VST Plugin?

Here is the code from the JuceDemoPlugin. But it’s not a great solution for my problem, because I also need to set the delay back and delay the other channel on runtime.

void JuceDemoPluginAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
    const int numSamples = buffer.getNumSamples();
    int channel, dp = 0;

    // Go through the incoming data, and apply our gain to it...
    for (channel = 0; channel < getNumInputChannels(); ++channel)
        buffer.applyGain (channel, 0, buffer.getNumSamples(), gain);

    // Now pass any incoming midi messages to our keyboard state object, and let it
    // add messages to the buffer if the user is clicking on the on-screen keys
    keyboardState.processNextMidiBuffer (midiMessages, 0, numSamples, true);

    // and now get the synth to process these midi events and generate its output.
    synth.renderNextBlock (buffer, midiMessages, 0, numSamples);

    // Apply our delay effect to the new output..
    for (channel = 0; channel < getNumInputChannels(); ++channel)
        float* channelData = buffer.getSampleData (channel);
        float* delayData = delayBuffer.getSampleData (jmin (channel, delayBuffer.getNumChannels() - 1));
        dp = delayPosition;

        File file("~/Log/PluginDemo.log");
        FileLogger fl(file, "Plugin Demo Log", 1000);
        fl.logMessage(T("Delay = ")+String(delay));
        for (int i = 0; i < numSamples; ++i)
            const float in = channelData[i];
            channelData[i] += delayData[dp];
            delayData[dp] = (delayData[dp] + in) * delay;
            if (++dp > delayBuffer.getNumSamples())
                dp = 0;

    delayPosition = dp;

    // In case we have more outputs than inputs, we'll clear any output
    // channels that didn't contain input data, (because these aren't
    // guaranteed to be empty - they may contain garbage).
    for (int i = getNumInputChannels(); i < getNumOutputChannels(); ++i)
        buffer.clear (i, 0, buffer.getNumSamples());

    // ask the host for the current time so we can display it...
    AudioPlayHead::CurrentPositionInfo newTime;

    if (getPlayHead() != 0 && getPlayHead()->getCurrentPosition (newTime))
        // Successfully got the current time from the host..
        lastPosInfo = newTime;
        // If the host fails to fill-in the current time, we'll just clear it to a default..