Creating a DSP system from scratch

Question

I love electronic music and I am interested in how it all ticks. I've found lots of helpful questions on Stack Overflow on libraries that can be used to play with audio, filters etc. But what I am really curious about is what is actually hapening: how is the data being passed between effects and oscillators? I have done research into the mathematical side of dsp and I've got that end of the problem sussed but I am unsure what buffering system to use etc. The final goal is to have a simple object heirarchy of effects and oscillators that pass the data between each other (maybe using multithreading if I don't end up pulling out all my hair trying to implement it). It's not going to be the next Propellerhead Reason but I am interested in how it all works and this is more of an exercise than something that will yeild an end product.

At the moment I use .net and C# and I have recently learnt F# (which may or may not lead to some interesting ways of handling the data) but if these are not suitable for the job I can learn another system if necessary.

The question is: what is the best way to get the large amounts of signal data through the program using buffers? For instance would I be better off using a Queue, Array,Linked List etc? Should I make the samples immutable and create a new set of data each time I apply an effect to the system or just edit the values in the buffer? Shoud I have a dispatcher/thread pool style object that organises passing data or should the effect functions pass data directly between each other?

Thanks.

EDIT: another related question is how would I then use the windows API to play this array? I don't really want to use DirectShow because Microsoft has pretty much left it to die now

EDIT2: thanks for all the answers. After looking at all the technologies I will either use XNA 4(I spent a while trawling the internet and found this site which explains how to do it) or NAudio to output the music... not sure which one yet, depends on how advanced the system ends up being. When C# 5.0 comes out I will use its async capabilities to create an effects architecture on top of that. I've pretty much used everybody's answer equally so now I have a conundrum of who to give the bounty to...

Answer 1

Have you looked at VST.NET (http://vstnet.codeplex.com/)? It's a library to write VST using C# and it has some examples. You can also consider writing a VST, so that your code can be used from any host application (but even if you don't want, looking at their code can be useful).

Signal data is usually big and requires a lot of processing. Do not use a linked list! Most libraries I know simply use an array to put all the audio data (after all, that's what the sound card expect).

From a VST.NET sample:

    public override void Process(VstAudioBuffer[] inChannels, VstAudioBuffer[] outChannels)
    {
        VstAudioBuffer audioChannel = outChannels[0];

        for (int n = 0; n < audioChannel.SampleCount; n++)
        {
            audioChannel[n] = Delay.ProcessSample(inChannels[0][n]);
        }
    }

The audioChannel is a wrapper around an unmanaged float* buffer.

You probably store your samples in an immutable array. Then, when you want to play them, you copy the data in the output buffer (change the frequency if you want) and perform effects in this buffer. Note you can use several output buffers (or channels) and sum them at the end.

Edit

I know two low-level ways to play your array: DirectSound and WaveOut from Windows API. C# Example using DirectSound. C# example with WaveOut. However, you might prefer use an external higher-level library, like NAudio. NAudio is convenient for .NET audio manipulation - see this blog post for sending a sine wave to the audio card. You can see they are also using an array of float, which is what I recommend (if you do your computations using bytes, you'll end up with a lot of aliasing in the sound).