从头开始创建 DSP 系统

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).

Answer 2

F# is probably a good choice here, as it's well fitted to manipulate functions. Functions are probably good building blocks for signal creation and processing.

F# is also good at manipulating collections in general, and arrays in particular, thanks to the higher-order functions in the Array module.

These qualities make F# popular in the finance sector and are also useful for signal processing, I would guess.

Visual F# 2010 for Technical Computing has a section dedicated to Fourier Transform, which could be relevant to what you want to do. I guess there is plenty of free information about the transform on the net, though.

Finally, to play samples, you can useXNA . I think the latest version of the API (4.0) also allows recording, but I have never used that. There is a famous music editing app for the Xbox called ezmuse+ Hamst3r Edition that uses XNA, so it's definitely possible.

Answer 3

With respect to buffering and asynchrony/threading/synchronization issues I suggest you to take a look at the new TPL Data Flow library. With its block primitives, concurrent data structures, data flow networks, async message prcessing, and TPL's Task based abstraction (that can be used with the async/await C# 5 features), it's a very good fit for this type of applications.

Answer 4

I don't know if this is really what you're looking for, but this was one of my personal projects while in college. I didn't truly understand how sound and DSP worked until I implemented it myself. I was trying to get as close to the speaker as possible, so I did it using only libsndfile, to handle the file format intricacies for me.

Basically, my first project was to create a large array of doubles, fill it with a sine wave, then use sf_writef_double() to write that array to a file to create something that I could play, and see the result in a waveform editor.

Next, I added another function in between the sine call, and the write call, to add an effect.

This way you start playing with very low-level oscillators and effects, and you can see the results immediately. Plus, it's very little code to get something like this working.

Personally, I would start with the simplest possible solution you can, then slowly add on. Try just writing out to a file and using your audio player to play it, so you don't have to deal with the audio apis. Just use a single array to start, and modify-in-place. Definitely start off single-threaded. As your project grows, you can start moving to other solutions, like pipes instead of the array, multi-threading it, or working with the audio API.

If you're wanting to create a project you can ship, depending on exactly what it is, you'll probably have to move to more complex libraries, like some real-time audio processing. But the basics you learn by doing the simple way above will definitely help when you get to this point.

Good luck!

Answer 5

I've done quite a bit of real-time DSP, although not with audio. While either of your ideas (immutable buffer) vs (mutable buffer modified in place) could work, what I prefer to do is create a single permanent buffer for each link in the signal path. Most effects don't lend themselves well to modification in place, since each input sample affects multiple output samples. The buffer-for-each-link technique works especially well when you have resampling stages.

Here, when samples arrive, the first buffer is overwritten. Then the first filter reads the new data from its input buffer (the first buffer) and writes to its output (the second buffer). Then it invokes the second stage to read from the second buffer and write into the third.

This pattern completely eliminates dynamic allocation, allows each stage to keep a variable amount of history (since effects need some memory), and is very flexible as far as enabling rearranging the filters in the path.

Answer 6

Alright, I'll have a stab at the bounty as well then:)

I'm actually in a very similar situation. I've been making electronic music for ages, but only over the past couple of years I've started exploring actual audio processing.

You mention that you have researched the maths. I think that's crucial. I'm currently fighting my way through Ken Steiglitz' A Digital Signal Processing Primer - With Applications to Digital Audio and Computer Music. If you don't know your complex numbers and phasors it's going to be very difficult.

I'm a Linux guy so I've started writing LADSPA plugins in C. I think it's good to start at that basic level, to really understand what's going on. If I was on Windows I'd download the VST SDK from Steinberg and write a quick proof of concept plugin that just adds noise or whatever.

Another benefit of choosing a framework like VST or LADSPA is that you can immediately use your plugins in your normal audio suite. The satisfaction of applying your first home-built plugin to an audio track is unbeatable. Plus, you will be able to share your plugins with other musicians.

There are probably ways to do this in C#/F#, but I would recommend C++ if you plan to write VST plugins, just to avoid any unnecessary overhead. That seems to be the industry standard.

In terms of buffering, I've been using circular buffers (a good article here: http://www.dspguide.com/ch28/2.htm ). A good exercise is to implement a finite response filter (what Steiglitz refers to as a feedforward filter) - these rely on buffering and are quite fun to play around with.

I've got a repo on Github with a few very basic LADSPA plugins. The architectural difference aside, they could potentially be useful for someone writing VST plugins as well. https://github.com/andreasjansson/my_ladspa_plugins

Another good source of example code is the CSound project. There's tonnes of DSP code in there, and the software is aimed primarily at musicians.

Answer 7

You could have a look at BYOND . It is an environment for programmatic audio / midi instrument and effect creation in C#. It is available as standalone and as VST instru and effect.

FULL DISCLOSURE I am the developer of BYOND.

Answer 8

Start with reading this and this .

This will give you idea on WHAT you have to do.

Then, learn DirectShow architecture - and learn HOW not to do it, but try to create your simplified version of it.