On Thursday, 28 November 2013 at 10:30:36 UTC, Chris wrote:
There are voice analysis and speech processing toolkits like Covarep and Voicebox (see links below) that were coded in Matlab, because they were originally only prototypes. There has been talk of porting them to C++. My first thought, as you might imagine, was why not use D? However, I don't know if there are any performance issues, especially for real time systems (in speech recognition), talking about GC, or in fact any other issues (number grinding etc.).

A lot of the analysis tools are based on some sort of HMM (http://en.wikipedia.org/wiki/Hidden_Markov_model) and I think D could handle that elegantly.

https://github.com/covarep/covarep
http://www.ee.ic.ac.uk/hp/staff/dmb/voicebox/voicebox.html

Hi, I have a little experience in dsp programming using oop languages, so I'll try to give you my mind, but my mind is more related to entertainment dsp softwares (asio, vst, etc...).

talking about GC
In "pseudo" real time (RT) audio (one or many buffer are overlapped) you are a in a loop (interesting example is bufferswitch in asio). It's time critical and performance critical, so you'll never create a class neither allocate a buffer here...The idea is: what does trigger the GC: memory allocation and dynamic class instance creation. It's like in GUI programming: you don't destroy and recreate many objects in the "resize/realign" message handler...So the GC problem is solved: there is no GC problem because in RT dsp you won't do something stupid that'll trig a GC pass.

In speech recognition you'll mostly use some frequency-domain technics (not to name the fft), so basically if you don't want to trigger a GC pass, don't use build-in array and make your own array using alloc/malloc/free. For the classes it's the same, you can still make your own class allocator/deallocator, like specified in the manual (even if they say it's deprecated). With user managed classes and array you'll avoid most of the GC passes...But it doesn't mean that the most important stuff is: not to allocate in the audio buffer loop.

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