[music-dsp] 2 postdoc opportunities

Dear all I'm new to this list, I have been recommended to post the following offers here. I've been successful in an ERC application and as part of the project, I am hiring 2 postdocs, both of which could be of interest to some of you who are almost done or just graduated PhDs? The project is

Re: [music-dsp] IIR filter efficiency

Even cheaper—no need to get fancy with TPDF, since the level can be -300 dB or so :-D (I’ve done things as simply as toggle sign on 1e-15 once per block) But watch out for highpass filters and add back in ;-) > On Mar 10, 2017, at 1:10 AM, Richard Dobson wrote: > > This

Re: [music-dsp] denormals on ARM processors ?

The functions here  https://github.com/BelaPlatform/Bela/blob/master/core/math_runfast.c will allow to manipulate "fast" mode for ARM vfp (making it non-IEEE-754 compliant). If you are using the NEON SIMD (you should!) then it always flushes to zero (which is one of the reasons why NEON is

Re: [music-dsp] IIR filter efficiency

Well, this is intended to be used on a system where many applications may run concurrently and therefore the need to not denormalise may not be universally shared. Some applications do want to allow computation on denormals in order to get a bit more precision in the result at the expense of a

Re: [music-dsp] IIR filter efficiency

How about *not* calling RESTORE_DENORMALS as a courtesy for the rest of the audio processing chain? > On 10 Mar 2017, at 10:28 , Philippe Wicker wrote: > > Yes, it is done that way in the Apple AudioUnit SDK. The code calls > DISABLE_DENORMALS at the beginning of the render

Re: [music-dsp] IIR filter efficiency

Yes, it is done that way in the Apple AudioUnit SDK. The code calls DISABLE_DENORMALS at the beginning of the render routine, and RESTORE_DENORMALS at the end. Those macros call asm instruction stmxcsr and ldmxcsr (in AUBase.cpp: line 53). > On 10 Mar 2017, at 10:17, Stefan Stenzel

Re: [music-dsp] IIR filter efficiency

I don’t get it - what prevents you from checking/setting DAZ/FTZ in your processing routine? > On 10 Mar 2017, at 10:10 , Richard Dobson wrote: > > This might not be possible if one is implementing e.g. a plugin to run in a > third-party host - where one does not have

Re: [music-dsp] IIR filter efficiency

This might not be possible if one is implementing e.g. a plugin to run in a third-party host - where one does not have full control over the host, nor indeed of fellow plugins. Whereas adding some ~very~ low level TPDF dither to a signal should be close to minimum cost. Richard Dobson On

Re: [music-dsp] IIR filter efficiency

That’s true. I’ve not been working with the windows compilers since a long time but I remember that it wasn’t possible to be 100% sure that the compiler uses only SSE instructions. What I’ve been said is that the compiler could always decide to mix scalar (i.e. use the FP unit) and the vectored

Re: [music-dsp] IIR filter efficiency

> On 10 Mar 2017, at 09:29, Stefan Stenzel > wrote: > > That document is from 2002 - today all these suggestions make little sense > unless you want your code to run explicitly on a CPU without SSE. > The best strategy for avoiding denormals is to enforce the

Re: [music-dsp] IIR filter efficiency

That document is from 2002 - today all these suggestions make little sense unless you want your code to run explicitly on a CPU without SSE. The best strategy for avoiding denormals is to enforce the compiler to use SSE and avoid the FPU, then set the Denormals-Are-Zero (DAZ) and Flush-To-Zero