Re: [Sursound] [allowed] Re: Recreating a 3d soundfield with lots of mics.....
Even "dead" concert halls in the relative sense have a lot of reverberation. A really dead hall still has a 1 second reverberation time say and most of what you hear in the audience is still reverberant sound. Robert On Mon, 20 May 2013, David Pickett wrote: At 00:50 18-05-13, Robert Greene wrote: Of course in those live versus canned experiments(also with AR) reverberation tended to make things sound pretty much the same to smooth out errors and so on. Reverberation in the RFH pre 1966? David ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] time variance...
Hi, So, there must be quite a lot going on in Focusrite's Liquid Channel. http://global.focusrite.com/mic-pres-channel-strips/liquid-channel Reputedly Focusrite license a system from Sintefex. http://www.sintefex.com/docs/appnotes/dynaconv.PDF Ciao, Dave Hunt Date: Mon, 20 May 2013 20:33:33 + From: Fons Adriaensen On Sun, May 19, 2013 at 10:18:35AM +0200, J?rn Nettingsmeier wrote: On 05/02/2013 01:26 AM, Richard Dobson wrote: I have always understood it to mean that the behaviour is not dependent upon ~when~ the signal is injected. Thus, a plain delay is TI because everything is always delayed the same way; while a modulated effect such as a flanger (maybe using a variable delay) is not TI as exactly what comes out depends in the time something goes in. ... for practical purposes, i guess fons' definition is more useful, because then the term "LTI" system is strictly limited to something that can be fully described with an impulse response. Despite what I wrote before, I tend to agree with Richard. If I interpret his formulation correctly, a system is time-invariant iff, when the output for input x(t) is y(t), then the output for x(t + T) is y(t + T), for any T. It's actually quite difficult to formulate a stronger version *unless* you assume that the system is linear as well. A linear time-invariant (LTI) system is fully defined by an impulse response, or by a transfer function in the frequency domain. Now consider three cases: 1. A filter, 2. A tremolo effect, 3. A compressor. The filter is LTI, while the tremolo and compressor are not. Do they fail to be LTI because they are not linear, or because they are not time-invariant ? The tremolo fails Richard's TI criterion. But it *is* linear in a very strong sense: for any a(t) and b(t) Tremolo (a(t) + b(t)) == Tremolo (a(t)) + Tremolo (b(t)). The compressor is time-invariant according to Richard's criterion. But it isn't linear in the way the tremolo is. It could be said to be linear 'at any instant', assuming attack and release times are non-zero. But that is a somewhat problematic definition of linearity, since apart from trivial cases (pure gain) linear processes depend on the input's or output's history, and are not defined by some relation at a single instant. So it seems that a stronger definition of TI is not necesssary. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] time variance...
On 2013-05-21, Dave Hunt wrote: So, there must be quite a lot going on in Focusrite's Liquid Channel. http://global.focusrite.com/mic-pres-channel-strips/liquid-channel Reputedly Focusrite license a system from Sintefex. http://www.sintefex.com/docs/appnotes/dynaconv.PDF There's altogether too much hype there. Yes, you can do what they do: characterize the short term near-LTI part of the system in steady state and then brute force apply that sample by sample in a time-variant convolution. That is a pretty powerful operation, but it's not really the form you need for compressors. I believe it's both underkill and overkill at the same time. (The input portion seems legit as far as I can tell, presuming they implemented it right.) That's because typical analog compressors are single band, which means they operate on instantaneous amplitude only. There's filtering complexity to be sure, but it's entirely in the side chain, whereas the main path is more or less just a voltage controlled amplifier. The Focusrite architecture gets it the other way around: a brutal amount of brute force processing power is being used for an operation which essentially ends up recreating a constant, more or less unity, EQ curve, while the side chain isn't being modelled at all beyond instantaneous nonlinearity. In analog compressors, and especially the better ones, the side chain which determines the eventual gain in the main one is exceedingly carefully tuned, stateful, and at longer time scales surprisingly nonlinear. It has different attack and decay time constants, it can employ slew rate limited ramps purposely, and sometimes it even has differently EQ's subbands with different constants. That is not something an architecture like Focusrite's can capture, not in analysis nor in synthesis. Mostly you're not going to notice it, of course, but a suitable mixture of steady state background, transients and silence will almost certainly show a definite difference to the original system being modelled. Additionally they say in the second link that they interpolate impulse responses linearly. That is a bad idea in itself, because it'll almost always lead to passband ripple between the endpoints, and if you're heavily into transient content like me, intermediate forms with time-variant allpass terms, muddying up the temporal structure of the signal. Combining the simulation of the nonlinear preamp and the compressor into a single, simple circuit like this buys them easy analysis, but it also makes their synthesis side unsuited to the task at hand and so nasty to analyze properly they don't even try it but resort to passing ad hoc intuitions to it. That sort of thing is Unclean. It will get the macroscopic stateless nonlinearity of the preamp more or less right, in steady state, for sparse quasiperiodic LF signals, the overall EQ curve more or less in the ballpark, and it'll capture the compression characteristic for slowly and smoothly varying envelopes. But it'll definitely not be a "precise" replica of any and all analog input stages. In fact I'm pretty sure you can even hear alias in the output, because when you do it the way they claim to, that first crucial coefficient of the impulse response, as a function of the input signal, will constitute an arbitrary table lookup/waveshaper of very high polynomial order. That sort of thing is very easy to drive into audible aliasing unless they employ truly exorbitant oversampling rates in the intermediate stages...which you can't really do without running into processing power and latency constraints. -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-50-5756111, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Naive question on MS and Ambisonics
Hello jörn, Thinking about what you say here, is this working by having pure M from the front and S from 90 degrees to the side, effectively 'mixing' the M S signals in the air as they reach the ears/brain? (Maybe I'm thinking about this too much, my brain is hurting.) If so, is there significant distortion/corruption of the effect from the two ears receiving different variations of the M and S signals? I realise that ear crosstalk effect is an issue with standard two speaker stereo as well, but the consequences with this kind of signal presentation seem to me to be quite different. As part of this, if the head turns say 45 degrees to the left, the ear difference would seem to be at a maximum, with the left ear receiving a significant amount of the opposite lobe of the figure 8 with little cancellation effect from the M in front. Perhaps this is all part of the plan….? Regards, Ray E. > > i sometimes use MS pairs in ambisonic mixing, where a first or > higher-order panner takes the M signal (i point it into the direction of > the mic as seem from the listening position) and a first-order panner > with W disconnected takes the S signal, which i point to mic position > minus 90?. > > > -- > J?rn Nettingsmeier > Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] [allowed] Re: Recreating a 3d soundfield with lots of mics.....
At 12:16 21-05-13, Robert Greene wrote: Even "dead" concert halls in the relative sense have a lot of reverberation. A really dead hall still has a 1 second reverberation time say and most of what you hear in the audience is still reverberant sound. Did you ever hear an orchestra playing in the RFH pre 1960??? David ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
[Sursound] [ot] hard to find papers
Does anybody happen to have the Dolby A and SR papers handy? They seem to be particularly difficult to find without access to the AES archives. -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-50-5756111, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] [ot] hard to find papers
They should be in your in-box by now. I trust you were able to find the patents? - Original Message From: Sampo Syreeni To: sursound-list Sent: Tue, May 21, 2013 5:27:36 PM Subject: [Sursound] [ot] hard to find papers Does anybody happen to have the Dolby A and SR papers handy? They seem to be particularly difficult to find without access to the AES archives. -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-50-5756111, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
