Meridian Unplugged - Powered by MrTechGuy.co.uk Forums Discussion Meridian - Systems Interesting Discussion 861v6 v ADA

Here's a thought:

On the one hand, there is the prevailing school of thought that multiple subwoofers can "even out" the bass response in a room.

On the other hand, Bob Stuart talks about bass (below 80Hz) being directional (contrary to the commonly held view) and that the brain takes important psychological cues from directional bass. He believes it is critical to reproduce the acoustic of the recording; and having heard wonders on 8ks at HQ, I'm not going to argue with him.

I would say that these two views are not mutually exclusive as long as the multiple subs are only used for the LFE channel (which contains no directional information).

If you use subs to assist with the bass from the non-LFE channels, as you move away from a full range 7.0 setup (or 7.1 with a dedicated LFE sub or subs), the more information you will destroy. This is one reason why I'm using stereo subs with my DSP33s.

Exactly VK.

So the higher the cross-over point from the non-LFE channels the more directional bass (moving trains, planes and automobiles) will be pushed to the sub, potentially damaging the image/acoustic.

Pure LFE effects (explosions, earthquakes, etc.) should shake the whole room and thus be directionless rather than located, therefore multiple subs are potentially a good thing for that application.

Actually Jon, it's the subtlety of a natural acoustic that I was talking about, not the artificially steered effects in a movie.

I used the moving vehicle examples as I couldn't think of a good musical example.

George:

I am not going to respond to what you are saying as I do not have the answers as this level of discussion is above my experience and knowledge level [and admittedly an aera where I need to do more reading and research]...that said, I do hope that DErskine again jumps in as he will give you the whys and wherefores to his thinking...

I generally agree with your observations. In principle, I think it is dangerous to play around too much with bass management, particularly beyond the LFE channel; however in practice, if one must do it, I have found using a very low corssover point to be largely benign.

It would make sense to me, that when changing a pre-amp in such a setup, you would want to do a new setup of the EQ.
Analog output levels (hard- and/or software) will vary between pre-amp models/makes. Thus the EQ could behave differently, and therefore the end-result.

When you say "what comes out of" are you referring to frequency response? If so, I think you are mistaken in your assertions. If not, I have no idea what you mean. Perhaps you could clarify.

George - you have a very fundamental misunderstanding of Nyquist and the sample rate. 48kHz indicates the analog signal is sampled 48,000 times per second (at Soundstream, we used 50kHz). Nyquist then says at 48,0000 times per second, 24kHz would be the highest frequency you can accurately resolve (prior to folding and other stuff ... note, I am keeping this in very simple terms). In actual practice, the implementation of the filters and the conversion back to analog also enter into sound quality.

What is magic about a 96kHz sample rate? The real magic is it is an exact multiple of a 48kHz sample rate. Makes life really easy for up/down conversion, etc. In fact, the primary reason was the fact that 96kHz is an exact multiple of 48kHz...what audio (or movie track) would have high frequency content at 48kHz? Rather outside the range of human perception. Now, there is a suggestion that "taking a picture" of the analog signal more frequently (say 96000 times per second) would allow a more accurate reconstruction of the original analog signal. Clearly, more is better, yes? That baits the question of exactly what sample rate is needed to reconstruct the analog signal in such a manner that the reconstructed signal is audibly identical to the original. All of this is before any consideration is given to the word length! 16bit? 20 bit? 24 bit? What are the differences between 16 bit 96 and 24 bit 48? The body of research in this area is a real high stack of papers and not fully conclusive. And, not fully conclusive since the "audible" word keeps coming into the picture and "audible" has so many components to it. Not the least of which is the horrid, terrible, mythological, misunderstood, folklore nonsense that permeates this field (usually by self professed professional lay people). (Resulting many being happy with poor quality sound, horrid acoustic environments in listening rooms, and sound engineers mixing tracks in noisy little rooms using nearfield monitors and absolutely refusing to mix to any base standard and therefore preventing anyone from properly setting up their music listening environment.) And, let's not forget the green felt tip markers.

I would refer you to http://www.Meridian-audio.com/w_paper/Coding2.PDF which was done roughly in the 1996 time period. The AES library is full of subsequent data on these topics for those interested.

In my judgement, one of the primary advantages of higher sampling rates is not the ability to capture higher frequencies per se, but to be able to use gentler filters around the Nyquist frequency to avoid the aliasing (distortion) which occurs below the NyQuist frequency when brickwall filters are used, such as with a 44.1kHz sampling rate to maintain a frequency response extending to 20kHz. Indeed, the audible differences among DACs can largely be attributed to differing filter implementations.

I do not run it full range. The ADA is crossed over at 80hz.
Ken

You stated that all 8 channels of the SSP where connected to the QSC, so which device (SSP or QSC) does output above 80Hz to your speakers?