Meridian Unplugged - Powered by MrTechGuy.co.uk Forums Discussion Meridian - Systems Reflections on Meridian

Probably won't happen any time soon. Just a guess of course. A single SPDIF, as I am told, can only carry 1-channel of 192kHz. So until they change the interface from processor to speaker, they probably won't do it. Then again perhaps they can just add a 2nd digital output card and then you can run an SPDIF to each speaker. That is a lot of cable, I bet cable companies would like that. ;-) Good thing Meridian includes the proper digital cable with their speakers.

Then there is the issue of 192kHz content. There is none! DVD-A appears to be going for mult-channel. I thought I had read somewhere once that lower resolution works when you have more channels. Besides we don't need 192kHz at home anyway. Let them use that in the studio.

Since their DSP speaker upgrades are costly, they might hold out until they believe 192kHz DACs are up to a certain level of performance.


They won't. Who say's is it a downgrade? Do you know what the effective resolution of DSD really is?


Perhaps what you should really be asking is, "How come the majority of high-end audio has announced some type of plan for DVD-A but nothing for SACD?" (Meridian, Mark Levinson, Proceed, Tag, EAD, Camelot, Arcam, Krell, Myryad, and Theta. Not to mention Toshiba, Panasonic, and Denon.)

[This message has been edited by sspears (edited 07-16-2001).]

Well... the point is not which format is better, but rather in which format my favourite artist of the moment is recorded on best. At this level of investment in audio equipment I would like to be able to enjoy high quality recordings from both camps in their own native formats. Converting from one format to the other at the user end sounds to me suspicious at best.

Now I don't mind having duplicate sources. Beyond that, as a would-be Meridian customer I could see three options in front of me; a) get DSP speakers for CD and DVD-A and another set of Preamp/Amps/speakers for SACD, b) get DSP speakers and wait for a solution to convert SACD to MHR/PCM which could be a non-event, unsatisfactory after all, or just too expensive, or c) forget about DSP speakers, route my processed PCM and DSD signals through an analog preamp and stay in the analog domain thereafter.

The last option appears to be the most reasonable. However, I am open for suggestions. I would appreciate it if anyone out there could recommend for me a complete two channel solution that is a) buildable within the next 6 months, b) allow playing SACD and 192kHz DVD-A in their native formats, and c) costs $20,000 to $30,000 with cables, power conditioning and all.

The system will be placed within a large multi-purpose hall that has a roughly trianglar shape with dimensions of 50' by 35' and a 10' ceiling and several open doorways. The expected speakers-listner distance is 15'.

Interestingly, the only high-end dealer available in Kuwait is a Meridian dealer who's a friend of mine. I have to travel for auditiong anything else (unlikely) or rely solely on published reviews and your kind recommendations.

I missed Kuwait on my last trip to the region. I spent several months in Saudi Arabia and a week or so in Iraq. That was late 90 early 91 if I recall correctly.

I honestly don't believe 192 kHz will be used much in the home. The people who claim it takes 192 kHz to sound as good as SACD are so full of it!

I understand your reason for wanting both formats. After all, you just want to listen to the music and probably don't care what format it is.

Personally I like DSP speakers and active speakers a lot. For movies, CD, and DVD-A, you can enjoy the DSP speakers. The only case you would want analog is for SACD. How many titles are out in the format that you want? If 2-channel SACD was enough, then a small analog system might work, but if you wan't multi-channel then that is another issue. No sense on having two complete 5.1 systems.

Remember that all analog inputs on the Meridian processor will get converted to digital. They do all of their work in the digital domain.

I agree with Stacy that support for 192mHz will be a long time coming, if at all, on Meridian gear. I believe Bob Stuart is on record that he thinks that 96mHz is overkill; that something between 48mHz and 96mHz is more than adequate. In fact, the 568 and 861 only upsample 44.1hMz material, i.e., your standard Redbook CD, to 88.2hMz. Only 48hMz material is upsampled to 96hMz. Meridian likes to keep the arithmetic simple, i.e., doubling the sampling rate, rather than upsampling 44.1hMz to 96hMz, which would introduce rounding errors. If he thinks 96hMz is overkill, I doubt he is going to spend a lot of time (not to mention DSP horsepower and bandwidth) on 192hMz.

>>What about 192 kHz support for the DSP speakers? How much will THAT cost? How will the potential increased jitter through the digital cable be handled? How would SACD signals be handled by DSP speakers without converting (downgrading) them to 96 kHz at the processor end? What if Meridian keeps dragging its feet on SACD support? Does that mean we have to buy a complete parallel high-end system from source to speaker to enjoy the full potential of SACD?<<

Well, there are no 192kHz Multichannel DVD-A discs and only a few 192kHz Stereo DVD-A tracks. So I don't think Meridian's plans to not support that format are going to be much of an issue for most folks.

As for SACD support, time will tell. There's supposedly a major upgrade of the 861 coming. So we may yet see things like a multichannel analog bypass in the 861 or perhaps SACD support in the 800 universal player.

Please don't get your hopes up, this is why they have the IA40.

I don't believe it is technically possible to offer an analog bypass on the 861. The back plane is digital. The analog input card converts the incoming signal to digital and then sends it down the bus. The analog output cards grabs the digital signal off the bus and then converts it back to analog.

The other little issue is the Meridian's Statement system, it consists of DSP speakers.

[This message has been edited by sspears (edited 07-19-2001).]

>>Please don't get your hopes up, this is why they have the IA40.

I don't believe it is technically possible to offer an analog bypass on the 861. The back plane is digital. The analog input card converts the incoming signal to digital and then sends it down the bus. The analog output cards grabs the digital signal off the bus and then converts it back to analog.<<

We'll see come September at CEDIA. If not, there are other ways to get the analog signal from a Multichannel SACD player to the amp besides the 861.

Adding such a thing as analogue pass through for support of SACD would be very foolhardy for current SACD players IMHO.

The 861 is a reference product, Meridian do DIGITAL, Meridian do MLP what possible benefit is there to Meridian or it's reputation by adding analogue passthrough? Especially is you are digital theatre, plus your passing analogue from todays current players, I think not.

If the content owners feel that the only way to protect their IP is to force analogue outputs then I for one will not be buy a SACD player anytime ever.

Is it just me that thinks analogue passthrough on a 861 is about as stupid as you can get (especially for SACD), Meridian may prove me wrong of course and I'll be eating these words(-:

Martin.

Or we will have to BBQ up some crow.

Meridian is one of the founders and creators of the DVD Audio specification. They of course will support DVD-A. I spoke with Meridian and they told me that if the market were to choose SACD over DVD-A then they would find a way to do it on the 861. However they were pretty sure that DVD-A content would far surpass SACD in quality and quantity. Especially since the only SACD's that I've seen are from Sony and they dont have enough SACD's available for the buying public with the artists that we are all looking for.

SACD/ DVD-A?

what was that which was playing when we entered the Meridian booth at CES and sounded so good?

not DVD-A
not SACD

it was MP3. MP3 is what the mass market wants.

Of course, then they put on the Doors DVD-A and, WOW!

No matter what happens with these formats- my CDs sound so good via the 861 today.

- Jerry

Just a quick note on sample rate conversion as my previous job involved plenty of this...

Converting from one sample rate to another integer multiple/divisor of that rate is relatively straightforward. To convert upwards, simply stuff zero samples between the samples you have and apply a low pass filter to fill in the gaps. To go the other way, you need to filter first, then throw away some of the samples.

This is trivial compared to what you have to do to convert between unrelated rates, eg. 44.1kHz to 96kHz. I'll give a couple of examples:

Take a 10ms snippet of music (ie. 0.01 seconds). This gives us nice round figures to use...

10ms at 44.1kHz is 441 samples.
10ms at 88.2kHz is 882 samples.
10ms at 96.0kHz is 960 samples.

Converting from 44.1 to 88.2 is fairly straightforward, as every even sample in the output corresponds exactly in time to a sample in the input. Another way of looking at it is that we only have to fabricate 1 sample for every two that we output.

Converting from 44.1 to 96 is horrendous. Take a piece of paper and mark out 441 points, spaced regularly. Below these marks, add 960 points, spaced regularly, OVER THE SAME DISTANCE. This piece of paper represents our 10ms snippet.

Look at the relationship between the input samples (the 441 points) and the output samples (the 960 points). You will see that the distance between a point on the 960 line and its nearest neighbours on the 441 line keeps changing.

What happens when we want to generate the 960 line from the 441 line? Well, we could copy samples from the 441 line if they are close to the required 960 point, filling the rest with zeros and doing the low pass filter (as we would to convert to 882), but you will see that the gaps between real samples is uneven, horribly uneven. This unevenness is known to us all as jitter. The sound would be terrible.

So, what we must do is implement a low pass filter that takes into account the relative distances from the required output sample to a number of input samples. In fact, we need a completely different filter for each distance. We noticed how the distances between the samples keeps changing, each change requires a new filter. In the worst case, we need 960 separate low pass filters (the problem is symmetrical, so we only really need 480). The sound quality of the output is directly related to the number of coefficients in each filter (say at least 40).

So, where are we? We have 480 filters of 40 coefficients each, that's 19200 coefficients we need to store somewhere, we also have to select the correct filter out of these 480, on top of all that, we need to do the DSP to actually implement the filter.

Bear in mind the fact that the product has many many other things to do as well. Also bear in mind that each of the 480 different filters will sound slightly different. Also bear in mind that buffer management is complete hell when your input and outputs are at unrelated rates. Also bear in mind that locking two unrelated clocks is difficult and jittery at best.

Believe me, you don't want to the processor to convert 44.1kHz to 96kHz.

I know I began with "just a quick note." I could have got into detail.


Forever yours,

Simon.