Meridian Unplugged - Powered by MrTechGuy.co.uk Forums Discussion Meridian - Systems Meridian 568 and DVD-A / SACD

Brian,

You understand wrong! 1 bit technology, oversampling! If there is an academic out there that want's to be a laughing stock, then attempt to publish, review normally weeds out the rubbish.

And where is this paper to be published at an AES conference! The loud noise you hear is me falling off my chair with laughter!

Martin.



[This message has been edited by mgarthwaite (edited 03-19-2001).]

The Sony 9k has newer DSD DACs. I believe these were from Sony.

Not sure about the rumor on the Pioneer converting DSD to PCM. From the 2nd hand info, I hear both BurrBrown and Crystal make DACs that will accept either format and will decode in the native format. (again 2nd hand info so I can't verify) This would mean that the AX10 may actually decode DSD without converting to PCM.

People have really chosen sides on this one. The one magazine that supports DTS now seems to also support DSD.

The only real loosers will be the consumers who get stuck with the dead format, which ever it may be.

Martin, hopefully you don't have hardwood floors, other wise you may hurt yourself.



[This message has been edited by sspears (edited 03-19-2001).]

More rumours. Just get these information from magazines.

SACD multichannel will also include a 2-channel down mix to allow all existing players (SACD) to play the new multichannel disc.

The accuphase do also convert the SACD to PCM now and they must use the RJ45 connector. May be they can add an extra card (not sure).

The coming Luxman DVD player will support DVD-A and SACD multi-channel decoding in their native decoding format (No DSD->PCM).

I hope Meridian will add the SACD multichannel digital support. No matter it's proprietary (rumoured MMHR) or standard(1394). Multiple stage A/D and D/A does not make much sense and it's costly for the cables.

Unless room correction software Meridian is going to have is not capable of handling DSD properly I strongly recommend them to keep DSD all the way. Otherwise, a multi-channel fully analog pre-amp must be on the way.

Just my 2 cents.

-Dennis

Dennis,


You absolutely right, but if I were Meridian I would not bother, the architecture of Meridian processors requires PCM.


DSD makes it almost impossible to post process the signal in digital form, as we all know Meridian software engineers work wonders with Trifield, DPLII and all the other modes, PCM is the ideal format for this.


IF enough people are prepared to pay for the multichannel analog inputs from SACD players to be turned back into PCM and processed great, but you would need a 6 channel input with high quality ADC, and the source signal is only as good as the SACD's player internal DAC's CRAP in Sony's case.


This picture does not look good.


Martin.

[This message has been edited by mgarthwaite (edited 03-20-2001).]

The papers on DSD vs PCM are mentioned in the Letters to the Editor section of Stereophile Magazine a few months back. The author of the letter was Sony's Super Audio Project Manager. So I assume that will be happening at one of the upcoming AES meetings.

As for the DACs, my understanding was that the Philips DACs mentioned were actually used in the Marantz SA-1. The DACs in the Sony SCD-1 are Sony made (CXA8042AS 837001).

Brian,

The more I dig the sadder this situation becomes.
http://www.aes.org/sections/uk/meetings/0499.html

This was not a published paper but a meeting in the UK in 1999 with one Sony and one Philips guy. I would loved to have been at this event because I would have given them hell

Peter Eastty, Chief Consultant Engineer at Sony Brodcast Europe.

"In PCM coding,the sample rate is generally only about twice the required signal banndwidth (although DVD-Audio allows sample rates up to 192kHz)"

I can not believe what I'm reading here, this guy is questioning Shannon! Information theory states that the bandwidth MUST be at least twice the sampling rate to prevent aliasing.

Shannon's law (even I can understand this, and I just got O'Level math)

C=BW log2(1 + S/N)
BW = Bandwidth
S = Signal power
N = Noise power

Peter then goes on to say

"A DSD coded audio signal consists of a data stream at 2.8224Mbit/s"

Followed by "albeit with a bandwidth of 1.4MHz" Call me stupid but this guy has got it round the wrong way, the bandwidth is 2.8224Mbit/s and the information rate is 1.4112Mbit/s.

If Peter is to present a paper I need to be there to see him laughed off the stage.

Martin.

I ran across this Sony DSD Technical Proposal several years ago. It speaks only to the technology itself and not to implmentation, but you may find it interesting.

http://www.sel.sony.com/SEL/consumer/dsd/dsd.pdf

Jeff

Sorry to call you stupid, Martin, but you have it around the wrong way.

He was stating that the 2.8224Mbit/s data stream gives an analogue bandwidth of 1.4112MHz (the Nyquist limit- sampling theory).

This statement is missing a fundamental point about DACs.

In the beginning, there was the R-2R ladder. This allows a digital code to be converted directly into an analogue voltage. The disadvantage is that the resistors have to be very accurate. The limit before the error in the resistors outweighs the LSB is about 12 bits (before things get stupidly expensive).

There had to be another way.

The answer was the oversampling sigma-delta DAC. This works by converting the incoming multi-bit PCM stream into a very fast 1 bit stream. This 1 bit stream has the property that its frequency spectrum looks rather like the original audio signal- with a big pile of quantization noise thrown in (you can't get from 24 bits to 1 bit without introducing noise). Taking care with the encoding mechanism allows most of this quantization noise to be pushed up to ultrasonic frequencies (noise shaping).

All we need now is a simple low pass filter to cut off all this ultrasonic rubbish and we have what seems to be a multibit DAC.

As we have seen, the theoretical maximum bandwidth of a sampled system is one half the sampling frequency. To be able to push the quantization noise out of the audible band into ultrasonic territory requires substantial oversampling. DSD uses 64 times oversampling, ie. the 1 bit sampling clock is 64 times the audio sampling clock (128 times the audio bandwidth).

Soooo.... In stating that the DSD audio stream represents an analogue bandwidth of 1.4MHz, our author making a true statement. This statement is (possibly deliberately) misleading, however. As I have mentioned, most of the high frequency information in this bandwidth is ultrasonic noise, which MUST be filtered out to prevent everyone's tweeters from melting.

The effective sampling rate of our 2.8224MHz DSD stream is exactly 44.1kHz. This is substantially lower than DVD-A's 96kHz or 192kHz.


A little note about oversampling: modern sigma-delta DACs have taken advantage of smaller and faster microchip technology and have moved on from 64 times oversampling to 256 times (who knows what the future holds?).

A little note about 1 bit conversion: I mentioned in a previous post that modern DACs use multibit conversion to achieve better linearity, and mentioned an AES paper on the subject. That paper proves that it is impossible to properly linearise (dither) the output of a 1 bit sigma-delta DAC. Indeed, DAC manufacturers have been using 2, 3 and 4 bit conversion for a number of years. This allows proper dither to be applied to make the output that bit more beautiful.

So there it is, in grey and black. My chief complaint with DSD is that it is stuck in the past. My other major gripe is that it is almost impossible to do any signal processing to it in the digital domain.


Cor - verbal diarrhoea.

Simon.

Simon,

Thank you for your very enjoyable post. Thanks for clearing that up, I thought that the bandwidth is the maximum that the given channel can support.

Martin.

>>SACD multichannel will also include a 2-channel down mix to allow all existing players (SACD) to play the new multichannel disc.<<

Actually SACD Multichannel discs have separate Multichannel SACD and Stereo SACD mixes. There is no downmixing.

If there's no downmixing, how does the studio produce the 2 channel track?

Simon,

If the DACs in our processors (or DSP's) are 1-Bit delta-sigma's what difference does it make if the source is DSD or DVD-A?