Chord Dac64 - differences of opinion?

Paul V said:
Are you saying that fed with a reasonable signal the DAC64 should sound identical whether set to 0, 2 or 4 second delay ??.
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That's what logic says should be the case, and it's what the designer says is the case.

Some people have of course reported hearing differences :)
 
Isaac - thanks, I was beginning to believe I had cloth ears !!. After blowing a year's spending money on new amp and speakers, it's a big relief I couldn't hear a big difference...

Paul
 
Paul V said:
Isaac - thanks, I was beginning to believe I had cloth ears !!. After blowing a year's spending money on new amp and speakers, it's a big relief I couldn't hear a big difference...
Paul
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But what about your amp and speakers?

Let's be totally brutal and also fair in good measure. We all generally have a system that does not reproduce accurately the original source, and more importantly, speakers that can never get you even close to realistic scales and dynamics. I would say that most specialist systems give somewhere between 5-10 % of a real-life performance. Indeed, these systems can be very entertaining when musically coherent, but never close to the effortlessness nature and ambient separation of a real life performance. I was only saying today, that an Active ATC setup probably gets you within 15-25%, but it's still a far cry and largely restricted to the recording as much as the speaker drivers and cabinet etc. In reality, the hope for us all is rather slim, and by learning to make compromises is the key to living happily with any system.

'A system that produces a smooth and enjoyable sound for £200 is probably more viable than a CD player that leaves 50% of your CD collection gathering dust on the shelf.'

Even the older ATC's I heard today couldn't reproduce the modern production of Damian Rice ââ'¬Å"Oââ'¬Â album without running into tweeter distortion problems. Who needs these problems when all we want is to enjoy the music? Maybe the Japanese were right about producing mid-FI, it works generally for everyone.
 
Shurely shome mishtake?

The Japanese are responsible for some of the greatest HiFi equipment on the planet!!!
 
s'right!

Ive had some lowfi british stuff through the years. low quality control and poor sound :S

There were times when Japanese products were generally thought to be superior..
I remember in the 80's when Curry's labelled its own brand 'saisho' to make it sound Japanese - so that people would think it was better.
 
I've been listening to the DAC 64 side-by-side to the Exposure CDP over the past few days, and struggle even more to understand how anyone could describe the DAC64 as 'harsh'. There's not much wrong with the Exposure, a very 'analogue' sounding player, but the DAC 64 betters it in terms of detail and digs out more bass. It has a bit more punch than the Exposure, but harsh? Geddouttahere!
 
The 64, or dac1 for that matter, are NOT harsh. They are an open window. There's no character to speak of. Any harshness is coming from elsewhere.

Some are just used to their previous system's warm fuzzy distortion. When that veil is lifted they assume it's the new dac's sound. Wrong! It's the lack of the previous garbage somothering things over.

It's like wandering up in the attic with a torch, and realising there's a load of crap up there. When before you assumed it was all tidy. It's not the torch's fault!!!
 
The best combination I heard with the DAC64 was when it was fed by an Accuphase dp-75V via optical cable (transports do make a difference even when using optical connection).

The sound of the DAC64 is to my ears compressed... like radio sound is... not necessarily harsh if the system is somehow veiled, which is usually the case with most of the "high-end" stuff.

Interesting that Isaac has said that 0, 2 and 4 sec. makes no difference whatsoever in the sound... and even more important that in 0 sec. position the DAC still is buffered.

In fact, the working principle of the DAC64 relays entirely on the buffer mecanism... it doesn't work without it!

The buffer "time" length does make difference... with 4 sec. you save more samples per second which are then successively delayed by one bit (from the first buffered sample to the latest) and finally added to form the output bitstream... it is like interleaving the samples' bits in order to have a more gentle wave shape and overcome the pronounced "digital steps"... the greater the number of samples, the less pronounced those steps will be...

What is the final result then?!

You gain some "artificial" low level resolution (because you are averaging over a certain amount of time or a number of consecutive samples) at the expense of dynamics... hence the word "compressed".
 
BerylliumDust said:
The buffer "time" length does make difference... with 4 sec. you save more samples per second which are then successively delayed by one bit (from the first buffered sample to the latest) and finally added to form the output bitstream... it is like interleaving the samples' bits in order to have a more gentle wave shape and overcome the pronounced "digital steps"... the greater the number of samples, the less pronounced those steps will be...

What is the final result then?!

You gain some "artificial" low level resolution (because you are averaging over a certain amount of time or a number of consecutive samples) at the expense of dynamics... hence the word "compressed".
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Not my understanding of how the DAC64 works. I understand the buffering to be there to remove the effects of clock variation/jitter - with buffering the data can be clocked out and into the D-A convertor based on the precision/stability of the DAC's clock. The clock accuracy no longer relies upon how well it is recovered from incoming spdif signal.

(As I understand it) Anything to do with filtering/taps etc. is then performed external to the 'buffer' length, within the DAC part of the chain (as it is with any traditional DAC) not as part of the 'clock' part of the chain.

Does this sound right Isaac?
 
BerylliumDust - no offence, but I think I will take Rob Watt's opinion of it over yours.

The thing that makes the DAC64 sound different to other things is the way that the filtering is done with the WTA filter. This is, as Chris suggested, entirely separate to the buffering function.
 
I read an article on how it worked ages ago. One important aspect was to do with the data stream length. Chord used their own unit, as it was a tidy multiple of the clock speed, or some such. Anyway the idea was to keep the maths simple for the DAC, so making a decent result far easier to achieve. Clear as mud. Sorry about my kack summary. Anyway, the long and the short of it was the 4 second buffer was best.
 
Isaac Sibson said:
The thing that makes the DAC64 sound different to other things is the way that the filtering is done with the WTA filter. This is, as Chris suggested, entirely separate to the buffering function.
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Isaac,

WTA filter is something very closely related to what I've just described... when you average (with a moving average scheme) by delaying and summing each data stream you are in fact filtering!

The buffer allows storing those data streams for ulterior processing. With a longer buffer you can process (not save, as I wrongly said before) more data streams per second. It only depends on the re-synchronization master clock.
 
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