Ayre DX-5 bluray player - Page 20 - AVS Forum
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post #571 of 1442 Old 02-25-2010, 08:41 AM
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Originally Posted by Charles Hansen View Post

Please refer to:

http://www.avsforum.com/avs-vb/showt...8#post18203168

Thanks.

OK. So it looks like from a clocking / jitter perspective your player will best the Marantz and be the best HDMI digital transport around. For DSD sources, did you also improve on the standard DSD to PCM conversion of the Oppo?
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post #572 of 1442 Old 02-25-2010, 09:55 AM
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Originally Posted by tyree91 View Post

Charles, all other things being equal which transmission method do you find to be the best for PCM: SPDIF, Toslink, USB, AES/EBU, Firewire, HDMI (sic) or some other I forgot?

Charles, thanks for the great summary of these interfaces. Tyree forgot to ask about I2S, which a few firms are using between the transport and DAC.

If I understand correctly, I2S is the format used internally within a CD player. It seems logical to extend to use between a transport and DAC, but apparently I2S can only be used over short distances. Using an HDMI cable to to transmit I2S is apparently helpful.
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post #573 of 1442 Old 02-25-2010, 01:03 PM
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Originally Posted by GGA View Post

Charles, thanks for the great summary of these interfaces. Tyree forgot to ask about I2S, which a few firms are using between the transport and DAC.

If I understand correctly, I2S is the format used internally within a CD player. It seems logical to extend to use between a transport and DAC, but apparently I2S can only be used over short distances. Using an HDMI cable to to transmit I2S is apparently helpful.

At best, I would think that I2S would work as well as asynchronous USB and the analogous Firewire implementation (I think Pioneer calls it PQLS, Sony something else. It's a feature in the IEEE394 spec, IIRC). I have a QB-9, so I'm all set for 2ch music.

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post #574 of 1442 Old 02-25-2010, 03:42 PM
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Originally Posted by edorr View Post

For DSD sources, did you also improve on the standard DSD to PCM conversion of the Oppo?

No, that would require a lot of additional expense and not many people would use the feature. I don't know what is out there in terms of SSP's, but they should either accept DSD or else perform their own high-quality conversion. I'm sure that the conversion of the Oppo is fine, but if you are looking for the ultimate SACD playback then you will want an SSP that does "DSD" natively.

But maybe there isn't any such animal, I don't know. If you are making an SSP, you can only do bass management and time delays with PCM. Therefore there has to be a conversion somewhere. If all of the sources include it, why include it again?

This is one of the problems with this arena -- there are so many different customers, each with a different set of expectations and desires. But it's like the sign on the wall of the woodworker's shop: "Quality, Service, Price. Pick Two." If you included every single feature that every single user wanted, you would end up with $50,000 processor. And 98% of the customers wouldn't even use 98% of those features.

Case in point. The Theta Casablanca has incredibly flexible bass crossover slopes for the subwoofers. Do you use those?
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post #575 of 1442 Old 02-25-2010, 03:48 PM
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Originally Posted by GGA View Post

If I understand correctly, I2S is the format used internally within a CD player. It seems logical to extend to use between a transport and DAC, but apparently I2S can only be used over short distances. Using an HDMI cable to to transmit I2S is apparently helpful.

The place where jitter matters is at the DAC chip. I2S is a big improvement over S/PDIF as the clock is separated from the data. This allows for big reductions in jitter.

But every implementation of I2S I've ever seen is sub-optimal. They should have the clock in the DAC and send it back upstream to the source. This would give much better results.

Sending a high-speed digital clock over 1" of PCB trace isn't so bad. If you try to send it down 25' of cable, it is going to suffer degradation. It is still better than S/PDIF or non-ARC HDMI, but not as good as asynchronous USB or FireWire where the clock is right next to the DAC chip.

Using HDMI cable is just a low-cost way to send the signals. They could have picked CAT-5 cabling or whatever. One company just happened to pick HDMI because it was readily available at low cost and had enough wires to do the job.
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post #576 of 1442 Old 02-25-2010, 03:55 PM
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Originally Posted by PooperScooper View Post

At best, I would think that I2S would work as well as asynchronous USB and the analogous Firewire implementation (I think Pioneer calls it PQLS, Sony something else. It's a feature in the IEEE394 spec, IIRC).

For a brief time, Sony and Pioneer used a very complex method of sending multi-channel high-resolution audio via FireWire. They called it iLink and the license was $15,000 per year. It was good, but expensive, complex, and didn't play well with other brands (Denon had their own version).

Modern computers often use FireWire for audio, especially in the pro world. Just as with USB, there is adaptive and there is asynchronous. Adaptive (for either) is about like regular HDMI. Asynchronous (for either, although special device drivers are required for FireWire) is as good as it gets.

I2S would fall somewhere between the "gold standard" of HDMI with ARC, or asynchronous USB or FireWire, or the old iLink and the lower performing S/PDIF and regular HDMI connections. Exactly where depends on how good the implementation is.

But remember that we are talking about only one tiny aspect of the sound of a product -- its jitter performance. There are a thousand other things that also affect the sound. So while it is fun to talk about all this stuff, in the end the only way to know is to listen to them.
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post #577 of 1442 Old 02-25-2010, 04:15 PM
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Originally Posted by Charles Hansen View Post

No, that would require a lot of additional expense and not many people would use the feature. I don't know what is out there in terms of SSP's, but they should either accept DSD or else perform their own high-quality conversion. I'm sure that the conversion of the Oppo is fine, but if you are looking for the ultimate SACD playback then you will want an SSP that does "DSD" natively.

But maybe there isn't any such animal, I don't know. If you are making an SSP, you can only do bass management and time delays with PCM. Therefore there has to be a conversion somewhere. If all of the sources include it, why include it again?

This is one of the problems with this arena -- there are so many different customers, each with a different set of expectations and desires. But it's like the sign on the wall of the woodworker's shop: "Quality, Service, Price. Pick Two." If you included every single feature that every single user wanted, you would end up with $50,000 processor. And 98% of the customers wouldn't even use 98% of those features.

Case in point. The Theta Casablanca has incredibly flexible bass crossover slopes for the subwoofers. Do you use those?

None of the high end SSPs does DSD natively. So for me one of the wildcards in the HDMI versus MC analog debate is how much degradation is created by converting DSD to PCM. We'll see. Clearly for the ultimate SACD MC playback you need something like EMM labs and a switchman. I'm OK with 90% of the performance for 20% of the price. And you are absolutely right - I have spend zero time tinkering with the Theta bass crossover slopes.
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post #578 of 1442 Old 02-25-2010, 05:04 PM
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Originally Posted by Charles Hansen View Post

But every implementation of I2S I've ever seen is sub-optimal. They should have the clock in the DAC and send it back upstream to the source. This would give much better results.

IIRC, UltraAnalog and SonicFrontiers developed something like this back in the late 90s = I2S-Enhanced.

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post #579 of 1442 Old 02-25-2010, 06:53 PM
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Originally Posted by Kal Rubinson View Post

IIRC, UltraAnalog and SonicFrontiers developed something like this back in the late 90s = I2S-Enhanced.

Yep. It's been around for years. At that time their version was doomed because it only supported 44/16 and DVD was just coming out with music all the way up to 96/24. Muse Electronics had a competing version which didn't last long as they were the only company to use it. Both of those had a now-obsolete connector used for analog computer monitors, although the electrical standards were different! Then a few years later Camelot (long since out of business) made a version that used an S-video cable. I think we all know who is making the latest version of an I2S system, this time with an HDMI interconnect cable...

Plus ça change, plus c'est la même chose.
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post #580 of 1442 Old 02-25-2010, 07:03 PM
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Originally Posted by Charles Hansen View Post

Yep. It's been around for years. At that time their version was doomed because it only supported 44/16 and DVD was just coming out with music all the way up to 96/24. Muse Electronics had a competing version which didn't last long as they were the only company to use it. Both of those had a now-obsolete connector used for analog computer monitors, although the electrical standards were different! Then a few years later Camelot (long since out of business) made a version that used an S-video cable. I think we all know who is making the latest version of an I2S system, this time with an HDMI interconnect cable...

Plus ça change, plus c'est la même chose.

I know. I used to be a real I2S fan, even to the point of writing a few DIY articles about it. Still have some devices and cables in the closet, gathering dust.

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post #581 of 1442 Old 02-26-2010, 07:52 AM
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Originally Posted by edorr View Post

Clearly for the ultimate SACD MC playback you need something like EMM labs and a switchman. I'm OK with 90% of the performance for 20% of the price. And you are absolutely right - I have spend zero time tinkering with the Theta bass crossover slopes.

If for music you don't need any time delays, then you might be able to combine the front channels of the Ayre with the other channels from the SSP. The trick would be matching the volumes.

If you get an Ayre preamp for the front two channels, the volume steps are 1.0 dB throughout the range (except the older K-1xe). So if your SSP were able to "move" in 1.0 dB increments also, you could write a macro for a system remote (or a Crestron) that would keep all the channels in sync.

The other thing required is that we would need to add a mode that changed the stereo output from the normal mixdown to just the front L&R outputs. We'll do it if there is enough interest. There was another guy early on in the thread that wanted to do it also.
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post #582 of 1442 Old 02-26-2010, 09:29 AM
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If for music you don't need any time delays, then you might be able to combine the front channels of the Ayre with the other channels from the SSP. The trick would be matching the volumes.

If you get an Ayre preamp for the front two channels, the volume steps are 1.0 dB throughout the range (except the older K-1xe). So if your SSP were able to "move" in 1.0 dB increments also, you could write a macro for a system remote (or a Crestron) that would keep all the channels in sync.

The other thing required is that we would need to add a mode that changed the stereo output from the normal mixdown to just the front L&R outputs. We'll do it if there is enough interest. There was another guy early on in the thread that wanted to do it also.

It is an interesting idea. Just outputting the L&R should be easy because you simply need to play the multichannel track in multi channel mode, but send just the L/R output to your analog outputs. I assume the audio only clock would be engaged for the HDMI signal in this mode, so this configuration would not only improve the L/R channel, but also the SW, surrounds and center.

Cost would be a concern though - the additional outlay (over just using an Oppo 83 over HDMI into the processor) would be an the Ayre BR player and preamp, for a grand total of 12K. Now you are in EMM labs territory.

I think this idea makes perfect sense for someone that is buying the DX5 as their reference 2 channel player and use it with a SSP with HT bypass through (preferably Ayre preamp) for MC. Now, by implementing your suggested architecture for a very modest additional cost and some fiddling with remotes, they can take their SACD MC audio playback up a few notches by taking advantage of the DACs in the DX5 and analog circuitry in the preamp.

If on the other hand, (like me), you go down a different path (music server in my case) for 2 channel, this configuration would be cost prohibitive.
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post #583 of 1442 Old 02-26-2010, 11:51 AM
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Originally Posted by Charles Hansen View Post

I don't know what is out there in terms of SSP's, but they should either accept DSD or else perform their own high-quality conversion. I'm sure that the conversion of the Oppo is fine, but if you are looking for the ultimate SACD playback then you will want an SSP that does "DSD" natively.

But maybe there isn't any such animal, I don't know. If you are making an SSP, you can only do bass management and time delays with PCM.

Why can't you use a 1-bit FIFO for time delay of DSD? Dolby did that for their Time Link delay back before LPCM was fashionable. Worked fine.
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HDMI is not the worlds best protocol by any means. However, a DAC ought to be able to establish a good sampling rate from it, if the delivery system can stand the necessary latency.

I don't have any real need for SACD over things like DTS Lossless, running at 96khz if you must, but that's a seperate issue.

When S/HDMI can come up with the same latency every time one opens an audio stream, get back to me.

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post #585 of 1442 Old 02-26-2010, 08:04 PM
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Originally Posted by Charles Hansen View Post

This is only true if the customer never listens to music.

But there is no HDMI D/A converter on the planet that sounds even half as good as the analog outputs of the DX-5.

So if the customer only wants it for movies, that is a great plan. For years we have all been listening to 5.1 channels of low-bit-rate MP3, and it has been just fine. The visuals make the sound quality less important for watching movies.

Until someone makes an SSP with Audio Rate Control, discrete analog circuitry, and volume controls at least as good as our $3500 preamp (FET switches and metal film resistors), then anybody who listens to music even once in a while shouldn't mind spending a couple of thousand dollars extra for great music playback.

And they will be still be saving money. To make an SSP with the features noted above would probably cost $25,000 or more. Why spend that much money to get eight channels of incredible sound when eight channels of pretty good sound is all you need for movies? Then you can save your money for where it counts -- those two channels of music playback.

Maybe we will build a "transport" version someday. But only when someone (I guess it will be up to us, as nobody else seems to be willing...) offers a video switcher/scaler that offers galvanically isolated audio outputs and an SSP with the above features. Otherwise, what would be the point?

Until then, the optimal solution for someone who wants multi-channel and listens to music is to get a good two-channel preamp with processor-passthrough and the DX-5.

to chime in, this has certainly been my experience, dollar for dollar have always found analog 2ch pre amps to be better than ssp. Something with the classe 800 as well where the cp700 does a better job of 2ch analog. Thats not to say the 800 doesnt do a pretty good job with 2ch analog, just that at $8k us or $15k aussie which is what they sell for in oz you can buy a pretty damn good 2ch analog pre amp

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Until then, the optimal solution for someone who wants multi-channel and listens to music is to get a good two-channel preamp with processor-passthrough and the DX-5.

is what I do even though have an av processor that is very decent over analog for 2ch

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post #586 of 1442 Old 02-27-2010, 11:24 AM
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Why can't you use a 1-bit FIFO for time delay of DSD? Dolby did that for their Time Link delay back before LPCM was fashionable. Worked fine.

Sorry, Roger, I shouldn't have said "you couldn't" do delays with SACD. I should have said "you couldn't easily" do delays with SACD.

In a previous post I wrote, "The problem with "DSD" is that it is completely unwieldy to work with for any signal processing. Even a simple time delay is a giant pain. The signal is only 1 bit wide, so it is a mile long and coming in at 64x speed. To build a time delay would require a special buffer. Then you would need six of them. (Thankfully, there is no such thing as 7.1 channel SACD!)"

Nearly all specialist manufacturers need to buy either SSP "engines" or pre-programmed DSP chips to build an SSP. To the best of my knowledge, there is no "off-the shelf" solution that includes the possibility of anything for "DSD" signals.

Doing a 1-bit wide FIFO in an FPGA is quite easy. Doing it in a DSP chip is a giant pain in the neck. There is no memory in the DSP chip, it's all off-chip. So the DSP chip would have to take "chunks" of 1-bit DSD data and format it into (say) 32-bit "blocks" to send to the off-chip memory. Then when it needed it, it would have to re-serialize it to a one-bit stream.

Again, if it were easy, everyone would be doing it.

The biggest problem is that 99% of the customers with big home theater rigs don't give a fig about multi-channel SACD. So there is very little incentive to put all the time and money into developing this kind of thing, especially when there are a million other new technologies that you are forced into keeping up with. 3D video, anyone???
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post #587 of 1442 Old 02-27-2010, 11:33 AM
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Originally Posted by jj_0001 View Post

HDMI is not the worlds best protocol by any means. However, a DAC ought to be able to establish a good sampling rate from it, if the delivery system can stand the necessary latency.

EDIT: Welcome to the thread, JJ!

Clearly the DAC's are able to establish the sampling rate. The question is whether they can regenerate a low-jitter audio master clock from the questionably jittery video (TMDS) clock.

The traditional way to do this has been to use PLL's with longer and longer time constants. Of course, these take longer and longer to lock onto the signal, and depending on what approach is taken may add greater and greater latency. Obviously latency is unacceptable for combined audio/video streams.

Definitely a sub-optimal system from the standpoint of low-jitter audio. Audio Rate Control effectively solves the problem, but the only solutions to date are proprietary ones from Sony and Pioneer.

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I don't have any real need for SACD over things like DTS Lossless, running at 96khz if you must, but that's a seperate issue.

I'll let you argue with your customers as to whether or not it is needed. But Sony created a demand for it when they were trying to keep from losing their $1 billion per year royalty stream from the CD patents. And since they are the biggest force in the Blu-Ray consortium, it is somewhat surprising that there is no provision for it under the Blu-Ray spec.

I guess they just decided to throw it under the bus, unlike the folks at the HDMI group (chiefly, Silicon Image, Intel, and oddly enough, Sony) who went out of their way to include a way to transmit "DSD"....
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post #588 of 1442 Old 02-27-2010, 11:38 AM
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Originally Posted by alebonau View Post

to chime in, this has certainly been my experience, dollar for dollar have always found analog 2ch pre amps to be better than ssp.

It just stands to reason.

The only places where opinion come in is:

a) Whether or not the SSP is close enough to the stereo preamp that the difference doesn't matter. This largely seems to be a question of whether one is willing to spend a few thousand dollars for a good stereo preamp and dedicate extra shelf space for the additional box.

b) Whether or not multiple lower quality channels will provide a better listening experience than two higher quality channels. This seems to boil down to a question of whether one is satisfied with perhaps 1000 titles of (largely) classical music or whether one wants to listen to all kinds of music.
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post #589 of 1442 Old 02-27-2010, 01:15 PM
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I would expect the Oppo/Theta combination to sound better based on two things:

a) The Six-Shooter is a great volume control, but to the best of my knowledge, is identical to the one that is already inside the rest of the Casablanca. The only reason they made the Six-Shooter was because they didn't have any way to send certain types of data digitally, such as "DSD", five years ago. But now they do -- HDMI.

I have a Marantz SA7s1 with the Six Shooter. The difference between the sound quality that produces compared to ANYTHING that Casablanca with Xtreme dacs can do even when feed a 24/96 signal is a beyond huge. I like the Casablanca and know the Six Shooter uses the same parts. However even using the two channel internal analog bypass of the Casablanca and the Marantz SA7 the sound is VASTLY inferior to the Six Shooter . I mean, as much as I like the Casablanca, compared to the Six Shooter and Marantz SA7s1, the sound quality of the CBIII is garbage.

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post #590 of 1442 Old 02-27-2010, 01:32 PM
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Originally Posted by Charles Hansen View Post

Sorry, Roger, I shouldn't have said "you couldn't" do delays with SACD. I should have said "you couldn't easily" do delays with SACD.

In a previous post I wrote, "The problem with "DSD" is that it is completely unwieldy to work with for any signal processing. Even a simple time delay is a giant pain. The signal is only 1 bit wide, so it is a mile long and coming in at 64x speed. To build a time delay would require a special buffer. Then you would need six of them. (Thankfully, there is no such thing as 7.1 channel SACD!)"

Nearly all specialist manufacturers need to buy either SSP "engines" or pre-programmed DSP chips to build an SSP. To the best of my knowledge, there is no "off-the shelf" solution that includes the possibility of anything for "DSD" signals.

Doing a 1-bit wide FIFO in an FPGA is quite easy. Doing it in a DSP chip is a giant pain in the neck. There is no memory in the DSP chip, it's all off-chip. So the DSP chip would have to take "chunks" of 1-bit DSD data and format it into (say) 32-bit "blocks" to send to the off-chip memory. Then when it needed it, it would have to re-serialize it to a one-bit stream.

Again, if it were easy, everyone would be doing it.

The biggest problem is that 99% of the customers with big home theater rigs don't give a fig about multi-channel SACD. So there is very little incentive to put all the time and money into developing this kind of thing, especially when there are a million other new technologies that you are forced into keeping up with. 3D video, anyone???

hi charles I have a sony scd-xa9000es cd/sacd player that I beleive utilises a DSP specially sony designed chip that "processes the Direct Stream Digital signal in its 1-bit form, using technology similar to the professional editing systems currently used in the studio to produce Super Audio CDs." for level setting and bass management.

see below

http://www.cdrinfo.com/Sections/Revi...13116&PageId=3

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- Multi Channel DSP

Home theater speaker configurations vary considerably. Some enthusiasts have built upon audiophile-grade stereo systems. These systems may have large, full-range Left and Right speakers that produce bass so deep that any subwoofer would be extraneous. In this case, the Left and Right speakers may well be considerably larger than the Center and Surround speakers. Other systems may have five matching satellite speakers, plus a subwoofer. Some systems may have no Center channel speaker. The SCD-XA9000ES has a Digital Signal Processing (DSP) Large Scale Integrated Circuit (LSI) to achieve optimal multi-channel reproduction with all these speaker configurations. The DSP processes the Direct Stream Digital signal in its 1-bit form, using technology similar to the professional editing systems currently used in the studio to produce Super Audio CDs. The DSP accomplishes three functions when using the analog outputs:

Bass redirection. You can optimize the SCD-XA9000ES output to work with your specific speaker configuration.

Channel balance. The Sony ES player can also accommodate differences in speaker efficiency, adjusting the balance between front/surround, front/center and front/subwoofer speakers.

Test tone. The SCD-XA9000ES is equipped with a test tone oscillator to confirm connection status and channel balance adjustments.

It's easy to access these functions through the front panel menu key. And you can make critical channel-balance adjustments from your actual listening position, using the supplied remote control.

If you have an ideal speaker layout, you can use the Direct Mode, which completely bypasses the DSP circuit. Other options include 2-Channel Direct and 2-Channel Direct + Subwoofer, ideal for stereo SA-CD playback.

Engaging Multi-Channel Management selects among seven different speaker configurations, each with adjustable channel balance.


It is a fantastic player over analog in my opinion. Something I will never let go

"Technology is a drug. We can't get enough of it."


Welcome to my lounge room :)
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post #591 of 1442 Old 02-27-2010, 04:12 PM
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Sorry, Roger, I shouldn't have said "you couldn't" do delays with SACD. I should have said "you couldn't easily" do delays with SACD.

Understood. Thanks for the explanation.

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The traditional way to do this has been to use PLL's with longer and longer time constants. Of course, these take longer and longer to lock onto the signal, and depending on what approach is taken may add greater and greater latency. Obviously latency is unacceptable for combined audio/video streams.

Techniques for achieving fast PLL lock up through the use of dual time-constants have been around for decades for radio tuners. Wouldn't the same idea work here?

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Sony created a demand for [SACD] when they were trying to keep from losing their $1 billion per year royalty stream from the CD patents. And since they are the biggest force in the Blu-Ray consortium, it is somewhat surprising that there is no provision for it under the Blu-Ray spec.

Pure speculation on my part, but that's the difference between a proprietary format (MiniDisc, SACD) and a consortium formulated format. Every added cost is given higher scrutiny and requires more stringent justification.
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post #592 of 1442 Old 02-28-2010, 07:14 AM
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Wmean "cost no object". One excellent choice would be to purchase four of these:

http://www.placetteaudio.com/passive_line.htm

for a mere $1,700 each for a grand total of $6,800. Another poster earlier in this thread had eight of the single-input mono versions and was quite pleased.


I have had one of those over at my house. They sound almost identical to the Six Shooter. Clean,transparent but not quite as dynamic as some other pre-amps which I think describes both of them.

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post #593 of 1442 Old 02-28-2010, 07:21 AM
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.


The Ayre HDMI audio-only is somewhat different, as it is not just for convenience. We have a super-low-jitter master clock 1/8" away from the HDMI transmitter chip. When the board is engaged, the audio master clock shuts down, and the entire player is slaved to the HDMI audio clock. This provides the absolute lowest jitter possible for conventional operation.

In addition, the Ayre board supports ARC. So when the SSP makers wake up and build something that supports ARC, jitter will be reduce by a factor of at least 10x and possibly 100x.

Ahhhh. This is why I always try to support the hi-end. No one but designers like you are going to really push the state of the art. I remember the first time I saw the D1. You redefined what could be done with DVD. I can't wait to see what you do with this player.

Never become so involved with something that it blinds you.
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post #594 of 1442 Old 02-28-2010, 09:40 AM
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I have a Marantz SA7s1 with the Six Shooter. The difference between the sound quality that produces compared to ANYTHING that Casablanca with Xtreme dacs can do even when feed a 24/96 signal is a beyond huge. I like the Casablanca and know the Six Shooter uses the same parts. However even using the two channel internal analog bypass of the Casablanca and the Marantz SA7 the sound is VASTLY inferior to the Six Shooter . I mean, as much as I like the Casablanca, compared to the Six Shooter and Marantz SA7s1, the sound quality of the CBIII is garbage.

OK, Edorr, I was wrong!

Don't get the HDMI option! I believe that the Six-Shooter operates as a "passive preamp" without any active circuitry. If this is the case, then it would explain why going into the two-channel internal analog bypass would sound worse than the Six-Shooter. The analog circuitry simply must not be up to snuff.

So clearly you will get better sound with the multi-channel audio with the Six-Shooter. As long as you can also digitize the analog signal for bass management and time delays for watching movies, then it sounds like that will be better than going with the HDMI route...

Good luck with your quest!
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post #595 of 1442 Old 02-28-2010, 09:48 AM
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hi charles I have a sony scd-xa9000es cd/sacd player that I beleive utilises a DSP specially sony designed chip that "processes the Direct Stream Digital signal in its 1-bit form, using technology similar to the professional editing systems currently used in the studio to produce Super Audio CDs." for level setting and bass management.

It's fun for the marketing types to make up silly techno-speak. After all, that's all that SACD ever was -- "Direct Stream Digital" or "DSD" is NOT an engineering term, it's a marketing term. It doesn't mean anything in particular except whatever the marketing department feels like on any given day.

But for the record, it is not possible to do any type of signal manipulation in the pure "DSD" mode except for time delays. Even changing levels will require a conversion to PCM and then back to "DSD". Some signal conversion sounds better than others, but it still requires two signal conversions.

The main thing is if you are happy with the sound quality. If so, then don't worry, be happy. Spend your money on concert tickets and other music.
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post #596 of 1442 Old 02-28-2010, 09:57 AM
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Techniques for achieving fast PLL lock up through the use of dual time-constants have been around for decades for radio tuners. Wouldn't the same idea work here?

Yes, of course. It just adds to the cost and complexity of the system.

With S/PDIF and HDMI audio, jitter is built into the transmission link. One can never completely eliminate jitter once it's there, but rather filter it out through various techniques. Basically, the more money you throw at it, the better the performance can get.

A big problem is that some of the best techniques introduce a lot of latency, which is unacceptable for A/V applications. That is why many companies were thrilled when Asynchronous Sample Rate Conversion was developed. Data processing power has become so cheap that a chip can throw out all of the original data and calculate new data for what it thinks it would have been if there were no jitter in the incoming signal.

The algorithms work well enough to fool the measuring instruments, but the audible results are not so clear-cut. But since it measures well and is cheap, its use is becoming very widespread.

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Pure speculation on my part, but that's the difference between a proprietary format (MiniDisc, SACD) and a consortium formulated format. Every added cost is given higher scrutiny and requires more stringent justification.

That plus the fact that by now SACD has been proven to be a commercial flop. Very few customers are offended by the fact that there is no support for SACD. I'm sure Sony wishes they had never thought up the idea in the first place. I'm sure they lost at least $100 million on that one -- money they could desperately use right now.
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post #597 of 1442 Old 02-28-2010, 10:02 AM
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Ahhhh. This is why I always try to support the hi-end. No one but designers like you are going to really push the state of the art. I remember the first time I saw the D1. You redefined what could be done with DVD. I can't wait to see what you do with this player.

That's when the rubber meets the road! We'll all collectively see how it is received. Some products are hits and some are misses.
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post #598 of 1442 Old 02-28-2010, 10:06 AM
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OK, Edorr, I was wrong!

Don't get the HDMI option! I believe that the Six-Shooter operates as a "passive preamp" without any active circuitry. If this is the case, then it would explain why going into the two-channel internal analog bypass would sound worse than the Six-Shooter. The analog circuitry simply must not be up to snuff.

So clearly you will get better sound with the multi-channel audio with the Six-Shooter. As long as you can also digitize the analog signal for bass management and time delays for watching movies, then it sounds like that will be better than going with the HDMI route...

Good luck with your quest!

I'm no engineer, but if the six shooter needs AC power it cannot be entirely passive - am I wrong? Either way, very timely advise - I just sold my Sony XA5400 and Theta Compli and was about to hedge my HDMI/Analog bet by ordering an Oppo Nuforce. Sounds like I should probably get the Marantz UD9004 after all and sell the Xtreme cards of my CBIII.....

Bass management and time delays can be done in the Marantz. It would be nice if the Marantz was 3D upgradeable. Not that I care for myself, but it will definitely help its resale value. I have not found a definitive statement about this.
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post #599 of 1442 Old 02-28-2010, 11:50 AM
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I'm no engineer, but if the six shooter needs AC power it cannot be entirely passive - am I wrong? Either way, very timely advise - I just sold my Sony XA5400 and Theta Compli and was about to hedge my HDMI/Analog bet by ordering an Oppo Nuforce. Sounds like I should probably get the Marantz UD9004 after all and sell the Xtreme cards of my CBIII.....

Bass management and time delays can be done in the Marantz. It would be nice if the Marantz was 3D upgradeable. Not that I care for myself, but it will definitely help its resale value. I have not found a definitive statement about this.

My understanding is that the Six-Shooter has FET switches. Basically, any kind of remote-controllable switch will need power. But I always thought that the Six-Shooter was essentially a passive preamp, just Bulldogger posted.

You still may want to get the Oppo SE. Has anyone done a comparison of that with the Marantz? Even if the Marantz is better, is it enough better to justify the price difference? I have no experience with either product, so I can't help you here.

I doubt that any existing player will be upgradeable simply by changing firmware. But that would be another reason to go with the Oppo SE. If the players actually do lose resale value because of some vaporware feature that people may or may not buy into, then you would lose less with the less expensive player.
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post #600 of 1442 Old 02-28-2010, 03:38 PM
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There have been requests for photos and so forth. We don't have any yet, as we won't schedule a photo shoot until we begin work on the brochure. In the meantime there are a couple of photos online. Stereophile has a picture from CES at:

http://www.soundstagenetwork.com/las...large/ayre.jpg

and SoundStage has a picture at:

http://www.soundstagenetwork.com/las...large/ayre.jpg

(I would have liked to link to the article, but it is buried inside some JavaScript that I can't link to directly.)

Finally, here is a drawing of the rear panel in PDF format.

 

dx5rear.pdf 27.7978515625k . file
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