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post #31 of 48 Old 12-07-2010, 11:31 AM - Thread Starter
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Tony,
You have a business relationship with Esoteric. Please call Mark at Esoteric and ask him to read the Soundonsound paper, referred to by Roseval earlier in the thread, and write a response in this thread. Also, please read it yourself and share your opinion.
I do believe you are enjoying your new equipment, don't let the cynical responses scare you away. Cynical behavior is not constructive. Those guys seem knowledgeable and could add a lot to this discussion if they stuck to the topic.

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They won't stay in business long if they don't know how far to mislead their customers and it varies depending on customers which has to be sensed out right away. That's what determines salesman's skill.

My dealer lets me try out equipment. If I don't like it I don't buy it.
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post #32 of 48 Old 12-07-2010, 11:48 AM
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I always wonder about the use of an external master clock with a single-unit transport-DAC combination.

An external DAC ought to generate its internal clock from the input signal with a high degree of accuracy — colour TV has been doing that at higher frequencies than digital audio uses since the 1950s, originally with very basic vacuum-tube circuitry — but I can see that a user might not trust that function completely. Of course, when using a master clock in that application, one has to be careful of the propagation delays between the external clock & the transport, external clock & the DAC, & the transport & the DAC.

When a single box is involved, though, the implication is that the internal master clock is no good, & needs an external reference of higher precision. Frankly, that's disturbing. Just build your one-box machine with a crystal oven in it, which isn't all that expensive in fact, & get it over with. A temperature-controlled crystal oscillator running at a good multiple of the necessary clock frequency, paired with a frequency divider chain, will give the highest stability you can get outside of an atomic clock (which can be implemented, in principle at least, by using a GPS receiver as a clock source).

The thing is, though, even a cheap ceramic resonator, not to speak of a crystal, has high enough short-term stability to prevent clock errors ocurring between the transport and the DAC in a single box, assuming the circuit layout is competent. Long-term drift will affect the pitch of the audio, if to an insensible degree, but it is irrelevant as far as DAC performance & perceptible output sound characteristics are concerned.


Unfortunately, the placebo effect in high-end audio is very strong.

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post #33 of 48 Old 12-07-2010, 11:36 PM
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Originally Posted by Sheer Lunacy View Post

I always wonder about the use of an external master clock with a single-unit transport-DAC combination.

An external DAC ought to generate its internal clock from the input signal with a high degree of accuracy

There seems to be a massive misunderstanding of modern technology in how such systems operate and what clock domains are present.

In the vast majority of modern "DAC" units i've seen, internally they look like:

input signal -> sampling rate converter -> DSP/microcontroller -> DAC -> output filters.

Under this structure, the clock of the input signal is generally not used for anything but clocking the data stream of the input signal. The SRC or DSP determine the sampling rate and bit resolution of the input signal by embedded header information in many formats or by empirical analysis of the signal in the case of something like pure PCM. At this point the DSP/microcontroller sets up the clocks for the actual DAC based on this information. The source of the clock for the DSP/microcontroller, the SRC, and ultimately the DAC is an on board oscillator, the input clock does not influence the system clock at all. Of course you could build a system that runs off the extracted input signal clock but there is little to no reason to do so unless you are really in dire need of saving $0.20 on the build cost by not putting a crystal/oscillator on board. The existence of modern SRC's and controller made such designs completely unnecessary.

The only place where this "external clock" concept even makes any sense is in the case of extremely low latency recording as running multiple clock domains generally implies buffers, however small they may be, this of course was mentioned like 30 post ago by ap1 but appears to have been ignored.

If you want to be constructive in this thread, how about someone provide a detailed analysis of what an external clock is supposed to do for sound quality?

Honestly i'd be really, really interested in what these devices even do with an external clock, i wouldn't at all be surprised if half of em don't even use it and the difference you see in performance is a result of changes in system noise as a result of having 2 similar, but not the same clocks bouncing around. If you think jitter is a problem, send a data stream and a clock through two completely different cables of different lengths at moderate speed. There is a reason to use encoding schemes that include the clock in the data stream.
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post #34 of 48 Old 12-08-2010, 02:33 PM
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Originally Posted by xianthax View Post

There seems to be a massive misunderstanding of modern technology in how such systems operate and what clock domains are present.

In the vast majority of modern "DAC" units i've seen, internally they look like:

input signal -> sampling rate converter -> DSP/microcontroller -> DAC -> output filters.

Under this structure, the clock of the input signal is generally not used for anything but clocking the data stream of the input signal. The source of the clock for the DSP/microcontroller, the SRC, and ultimately the DAC is an on board oscillator, the input clock does not influence the system clock at all. Of course you could build a system that runs off the extracted input signal clock but there is little to no reason to do so unless you are really in dire need of saving $0.20 on the build cost by not putting a crystal/oscillator on board.

Unless the clock for the DAC is locked to the average clock rate of the input signal, though, you're going to get a slight frequency mismatch, resulting in the equivalent of buffer underruns or overruns. If the source is clocking out 44 099.9 samples per second, & the DAC is clocking out 44 100.1 samples per second, on a long-time average, you're going to have a glitch every five seconds. Maybe that's covered up by interpolation, but it's easy to prevent that in the design phase.

True synchronous operation isn't required, so the two clocks can drift quite a bit on a short-time average (although that would result in various forms of distortion) as long as the input clock is regenerated from the input signal. Again, there's no difficulty at all in doing this. An SPDIF signal is about 1.5 MHz total, whereas the NTSC subcarrier is 3.58 MHz with 1.5 MHz of (asymmetrical) modulation riding on it, & while it takes at least 90 degrees of phase error to generate a bit error in a PCM system, 15 degrees of chroma phase error is extremely noticable.

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post #35 of 48 Old 12-08-2010, 07:21 PM
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Here is a good article I found reg clock jitter/DACs etc.

http://www.tnt-audio.com/clinica/diginterf2_e.html

Normally the clock of the DAC is the same clock that is recovered from incoming data stream.
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post #36 of 48 Old 12-08-2010, 09:02 PM
 
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Quote:
Originally Posted by dear.chap View Post

Here is a good article I found reg clock jitter/DACs etc.

http://www.tnt-audio.com/clinica/diginterf2_e.html

Normally the clock of the DAC is the same clock that is recovered from incoming data stream.

He also wrote these.
http://www.tnt-audio.com/sorgenti/lasergde.html
http://www.tnt-audio.com/accessories...iso_pt5_e.html
http://www.tnt-audio.com/accessories/themat_e.html
http://www.tnt-audio.com/accessories/shakti_e.html

And many more. http://www.tnt-audio.com/accessories/accessories.html
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post #37 of 48 Old 12-09-2010, 05:57 AM
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post #38 of 48 Old 12-09-2010, 06:56 AM
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Originally Posted by dear.chap View Post
And your point is ?
Did you read any of the links? The guy's just another nut job that believes in anything. CD mat's, Shakti stones, etc.

Here's a quote from your link:

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Now the big question: how much a digital interconnect can influence the sound of transport + dac couple? IMHO I can just say that the influence of digital cables on sound is far, far heavier than anyone could expect! So, we will discuss the issue too in a specific article.
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post #39 of 48 Old 12-09-2010, 07:47 AM
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^^^

i've found that ones that are too short to reach inbetween the transport and dac have a tremendous effect on sound quality...

- chris

 

my build thread - updated 8-20-12 - new seating installed and projector isolation solution

 

http://www.avsforum.com/t/1332917/ccotenj-finally-gets-a-projector

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post #40 of 48 Old 12-09-2010, 03:13 PM
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Quote:
Originally Posted by Sheer Lunacy View Post

Unless the clock for the DAC is locked to the average clock rate of the input signal, though, you're going to get a slight frequency mismatch, resulting in the equivalent of buffer underruns or overruns. If the source is clocking out 44 099.9 samples per second, & the DAC is clocking out 44 100.1 samples per second, on a long-time average, you're going to have a glitch every five seconds. Maybe that's covered up by interpolation, but it's easy to prevent that in the design phase.

True synchronous operation isn't required, so the two clocks can drift quite a bit on a short-time average (although that would result in various forms of distortion) as long as the input clock is regenerated from the input signal. Again, there's no difficulty at all in doing this. An SPDIF signal is about 1.5 MHz total, whereas the NTSC subcarrier is 3.58 MHz with 1.5 MHz of (asymmetrical) modulation riding on it, & while it takes at least 90 degrees of phase error to generate a bit error in a PCM system, 15 degrees of chroma phase error is extremely noticable.

The clock mismatch isn't an issue anymore, there are lots of algorithms that correct for such problems, specifically its usually handled in the sampling rate converter and no interpolation is done. For instance in the Berkley Alpha DAC thread there are some pictures of the PCB and they are using a Cirrus CS8421(family) SRC. This SRC, like most, has an internal buffer and supports the input sampling rate varying up to 10%/sec before there is any buffer under or overflow condition. Such a condition or lack of input clock lock is indicated to the host controller with output pins.

In addition, there are various ways to clock the entire circuit. In an audio application that is just a DAC its normal for the system crystal / oscillator to feed the DAC chip and the DAC to act as the system master which the DSP/system controller and SRC derive their clocks from. As such the DAC's clock is the master for the system.

There is no distortion caused by small amounts of drift in a digital signal. Your example from NTSC is not applicable as it just uses a digital edge to sample an analog signal, the edge being in the wrong place samples the analog signal at the wrong place. This doesn't matter in a digital transmission assuming the clock isn't so far off that it misses a high or a low condition, which is unlikely, near impossible in a modulation scheme with an embedded clock such as SPDI/F unless the entire stream is so distorted it can't be read at all.
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post #41 of 48 Old 12-09-2010, 06:37 PM
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Quote:
Originally Posted by xianthax View Post

The clock mismatch isn't an issue anymore, there are lots of algorithms that correct for such problems, specifically its usually handled in the sampling rate converter and no interpolation is done. For instance in the Berkley Alpha DAC thread there are some pictures of the PCB and they are using a Cirrus CS8421(family) SRC. This SRC, like most, has an internal buffer and supports the input sampling rate varying up to 10%/sec before there is any buffer under or overflow condition.

You've still got to lock the system clock to the input clock on a long-time average, though. Well, not strictly, if the buffer's large enough a 1-second buffer would compensate a 0.02% mismatch, which is actually fairly large, over the maximum length of an audio CD but it's certainly good engineering practice not to depend on such crutches when it's so easy to do it right.

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Originally Posted by xianthax View Post

There is no distortion caused by small amounts of drift in a digital signal. Your example from NTSC is not applicable as it just uses a digital edge to sample an analog signal, the edge being in the wrong place samples the analog signal at the wrong place. This doesn't matter in a digital transmission assuming the clock isn't so far off that it misses a high or a low condition, which is unlikely, near impossible in a modulation scheme with an embedded clock such as SPDI/F unless the entire stream is so distorted it can't be read at all.

Which is exactly what I said : instantaneous phase errors typically have to approach 90 degrees before you get bit errors, unless the eye pattern is nearly shut due to signal degradation or noise on the line. It's not nearly as sensitive, ie difficult to implement, as analog composite video, & electronics engineers solved that problem cheaply enough for use in consumer products back in the 1950s, with the crudest sort of linear analog electronics.

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post #42 of 48 Old 12-15-2010, 04:35 PM - Thread Starter
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I received my new Esoteric K-01 CD/SACD player today and a loaner Esoteric G-ORB master clock (to try out) today.
My early impressions:
The K-01 is absolutely outstanding, much better than the Esoteric X01D2, and this is cold, right out of the box. I noticed the improvement immediately. It is the best sounding CD player I have ever heard.
The Esoteric G-ORB can be easily swithched on/off instantly for A-B comparison. My dealer, who I have known for many years and trust completely feels the G-ORB improves the sound significantly. I cannot detect a difference; the K-01 sounds the same to me (superb) with and without the G-ORB. I will continue to evaluate the G-ORB. Unless I can detect a significant improvement with the G-ORB, I won't buy it. But, I already love the
Esoteric K-01.
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post #43 of 48 Old 12-29-2010, 12:44 AM
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My studio is stuck using master clocks because we do work with video (not often but we do) but we also have a large amount of AD/DA.

can't say that i or anyone else has heard any real difference using clocks or not. the nice thing about the studio is we are all gear heads and like to swap things in and our and play around.

Matt

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post #44 of 48 Old 12-29-2010, 04:59 AM - Thread Starter
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After 2 weeks of experimentation using the Esoteric G-ORb with the K-01 I have concluded that the clock has absolutely no detectable audio effect on this amazing single box player. The Esoteric K-01 sounds fantastic and the same with and without the G-ORb. I will keep the K-01, but the G-orb goes back.
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post #45 of 48 Old 10-13-2011, 03:29 AM
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The Sound on Sound article is referring to master word clocks being synched in a studio. Not necessarily connected to what you would do in a home audio environment. I asked someone who works professionally with reclocking about the article, this is his response:

Studio master clocks are word clocks. They are generally used for synchronizing events for editing. They are generally not used for playback. The article confuses the word-clock with the frequency I would call the master clock, which is generally 128 or 256 times the word-clock. Some of the things he says about master clocks are true, like with the A/D converter. However, all of the "master-clocks" that he tests are actually word-clock generators.

The reason that I use the term "master-clock" is that this is the term that is used at the D/A chip. Unfortunately, the word also describes a system word-clock in a studio environment. Totally different.

The tests that he performs are interesting primarily for studios. IMO, providing even a superb low-jitter word-clock is not very effective for reducing system jitter. The clocks need to be low-jitter true master clocks at 256X the word-clock frequency. When you provide only word-clocks, then the device must synchronize its internal clock to this and then the jitter is a function of the internal high-frequency clock, not so much the external word-clock.
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post #46 of 48 Old 10-25-2011, 09:28 PM
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the denon dcd2010 has a master clock where the denon dcd1510 does not as far as i know.So what would be the difference in these two players when it comes to playback of sacd an standard cd due to one having a master clock and the other without.

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post #47 of 48 Old 10-26-2011, 05:11 AM
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Quote:
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the denon dcd2010 has a master clock where the denon dcd1510 does not as far as i know.So what would be the difference in these two players when it comes to playback of sacd an standard cd due to one having a master clock and the other without.

No difference at all. As was mentioned above master clock is needed mostly to synchronize digital mixers. If you are not planning to do real time sound editing, master clock has no value for you.
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post #48 of 48 Old 10-26-2011, 11:13 PM
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Thanks, i now know that i dont have to waste money. I am not very switched on when it comes to cd players ,its been years and years since i have owned one. I was looking at the a1ud for an all purpose solution as i could pair it with my avpa1 but this could be a cheaper option and then i was going to wait and see what happens in 2012 for a video solution perferably with denon link also. thanks for the reply

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