Sound with COAX is brighter than with OPTICAL.

Optical and coaxial both carry identical S/PDIF signals (same source). Any differences you hear is attributed to other factors (settings, levels, processing, adjustments, malfunctions...).

Do I understand you correctly? You didn't change a cable or an input. Rather, you changed a setting from one called Optical to another called Coax. How does the manual describe those two settings? What piece of equipment is this?

I have read many statements by experts in the field stating that the Optical (Toslink) connection does not give good results, and that the coaxial is much to be preferred.

Any connection, however, is impossible to isolate from the quality of the electronic circuitry sending and receiving the signal; these can vary all over the place in how well they process the sound.

Also, you should not be using an ordinary RCA audio cable for the coaxial signal. The specification for a coaxial audio cable calls for a 72-ohm coaxial cable that is specifically designed for proper impedance matching. Use of the wrong cable causes data corruption and distortion due to an impedance mismatch. I have experience with this and it is critical to use a correctly designed cable.

I suspect that you are not using the correct cable and are hearing a distorted signal as a result. When you say "brighter", is that a positive or negative characterization?

I have several spare 72 ohm coaxial cables and would send one to you if you want one. They are only about one foot long, however.

Send a PM for follow-up if you wish.
Edited by commsysman - 12/29/12 at 7:53am

Quoted for the fun of it. Yes, again.

The standard is 75ohms, not 72. That goes for video, CATV RF as well. Professional RF gear is typically 50 ohms. While I also agree you should use a 75ohm cable, a 6 foot basic audio grade cable usually works just as well. The -3db point of AES/SPDIF is 6mhz. Not much of an impedance matching issue at 6 feet of cable length.

As for what the OP heard? Could be a defective optical cable inducing errors. But errors are ticks and pops, not altered frequency response. I suspect the receiver has a different tone control setting programmed into the coax input versus the optical. Some receivers today offer a separate tone control setting for each input.

Important note: It appears that according to the author of the above paragraph, scientific truth is determined by some sort of unofficial voting process that he is privy to. Those of us who thought that some application of the Scientific Method is involved are apparently according to him to be badly mistaken.

Furthermore, he appears to have no personal experience of his own to report.

In fact just about every music modern player and AVR with pretensions to quality has both optical and coax digital audio connections. The author apparently wants us to believe that all of the optical connections are in fact useless, and are apparently placed on this equipment in order to confuse audiophiles. He wants us to disregard the expertise of the engineers of the audio equipment with optical connections, since they obviously see some functional purpose for including them.

Now for some relevant facts:

(1) Based on numerous technical measurements that I've done, optical digital connections provide the essentially same frequency response, jitter, noise and distortion performance as coaxial connections.

(2) The following is a schematic covering coax and optical digital inputs, from the service manual of a popular AVR (Onkyo tx-nr 3009):



Note that no authority is cited for the above exceptional claim. The schematic diagram above shows that the vast majority of the electronic circuitry for receiving optical and coax digital signals are the same. Therefore their sonics can't vary since they are the identical same components.

HDMI carries both audio and video. Why do you have an optical connection as well? Beyond that, It's impossible to know what's going on without answers to the questions in my initial post.

My hypothesis:

Your WD was sending audio to your receiver through both HDMI and optical. When your receiver was set to "Optical", it took the audio from there (ignoring the audio over HDMI), and that audio is limited to Stereo PCM, Dolby-Digital 5.1, or core-DTS. When you switched to "Coax" there was no audio present over the coax (as it wasn't connected) or optical (as it was now disabled), so the receiver defaulted to playing the audio from HDMI, which is not limited to the afore-mentioned three formats.

The fact that you singled-out the improved sound of the center speaker leads me to believe you may have initially been listening to stereo PCM (no center channel) over the optical line, and then later heard surround over HDMI once the optical was disabled.

As BIslander implies, you do not need any audio cable, as the HDMI cable is more suitable for your audio.

Arny says, "The schematic diagram above shows that the vast majority of the electronic circuitry for receiving optical and coax digital signals are the same."

What he posts is a block diagram, not schematic. Block diagrams are designed to familiarize the technician with the general architecture of the device. You can tell it is a block diagram and not a schematic when there are generic blocks such as "PLL Jitter Recoverer" and "Crosspoint SW" inside some of the squares:



The little side-ways triangles after the input ports on the left are buffers/amplifiers. We see that there are no part numbers or circuit components there. Here is what a detailed schematic for a DAC comparable to above would show:



Here is another random one (optical input top left, coax bottom left)



You see how detailed they are relative to what Arny post? Of course if one gets rid of the details, at some level things look the same. The block diagram of Porsche showing an engine and transmission will look like any other car but obviously its performance will not be the same as a result. You can't remove detail and then claim, "look, they are the same."

All of this is a hat trick anyway as the incoming signals vary before they even get to the AVR. It is there that differences exist. The amplified signals driving into the AVR are inherently different with respect to their jitter and waveform and hence it matters not that the rest of the signal path is the same. Unless those differences are taken out, the two inputs will act different and *will* as a matter of engineering and science, produce different analog signals out of the DAC/AVR. See http://www.madronadigital.com/Library/DigitalAudioJitter.html

Yes, but is the difference audible?

Interesting how resistant some people are to learning digital audio 101, which states that it is impossible make unintentional audible alternations to a digital audio signal other than obliterating it, and that it is the responsibility of the device receiving the digital signal to remove the audible effects of any noise or timing errors that are introduced in transmission right up to the point where they are so severe as to make recovery of data completely impossible.

If you want to talk about digital audio signals that have massive timing errors as received, consider the packets of data that are routinely received when movies and music are downloaded, particularly streaming media. The nature of the TCP/IP data transfer process that is universally used is that data packets may be received out of order, via different paths, and with random and large timing errors. Yet, we receive these movies and they sound great!

Unlike many people around here I have hands-on experience in these matters in terms of actual reliable listening tests and bench tests of actual audio gear.

Amir has revealed that he personally owns some of the AVRs that he has criticized for high jitter, but yet he seems to continue to be empty handed when it comes to any reliable listening tests of his own. All of the information that he has provided in the past was gathered by others and related to equipment that is now mostly completely obsolete in the new equipment marketplace.

Amir since you claim to have Audio Precision test equipment at your disposal, why not show us how you can vary the performance of a real world digital audio component by varying the source or quality of any of the parts in the above schematics?

For example the 74HCU04 chips are made by dozens of different chip houses and are sold for a variety of different prices. Please show us how a cheap 74HCU04 provides audio with degraded frequency response or noise as compared to a premium-priced one. Ditto for the TORX173 optical receiver chips. What about the resistors? Can you show that replacing all of the resistors with 1% or 5% parts changes the sound quality as has been asserted by others and as you appear to be supporting?

Or, why not show how the optical and coax inputs on some real world AVR actually produces audio at the speaker terminals with different sound quality in a DBT?

Amir, the above looks like yet another lame attempt to mire the discussion in irrelevant details. If they aren't irrelevant, where is the objective evidence that they really matter?

To pick up a drum that Amir seems to want to beat lately, where are the DBTs that support his recent claims about parts quality in digital audio circuits?

Two words: sighted evaluation.

Obviously this is a low cost consumer product.

I would have at least used a 74HC14 Schmitt trigger. They are making a poor man's Schmitt trigger with R3 and the biasing network in IC1.B . But a 74HC04 and a few added resistors are a few cents cheaper then a 74HC14. However the induced jitter would be a few pico seconds lower. Audible? I highly doubt it.

Me, I would have used a CS8412 or modern variant which is a specifically designed AES/SPDIF receiver chip!
Edited by Glimmie - 12/30/12 at 10:57am

I address the issue of audibility, or better said, inaudibility, in the second part of my latest article in WSR. Here is an online copy: http://www.madronadigital.com/Library/AudibilityofSmallDistortions.html

As to signals being digital and therefore it doesn't matter, I addressed that already in my previous article. If someone wants to keep arguing, then maybe they can explain why jitter over HDMI can be 10X higher than jitter over S/PDIF *in the exact same AVR*. Both of these inputs are ostensibly "digital." Yet there are these measured differences all the way through the AVR at its analog outputs. The answer is that digital audio is not digital. It conveys digital samples and analog timing data. So you can argue all day long about the digital part but you can't do anything about the analog.

As to this statement by Arny: "If you want to talk about digital audio signals that have massive timing errors as received, consider the packets of data that are routinely received when movies and music are downloaded, particularly streaming media. The nature of the TCP/IP data transfer process that is universally used is that data packets may be received out of order, via different paths, and with random and large timing errors. Yet, we receive these movies and they sound great!"

This is a totally different architecture and the way it should have been for our consumer devices but isn't. Computer guys did not design S/PDIF or we would not have the issues that we have. The architecture in streaming environment is such that the audio device/DAC is the master. It demands audio from layers up stream to it. To keep up with it, the layers above maintain what is called a "leaky bucket" where they try to keep a buffer with enough data as to make sure the audio device/DAC does not run dry. The "buffering" message you see at the start of say, Youtube video, is for this. The buffer allows the upstream data to be non-real-time and very jittery. While the data out of the buffer to the DAC remains real-time and with high precision. Variations in the timing of source data is not a factor since the output device has its own source of timing.

It is of course possible for the above system to break down. If you can't feed the buffer fast enough, playback will by definition stop as you run out of data to play. I am sure you have seen this where you video stops playing and you get the dreaded "buffering" message. Have you ever seen that happen with your CD player? Nope. Why? Because it doesn't work this way (see below).

You can also get packets out of order over the Internet. If the out of packet data arrives in time for the audio device/DAC to play them, they they will be used. Otherwise, it is too late and we throw them away. The result is an audible mute, glitch, or pop because we don't have the samples to play. Again, have you see your CD player do that mid-span of playing some content? Nope. Why? Because it doesn't work that way.

The architecture of our home consumer systems is one where the source is the master, not the DAC/receiver. The source shoves down the bits into the throat of the receiver whether it wants it that fast or not. The receiver can't therefore decide to use its own (high precision) clock and run at its own pace like we did in the streaming case above. The receiver needs to instead "lock" to the rate of incoming bits. If those bits change in timing, the receiver will track some of that change as that is what it is supposed to do (figure out the rate of incoming data). Since all transmission lines suffer from such timing errors, then by definition we let some amount of distortion through. Unfortunately, as I explained in the above article, the threshold of audibility for timing errors can be surprisingly low.

There are fixes to this problem such as using asynchronous USB which transforms the system to work like the streaming case bove. Or as OP has done, use a different input with less jitter. So this doesn't have to cost money. There are also better and worse implementations.

You can also pretend the problem is always inaudible which is also fine. Just don't say it is so because you post a block diagram and say the circuits are the same. Or bring analogies like streaming which do not at all work the same way as this system. We either care about how these systems work or dont'.....

But you can clean this up to some extent with a FIFO. The FIFO is analogous to your "leaky bucket" example above. And it will reduce the jitter of the incoming audio signal to that of the FIFO output clock.

However, it WILL NOT eliminate the timing error with respect bit alignment because the FIFO input must still have a local PLL that is tracking the input signal to phase lock. But now that we have this buffer we can allow the input clock to be much looser and track the jitter more closly yet still provide the DAC with a tighter jitter spec. Net result, less jitter reproduced.

This is TRUE RECLOCKING IMO. Just reshaping the bit stream like many cheaper circuits does little good. Now is this at a consumer level price point? Depends. If the product has a DSP, like an AVR, it's pretty easy to implement. A stand alone lower cost DAC, probably not.

You are tempting me to deconstruct that piece.

Addressing and responding believably are two different things.

Furthermore Amir, you are as seems to be your habit making up statements, stuffing them into my mouth and then answering your ow dumbed-down creations instead of answering the more difficult questions that I actually asked.

Just in case they have escaped your memory, here they are again:

Amir since you claim to have Audio Precision test equipment at your disposal, why not show us how you can vary the performance of a real world digital audio component by varying the source or quality of any of the parts in the above schematics?

For example the 74HCU04 chips are made by dozens of different chip houses and are sold for a variety of different prices. Please show us how a cheap 74HCU04 provides audio with degraded frequency response or noise as compared to a premium-priced one. Ditto for the TORX173 optical receiver chips. What about the resistors? Can you show that replacing all of the resistors with 1% or 5% parts changes the sound quality as has been asserted by others and as you appear to be supporting?

Or, why not show how the optical and coax inputs on some real world AVR actually produces audio at the speaker terminals with different sound quality in a DBT?


Amir, I would like you to start arguing with someone other than yourself!



The answer is obvious - there are vast differences in the areas of signal processing that I referenced in post 7 and you sloughed.

As usual Amir you are avoiding the point which is that any competent digital receiver is supposed to resolve any likely data timing issues that arise. If certain obsolete products didn't do this, that must be one reason why improved models were produced.

Again Amir you seem to have no up-to-date technical measurements of the HDMI performance of AVRs that are currerntly on the market as new products. You have no reliable listening test results to report? Why not? Doesn't your store carry them? Don't you claim to have the necessary test equipment and AVR at your disposal? Aren't you competent to use your equipment and set up your own DBTs to support your claims?

The silence is deafening.

Can you still hear me? Oh good. You are not deaf yet.

Arny's suggestion was good for the most part. Some fresh data would add more spice to the conversation . So I have been working on that in the last few days. I don't have a prediction for when I will finish it given that I am going to CES next week. The early data is exciting though! So much so that my WSR magazine editor wants it for the next issue. If so, then out of courtesy I let the print issue come out first before putting the data online. So it will be a couple of months. I may also publish part of it online prior to that. Look for it in the Technical section of WBF Forum. If I can remember and find this thread then, I will post a link here too.

BTW, none of this is to back my "claim." What I have said I have published in an article that countless manufacturers have already read. And not one has come to me/editor and say, "oh wait, we don't have this problem with HDMI anymore." And what I wrote in my article, *is* backed by listening tests. That is how we determine the threshold of hearing and then compare the results to measurements. If I measured your heating threshold in your room to fan noise at 30 db SPL, then you would have pretty good idea that a projector with 40 db SPL would be audible to you (I am making up these numbers). That is the analysis that is in my article and referenced papers. In addition, there is work in the industry to bring more attention to it with the hopes that someone will do something about it. So the data I am gathering is more to convince the non-believers than anything else.

Finally, there are things that Arny is asking for that are darn near impossible for me to do. If you don't know what they are, then you don't know what he is really asking . But per above, some of it are very doable and I am in the process of collecting the data for them. So far it is more fun than I expected!

Even including post 23! ;-)

Logical fallacy: Argument from ignorance

Indeed. One of the lamest arguments I've ever seen until you get into the "Made so much difference that my wife heard the difference in the kitchen" level of claims.

Basically, the writer seems to want us to think that he's so influential that the manufacturer's are going to waste the time of someone in their public relations department to rebut his baseless claims.

We have pointed out this logical fallacy in his arguments so many times that I can only conclude that he is being intentionally deceptive. It is all a part of his pathological intellectual dishonesty. He continually and deliberately misleads the readers of AVS in an effort to inflate his own importance. Unfortunately, no one cares enough to take part in an intervention to get him the help he so desperately needs.

Widescreen Review is so desperate for content that they don't have to pay for that they will publish anything. Of course it doesn't hurt that amirm's store, Madrona Digital, is also an advertiser. I doubt any manufacturers even saw the article, much less read it.
Edited by audiophilesavant - 1/10/13 at 8:04pm