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#1
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I've been poking around the net for information about word clock generators,
internal and external, rumored improvements, etc. One of the patterns I've noticed is a lot of statements to the effect of "when I played back my tracks I could hear a big improvement" - which strikes me as almost irrelevant. It seems to me that improving playback is a minor benefit, unless I plan on having everyone come over to my house to listen to my tracks. Yes I understand the importance of accurate playback during mixing so you can make better decisions, but it seems to me that the speakers and the room is going to blur the sound a lot more than clock jitter, at least in that context. My concern is whether better clocking improves the tracks during recording, and will this improvement be audible for other people? What led me here was that I've been kicking around the idea of trying to get some improvements out of my Motu 1296. It's done what I need well enough that I'm more interested in getting it upgraded than replacing it outright - recording is just one of my hobbies. I've heard that it has room for improvement with the clock, and this is one of the mods that Black Lion does. I was also interested in the idea of external word clock, but after reading a long thread on this topic on Dan Lavry's PSW forum, I can see where the benefits of this are dubious (reclocking from an external clock shouldn't sound any better than the internal clock, and possibly worse). I don't have any other gear to synch with, so I have no other reason to use an external clock anyway. I guess I don't have a specific question, I'm just curious what other think on this topic. Sean |
#2
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Sean Conolly wrote:
I've been poking around the net for information about word clock generators, internal and external, rumored improvements, etc. One of the patterns I've noticed is a lot of statements to the effect of "when I played back my tracks I could hear a big improvement" This is usually an indication that the D/A converter isn't very good. A good converter won't be improved by an external clock. But if the playback really does sound better, don't knock it. It seems to me that improving playback is a minor benefit, unless I plan on having everyone come over to my house to listen to my tracks. In that respect, yes, but it might help you to mix better if you can hear more accurately. My concern is whether better clocking improves the tracks during recording, and will this improvement be audible for other people? Again, if it does, it means that it's improving a converter that isn't very good. Chances are a better converter would be better overall than a converter that was poor enough to be improved by an external clock. What led me here was that I've been kicking around the idea of trying to get some improvements out of my Motu 1296. I've heard that it has room for improvement with the clock, and this is one of the mods that Black Lion does. I don't know what he does, but it's likely he replaces the A/D converter chip with something that has a better phase locked loop than the original, or if it's based on parts rather than a chip, rebuilds the PLL. I suspect that whatever he does improves its performance with the internal clock, which most likely makes the improvement, if any, with an external clock, negligible. I was also interested in the idea of external word clock, but after reading a long thread on this topic on Dan Lavry's PSW forum, I can see where the benefits of this are dubious The benefit of an external clock, and indeed what makes one necessary, is to synchronize clocks within a system. That's its primary purpose, not to make a badly clocked converter sound better. I guess I don't have a specific question, I'm just curious what other think on this topic. P'tooey! -- If you e-mail me and it bounces, use your secret decoder ring and reach me he double-m-eleven-double-zero at yahoo -- I'm really Mike Rivers ) |
#3
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Mike Rivers wrote:
Sean Conolly wrote: I've been poking around the net for information about word clock generators, internal and external, rumored improvements, etc. One of the patterns I've noticed is a lot of statements to the effect of "when I played back my tracks I could hear a big improvement" This is usually an indication that the D/A converter isn't very good. A good converter won't be improved by an external clock. But if the playback really does sound better, don't knock it. Word clock is invaluable in systems that synchronize multiple external interfaces or where tracking may involve a bi-directional flow between recorders or external effects. But I'm hard pressed to come up with any reason why the sound would audibly differ in a "one way" system with a single interface. The worst that one might discover in a laboratory is that a poorly designed internal clock, PLL, etc. might deliver samples with timing that spreads a bit, but it would probably be next to impossible to detect such a thing aurally unless the timing is so bad that it is basically broken. -- Neil |
#4
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"Neil Gould" wrote:
I'm hard pressed to come up with any reason why the sound would audibly differ in a "one way" system with a single interface. I'm sure this is the correct answer. Jitter is irrelevant and inaudible in modern gear. I'm not convinced it was ever a problem, even at the dawn of affordable digital in the 1980s. Even a 50-cent crystal oscillator is highly stable. I'm certain that when people believe they hear an improvement from changing clocks it is entirely in their mind, or due to comb filtering, and is exactly the same as improvement audiophiles claim to hear when they change speaker cables etc. As Sean said, "speakers and the room is going to blur the sound a lot more than clock jitter." No kidding. Room reflections and their "timing errors" are literally 1,000 times larger and stronger than anything possible from clock jitter. Much more he http://www.ethanwiner.com/audibility.html http://www.ethanwiner.com/believe.html http://www.ethanwiner.com/dither.html http://www.ethanwiner.com/audiophoolery.html --Ethan |
#5
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"Ethan Winer" wrote ...
I'm sure this is the correct answer. Jitter is irrelevant and inaudible in modern gear. I'm not convinced it was ever a problem, even at the dawn of affordable digital in the 1980s. Even a 50-cent crystal oscillator is highly stable. I'm certain that when people believe they hear an improvement from changing clocks it is entirely in their mind, or due to comb filtering, and is exactly the same as improvement audiophiles claim to hear when they change speaker cables etc. And caused by exactly the same effect -- namely hucksters trying to peddle "up-market" goods to technically illeterate customers. If you can convince someone that a $20 cable will fix the "jitter problem" caused by a $5 cable, you can make an easy $15. |
#6
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Ethan Winer ethanw at ethanwiner dot com wrote:
I'm sure this is the correct answer. Jitter is irrelevant and inaudible in modern gear. I'm not convinced it was ever a problem, even at the dawn of affordable digital in the 1980s. Even a 50-cent crystal oscillator is highly stable. I'm certain that when people believe they hear an improvement from changing clocks it is entirely in their mind, or due to comb filtering, and is exactly the same as improvement audiophiles claim to hear when they change speaker cables etc. A 50-cent crystal oscillator is highly stable, but it's also only free-running. There are a lot of pieces of equipment out there that use derived clocks generated by PLL oscillators that are locked to the clock of the incoming datastream. You can argue that this is not competent design practice today in an age where buffering is cheap, but it still goes on a lot. As Sean said, "speakers and the room is going to blur the sound a lot more than clock jitter." No kidding. Room reflections and their "timing errors" are literally 1,000 times larger and stronger than anything possible from clock jitter. Much more he That doesn't mean that having a good clock isn't important. BUT, it is only important at the converters themselves. If the converter is free-running it's probably stable... if it has to lock to something else, make sure you have a good quality piece of equipment. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#7
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Ethan Winer wrote:
I'm sure this is the correct answer. Jitter is irrelevant and inaudible in modern gear. I'm not convinced it was ever a problem, even at the dawn of affordable digital in the 1980s. Even a 50-cent crystal oscillator is highly stable. There's a little more to it than frequency stability. That's easy. Getting modulation out of the clock is a little harder. The PLL does its best to deal with the crappiest clocks, but it does so by making some assumptions about what it has to filter out. It turns out that if you know the characteristics of the PLL in a given converter, and you're a good enough designer to be able to control the noise spectrum of the clock, you may be able to improve on a particular converter with an external clock. What this means is Clock Generator A might make Converter A sound better (because Company A designed both of them and they're good designers), but not Converter B or C, but maybe Converter D because it's similar enough to Converter A. Honestly, I don't believe the "I hooked up the Westclox and just started playing some old tracks, and my grandmother ran in from another room asking what I did because it sounds so much better." stories, but I believe that a clever designer can improve a particular converter with an external clock if he knows enough about the characteristics of the converter's PLL and he knows how to design for it. Of course a better speaker or a better broadband trap can make more difference than a new clock, so when it comes to bang-for-buck, this is not the way to improve your sound (unless you need it for system synchronization - in which case you should be prepared for a possible minor degradation). But if you already have the best speakers you can afford, you might get a bit of an improvement (or not) with a new clock in the system. Here's a white paper on the subject of external clocks, toned down for the general public - no math, but based on too much research: http://www.drawmer.com/uploads/File/...clock-sync.pdf -- If you e-mail me and it bounces, use your secret decoder ring and reach me he double-m-eleven-double-zero at yahoo -- I'm really Mike Rivers ) |
#8
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![]() "Mike Rivers" wrote in message ... Sean Conolly wrote: What led me here was that I've been kicking around the idea of trying to get some improvements out of my Motu 1296. I've heard that it has room for improvement with the clock, and this is one of the mods that Black Lion does. I don't know what he does, but it's likely he replaces the A/D converter chip with something that has a better phase locked loop than the original, or if it's based on parts rather than a chip, rebuilds the PLL. I suspect that whatever he does improves its performance with the internal clock, which most likely makes the improvement, if any, with an external clock, negligible. I was doing some tests today with my Motu, and I rediscovered that I have to use the clock on the PCI card, or I get these weird glitches - every fourth sample will hold the same value for a few cycles. Picture the output as if it was a four sample circular buffer, and then stop updating one of the buffer slots so it just hold the same value for 10-20 cycles. When I switch the master to the the PCI card it runs perfectly, but if I can't use the 1296 as the master then I'm not sure if I'm going to get any benefit from modding that clock. And in the process of really listening and looking critically at what the Motu produces - I realized it's probably the best piece of gear in my little office here. The only thing I might need is to pick up a 421 card as a replacement or spare for the 321 card. Sean |
#9
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"Scott Dorsey" wrote in message
... Ethan Winer ethanw at ethanwiner dot com wrote: I'm sure this is the correct answer. Jitter is irrelevant and inaudible in modern gear. I'm not convinced it was ever a problem, even at the dawn of affordable digital in the 1980s. Even a 50-cent crystal oscillator is highly stable. I'm certain that when people believe they hear an improvement from changing clocks it is entirely in their mind, or due to comb filtering, and is exactly the same as improvement audiophiles claim to hear when they change speaker cables etc. A 50-cent crystal oscillator is highly stable, but it's also only free-running. There are a lot of pieces of equipment out there that use derived clocks generated by PLL oscillators that are locked to the clock of the incoming datastream. You can argue that this is not competent design practice today in an age where buffering is cheap, but it still goes on a lot. I'd expect that practice is intended to keep the converter running at the same pace as the source, more than just syncing samples. Today I compared my two A/D boxes and found that the little E-Mu is slower than the Motu by a millisecond per minute - in a D/A converter that kind difference would eventually overflow the buffer if the clock wasn't sync'ed to the incoming data. That doesn't mean that having a good clock isn't important. BUT, it is only important at the converters themselves. If the converter is free-running it's probably stable... if it has to lock to something else, make sure you have a good quality piece of equipment. Isn't also true that in a multi-channel converter, the internal clock is a separate circuit driving all the converters? A single converter with a clock on the chip is fine, but when there's more than one in the same box an internal master clock is needed. Sean |
#10
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Sean Conolly wrote:
I was doing some tests today with my Motu, and I rediscovered that I have to use the clock on the PCI card, or I get these weird glitches - every fourth sample will hold the same value for a few cycles. If this is MOTU-speak for "I have to use its internal clock. I get glitches when using the external clock" this suggests that it's not really synchronizing to the external clock. Is there an indicator, either an LED on the front panel or something on the MOTU software control panel that indicates PLL lock? There's no published standard for the voltage level of word clock signal level (either as output level or input sensitivity), only the rise time of the waveform. I've encountered hardware pairs (a source and destination) that aren't compatible. When connected, the clock source doesn't have enough poop to reliably trigger the sync input. I first encountered this when trying to synchronize a Mackie hard disk recorder and digital console. With the console's word clock output properly terminated at the HDR end, the clock voltage dropped below the input threshold of the HDR and the HDR's "sync" light would blink intermittently. Swap the word clock master/slave relationship and the console would lock up solidly to the HDR's clock output. It was possible to make the HDR work with the console's word clock output by turning off the termination (there's a switch) on the HDR but that's not the ideal situation. This is one of those problems that you just have to analyze. Without having exactly the same setup that you have, it's impossible for anyone to make the right guess. -- If you e-mail me and it bounces, use your secret decoder ring and reach me he double-m-eleven-double-zero at yahoo -- I'm really Mike Rivers ) |
#11
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In article ,
Sean Conolly wrote: "Scott Dorsey" wrote in message ... Ethan Winer ethanw at ethanwiner dot com wrote: I'm sure this is the correct answer. Jitter is irrelevant and inaudible in modern gear. I'm not convinced it was ever a problem, even at the dawn of affordable digital in the 1980s. Even a 50-cent crystal oscillator is highly stable. I'm certain that when people believe they hear an improvement from changing clocks it is entirely in their mind, or due to comb filtering, and is exactly the same as improvement audiophiles claim to hear when they change speaker cables etc. A 50-cent crystal oscillator is highly stable, but it's also only free-running. There are a lot of pieces of equipment out there that use derived clocks generated by PLL oscillators that are locked to the clock of the incoming datastream. You can argue that this is not competent design practice today in an age where buffering is cheap, but it still goes on a lot. I'd expect that practice is intended to keep the converter running at the same pace as the source, more than just syncing samples. Today I compared my two A/D boxes and found that the little E-Mu is slower than the Motu by a millisecond per minute - in a D/A converter that kind difference would eventually overflow the buffer if the clock wasn't sync'ed to the incoming data. Right. So you have an internal clock that is synched to the source clock so they all run at the same rate overall. Imagine having four 2-channel A/D boxes feeding an 8-channel recorder, for instance. They all need to lock off some master clock because otherwise they wouldn't be sending data at quite the same rate. Normally that master clock is in the recorder. That doesn't mean that having a good clock isn't important. BUT, it is only important at the converters themselves. If the converter is free-running it's probably stable... if it has to lock to something else, make sure you have a good quality piece of equipment. Isn't also true that in a multi-channel converter, the internal clock is a separate circuit driving all the converters? A single converter with a clock on the chip is fine, but when there's more than one in the same box an internal master clock is needed. Actually, even on two-channel converters, there is an internal clock which is usually not on the same chip as the converters. Sometimes it's actually built into the S-PDIF receiver chip. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#12
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Scott,
There are a lot of pieces of equipment out there that use derived clocks generated by PLL oscillators that are locked to the clock of the incoming datastream. Yes, and I was addressing only a single clock as used in a sound card. --Ethan |
#13
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Mike,
Honestly, I don't believe the "I hooked up the Westclox and just started playing some old tracks, and my grandmother ran in from another room asking what I did because it sounds so much better." stories Exactly, and that's all I'm addressing, which in turn is what Sean (OP) was asking about. Here's a white paper on the subject of external clocks In that paper I see exactly what I object to: "People have installed an external word clock and have reported an apparent sonic improvement, generally described as more open and less harsh." Jitter manifests as FM sidebands some number of dB below the music. Typical values I see are -110 to -140 dB, with is WAY below the noise of 16-bit digital, which in turn is WAY below the room noise of 16+ tracks mixed together. Even one microphone in a quiet room capturing an acoustic guitar will have ambient noise well above the noise floor of 16 bits. I'm just trying to keep this stuff in perspective. :-) --Ethan |
#14
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On Nov 3, 10:29 am, "Ethan Winer" ethanw at ethanwiner dot com
wrote: Jitter manifests as FM sidebands some number of dB below the music. Typical values I see are -110 to -140 dB Is that with an external clock source? Remember, when you connect two boxes, and defect in one (like a ground problem or a noisy power supply) can be imposed on the other. If there are sidebands 60 Hz removed from a given frequency, that probably won't hurt anything. It might even make a bass part sound better. But if they're 2-3 kHz removed, they can get in the way of some pretty useful stuff. Now don't get me wrong. I'm not a strong advocate of using an external clock unless you need one. But I can understand the principle that predicts the problem, even though the magnitude of the problem may be tiny in some circumstances. I've not heard any problems even when I run my Mackie HDR24/96 using a 45+ year old Wavetek function generator as the word clock source when I need to change the speed/pitch of a track, but I wouldn't be surprised if some golden ears would tarnish at the thought, if not the sound. Remember, too, that the better the converter's clock design, the more immune it is to effects of external noise. |
#15
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Mike,
Is that with an external clock source? I dunno, it's from tables in Ken Pohlmann's book Principles of Digital Audio and a few other places. If there are sidebands 60 Hz removed from a given frequency, that probably won't hurt anything. It might even make a bass part sound better. But if they're 2-3 kHz removed, they can get in the way of some pretty useful stuff. Yes, but only if the side-bands are audible. If they're 100+ dB below the music, they're not audible no matter how golden one's ears are. Now don't get me wrong. I'm not a strong advocate of using an external clock unless you need one. But I can understand the principle that predicts the problem, even though the magnitude of the problem may be tiny in some circumstances. Actually, one of the famous designers (Dan Lavry?) argues strongly the opposite, that an external clock can only make jitter worse. The explanation involves math that's over my head, but I believe him. However, all that stuff is still just mental masturbation because it's always 100+ dB down and thus irrelevant. I've not heard any problems even when I run my Mackie HDR24/96 using a 45+ year old Wavetek function generator as the word clock source when I need to change the speed/pitch I rest my case. :-) --Ethan |
#16
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![]() I've not heard any problems even when I run my Mackie HDR24/96 using a 45+ year old Wavetek function generator as the word clock source when I need to change the speed/pitch I rest my case. :-) --Ethan and the effects of jitter are EASILY detected on a spectrum analyzer looking at simple test tones... anyone who thinks they hear a change of one clock configurartion vs another can easily verify what they think they hear with a spectrum analyzer. If you are using a function generator as a clock source, as a test you can apply intentional FM of various deviations and frequencies to the clock and create HUGH amounts of jitter to see how much it really takes to become audible. Mark |
#17
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Ethan Winer wrote:
Actually, one of the famous designers (Dan Lavry?) argues strongly the opposite, that an external clock can only make jitter worse. The explanation involves math that's over my head, but I believe him. However, all that stuff is still just mental masturbation because it's always 100+ dB down and thus irrelevant. His explanation is correct in the "all things as expected" case. However, there can be things wrong (we're not just talking bad designs here, though bad design or implementation is a factor) where sidebands resulting from jitter can be much greater than -100 dB. And then there's the issue of putting them in places where they won't do a lot of harm. One of the things that Lavry is very careful about is not just low jitter clocking, but clock circuits that are immune to being buggered by outside influences such as external clock generators, power supply hum, stray EMI, and ground loops. The clock might be find by itself, and then the real world comes along and screws things up. That's where theory and practice are not the same, which, in theory, they should be. g -- If you e-mail me and it bounces, use your secret decoder ring and reach me he double-m-eleven-double-zero at yahoo -- I'm really Mike Rivers ) |
#18
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"Ethan Winer" ethanw at ethanwiner dot com writes:
[...] Actually, one of the famous designers (Dan Lavry?) argues strongly the opposite, that an external clock can only make jitter worse. The explanation involves math that's over my head, but I believe him. Hi Ethan, Do you have a reference? -- % Randy Yates % "She's sweet on Wagner-I think she'd die for Beethoven. %% Fuquay-Varina, NC % She love the way Puccini lays down a tune, and %%% 919-577-9882 % Verdi's always creepin' from her room." %%%% % "Rockaria", *A New World Record*, ELO http://www.digitalsignallabs.com |
#19
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Randy Yates wrote:
"Ethan Winer" ethanw at ethanwiner dot com writes: [...] Actually, one of the famous designers (Dan Lavry?) argues strongly the opposite, that an external clock can only make jitter worse. The explanation involves math that's over my head, but I believe him. Hi Ethan, Do you have a reference? I'm not Ethan, but.... http://www.lavryengineering.com/white_papers/jitter.pdf Briefly, jitter is based on error in absolute time. As the time goes down, the effect of a change on a unit circle at a higher Fs goes up - a 1 nanosecond hit causes twice as much error at 2Fs than at 1Fs. Jitter also has a different spectrum at twice the clock rate. -- Les Cargill |
#20
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Les Cargill writes:
Randy Yates wrote: "Ethan Winer" ethanw at ethanwiner dot com writes: [...] Actually, one of the famous designers (Dan Lavry?) argues strongly the opposite, that an external clock can only make jitter worse. The explanation involves math that's over my head, but I believe him. Hi Ethan, Do you have a reference? I'm not Ethan, but.... http://www.lavryengineering.com/white_papers/jitter.pdf Thanks Les. Ethan, is this the paper you were referring to? -- % Randy Yates % "How's life on earth? %% Fuquay-Varina, NC % ... What is it worth?" %%% 919-577-9882 % 'Mission (A World Record)', %%%% % *A New World Record*, ELO http://www.digitalsignallabs.com |
#21
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"Mike Rivers" wrote in message
news ![]() Sean Conolly wrote: I was doing some tests today with my Motu, and I rediscovered that I have to use the clock on the PCI card, or I get these weird glitches - every fourth sample will hold the same value for a few cycles. If this is MOTU-speak for "I have to use its internal clock. I get glitches when using the external clock" this suggests that it's not really synchronizing to the external clock. Is there an indicator, either an LED on the front panel or something on the MOTU software control panel that indicates PLL lock? I have three basic choices: using the 1296 box as the master, using the PCI card as the master, or using the external clock input on the 1296. I'm assuming that using the 1296 as the master would be the better choice since it's part of the same hardware as the converters, but in practice it turns out that I have to use the PCI card to avoid the glitches I described. It may be that the card clock should take priority since it's doing the actual recording to disk, or maybe there's little too much spread between the internal clocks of the card and the box, or that the box is better at sync'ing to an external clock (external to the box) than the card is. There's no obvious indicator - it does flash the chosen sample rate for a while and stop, but the pattern is the same using either the box or the card as the master. I've never seen the display flash during recording, but I don't watch it that much after I get my levels set. Not a big deal to me, other than every couple of years I forget why I had the master set to the card and go through the discovery process again. Sean |
#22
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"Mark" wrote in message
... I've not heard any problems even when I run my Mackie HDR24/96 using a 45+ year old Wavetek function generator as the word clock source when I need to change the speed/pitch I rest my case. :-) --Ethan and the effects of jitter are EASILY detected on a spectrum analyzer looking at simple test tones... anyone who thinks they hear a change of one clock configurartion vs another can easily verify what they think they hear with a spectrum analyzer. If you are using a function generator as a clock source, as a test you can apply intentional FM of various deviations and frequencies to the clock and create HUGH amounts of jitter to see how much it really takes to become audible. I think that the effects of clock jitter are going to be in the time domain much more than the frequency domain. The most common benefit that people report with improved word clocks is tighter stereo imaging, which of course is how the listener's brain interprets the signals from each ear into a spatial point. It can be easily shown that the position of a source can be moved in the stereo field just by manipulating the timing, so I can understand how the accuracy of the timing errors could influence where the source sits in the field. It also seems that the nature of the jitter is a factor: if it's a slow modulation it should be imperceptible, and the same if it's random - at least in relation to audio frequencies. Just as a guess, I'll speculate that jitter would have to be in the 100 microsecond range to perceptually effect the imaging, and even then only if the average is modulating at something near audible frequencies. If it's jumping around too much the brain will ignore it, and if it's drifting slowly enough then you're not conscious of it, at least until someone points it out. Obviously if it's a constant offset then that would just change where the source sits in the field, not the size of it. Again this is just my guess: the brain is a strange organic DSP and may be far more sensitive than I'm imagining here. The very fact that we perceive a sound coming from multiple speakers as a single point says a lot about how adaptive our hearing is. Sean |
#23
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Sean Conolly wrote:
I have three basic choices: using the 1296 box as the master, using the PCI card as the master, or using the external clock input on the 1296. I'm assuming that using the 1296 as the master would be the better choice since it's part of the same hardware as the converters That's a good starting point, but . . you have to do what works in the system. but in practice it turns out that I have to use the PCI card to avoid the glitches I described. It may be that the card clock should take priority since it's doing the actual recording to disk, or maybe there's little too much spread between the internal clocks of the card and the box, or that the box is better at sync'ing to an external clock (external to the box) than the card is. I suspect the latter. But if you want to pursue it, look at the driver situation. Are you up to date? And more important, is the manufacturer up to date? And are all your terminations and non-terminations correct? I'll bet the card's word clock input has a fixed termination, and what you're feeding it with may not develop enough voltage across that load to properly synchronize the card's clock input. My Lynx L22's software control panel has a set of indicators including one labeled "Lock" that's red if it doesn't have a good clock signal from whatever source you've selected. -- If you e-mail me and it bounces, use your secret decoder ring and reach me he double-m-eleven-double-zero at yahoo -- I'm really Mike Rivers ) |
#24
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"Sean Conolly" wrote in message
I think that the effects of clock jitter are going to be in the time domain much more than the frequency domain. Ask anybody whose taken any EE courses in the last 50 years. The time domain and the frequency domain are very closely related. It's darn hard to do something in one domain without doing some analogous thing(s) in the other. Remember, F = 1/T where F is frequency and T is time. Relationships don't get much simpler than that! The most common benefit that people report with improved word clocks is tighter stereo imaging, which of course is how the listener's brain interprets the signals from each ear into a spatial point. That may be intuitive to you, but it doesn't hold up in reality. What FM distortion (jitter's technical name) does at modest levels is add sidebands around every tone in the recorded sound. These tones are aharmonic, so if they are loud enough, they are going to make the music sound kind of sour. At higher levels, jitter simply keeps digital receivers from ever locking into the input signal. If the jitter level is inconsistent, you may have clicks and pops as the digital receiver falls in and out of lock. At really high levels, jitter keeps the digital receiver from ever locking into the signal, and you have silence. The reason why I can say these things so boldly is that one day maybe 5-7 years ago I built a device that added jitter to a digital audio signal in ways and amounts that I could control from mild to wild. I then tried out a bunch of digital audio gear with different amounts and kinds of jitter, and carefully listened to it and also measured its effects on the equipment. One of the devices I had that did very little to correct jitter was a fairly pricey (in its day) Denon DAC. Another device that did wonders with a jittery input signal was essentially a cheap Technics surround receiver. It can be easily shown that the position of a source can be moved in the stereo field just by manipulating the timing, Yes, but that requires timing changes on the order of tens of milliseconds and even fractions of a second. Jitter at its worst involves changes on the order of nanoseconds or picoseconds. So, jitter involves changes that are on the order of a thousandth or a millionth of what it takes to cause image shifting. so I can understand how the accuracy of the timing errors could influence where the source sits in the field. It also seems that the nature of the jitter is a factor: if it's a slow modulation it should be imperceptible, and the same if it's random - at least in relation to audio frequencies. Jitter is usually imperceptible because at this time, it is pretty trivial and inexpensive to make equipment that simply reduces it to such low levels that the next trick is to accurately measure it, let alone actually hear it. That's one of the nice things about digital, a good DAC can make horrendous amounts of jitter simply and effectively disappear. An example of a piece of audio gear that has a DAC that makes huge amounts of jitter disappear might be a $150 surround receiver. Just as a guess, I'll speculate that jitter would have to be in the 100 microsecond range to perceptually effect the imaging, and even then only if the average is modulating at something near audible frequencies. Think 10-100 times larger. That much jitter takes a major screw-up to actually have. Not to say that people just plugging cables and flicking switches in a studio haven't done this to themselves at one time or the other. If it's jumping around too much the brain will ignore it, and if it's drifting slowly enough then you're not conscious of it, at least until someone points it out. Low frequency FM distortion is like flutter and more even more likely wow. Play a good piano recording with audible amounts of flutter and/or wow, and nobody needs to tell you that is there. The ear is very sensitive to low frequency FM distortion, and its sensitivity to it goes down as the jitter frequency goes up. Obviously if it's a constant offset then that would just change where the source sits in the field, not the size of it. No, a constant amount of FM distortion is called a pitch change. If you want to change the imaging you *have* to do different things to each of the channels that you are listening to. Since most jitter gets tacked onto a digital signal that is also multichannel (degenerate case: Stereo) all the embedded channels get the same bad treatment. bottom line, no image shift. Again this is just my guess: I'm sure you mean well, but unfortunately you've got just about everything wrong. I'll bet money that you've been reading lots of subjective comments about jitter. Since there are a lot of pro audio ragazines that employ some writers are actually no more savvy than your average consumer high end audio ragazine, a lot of really weird stuff shows up where it shouldn't. the brain is a strange organic DSP and may be far more sensitive than I'm imagining here. Actually, what the brain does with sound and music is pretty well understood. Music, Your Brain, and Ecstasy by Jourdain would be a good starter, and then for the heavy lifting there are the writings of Zwicker and Fastl. |
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"Ethan Winer" ethanw at ethanwiner dot com wrote in
message Mike, Is that with an external clock source? I dunno, it's from tables in Ken Pohlmann's book Principles of Digital Audio and a few other places. If there are sidebands 60 Hz removed from a given frequency, that probably won't hurt anything. It might even make a bass part sound better. But if they're 2-3 kHz removed, they can get in the way of some pretty useful stuff. Agreed, but for more details, please see the post to another subthead that I just posted. 60 Hz (and 120 Hz) are good frequencies to use when talking about jitter because they are very common in the real world. One less-obvious that this happens is when a digital signal gets mixed with a lot of hum. Obviously, you can't hear the hum directly. And, a well-designed digital input will filter low frequencies out. But there is gear that is not well-designed. ;-( At any rate, some place along the way, most digital signals pass through some kind of a level-sensitive logic device like a Schmidt Trigger. The hum makes the trigger points shift back and forth at the frequencies contained in the hum which of course are generally power-line related. Bang! Jitter. BTW, this is the general approach I used to make a jitter-inducing device. Yes, but only if the side-bands are audible. If they're 100+ dB below the music, they're not audible no matter how golden one's ears are. Agreed. Now don't get me wrong. I'm not a strong advocate of using an external clock unless you need one. But I can understand the principle that predicts the problem, even though the magnitude of the problem may be tiny in some circumstances. Actually, one of the famous designers (Dan Lavry?) argues strongly the opposite, that an external clock can only make jitter worse. I suspect that most of that is because its external. It suffers the slings and arrows of equipment grounding, and bad cabling, and the like. External DACs for CD players were always a disaster waiting to happen. Put as much as you can inside a box and engineer the box carefully (Dan's your guy, there!) and all is well. The explanation involves math that's over my head, but I believe him. However, all that stuff is still just mental masturbation because it's always 100+ dB down and thus irrelevant. Exactly. Jitter often stops being a problem when it is *only* 80 dB down, but 100 dB is both readily doable and gives you lots of insurance. I think that 120 dB down is not all that rare. One of the cheap surround receivers I tested put any jitter in a signal that it could lock on, about 115 dB down. ;-) I've not heard any problems even when I run my Mackie HDR24/96 using a 45+ year old Wavetek function generator as the word clock source when I need to change the speed/pitch Been there, done that. I rest my case. :-) As is usual, exactly right. ;-) Back to things that actually matter like room acoustics! |
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![]() Again this is just my guess: the brain is a strange organic DSP and may be far more sensitive than I'm imagining here. The very fact that we perceive a sound coming from multiple speakers as a single point says a lot about how adaptive our hearing is. I say again.. use a spectrum analyzer!! You can SEE jitter well below the level anyone can hear it. There is no guesswork needed.. Mark |
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Sean,
The most common benefit that people report with improved word clocks is tighter stereo imaging Yes, I know that's the common report, but I think the reports are just wrong and those people would never identify low jitter from high jitter in a blind test. This article I linked to earlier explains my best guess as to why people report changes in "imaging" and clarity etc when changing clocks, or speaker cables, or other things that are unlikely to make an audible change: http://www.ethanwiner.com/believe.html The very fact that we perceive a sound coming from multiple speakers as a single point says a lot about how adaptive our hearing is. It's actually much simpler than that. If a sound is directly in front of you, it has the same volume and arrival time at both ears. So whether the sound really is in front of you, or arrives from two separate loudspeakers, both ears receive the same signal. So it's not any mystique in the ears or brain, but simple physics and acoustics. --Ethan |
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Do you have a reference?
I know I've seen it several times at Gearslutz, both by the author (pretty sure it's Dan Lavry) and by other people referring to him saying that. Maybe go to Gearslutz.com and search for posts by Dan. As I recall he posts under his own name there. --Ethan |
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Thanks Les. Ethan, is this the paper you were referring to?
I've mostly seen him explain it in forum posts. --Ethan |
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One of the cheap surround receivers I tested put any jitter in a signal
that it could lock on, about 115 dB down. Yep. A total non-issue. Back to things that actually matter like room acoustics! LOL - okay then! --Ethan |
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"Arny Krueger" wrote in message
... "Sean Conolly" wrote in message I think that the effects of clock jitter are going to be in the time domain much more than the frequency domain. Ask anybody whose taken any EE courses in the last 50 years. The time domain and the frequency domain are very closely related. It's darn hard to do something in one domain without doing some analogous thing(s) in the other. Remember, F = 1/T where F is frequency and T is time. Relationships don't get much simpler than that! The most common benefit that people report with improved word clocks is tighter stereo imaging, which of course is how the listener's brain interprets the signals from each ear into a spatial point. That may be intuitive to you, but it doesn't hold up in reality. What FM distortion (jitter's technical name) does at modest levels is add sidebands around every tone in the recorded sound. These tones are aharmonic, so if they are loud enough, they are going to make the music sound kind of sour. At higher levels, jitter simply keeps digital receivers from ever locking into the input signal. If the jitter level is inconsistent, you may have clicks and pops as the digital receiver falls in and out of lock. At really high levels, jitter keeps the digital receiver from ever locking into the signal, and you have silence. The reason why I can say these things so boldly is that one day maybe 5-7 years ago I built a device that added jitter to a digital audio signal in ways and amounts that I could control from mild to wild. I then tried out a bunch of digital audio gear with different amounts and kinds of jitter, and carefully listened to it and also measured its effects on the equipment. One of the devices I had that did very little to correct jitter was a fairly pricey (in its day) Denon DAC. Another device that did wonders with a jittery input signal was essentially a cheap Technics surround receiver. It can be easily shown that the position of a source can be moved in the stereo field just by manipulating the timing, Yes, but that requires timing changes on the order of tens of milliseconds and even fractions of a second. Jitter at its worst involves changes on the order of nanoseconds or picoseconds. So, jitter involves changes that are on the order of a thousandth or a millionth of what it takes to cause image shifting. so I can understand how the accuracy of the timing errors could influence where the source sits in the field. It also seems that the nature of the jitter is a factor: if it's a slow modulation it should be imperceptible, and the same if it's random - at least in relation to audio frequencies. Jitter is usually imperceptible because at this time, it is pretty trivial and inexpensive to make equipment that simply reduces it to such low levels that the next trick is to accurately measure it, let alone actually hear it. That's one of the nice things about digital, a good DAC can make horrendous amounts of jitter simply and effectively disappear. An example of a piece of audio gear that has a DAC that makes huge amounts of jitter disappear might be a $150 surround receiver. Just as a guess, I'll speculate that jitter would have to be in the 100 microsecond range to perceptually effect the imaging, and even then only if the average is modulating at something near audible frequencies. Think 10-100 times larger. That much jitter takes a major screw-up to actually have. Not to say that people just plugging cables and flicking switches in a studio haven't done this to themselves at one time or the other. If it's jumping around too much the brain will ignore it, and if it's drifting slowly enough then you're not conscious of it, at least until someone points it out. Low frequency FM distortion is like flutter and more even more likely wow. Play a good piano recording with audible amounts of flutter and/or wow, and nobody needs to tell you that is there. The ear is very sensitive to low frequency FM distortion, and its sensitivity to it goes down as the jitter frequency goes up. Obviously if it's a constant offset then that would just change where the source sits in the field, not the size of it. No, a constant amount of FM distortion is called a pitch change. If you want to change the imaging you *have* to do different things to each of the channels that you are listening to. Since most jitter gets tacked onto a digital signal that is also multichannel (degenerate case: Stereo) all the embedded channels get the same bad treatment. bottom line, no image shift. Again this is just my guess: I'm sure you mean well, but unfortunately you've got just about everything wrong. I'll bet money that you've been reading lots of subjective comments about jitter. Since there are a lot of pro audio ragazines that employ some writers are actually no more savvy than your average consumer high end audio ragazine, a lot of really weird stuff shows up where it shouldn't. the brain is a strange organic DSP and may be far more sensitive than I'm imagining here. Actually, what the brain does with sound and music is pretty well understood. Music, Your Brain, and Ecstasy by Jourdain would be a good starter, and then for the heavy lifting there are the writings of Zwicker and Fastl. I'm going to politely disagree on most of your points, probably because I didn't I meandered around instead of making a plain statement: The brain isn't like a piece of hardware, we interpret imaging mostly by the time difference in when the event reaches each ear, and that our ears are far more sensitive to this than to sideband frequencies buried in the signal. That's my personal assertion - I can offer nothing to back it up and may well be dead wrong. I also disagree that aural perception is well understood, especially in regards to timing. Well studied yes, but not understood. If aural perception were an exact science we wouldn't need listening tests for gear, would we? Sean |
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"Sean Conolly" wrote in message
"Arny Krueger" wrote in message ... "Sean Conolly" wrote in message I think that the effects of clock jitter are going to be in the time domain much more than the frequency domain. Ask anybody whose taken any EE courses in the last 50 years. The time domain and the frequency domain are very closely related. It's darn hard to do something in one domain without doing some analogous thing(s) in the other. Remember, F = 1/T where F is frequency and T is time. Relationships don't get much simpler than that! The most common benefit that people report with improved word clocks is tighter stereo imaging, which of course is how the listener's brain interprets the signals from each ear into a spatial point. That may be intuitive to you, but it doesn't hold up in reality. What FM distortion (jitter's technical name) does at modest levels is add sidebands around every tone in the recorded sound. These tones are aharmonic, so if they are loud enough, they are going to make the music sound kind of sour. BTW, I find it very interesting that so much has been written about the alleged sound of jitter, but so few have come up with anything that resembles this characterizations what jitter sounds like. If one actually ever hears audible amounts of jitter, it literally jumps out at you. At higher levels, jitter simply keeps digital receivers from ever locking into the input signal. If the jitter level is inconsistent, you may have clicks and pops as the digital receiver falls in and out of lock. At really high levels, jitter keeps the digital receiver from ever locking into the signal, and you have silence. The reason why I can say these things so boldly is that one day maybe 5-7 years ago I built a device that added jitter to a digital audio signal in ways and amounts that I could control from mild to wild. I then tried out a bunch of digital audio gear with different amounts and kinds of jitter, and carefully listened to it and also measured its effects on the equipment. One of the devices I had that did very little to correct jitter was a fairly pricey (in its day) Denon DAC. Another device that did wonders with a jittery input signal was essentially a cheap Technics surround receiver. It can be easily shown that the position of a source can be moved in the stereo field just by manipulating the timing, Yes, but that requires timing changes on the order of tens of milliseconds and even fractions of a second. Jitter at its worst involves changes on the order of nanoseconds or picoseconds. So, jitter involves changes that are on the order of a thousandth or a millionth of what it takes to cause image shifting. so I can understand how the accuracy of the timing errors could influence where the source sits in the field. It also seems that the nature of the jitter is a factor: if it's a slow modulation it should be imperceptible, and the same if it's random - at least in relation to audio frequencies. Jitter is usually imperceptible because at this time, it is pretty trivial and inexpensive to make equipment that simply reduces it to such low levels that the next trick is to accurately measure it, let alone actually hear it. That's one of the nice things about digital, a good DAC can make horrendous amounts of jitter simply and effectively disappear. An example of a piece of audio gear that has a DAC that makes huge amounts of jitter disappear might be a $150 surround receiver. Just as a guess, I'll speculate that jitter would have to be in the 100 microsecond range to perceptually effect the imaging, and even then only if the average is modulating at something near audible frequencies. Think 10-100 times larger. That much jitter takes a major screw-up to actually have. Not to say that people just plugging cables and flicking switches in a studio haven't done this to themselves at one time or the other. If it's jumping around too much the brain will ignore it, and if it's drifting slowly enough then you're not conscious of it, at least until someone points it out. Low frequency FM distortion is like flutter and more even more likely wow. Play a good piano recording with audible amounts of flutter and/or wow, and nobody needs to tell you that is there. The ear is very sensitive to low frequency FM distortion, and its sensitivity to it goes down as the jitter frequency goes up. Obviously if it's a constant offset then that would just change where the source sits in the field, not the size of it. No, a constant amount of FM distortion is called a pitch change. If you want to change the imaging you *have* to do different things to each of the channels that you are listening to. Since most jitter gets tacked onto a digital signal that is also multichannel (degenerate case: Stereo) all the embedded channels get the same bad treatment. bottom line, no image shift. Interesting that so many are so far out in left field when it comes to characterizing the subjective effect of a steady-state clock frequency error. It is really where music 101 hits digital technology 101. Simple stuff over which there is actually no controversy at all among knowlegable people. Again this is just my guess: I'm sure you mean well, but unfortunately you've got just about everything wrong. I'll bet money that you've been reading lots of subjective comments about jitter. Since there are a lot of pro audio ragazines that employ some writers are actually no more savvy than your average consumer high end audio ragazine, a lot of really weird stuff shows up where it shouldn't. the brain is a strange organic DSP and may be far more sensitive than I'm imagining here. Actually, what the brain does with sound and music is pretty well understood. Music, Your Brain, and Ecstasy by Jourdain would be a good starter, and then for the heavy lifting there are the writings of Zwicker and Fastl. Note that our correspondent regrettably instnatly this well-known and generally-accepted knowlege base competely out of hand I'm going to politely disagree on most of your points, Your gun, your bullet, your foot. Well studied yes, but not understood. If aural perception were an exact science we wouldn't need listening tests for gear, would we? Let's cut to the chase. You can disagree with most of my techical points, but you do so at your own risk. Most of the ideas you put foreward, one of which was more erroneious than the next, were developed by people who knew little about digital technology, and did only casual listening tests. That's easy for me to say, because almost *nobody* bites the bullet like say Ethan and I, and puts any of these wild assertions to a reliable test. Audio is both Art and Science and one ignores either one at considerable risk to their own personal development. |
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almost *nobody* bites the bullet like say Ethan and I
Yep, you can identify the pioneers by the arrows in their backs. |
#34
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![]() I'm going to politely disagree on most of your points, probably because I didn't I meandered around instead of making a plain statement: The brain isn't like a piece of hardware, we interpret imaging mostly by the time difference in when the event reaches each ear, and that our ears are far more sensitive to this than to sideband frequencies buried in the signal. That's my personal assertion - I can offer nothing to back it up and may well be dead wrong. I also disagree that aural perception is well understood, especially in regards to timing. Well studied yes, but not understood. If aural perception were an exact science we wouldn't need listening tests for gear, would we? Sean- Hide quoted text - - Show quoted text - Sean, the fact that you fail to realize is that the "time distortions" that you are concerned about are mathematically tied to the "sidebands in the frequency domain". If you phase or frequency modulate a signal (what you call time distortion) it WILL create sidebands in the frequency domain that a spectrum anlyzer is perfectly capable of displaying. So while you may consider it an indirect observation, any modulation in the phase/ time domain WILL also create sidebands that can be seen on a spectrum analyzer. Therefore one of the best ways of looking for phase/frequency distortion is to look in the frequency domain. This has nothing to do with human hearing, it has everything to do with modulation theory. The time doman and frequency domain are related. Mark |
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"Ethan Winer" ethanw at ethanwiner dot com wrote in
message almost *nobody* bites the bullet like say Ethan and I Yep, you can identify the pioneers by the arrows in their backs. The good news is that there are now places like this, where proper reliable listening tests are *The Standard*: http://www.hydrogenaudio.org/ The other kind of news is that while the true pioneering work in reliable listening tests was done ca. 1975, some 33 years later, so many people are still wandering in the dark. |
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Arny Krueger wrote:
BTW, I find it very interesting that so much has been written about the alleged sound of jitter, but so few have come up with anything that resembles this characterizations what jitter sounds like. If one actually ever hears audible amounts of jitter, it literally jumps out at you. I suspect that this is because jitter, other than in the extreme, doesn't really hear a sound. You don't hear a stray tone or an increase in distortion. What you perceive is the effect of jitter on what you're hearing. This is where the "shrinking sound field" description comes from, I think, though I can offer no psychoacoustic or electronic reason for it to be so. -- If you e-mail me and it bounces, use your secret decoder ring and reach me he double-m-eleven-double-zero at yahoo -- I'm really Mike Rivers ) |
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Mike,
This is where the "shrinking sound field" description comes from, I think, though I can offer no psychoacoustic or electronic reason for it to be so. Right, I don't buy it either. As soon as someone says some tweak or other affected the sound stage or fullness, I'm sure it's comb filtering in their listening room. Comb filtering is proven to exist in huge amounts even in treated rooms, so that's the most sensible explanation. I'm sure Occam will agree. :-) --Ethan |
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"Mark" wrote in message
... I'm going to politely disagree on most of your points, probably because I didn't I meandered around instead of making a plain statement: The brain isn't like a piece of hardware, we interpret imaging mostly by the time difference in when the event reaches each ear, and that our ears are far more sensitive to this than to sideband frequencies buried in the signal. That's my personal assertion - I can offer nothing to back it up and may well be dead wrong. I also disagree that aural perception is well understood, especially in regards to timing. Well studied yes, but not understood. If aural perception were an exact science we wouldn't need listening tests for gear, would we? Sean- Hide quoted text - - Show quoted text - Sean, the fact that you fail to realize is that the "time distortions" that you are concerned about are mathematically tied to the "sidebands in the frequency domain". If you phase or frequency modulate a signal (what you call time distortion) Sorry - I'm just not being clear with what I'm thinking, and I can't seem to find the right way to describe it. So while you may consider it an indirect observation, any modulation in the phase/ time domain WILL also create sidebands that can be seen on a spectrum analyzer. Therefore one of the best ways of looking for phase/frequency distortion is to look in the frequency domain. This has nothing to do with human hearing, it has everything to do with modulation theory. The time doman and frequency domain are related. I do understand that, but I'm discussing the perceived effect, specifically in spatial perception. I'm not disputing what you or Arny are saying, just that's it's not really related to my line of thought - which was speculating about how jitter could influence the stereo image as some have claimed. But the real killer for my line of thought, as pointed out by Arny, is that jitter should have the same effect on both channels. It can't be influencing the stereo imaging unless it affects the channels differently. So now I can't imagine any way for jitter or lack of to have the effects people have reported. Sean |
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"Sean Conolly" wrote ...
But the real killer for my line of thought, as pointed out by Arny, is that jitter should have the same effect on both channels. It can't be influencing the stereo imaging unless it affects the channels differently. So now I can't imagine any way for jitter or lack of to have the effects people have reported. People report all sorts of things that can't be measured or explained scientifically. Whether you believe that the perceptions are real and undefined, or psychological and imagined depends on your philosophy. |
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Richard Crowley wrote:
"Sean Conolly" wrote ... But the real killer for my line of thought, as pointed out by Arny, is that jitter should have the same effect on both channels. It can't be influencing the stereo imaging unless it affects the channels differently. So now I can't imagine any way for jitter or lack of to have the effects people have reported. People report all sorts of things that can't be measured or explained scientifically. Whether you believe that the perceptions are real and undefined, or psychological and imagined depends on your philosophy. Sometimes I believe one, sometimes I believe another. The thing about stereo imaging is that all kinds of tonal changes can change perceived imaging. Add low order even harmonic distortion and folks will hear it as the image fuzzing out. Increase the top end and people perceive it as the image getting tighter, on top of the tonal difference. It's all happening in your brain. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
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