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Do you use different AD Converters tracking same song?
Old 7th January 2013
  #91
From my understanding, clocking has to do with cadence, sample rate has to do with word length, bit depth is resolution IIRC.

Been WAY too long and I am running on three days no sleep.

However, the finer the internal resolution of a system the more steps it takes to draw a word.

So long as the steps take place simultaneously, the systems are running in synch and so long as the word length is identical in a linear path, the sample rates are identical.

The finer the internal resolution though, the more time is needed for throughput.

Latency is greater in higher resolution systems

BTW, this has to be one of the most nitpicky threads I have ever encountered on GS.

I would say this thread is operating at great latency.

Also in agreement that high latency during mastering or even playback for that matter is no problem.

It is not till we try to play along (overdub) with latent playback do we hear a problem.

Even if we monitored input for the track in record, the playback will have a latent
fresh track.

This would be like having your MTR in repro while overdubbing.

Fun for gags, but not productive for sessions!
Old 7th January 2013
  #92
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Quote:
Originally Posted by pultech View Post
From my understanding, clocking has to do with cadence, sample rate has to do with word length, bit depth is resolution IIRC.
Sample rate and word length are independent. Word length is the number of bits per word. This could be 16 or 24, 32 or even 1 in some cases. Sample rate is how many of those words per second (or pick your time period).

Quote:
Originally Posted by pultech View Post
However, the finer the internal resolution of a system the more steps it takes to draw a word.
Perhaps... but maybe a little misleading in the current discussion - and not necessarily true in all cases. I think really when this is true, you're speaking of ADCs that use successive approximations rather than anything relating to word length.

Quote:
Originally Posted by pultech View Post
Latency is greater in higher resolution systems
This is not true.

Quote:
Originally Posted by pultech View Post
BTW, this has to be one of the most nitpicky threads I have ever encountered on GS.
Perhaps. Call it pedantic if you like, but I think this is important. There is little that is so fundamental about our craft (in this day and age) and it can either be treated as a magic black box or not. If you don't want to know what's really going on in there, you'll probably be fine, but if you want to talk about it, then you have to understand some fundamentals and the basic language. It pains me to read some of the posts and I hate to be that way, but in this case I feel some things need to be corrected.

I feel there's value in knowing the real difference (besides price tag) in an mbox, an HD IO, a Lavry, and a consumer mp3 player.

-s
Old 7th January 2013
  #94
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Quote:
Originally Posted by scottwilson View Post
Clocking does not sync the audio. It simply syncs the words coming out of the digital side of the converter.
But words coming out of the digital side of a converter are audio, they are just the digital representation of it.
This is how an A/D converter operates. It takes an incoming voltage and outputs a binary representation of it where
sample rate can be thought of as bandwidth of this input and bit rate can be thought of as range of the original signal. In the context of audio it is classified as dynamic range. But in A/D conversion in general the term resolution is common.

Clocking of course syncs the audio, it syncs the digital representation of it. Clocking does not only sync the words coming out, it also syncs the entire circuitry of the unit. This is the point I think may being overlooking here?

Do you understand that clocking is a big part of how digital in general operates? An A/D audio converter or any dynamic digital converter for that matter requires a clock to function.

What I'm trying to inform you is clocking does in fact cause latency with regards to ADCs. Maybe wikipedia doesn't supply this valuable info? Maybe read up on what a digital clock is but not in the context of audio, but in the context of digital in general.
Old 7th January 2013
  #95
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Quote:
Originally Posted by emitsweet View Post
but words coming out of the digital side of a converter are audio. Just the digital representation of it.
Yup. so when converters are clocked together, you're going to get words at the same time, but the audio represented by those words are going to be shifted in time from one converter to the next.

Quote:
Originally Posted by emitsweet View Post
Clocking does not only sync the words coming out it also syncs the entire circuitry of the unit. this is the point I think may being overlooked here.
It's being overlooked because it's not true. Clocking a Lavry to a Big Ben does not magically change the Lavry's decimating filters to the same as an Avid converter's decimation filters. Again, all the clocking does is ensure that the output signal that each of the converters are spitting out have the same bit 0 timing for each word and that there are the same number of samples per second across your set of devices.

s
Old 8th January 2013
  #96
Quote:
Originally Posted by scottwilson View Post
Sample rate and word length are independent. Word length is the number of bits per word. This could be 16 or 24, 32 or even 1 in some cases. Sample rate is how many of those words per second (or pick your time period).

Thanks for clarifying that.

When internal oversampling is taking place, doesn't this place more steps along the wave shape during the drawing process, i.e. a smoother and more accurate representation of the picture?

For instance, when I use my negative scanner, the more oversampling It does the more detail appears in the digital image.

Higher dynamic range (up to the max of the scanner's ability) as well as less noise is converted even though the DPI is not changed.


Perhaps... but maybe a little misleading in the current discussion - and not necessarily true in all cases. I think really when this is true, you're speaking of ADCs that use successive approximations rather than anything relating to word length.

So oversampling in a converter interpolates the wave shape rather than checking and rechecking for errors while correcting them?

I thought this was where the latency of throughput typically comes from, the time to check and recheck as well as error correction. *scratching head* lol





This is not true.

I was referring to higher internal resolution (I am most likely using the wrong term), or rather the oversampling.

Doesn't oversampling and error correction create latency the more you do, or do some ADCs run faster to compensate?



Perhaps. Call it pedantic if you like, but I think this is important. There is little that is so fundamental about our craft (in this day and age) and it can either be treated as a magic black box or not. If you don't want to know what's really going on in there, you'll probably be fine, but if you want to talk about it, then you have to understand some fundamentals and the basic language. It pains me to read some of the posts and I hate to be that way, but in this case I feel some things need to be corrected.

Correct away!
The last time I discussed this stuff with any detail was 30 plus years ago with a guy who worked for Phillips and helped create digital recording.

I wish I could remember his name, but we called him the axe murderer because he had wild hair and slightly crazy eyes.

Most likely from too many late nights and too much coffee.

I think we all ended up looking that way after years of installs

I feel there's value in knowing the real difference (besides price tag) in an mbox, an HD IO, a Lavry, and a consumer mp3 player.

Without a doubt, knowledge is fun and exciting as well as even useful in some cases *grin*

-s
Where did you gain your understanding of this stuff Scott?

Thanks,

John
Old 8th January 2013
  #97
Emit,
What types of converters have you built, and was this for fun and learning or as in operation DIY?

As for not encountering any phase differences between your converters, have you tried putting a stereo pair across both as well as keeping them on one and A/B ing the two?

I'm sure you know that time smear can be subtle, but when critically listening on a rig that lets you hear it, the difference in imaging can make a difference in the size of the mix as well a pin point placement of sounds.

I think one problem these days can be the end product, i.e. MP3.

If the listening public is happy with these, it says something about the quality expectations of our hard work.

I for one love listening to very high quality product where you feel as if you could walk into the mix.

It's a lot of fun when this happens!

Thanks,

John
Old 8th January 2013
  #98
Quote:
Originally Posted by scottwilson
It's being overlooked because it's not true. Clocking a Lavry to a Big Ben does not magically change the Lavry's decimating filters to the same as an Avid converter's decimation filters. Again, all the clocking does is ensure that the output signal that each of the converters are spitting out have the same bit 0 timing for each word and that there are the same number of samples per second across your set of devices.

s
+1....exactly what converters have you designed emitsweet?! This sounds like a fairly basic misunderstanding...
Old 8th January 2013
  #99
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Mo Facta's Avatar
This thread interests me because I am running a Lynx AES16e with an Aurora 8 and an old Apogee AD8000 clocked together via the AES stream from the AES16e. Someone earlier said that the native latency of the Lynx is on par with that of an Avid interface (anyone know what the figure is, exactly?) and that Apogee converters have higher native latency. This concerns me.

As far as I understand, what set Apogee apart in the early days was that Bruce Jackson came up with a clever way to improve on the anti-aliasing filters that most digital audio hardware companies were using at the time. I also understand that the native latency of any given converter is determined by the anti-aliasing/anti-imaging digital linear phase filters employed. Could this be why Apogee converters have a higher native latency?

If so, anyone know what kind of figures we're talking about here?

Cheers
Old 8th January 2013
  #100
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Quote:
Originally Posted by pultech View Post
Where did you gain your understanding of this stuff Scott?
Well, there's folk who know a lot more about this than I do... I don't consider myself to have any more than a basic understanding of this stuff. I went to school in EE and tried to concentrate on audio/DSP in a school that cranked out power engineers for the TVA.

There's plenty of information for the gleaming in the white papers and manuals from the chip manufacturers, pro-audio converters, and so on. These are very interesting reading even if they are a bit obtuse at first. Those two books I linked to earlier are great as well. Then there's an occasional meaty thread on GS as well.

Lots of places to learn from for those of us who won't be converter designers, but want to make educated decisions about what we're buying.

s
Old 8th January 2013
  #101
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Quote:
Originally Posted by Mo Facta View Post
I also understand that the native latency of any given converter is determined by the anti-aliasing/anti-imaging digital linear phase filters employed. Could this be why Apogee converters have a higher native latency?
I think that's a logical conclusion since we don't know much more than what you've described. There will likely be some additional latency due to other differences in architecture but without knowing more, it's hard to say.

Quote:
Originally Posted by Mo Facta View Post
If so, anyone know what kind of figures we're talking about here?
If you look in the Apogee X-HD manual, the figures for additional latency are in there for the various configurations. I believe they are round-trip times though, so I'm unsure of what the one-way latencies would be, but it's easy to measure the differences of different devices with nothing more than a click track or a metronome with a line level signal that you can route to them at the same time.
Old 8th January 2013
  #102
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Ok, so I got some figures from Lynx support. Apparently, the group delay (throughput latency) of the Aurora (8 and 16) is as follows:

12 samples @ 1X rate (44.1k, 48k)
9 samples @ 2X
5 samples @ 4X

The latency through the D/A is:

9.4 samples @ 1X rate (44.1k, 48k)
4.6 samples @ 2X
4.7 samples @ 4X

The delay in and out of the FPGA is 3 samples at any rate.


So, at 44.1/48, we have a total throughput latency, of 0.338 ms, including the FPGA.

Interestingly, I looked in the AD8000 tech specs listed in the manual and it claims a group delay figure (in ADC passband) of 38.7 samples or .806ms at 44.1/48k.

Now, does this mean that there will be a discrepancy of almost 30 samples between converters when capturing identical sources? I will have to experiment.

Cheers
Old 8th January 2013
  #103
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Cool. Look forward to seeing experimental results.
Old 8th January 2013
  #104
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Quote:
Originally Posted by pultech View Post
... BTW, this has to be one of the most nitpicky threads I have ever encountered on GS.

I would say this thread is operating at great latency.
....
What have i done...
Old 8th January 2013
  #105
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Quote:
Originally Posted by superburtm View Post
I will sometimes use a different (and even cheaper) converter for certain trks. There is some applications where a toppy crispy hi end converter is just to aggressive sounding ime.
Converters add a sonic thumb print.
THANK YOU!




So there was no issues mixing the song, right?
Old 8th January 2013
  #106
Quote:
Originally Posted by emitsweet View Post
But words coming out of the digital side of a converter are audio, they are just the digital representation of it.
This is how an A/D converter operates. It takes an incoming voltage and outputs a binary representation of it where
sample rate can be thought of as bandwidth of this input and bit rate can be thought of as range of the original signal. In the context of audio it is classified as dynamic range. But in A/D conversion in general the term resolution is common.

Clocking of course syncs the audio, it syncs the digital representation of it. Clocking does not only sync the words coming out, it also syncs the entire circuitry of the unit. This is the point I think may being overlooking here?

Do you understand that clocking is a big part of how digital in general operates? An A/D audio converter or any dynamic digital converter for that matter requires a clock to function.

What I'm trying to inform you is clocking does in fact cause latency with regards to ADCs. Maybe wikipedia doesn't supply this valuable info? Maybe read up on what a digital clock is but not in the context of audio, but in the context of digital in general.
Quote:
Originally Posted by scottwilson View Post
Cool. Look forward to seeing experimental results.
also looking forward to it Mo!

Sima,
You really,stepped in it

This is a great subject

Scott,
I'll check out the info, thanks!

John
Old 8th January 2013
  #107
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Quote:
Originally Posted by scottwilson View Post
Cool. Look forward to seeing experimental results.
Well, next time I'm in the studio is Friday so I'll try it out then.

Funnily, this is actually somewhat exciting for me to test, despite the horror of my recordings being offset 30 samples. Don't ask me why. I'm a geek like that.

In any case, I don't really ever use more than eight mics on a drum kit and usually use the Apogee in overdubs or to take scratch tracks. I think in those cases 30 samples isn't a big deal.

Well, I hope not.

Cheers
Old 8th January 2013
  #108
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Quote:
Originally Posted by pultech View Post
BTW, this has to be one of the most nitpicky threads I have ever encountered on GS.
I'm actually finding the techinical discussion interesting. What I think is nitpicky to the n'th degree is the original suggestion of choosing from a variety of AD's like you'd choose from a variety of mic's. Sure, latency issues aside, different AD's will sound slightly different, but seriously... it's slight. Moving your mic a 1/4" would generally change the sound to a greater degree. Having an arsenal of AD's at your disposal so that you could pick the right one for the source seems like an extreme form of OCD to me.
Old 8th January 2013
  #109
Quote:
Originally Posted by bonestar View Post
I'm actually finding the techinical discussion interesting. What I think is nitpicky to the n'th degree is the original suggestion of choosing from a variety of AD's like you'd choose from a variety of mic's. Sure, latency issues aside, different AD's will sound slightly different, but seriously... it's slight. Moving your mic a 1/4" would generally change the sound to a greater degree. Having an arsenal of AD's at your disposal so that you could pick the right one for the source seems like an extreme form of OCD to me.
I was really being goofy about the nitpicking...
Most GS threads are nitpicky amd are supposed to be necause this is Gear Slutz

No, my microphone is better than yours!

There is so much amazing gear out there and has been for way longer than I have been doing this which is only 30 some odd years.

I fantasize about using those old EMI consoles to track some sessions, all the amazing old mics, so,e of which I have had the joy to use many times but am not rich enough to own.

Now we even have digital conversion that has become fairly affordable to own a few channels of the very finest.

This is an exciting time to be in audio in many ways, and a difficult time as well.

The bussines has changed a lot in my brief time in it and many of the great studios are now gone.

I miss those rooms!
Old 8th January 2013
  #110
This Dude Abided's to these three rules, when using different branded AD converters.

1) Overdubbing with different AD converters is more than fine
But Would I? It is really no big deal, if you must or "have to"..Lets just say I would not change them out, for different sounds while recording. I just would not.

2) Spreading phase/time related sources between brands is a no-no
I've got enough time related shift to struggle with on my multi-miked setups. I don't need ANY of my mic signals getting delayed by ANY amount of time.

3) Make sure all Digital Audio Clocks are in sync, system wide. Which has absolutely nothing to do residual latency delay time on any AD converter.
Old 8th January 2013
  #111
Have to agree there Doc!

Running an Antelope Isochrone OCX for clocking over here and have UADs Little Labs IBP for phase correcting stuff when needed.

It's a very neat tool to have available and works well.

I remember a session I walked in on while working at Center For The Media Arts as a tech and substitute teacher.

The mix a teacher had going with his class sounded spacially really wild and I asked if he had checked the mix in mono as they were going along.

The reply was "of course I do that!!!", so we hit the switch and the mix dissapeared except for some of the vocal, the cymbals and very tinkly sounding guitars and keys.

I got the stinkeye and then we had a good chuckle.

A friend said, "At least having good phase does not kill a stereo mix on a CD, imagine if they were cutting to vinyl"

I think critical listening and checking for phase/hearing phase problems is something that has fallen by the wayside with some of the new garde as it falls under the technical side of mixing/tracking.

There are so many plugins now that we just HAVE to use them all in a mix.

HAAS effect?
Is this related to avocadoes?

Move a mic a little bit?
We can fix that with EQ!

Do you run a stand alone master clock in your place and if so, what did you hear as the benefits?

Thanks,

John
Old 8th January 2013
  #112
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Quote:
Originally Posted by pultech View Post
Emit,
What types of converters have you built, and was this for fun and learning or as in operation DIY?
Hi John,

Did some work with 8 bit successive approximation A/D about 20 years ago in Sunnyvale in the Valley. Worked in photolithography, doping, did some photomasks and worked manufacturing bare reticles for semi.

Quote:
Originally Posted by pultech View Post
I'm sure you know that time smear can be subtle, but when critically listening on a rig that lets you hear it, the difference in imaging can make a difference in the size of the mix as well a pin point placement of sounds.
never had an issue with latency to the point it was something I noticed. Maybe I'm not as seasoned as some here with my ears?

My only question 20 posts ago here, was really about the clocking and latency relationship. According to gs experts here there is no correlation. But however I do not know much specifically about Audio A/D which are more complex than most conversion since there is more error detection going on, presumably? maybe? I guess that's what causes this delay as stated previously by someone?

However I always thought when you clock 2 digital devices you are clocking 2 digital devices in tandem sharing one clock for all circuitry in both units. With an SA, the master clock is in series with the SAR then any latency that occurs is within the clock cycles of the SAR input signal, as they converge to the output register data structure/word. Sampling at a 48k frequency there is going to be some buildup and it is a result of waiting for the next edge of the clock. If SA based A/Ds are clocked they both have a phase accurate output. So in this type of scenario the latency is in fact based on clock pulse accumulated from each cycle of the frequency, collectively.

Apparently with audio converters which I learned here yesterday, this isn't the case and clocking will not in fact guarantee phase accurate digital audio data on the A/D output? So apparently the delay with a lavry gold is result of processing that takes place independent of clock pulse and any slave will not actually be sample accurate is what is confirmed here by others.

news to me
Old 8th January 2013
  #113
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Mo Facta's Avatar
Does a phase tool do much else than nudge samples around? What trickery is beneath the hood?

Cheers
Old 8th January 2013
  #114
Quote:
Originally Posted by pultech View Post
Have to agree there Doc!

Running an Antelope Isochrone OCX for clocking over here and have UADs Little Labs IBP for phase correcting stuff when needed.

It's a very neat tool to have available and works well.
I am a huge fan of running the IBP hardware. For me, It works better to get these things worked out in the analog realm, while I am tracking. I find you just cannot do the same techniques with the software. Because you cannot record with them. I cannot split an instrument to both an Amplifier and a Recorder with them. But they are fine for playing around inside the box. The hardware works so much better for me, because I would rather take the time on the front end of the recording, than mess with it on the back end. That's just me though, I am sure we can all agree each tool has its own place and "time"..

Quote:
I remember a session I walked in on while working at Center For The Media Arts as a tech and substitute teacher.The mix a teacher had going with his class sounded spacially really wild and I asked if he had checked the mix in mono as they were going along.The reply was "of course I do that!!!", so we hit the switch and the mix dissapeared except for some of the vocal, the cymbals and very tinkly sounding guitars and keys.

I got the stinkeye and then we had a good chuckle.A friend said, "At least having good phase does not kill a stereo mix on a CD, imagine if they were cutting to vinyl" I think critical listening and checking for phase/hearing phase problems is something that has fallen by the wayside with some of the new garde as it falls under the technical side of mixing/tracking.

There are so many plugins now that we just HAVE to use them all in a mix.

HAAS effect?
Is this related to avocadoes?

Move a mic a little bit?
We can fix that with EQ!
Too funny and honestly a little sad at the same time. If every bad mutha$%$$#% mentored someone, we'd have a lot more bad mutha%$%^$## engineers out there, passing it down.

Quote:
Do you run a stand alone master clock in your place and if so, what did you hear as the benefits?

Thanks,

John
Right now I am running the Symphony I/O on Internal Clock. I also have a Rosetta 200, which I clock via the Symphony I/O. This is the main ADA I use at my studio. Though we also have a DM2000, an array of MOTU stuff, and a Big Ben clocking all of that. I pretty much for the MOST part use my Apogee gear. I feel the Symphony Clock output, [with a real short WIDE EYE cable] makes a difference to the Rosetta, after listening back and forth, so that's how its rolling.

I have posted most all of my experiences hearing clocking artifacts on gearslutz, and I would say that it has got to the point, where it feels absolutely nauseating to continue. There is so much nonsense out there these days, and as a Sales Rep for a company that Sells High End Converter, and Clocking Products of varying degrees, I only recommend what I believe to be correct for the applications of the end users.

This means, that if someone asks me, if they should purchase a Master Clock, for only Sonic Reasons, I simply reply that is waste of money and an incorrect use of the tool. Whatever artifacts you can identify are subjective. Meaning that not everyone will hear them, nor understand what they are hearing. But if someone has a legit reason to employ a master clock, you simply have to evaluate what it does to your system.

Any change may or may not be present. There is too much variance to explain one single correct answer, unless you've had some experience clocking these things together and listening to them [which is kind of a waste of time IMO] I believe I am super sensitive when it comes to things of this nature, and I hear what is happening just fine. But it's reeeeeeeaaaaaallllllllyyyyyyy small amount of spectral change/distortion.

Me, personally? I think comparing clocks, and esoteric clocking stuff, is a bit like comparing dithers on a mix. Boring as Sin and a waste of time for me. If it takes me longer than 10 minutes, to figure this stuff, there is a serious problem. I've got other fish to fry up!
Old 8th January 2013
  #115
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Quote:
Originally Posted by psycho_monkey View Post
This sounds like a fairly basic misunderstanding...
seems like a trend , everything appears to be a misunderstanding when I step in, but luckily you have the understanding of thing others miss but we appreciate your knowledge it has made me a better engineer for sure. I recall 3 weeks ago we went back and forth about "signal" and DAWs. Good thing you set me straight on that too. Now I know data structures and registers process signal.

I will add, clocking in all A/D converters I have worked on will add additional latency due to the zero cycle void. This is very very very small delay. No where near 5ms. So it would not impact a phase/time/delay relationship except for very extreme
RT applications. used in conjunction with a RTOS.

So I suppose with higher end A/D such as lavry, additional jitter/error detection filtering, noise filtering cause increases delays is what you are saying?

As long as you don't claim what is coming out of a A/D output is a "signal" we are good here. If you can acknowledge it is in fact data, I can concede that based on your experience, expertise and knowledge of A/D conversion that only Audio A/D are exempt form zero cycle latency. Is that true?

I am happy I learned something here. It is no longer a misunderstanding. It is now data I can store in my neurological unit. I guess I was a little overzealous (falsely) assuming ADCs used in Audio converters are not that much different than other ADCs used in other applications and have similar characteristic specifically with regards to clocking and latency. Since every ADC I have used whether SA, Counter ramp, ladder, Binary weighted and or parallel encoded used in data acquisition, have latency due to gaps in clock cycle respective to analog input signal being encoded into binary data. I guess ADCs used in Audio A/D are exempt from this inherit flaw?

Based on your expertise and vast knowledge of A/D conversion what specific design functional characteristics are used for ADCs commonly used in, and specific to just audio converters? What type of sequential encoding methodology are these specific ADCs based on? Or do they even use encoding in these?
Old 9th January 2013
  #116
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Emmit
THis is condescender #01 ---For the last time
Digital Audio CLock is nothing but a square wave theat keeps the sample frequency in phase between multiple devices --The sample frequency has nothing to do with the audio or real time of the audio being transfered only that the audio has been transferred properly (wavforms reconstructed properly)-WORD CLOCK IS A WORD CLOCK NOT A TIME CLOCK---
No matter how many ridiculous confusing statements you utter --think about this and maybe just maybe you will understand.
You seem to try to find a reason to make yourself sound correct no matter how much BS your trying to feed people on here--
Old 9th January 2013
  #117
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Quote:
Originally Posted by FestivalStudios View Post
Emmit
THis is condescender #01 ---For the last time
Digital Audio CLock is nothing but a square wave theat keeps the sample frequency in phase between multiple devices --The sample frequency has nothing to do with the audio or real time of the audio being transfered only that the audio has been transferred properly (wavforms reconstructed properly)-WORD CLOCK IS A WORD CLOCK NOT A TIME CLOCK---
No matter how many ridiculous confusing statements you utter --think about this and maybe just maybe you will understand.
You seem to try to find a reason to make yourself sound correct no matter how much BS your trying to feed people on here--
lol, take it easy mate

with any digital converter (not necessarily and audio converter) the clock controls more than the data output, it is controlling the pulse of the ENTIRE unit's digital circuitry. All of it.

You are not being condescending you just appear to be completely confused on exactly what a DIGITAL CLOCK is and what it is used for collectively. BS? stop it

you guys don't even know what zero sample latency is and you are trying to explain A/D in a serious manner? come on guys get with it. You read some sampling article in a proaudio fanzine and you think you know all about converters.

It's as if some of you chaps think A/D conversion is unique only to pro audio. I'm talking about A/D converters in general.
Forget about audio applications for a moment. Some latency inherent to ADCs is in fact caused by clocking edge intervals. It's not BS, it's only ridiculous confusing statements I utter because you don't understand digital clokcing, it's fact
ADCs can incur latency/delay/waitstate/timegap whatever you want to call it

lol

The reason I put down some statements that you found "ridiculous confusing" was to test your knowledge. You pawn off digital operational fact as confusing, well that's not my problem, maybe learn how digital works before you insult someone and accuse them of BS. So my knowledge in the area of latency due to Audio filtering is non existent, no reason to be nasty about it. It doesn't make my statements incorrect.
Old 9th January 2013
  #118
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Quote:
Originally Posted by emitsweet View Post
The reason I put down some statements that you found "ridiculous confusing" was to test your knowledge.
Old 9th January 2013
  #119
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Rick Sutton's Avatar
 

Quote:
Originally Posted by emitsweet View Post
I'm talking about A/D converters in general.
Forget about audio applications for a moment.
We are audio people. You should understand that.
Old 9th January 2013
  #120
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Quote:
Originally Posted by Rick Sutton View Post
We are audio people. You should understand that.
hey Rick,

yeah I get it, but everyone is jumping down my throat for stating clocking cycles can cause delays. It's not really a big deal
however when people start calling someone clueless it sort of eats at me personally.

A/D is actually a very simple concept. In reality you can design a very primitive A/D converter with some archaic opamps wired in parallel to the inputs of an encoder and a bunch of resistors in series in front of the opamp, collectively acting as comparator. You could build it with ratshack parts for less than $5. You don't even need a clock. It's called a combinational parallel encoded A/D converter.

Pretty useless in audio. But you could build an audio one with not much more in actuality. It would be useless in pro audio but technically all A/D takes an analog signal determines it's voltage and assigns a digit accordingly. The analog voltage could originate from a sound, light, a mechanical device like a switch, a pedal a piezo even a transducer like a digital mic would do.

All the stuff that makes it hugely expensive is for error correction and signal quality and finally data integrity coming out
anyway enough babble, All I stated 4 pages back in one little blip of a comment is; when the analog voltage coming from the opamp section of the comparator circuitry; when controlled by a pulse (clock) for sequential capabilities; has to wait for a clock cycle each time the voltage is encoded for each input (sample), while each parallel bit is subsequently output. Not a big deal but because a clock is controlling this part of the process there is technically a delay as it waits for the next edge to rise or fall. Does this delay matter? does it cause enough delay to cause phase issues? I don't think so, but the fact of the matter is this is called zero cycle latency and it is 100% caused by the edge triggering clock cycles in the controlling of D, SR, JK or whatever FF/latch used.

Again NO BIG DEAL, but when you get 5 angst replies to a post stating clocking has nothing to do with latency, or latency has nothing to do with clocking when in fact it, in actuality can and it does at least with some applications, it's a little irritating getting bashed; unwarranted I might add.

Admittedly this latency caused by clocking perhaps is not something audio engineers or any human need to worry about, but it is something that occurs. I stated it of the cuff, it is true and I don't think it's ok for people to state it's wrong when they may not really understand the very simple basics of A/D. And the way it was being contested here was like I'm some kind of idiot and I don't think that's fair on a messageboard. Also defending something you believe to be true even if you are wrong is NOT being argumentative. It is called debate. If it's not applicable to this topic, then state that it is not applicable instead of implying"I'm stupid" "you know nothing" "they are not related" "BS"........................

Admittedly I know nothing about highspeed cutting edge audio converters electronically with their robust error detection filtering, oversampling yad yada yada, but I did program a few primitive A/D converters on FPGA and had some ICs fabricated of the design, and as much as a joke as they are compared to a modern audio converters or anyhting you guys have layed out, the basic principles still apply however if only to some very very small degree.

I was just adding a few observations I had many years ago, My apologies guys I am not up to your intellectual standards




Go niners!
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