Without Prejudice:
I want to be very clear about this post. I appreciate good gear that helps the mutual advancement toward better recording. I will try to be careful with my words.
Fact or fiction? Hype or BS?
Demystifying Super Clocks. Who needs an external clock?
I used the 10m for over a year. I am a guy looking for any improvements at this point so when someone says, hey, this thing makes "Huge" (not subtle") improvements to your sound... I am all over it!
I understand that once you are up to a certain level of sound, the last 5% towards pristine sound is baby steps that come at a price$.
I put this super clock to the test using it with the Orion 32 (including RME, Lavry and Prism converters).
After a year in my rack, I must be honest here and say, I never heard a bit of difference with or without the 10M.
I even doubted my ears so I had young ears (better than 18k hearing) spend time on this test and not one person heard the value with the 10M. We did numerous advanced null tests, all of which proved worthless. If this clock intended to "improve" the sound quality of your existing converters, it sure didn't do that for me.
What could be wrong I ask so I called the rep and began searching forums for answers on this product.
My finding and gut feeling about this clock look like snake oil and I will tell you why.
When I called the rep in this video he exclaimed I needed better speakers or better treatment. He cared nothing about what I heard or had any interest in the system I use, which btw ranks as a world-class hybrid tracking and summing system.
I'm disturbed by how this product is being marketed and shilled. It has me questioning the integrity of company marketing and why top-level engineers are promoting it to the mass as a sound improvement tool just by connecting it to your converters.
Well-respected people are claiming huge improvements so much so, that they are suggesting people doing electronic music get on the chain with this.
In fact... this was how the rep sold it to me too. He sent me a mastered track that was a before/after of electronic music :cautious: The difference between both versions was HUGE! This was electronic music, ITB music so I am thinking. WOW! This thing is amazing. I want one.
But... the videos, forum and direct marketing say a much different story.
As we see in the above video, here are world-class guys claiming huge improvements in their sound using the 10M, yet I am using it, running a very sophisticated mixing and mastering system and hearing no change. Why them, not me? How come the rep sent me a before-after example and said...
The rep says to me: "don't tell anyone about this track but you have to hear this... this is what tracks sound like with the 10m".
Let's get a few scenarios out of the way.
- Could the reason I don't hear any change be that I use Sequoia and do not have added ADDA products messing, or reducing the stability of my internal clock?
- Could it be because I do not round trip process back to the tracking DAW?
- Could it be that my system is simply telling the truth and exposing something?
Internal vs external clocking
Numerous top-level testimonials say internal clocking is a choice.
Something worth mentioning. I notice the majority of users who claim this clock is improving their sound are either on Pro Tools or Logic.
I have read less trusting comments from people claiming huge improvements in PC systems.
Being said, I feel there is a placebo effect and/or support of purchase going on here more than anything though, and let's not forget that the rep also sent me an audio example that he did, claiming it was the difference between the 10M in and out.
To even put more questions out, I have asked for proof it in a public A/B and no one has stepped up. We just read the hype without hearing the comparisons online.
This product is only good for those having seriously bad clocking, to begin with. It's certainly not for the guy with one converter in a simple system. Especially if all your mixing is ITB.
My feelings are, if a hybrid system is clocked properly, to begin with, your clock should be just fine. This is suited for a studio running all sorts of digital products in a rats nest going round and round.
Thoughts?
Topic Tags
Comments
kmetal, post: 435418, member: 37533 wrote: but I think a lot of
kmetal, post: 435418, member: 37533 wrote: but I think a lot of the online attitudes come from people looking for approval of their hair brained ideas,
I don't even mind the "hair-brained" ideas - because while some of the ideas truly are wacky, and serve no purpose other than to damage the fidelity of a recording, at the same time, new recording and mixing methods need to come from somewhere, right?
And not all of the good ideas are (or have been) initially "accepted", either ...
Like the first guy to find out that by recording above 0db on a tape machine, up around +3, that by "saturating" the tape with signal, some pretty sweet-sounding tones could result...
( and doing so during a time in audio history when this was actually a terminable offense at many studios).
Or the guy who first thought, "I wonder what would happen if sent a signal - like maybe a vocal - to a speaker that's in a large hall... then put a couple microphones in that same hall, to pick up the sound of the voice through that speaker in that hall...
Or the guy who decided to record a multitude of instruments, all playing the same musical parts together at the same time, and then recording it, and then doing it again ... on top of the first performance ... to create a big sound, a huge sound, kinda like a "wall". Yeah..."wall" is a good word to describe it ... like a big wall ... a "Wall Of Sound".
Or the guy who decided to record sound with a pair of microphones that were placed so that their axis's were at right angles to each other... his name was Blumlein.
Or even more recently, a couple of members of an internet audio recording forum named Recording.org, who broke ranks from the masses of other digital/analog hybrid workflow fanatics, and determined that, instead of using the more popular and more accepted process of returning the signal from external processing back to the same DAW, or that instead of rendering a mix through the DAW's internal stereo bus, that maybe , just maybe... using a second DAW instead - as a sort of "mix down DAW" - would sound better? ... ( audiokid Boswell )
Once in awhile, there's that one idea that might sound a little wacky initially, but that at the same time, is also a little intriguing, and turns out to be something that someone might find useful. ;)
IMHO of course.
audiokid, post: 435419, member: 1 wrote: Remy [SPOILER=Remy]star
audiokid, post: 435419, member: 1 wrote: Remy
[SPOILER=Remy]started stocking me on FB a few weeks back. I had to report her. Here I was, just posting some threads and what my hybrid studio looked like (specifically the beautiful Pultec rack). The next time I returned, there she was, disturbed as usual, yelling out that I was an impostor and clueless about recording. That was terrible, undeserving, mean.She is nice to some, and mean to others. There is no in between for people to me. You are either one way or the other (good or not good). With us or against us.
If only I had know, I would have put up an sm58 and she would have had less trash to say to me lol.
What a flip case.[/SPOILER]
Was this FB member also an RO member?
I'm certainly not defending her actions by any means... along with Chris, I was one of the people here that she put into her cross-hairs at the end, after the mods all got together and decided that, based on what was best for the forum on the whole, that she needed to be cut loose. She went off on both Chris and myself pretty good, taking final - and very personal - shots at us, as she was being shown the door. I don't blame her for that, though. It's a natural response for some people to do that, they have to point their anger at someone.
But... some of her behavior can at least be explained ... although that doesn't justify it - much of it was due to the slew of meds that she was on; she suffered from bipolar-ism and depression, and her behavior was classic of those illnesses.
She could be very nice ( and very helpful) at times - she was quite knowledgeable about many things in audio - Remy was certainly no dummy. To the contrary, she was incredibly intelligent. But, at other times, and with no warning at all, she could turn psycho, and become offensive, and ramble nonsensically at the flip of a switch. She could be brutal with new members, or with those who were just entering the craft, and who were simply looking for knowledge.
I'm fairly certain that she also suffered from some form of brain injury - the result of a very serious physical beat-down she took because of her lifestyle ... Remy was a transsexual ( which, BTW, I neither judge her for or care about not one bit)... I find it very sad that there are still people out there who want to hurt or kill that which they find different, or don't understand...
In the interest of peace, and to keep RO's reputation as a forum that welcomed new members, it was decided that she had to go. But no one made that decision lightly. We sweated over that one for several days.
FWIW
-d.
DonnyThompson, post: 435422, member: 46114 wrote: Or even more r
DonnyThompson, post: 435422, member: 46114 wrote: Or even more recently, a couple of members of an internet audio recording forum named Recording.org, who broke ranks from the masses of other digital/analog hybrid workflow fanatics, and determined that, instead of using the more popular and more accepted process of returning the signal from external processing back to the same DAW, or that instead of rendering a mix through the DAW's internal stereo bus, that maybe , just maybe... using a second DAW instead - as a sort of "mix down DAW" - would sound better? ...( audiokid Boswell )
Nice way of writing it, Donny, but my approach came from a slightly different direction:
Boswell (8 Dec 2014) wrote:
In my case it started when I was much younger (and that is a long time ago) when some newer jazz stereo LP releases were billed as being recorded "direct to disc", and I recognised that the sound on these discs was different from that of conventional studio tape-edit-tape-master recordings. The direct-to-disc releases were being mixed, mastered and cut in real time, so had no tape or other storage medium in the loop. Don't get me wrong, this is nothing against tape, as it has pleasing dynamic characteristics all of its own that even today we are struggling to emulate in a convincing manner. This was all about purity of path, and although the direct-to-disc experiments did not really transfer out of the jazz realm, they did lay down a tantalizing standard that kept me thinking.Phase forward to this century, and I kept on trying to get my 2-track mixed digital recordings to have something of the sound that I remembered from years ago. The mental breakthrough I had was purely that: a mental concept. I had been so immersed in and tied up with the intricacies and possibilities of multi-track recordings, DAW mixes and digital processing, that I had lost sight of the obvious - whatever does the final 2-track capture should see what it thinks is an analog stereo mix, just as the disk cutter used by those skilled direct-to-disk engineers did. Once I used my 96KHz recordings (on a bank of HD24XR recorders) and replayed those tracks together with any effect and dynamic units needed into an analog mixing desk with external stereo ADC capture, I suddenly got the sound I had been looking to reproduce for over 30 years. It didn't really matter what 2-track capture unit I used because the sonic quality I was looking for was part of the mix and not some sort of magic that had to be added later. I also found the whole system was much less sensitive to each component not being perfect, although of course there were pieces such as the 2-track capture ADC where sonic quality was more important than others.
Once you look at a system this way, you can see that an uncoupled 2-track capture DAW is your target, and you work backwards from there to feed it with an analog mix that the capture system thinks is basically a stereo microphone. That's the picture to retain in your head, and the more work you can do to achieve this aim, the better the result. For example, I mentioned 96KHz recording. This went along with another of my strong beliefs that the top octave of the multi-track recordings should not be represented in the final captured mix (see threads passim). What comes out of the analog mixer is a stereo signal of essentially 40KHz bandwidth, but with any phase effects due to anti-aliasing filters on the 16 or 24 source channels kept out of human audio range. Capture the mix at 44.1KHz for your target CD and you suffer only the effects of the capture system's filters.
It makes a real difference, particularly in the smoothness of both phase and frequency responses in the final top octave (10 - 20KHz). Chris and one or two others here at RO tried it, and although not everyone said they would instantly re-build their studios to encompass the method, they did notice the difference in sound.
DonnyThompson, post: 435126, member: 46114 wrote: It's not my in
DonnyThompson, post: 435126, member: 46114 wrote: It's not my intention to come off as sounding sarcastic here, Ric ... I'm sincere in my comments and curiosity ).
1st It's all good,
2nd ) I should clarify the 6K POS comment, I didn't mean for it to come off as a dis towards the 10M, it was meant as a knock on the V-amp, in that it takes a 6k clock to make the V-amp's DSP accurate.
hell my little old ART wordclock is just as good..( I really should of said, adequate). once you start putting a heavy load on the ART, I've had some issues with tracks not lining up when I go to edit but here again it's depends on Length and count, and which DAW..... with POO tools, I can see why the ART gets Dissed as much as it does, but Reaper and SonarX1 it's good....
The ART is fine for Clocking the HD24(s) to the Tascam MX2424 and the XP(32bit) reaper DAW, Its good for the Presouns Pre's in to an HD24 and then off to the DAW, it's when we load up the little guy with every else that it shows it's $99 price tag,
So basically, what you're saying is, that for 6 Large, you found that "it made some deep editing a bit easier", and some of the DSP "lined up tighter with less quantizing over a 6-8 min time frame..." and this improvement was based on visual markers, and showed no obvious sonic advantage; and that in the end, you found that your cheaper Art word clock device worked just fine, and accomplished the exact same thing, as the second-mortgage 10m clock sync did...
(I wouldn’t say “exact same thing” but close, under 4-5 Min, half the amount of tracks…. Yeap)
Have I got this right? (basically, Yeap)
If so, and if I hadn't already been convinced that this 10m was unnecessary, your comments pretty much sewed it up for me.
(It's not my intention to come off as sounding sarcastic here, Ric ... I'm sincere in my comments and curiosity ). See above No 1
So then, how does a device such as the 10m benefit those of us who are 100% ITB, 90% of the time? ………………………
Honestly I think that is something that People in that scenario have to decide themselves. I decided to add my $.02 worth for two reason, one was I saw a thread over at GS, about how the 10M had changed this individuals’ studio, and I know I’m going to get flack for this, but I’ve meet him twice, and have listen to him talk, without a doubt he has done some great work, but sonically in my opinion, if this clock woke up his studio in the manor(is this the right spelling) that “gorgeously musical it makes my mixes”
then he needs to step away and take a vacation to clear his senses, and/or dump Poo tools
2nd is hard to explain, but it has to do with we need more Voltaires and less Saul Alinskys in this world.
( okay... so, does UAD's DSP/plug-in system, or the Waves Digigrid, or any of the other external DSP processors that hold all the plugs/ FX and the processing power for these various plugs - require (or even have) a word clock connection? I'm asking because I don't know...)
I honestly I can’t answer that, I know UAD’s satellite hooks up via firewire so I would assume that the DAWs Internal clock(soundcard would handle clocking chores and line up the algorithms
Now Hardware DSPs like Eventide V4 , Eventide H7600, Lexicon PCM92 I’ve found they work better when it’s all clocked as a unit and not individual
Or, how does the 10M help those who use a 2-DAW system, "mixing down" in real-time from DAW 1 ( the production "multi-track"), to another computer, without any external digital FX being inserted into the workflow, and which is being used simply to capture that real-time playback, similar to what we used to do with a 2 track deck??
I don’t think it would help at all,
Personally, I can think of many other things ( speaking for me) to invest 6 G's into gear-wise, that would show an obvious and substantial difference in sonic quality... especially since I so very rarely work with any external DSP signal processing.
I’m with you here…I have a 10m only because when I was done with it and went to return it, 1 I was offered a price that I was not going to turn down…..and the kid used those big Doe like eyes.
Mics, Pre's, a quality monitor controller, room treatment... (I'm just kinda thinking off the top of my head, here) ... these are the things that would matter more (to me), and that I think would make a substantial ( and audible) difference to a modern DAW Studio....
If I purchased a $ 6000 clock, it'd better do a whole lot more than just make some of my editing "easier", or correct slight drift/quantize past a 6 min project length.
What I'm lead to understand, especially from Chris's ( [[url=http://[/URL]="http://recording.or…"]audiokid[/]="http://recording.or…"]audiokid[/] ) various posts on the subject, ( and now yours as well) is that the only real difference the 10M makes, is if you integrate it into a system that is already "clock-whacked" to begin with - and of course, it should improve things at that point, because that's what it's supposed to do, right?
Yeap, it should
But then, the other side of the debate kicks in - discussing easier ( and far less expensive) ways to fix the original clocking issues, instead of continuing on with the original clocking problems, and simply "camouflaging it" with a $ 6000 band-aid.
Just trying to keep all this straight in my head...
Rosendahl Nanosyncs, was the 1st clock I had, and honestly I think the best bang for the buck when it comes to clocks, especially if you do video work.
1st I don’t want any one to think I’m say you should by an exter
1st I don’t want any one to think I’m say you should by an external clock, but if you do buy the best you can afford and keep in mind the more load you put on the cheaper clocks the more they show their price.
Electricity/ electric current travels roughly +/- .00233 cm/sec Depending on a LOT of things.
This will be important in a few Mins…… we all know a sample rate of 44.1kHz is equivalent to say a slicer taking 44100 slices out of a sound wave per second 00:00.44100
For conversation, Recording Dude, has a 1 Pm session, 24 tracks, turn on the power conditioner feeding 3, 8 channel preamps hooked up to the DAW via lightpipe (firewire, what have you) turns on the DAW then goes and turns on each Pre, none of these 4 items are externally clocked, they all rely on internal factory clocks each item’s clock starts when it is turned on,
DAW = 11:00.0003 ….1st Click(slice) of the Clock = 11:00.44130
Pre (1) 11:03.0004….1st Click(slice) of the Clock = 11:.03.44140
Pre (2) 11:08.0045….1st Click(slice) of the Clock = 11:08.44551
Pre (3) 11:10.0034….1st Click(slice) of the Clock =11:10.44440
Now reality is these 4 items are constantly taking slices now, doesn’t matter what they are slicing at this moment they just know they are doing their job. Also 11:??. Doesn’t matter, it’s the. 00.????? that matters.
Band comes in Gets set up. Mic’ed up ready to go, we have 12 tracks of drummer (keep in mind now that 12 tracks = 2 Pre’s and the DAW, 3 different clock start times)
Keyboard player has 3 Keyboards, 1 Runs stereo ,2 and 3 run mono 4 tracks (2 different start times, DAW’s and Pre#2
Bass takes 2 tracks, 1 DI 2nd live iso room Mic’ed 15in cab (2 Different Start times DAW’s and Pre 3
Guitar 1 Stereo ,1 Mic Left 2nd Mic Right Cabs (2 Different Start times DAW’s and Pre 3
Guitar 2 Mono, but we use 2 mics, 1st off the speaker, 2nd somewhere in the room (pick a spot) again we have 2 Different Start times DAW’s and Pre 3
22 tracks. Record is Pressed at 1:16.03.00000(trust me here) , why does this mater, what is the point to this ramble
DAW takes its 1st slice at 1:16.03.0000 which is .00010 BEFORE Pre (1) takes its next slice, which is Pre 1 1st slice of sound wave ,Pre(2) is now .00541 behind ,Pre(3) is.0034 so we have
DAW=.00000/.44100
Pre1 =.00010/.44110
Pre2=.00541/..44641
Pre3=.00340/.44440
….we can NOT hear this but the DAW is looking at Pre1 1st 00010 as silence, Now when Play is pressed on the DAW, to play back ,what does it do, It lines up each track bassed(yeah I’m a bass player) on the DAW’s clock.
So the 1st 8 drum tracks are moved .00010, to line up with the DAW’s 1st recorded marker, next 4 drum tracks are moved .00541, so now we have a time discrepancy of -.00531. between 1-8 and 9-12,
Now the Keys are off by -.00531 and the guitars and bass are off.-0033
And we haven’t even tossed the monkey wrench of Eventide V4 , Eventide H7600, Lexicon PCM92
Which all have their own clocks,
Now this matters if you have drummer whose every 4th and 6th kick drum hit is weak and you have to replace the WHOLE dam Kick drum track over a 7 min Doo Woop medley with triggered drum sounds (dude is 79 I’m hoping I still am around to play at that age)
The over heads are on track 11/12, and snare is 4, so we have some original kick drum bleed on 3 tracks, 1 in time with the Kick track and 2 slightly out…again we can’t hear this deference but we can see it in the markers…and let us not forget if one of these Pre’s decides it wants to drift and sample 2 mins @.44099 yes it does happen,( it will not go the other way, so No .44101 reason for this is buffering, it will compress, but never expand).
ric3xrt said: [[url=http://[/URL]="http://recording.or…"]↑[/]="http://recording.or…"]↑[/]
I'm not talking sonically here , I'm talking strictly in the box(es) what is seen when wave form is against markers and bars.
can you see the mess now that needs to be cleaned up, and edited….and why an external clock should be used, but not needed……..and please understand my tone here is not to be meant as combative or dismissive
Or let’s get simple here, take a straight edge lay it on the ground, take 4 sheets of paper, line one up on the edge, place the other 3 a small distance away from the edge, draw a straight line down the papers, , then line the 3 sheets against the straight edge, does the line now line up?
Now we all have heard someone, or have read somewhere that someone said they Noticed recordings were brighter, Clearer when they hooked up an external clock, all that clock is doing is lining up start times., so now everything is “lined Up”
And if we are using an Alesis HD24 @ 44.1, these things are Notorious for dropping tracks or just plan old wiping out HDs if they come out of sync with the Infernal clock. Google HD24 44.1 track issues,
Electricity/ electric current travels roughly +/- .00233 cm/sec Depending on a LOT of things.......................I promises I will explain this and it's importance ...Have to jet
Work is here early today.....
Sorry, Ric, but you have a lot of things mixed up in your last t
Sorry, Ric, but you have a lot of things mixed up in your last two posts. I won't itemise all the points, but suffice to say that any serious system where digital gear is interconnected uses a single clock source. This clock does not have to be a separate external box (of whatever cost and quality), but could instead be the internal clock of one of the boxes that is fed around to the other boxes in the system. The point is that no samples are lost, as all the pieces of digital gear stay in synchronism, so there is no loss of sonic quality. The clock is used only to pass digital information from one piece of gear to another (sometimes within the gear), so clock phase is not an issue.
A related, but more important function to fulfill is that of the sampling clock. This is the clock that times the conversions at the boundaries between digital and analog parts of the system. The stability of this clock is a prime factor in governing the quality of the conversion.
In a self-contained system, the accuracy of the sampling clock does not affect the quality of the conversion. This is maybe where your reference to the Alesis HD24 comes in, since its internal clock runs fractionally fast at 44.1 and 88.2 KHz sampling rates. Although the speed inaccuracy makes no difference to the quality of the conversion, engineers usually feed the HD24 with an external accurate clock (e.g. by sending the audio data in by lightpipe) so that the tracks recorded on an HD24 replay exactly the correct time on a different replay system, or when the recordings have to sync to video (rare at 44.1KHz).
In a system that has both analog and digital components, it's important to look after the conversion clock first and make sure that is of the highest quality. A version of this clock is subsequently used to move the digital samples around the system, but, as stated above, there are no audio quality implications involved in this.
Don't confuse this need for digital clock interconnection with the references to the two-box mix process that I and Chris (Audiokid) use. In the two-box method, it's important that the clocks of the source system and the mix capture system are not synchronised or interconnected. In my post above, I referenced an earlier thread where I emphasised the need to think of the capture system as seeing the analog output of a stereo microphone (suitably pre-amped). Everything that comes before the capture system goes into emulating the stereo microphone. Drawing a dotted line through the analogue stereo link between the systems helps inform decisions about how to record and mix the material on the one side and how to capture and master it on the other.
DonnyThompson, post: 435422, member: 46114 wrote: I don't even m
DonnyThompson, post: 435422, member: 46114 wrote: I don't even mind the "hair-brained" ideas - because while some of the ideas truly are wacky, and serve no purpose other than to damage the fidelity of a recording, at the same time, new recording and mixing methods need to come from somewhere, right?
And not all of the good ideas are (or have been) initially "accepted", either ...
Like the first guy to find out that by recording above 0db on a tape machine, up around +3, that by "saturating" the tape with signal, some pretty sweet-sounding tones could result...
( and doing so during a time in audio history when this was actually a terminable offense at many studios).Or the guy who first thought, "I wonder what would happen if sent a signal - like maybe a vocal - to a speaker that's in a large hall... then put a couple microphones in that same hall, to pick up the sound of the voice through that speaker in that hall...
Or the guy who decided to record a multitude of instruments, all playing the same musical parts together at the same time, and then recording it, and then doing it again ... on top of the first performance ... to create a big sound, a huge sound, kinda like a "wall". Yeah..."wall" is a good word to describe it ... like a big wall ... a "Wall Of Sound".Or the guy who decided to record sound with a pair of microphones that were placed so that their axis's were at right angles to each other... his name was Blumlein.
Or even more recently, a couple of members of an internet audio recording forum named Recording.org, who broke ranks from the masses of other digital/analog hybrid workflow fanatics, and determined that, instead of using the more popular and more accepted process of returning the signal from external processing back to the same DAW, or that instead of rendering a mix through the DAW's internal stereo bus, that maybe , just maybe... using a second DAW instead - as a sort of "mix down DAW" - would sound better? ... ( audiokid Boswell )
Once in awhile, there's that one idea that might sound a little wacky initially, but that at the same time, is also a little intriguing, and turns out to be something that someone might find useful. ;)
IMHO of course.
I agree d no rules in audio. Should
Been more clear with my 'hair brained' comment. I was referring specifically to the acoustics end of things, where they're tends to be some very specific ways of doing certian things, particularly with isolation construction.
Lol as an aside- Sylvia Massey pushed a piano off a cliff and mic'd it as oart of the sessions on tools' undertow album.
ric3xrt, post: 435436, member: 47701 wrote: 1st I don’t want any
ric3xrt, post: 435436, member: 47701 wrote: 1st I don’t want any one to think I’m say you should by an external clock, but if you do buy the best you can afford and keep in mind the more load you put on the cheaper clocks the more they show their price.
Electricity/ electric current travels roughly +/- .00233 cm/sec Depending on a LOT of things.
This will be important in a few Mins…… we all know a sample rate of 44.1kHz is equivalent to say a slicer taking 44100 slices out of a sound wave per second 00:00.44100
For conversation, Recording Dude, has a 1 Pm session, 24 tracks, turn on the power conditioner feeding 3, 8 channel preamps hooked up to the DAW via lightpipe (firewire, what have you) turns on the DAW then goes and turns on each Pre, none of these 4 items are externally clocked, they all rely on internal factory clocks each item’s clock starts when it is turned on,
DAW = 11:00.0003 ….1st Click(slice) of the Clock = 11:00.44130
Pre (1) 11:03.0004….1st Click(slice) of the Clock = 11:.03.44140
Pre (2) 11:08.0045….1st Click(slice) of the Clock = 11:08.44551
Pre (3) 11:10.0034….1st Click(slice) of the Clock =11:10.44440
Now reality is these 4 items are constantly taking slices now, doesn’t matter what they are slicing at this moment they just know they are doing their job. Also 11:??. Doesn’t matter, it’s the. 00.????? that matters.
Band comes in Gets set up. Mic’ed up ready to go, we have 12 tracks of drummer (keep in mind now that 12 tracks = 2 Pre’s and the DAW, 3 different clock start times)
Keyboard player has 3 Keyboards, 1 Runs stereo ,2 and 3 run mono 4 tracks (2 different start times, DAW’s and Pre#2
Bass takes 2 tracks, 1 DI 2nd live iso room Mic’ed 15in cab (2 Different Start times DAW’s and Pre 3
Guitar 1 Stereo ,1 Mic Left 2nd Mic Right Cabs (2 Different Start times DAW’s and Pre 3
Guitar 2 Mono, but we use 2 mics, 1st off the speaker, 2nd somewhere in the room (pick a spot) again we have 2 Different Start times DAW’s and Pre 3
22 tracks. Record is Pressed at 1:16.03.00000(trust me here) , why does this mater, what is the point to this ramble
DAW takes its 1st slice at 1:16.03.0000 which is .00010 BEFORE Pre (1) takes its next slice, which is Pre 1 1st slice of sound wave ,Pre(2) is now .00541 behind ,Pre(3) is.0034 so we have
DAW=.00000/.44100
Pre1 =.00010/.44110
Pre2=.00541/..44641
Pre3=.00340/.44440
….we can NOT hear this but the DAW is looking at Pre1 1st 00010 as silence, Now when Play is pressed on the DAW, to play back ,what does it do, It lines up each track bassed(yeah I’m a bass player) on the DAW’s clock.
So the 1st 8 drum tracks are moved .00010, to line up with the DAW’s 1st recorded marker, next 4 drum tracks are moved .00541, so now we have a time discrepancy of -.00531. between 1-8 and 9-12,
Now the Keys are off by -.00531 and the guitars and bass are off.-0033
And we haven’t even tossed the monkey wrench of Eventide V4 , Eventide H7600, Lexicon PCM92
Which all have their own clocks,
Now this matters if you have drummer whose every 4th and 6th kick drum hit is weak and you have to replace the WHOLE dam Kick drum track over a 7 min Doo Woop medley with triggered drum sounds (dude is 79 I’m hoping I still am around to play at that age)
The over heads are on track 11/12, and snare is 4, so we have some original kick drum bleed on 3 tracks, 1 in time with the Kick track and 2 slightly out…again we can’t hear this deference but we can see it in the markers…and let us not forget if one of these Pre’s decides it wants to drift and sample 2 mins @.44099 yes it does happen,( it will not go the other way, so No .44101 reason for this is buffering, it will compress, but never expand).
ric3xrt said: [[url=http://[/URL]="http://recording.or…"]↑[/]="http://recording.or…"]↑[/]
I'm not talking sonically here , I'm talking strictly in the box(es) what is seen when wave form is against markers and bars.
can you see the mess now that needs to be cleaned up, and edited….and why an external clock should be used, but not needed……..and please understand my tone here is not to be meant as combative or dismissive
Or let’s get simple here, take a straight edge lay it on the ground, take 4 sheets of paper, line one up on the edge, place the other 3 a small distance away from the edge, draw a straight line down the papers, , then line the 3 sheets against the straight edge, does the line now line up?
Now we all have heard someone, or have read somewhere that someone said they Noticed recordings were brighter, Clearer when they hooked up an external clock, all that clock is doing is lining up start times., so now everything is “lined Up”
And if we are using an Alesis HD24 @ 44.1, these things are Notorious for dropping tracks or just plan old wiping out HDs if they come out of sync with the Infernal clock. Google HD24 44.1 track issues,
Electricity/ electric current travels roughly +/- .00233 cm/sec Depending on a LOT of things.......................I promises I will explain this and it's importance ...Have to jet
Work is here early today.....
15 years of digital audio and I've never experienced this. Whether clocking internally or to a digital mixer. MTC and MMC I think are more prone to drift. Especially when syncing tape, but a lot of that has to do w tape stretching and mechanisms not being quite perfect.
I don't see where pro tools has anything to do w what your describing either. I belive a hear sonic differences among diffennt programs but that has to do with the software coding not the clocking.
Boswell, post: 435433, member: 29034 wrote: Nice way of writing it, Donny, but my approach came from a slightly different direction:
Not the section I wanted to quote bos, but do you feel the same way about SRC from 192 to 96, as you do from 96 to 44.1?
kmetal, post: 435444, member: 37533 wrote: ... do you feel the s
kmetal, post: 435444, member: 37533 wrote: ... do you feel the same way about SRC from 192 to 96, as you do from 96 to 44.1?
In principle, no, as the top octave of both 192 KHz and 96 KHz sampling is outside the human hearing range. However, it may be just me, but I have a mistrust of software SRCs at any frequency.
Ric, like Boswell I too am sorry that you seem to have many mis
Ric, like Boswell I too am sorry that you seem to have many misunderstandings of how sampling and timing works in a DAW.
I think that a lot of your misunderstanding may arise from our use of the word "clock". We use the word "clock" very widely in electronics but this does not mean that it is a true clock in that it knows the time and date. It is, more accurately, just a reference frequency that is a simple square wave of a known and controlled frequency. These clocks also have frequency errors dependent on their type and I mentioned the typical magnitude of these frequency errors for various "clock" types in my earlier post. For this, clock jitter is completely irrelevant.
What this means is that a sound-card does not know inherently what time of day it was turned on (nor does it need to) so your various points using time of day are somewhat wide of the mark. What is important is, when you hit "record", that all interfaces start recording at the same time and the DAW records the very next sample available from all the different sources. These samples will all be within 1/(sample frequency) of each other i.e. for 44100 sampling, within 2.26 microseconds. If the various sound-cards are NOT synchronised then from that point onwards, they can drift relative to each other. Let's take a "worst case" where one clock is +50 ppm and another is -50 ppm. The relative drift is 100 ppm so after an hour of a single take, they will be 0.36 seconds different and hence there will be issues. This is exactly why we synchronise clocks. If we lock the clock frequency of one sound-card to another then, even when the master drifts, the slaved sound-card will still be exactly in sync. That means there will be NO difference in timing between the samples other than the possibility of a phase error which again would have a maximum error of 2.26 microseconds (as per the start of the recording).
This is the way it REALLY works and hence I think why all of your subsequent conclusions about drift are incorrect.
Just as a side point about your comment of
ric3xrt, post: 435436, member: 47701 wrote: Electricity/ electric current travels roughly +/- .00233 cm/sec Depending on a LOT of things.
I'm not sure where you got this figure? The +/- seems to indicate it refers to an AC signal but I suspect your figure relates to electron drift rather than the "effective" current flow which, in a vacuum is ~3 x 10^8 m/s (and admittedly less in a cable - but not that much!). If you don't trust this point, just go and turn on a light in your house! If your figure was "real" then it would, for a 5 metre run between switch and the light fitting, take over 59.6 hours for the light to come on! Maybe this also contributes to your misunderstandings as you must still be working in the dark! :
And here is a video to add. Ted Smith explains clocks and their
And here is a video to add.
Ted Smith explains clocks and their importance in the audio chain. All audio systems run on clocks and the quality of those clocks contributes to one of the major problems in sound, jitter. How do expensive, specialized clocks like rubidium devices affect the jitter? You might be surprised at the answer.
thoughts?
Boswell, post: 435474, member: 29034 wrote: In principle, no, as
Boswell, post: 435474, member: 29034 wrote: In principle, no, as the top octave of both 192 KHz and 96 KHz sampling is outside the human hearing range. However, it may be just me, but I have a mistrust of software SRCs at any frequency.
Cool I'll take your word for it. It may all be in my head but even when staying at the same sample rate 44.1 in DP, but bouncing/rendering to 16 bit I've noticed a mildly significant difference in the size of the mix compared w the multitrack. This is not the case with all programs I've used. This also seems to be the case even when the bit rate stays the same, like for instance when I mixdown for import into a mastering type 2track program. I'm wondering if it's jus bottlenecking of the master bus? Or maybe program code? Jw if anyone else had experienced this. I'm guessing the 2daw method doesn't exhibit this??
kmetal, post: 435491, member: 37533 wrote: That's what I figured
kmetal, post: 435491, member: 37533 wrote: That's what I figured Chris. I'm wondering if this would be an an example of phase distortion and/or jitter? I'm still a newb on this topic.
From your explanation, it doesn't sound like it but maybe. I've never experienced that to learn why.
Maybe it's all in my head. Maybe some samples get thrown out d
Maybe it's all in my head. Maybe some samples get thrown out during the internal summing process or perhaps cancel? I dunno, again it could just be in my brain. Nobody else at the studio has ever said anything about it.
I searched for audio examples of jitter but none came up, only explanations an graphs. In feel like it's one of those things that as soon as you can identify it, you always hear it. i.e no turning back.
kmetal, post: 435495, member: 37533 wrote: searched for audio ex
kmetal, post: 435495, member: 37533 wrote: searched for audio examples of jitter but none came up, only explanations an graphs. In feel like it's one of those things that as soon as you can identify it, you always hear it. i.e no turning back.
kmetal, post: 435495, member: 37533 wrote: I searched for audio
kmetal, post: 435495, member: 37533 wrote: I searched for audio examples of jitter but none came up, only explanations an graphs. In feel like it's one of those things that as soon as you can identify it, you always hear it. i.e no turning back.
You're unlikely to find any "clear examples" of jitter simply because (a) it really should not be a problem nowadays and (b) there is no single characteristic sound it would generate due to the many causes of the jitter itself. Any examples would have to be artificially induced and would only produce a "sound" that is a characteristic of the particular way it was induced.
As well as the reference from Donny, have you read the thread "What is clock jitter?" at the top of this forum? I would hope that will help too...
The problem with identifying jitter, as Ease mentioned, is that
The problem with identifying jitter, as Ease mentioned, is that it doesn't present itself as sounding like just one identifiable sound. It's not like "phasing" or "flanging" or "reflection"... things that all have their own consistent "sound" that is recognizable.
I'm not suggesting that you don't listen to the samples provided on that link I posted, pal; you mentioned that you had never heard it before, so I just gave you a link to some examples of it... but those are, as you'll hear, all very different in the way(s) that they sound, and are sounds that are indigenous to those particular situations of jitter individually.
I've heard an example of clock jitter that sounded like "distortion", where other examples didn't sound like that at all; one sounded more like a "whistling", whereas another presented itself as a kind of "pinging". And still another I heard - when working once back in the early days of digital, using a digital console with digital tape decks, where it sounded similar to "clicking", with each "click" sounding like a short, quick blast of SMPTE chatter... but, not exactly like that. It's difficult to describe because it's not something I'd ever heard before - or since - that I could identify.
Interesting. I wonder if, even though it could be a different t
Interesting. I wonder if, even though it could be a different type of noise, if when it does occur it's at least a consistent noise in each case. Or would a whistle turn into a pop into something else. I'm guessing it's probably a case by case basis as it's an inconsistency we're talking about.
kmetal, post: 435553, member: 37533 wrote: I'm guessing it's pro
kmetal, post: 435553, member: 37533 wrote: I'm guessing it's probably a case by case basis as it's an inconsistency we're talking about.
That's been my understanding.
Ease or Bos would be able to tell you more about it, they know way more about this stuff than I do; but as far as I've been able to figure out, it's so different in every case because it depends so much on so many different things; jitter might sound completely different on an acoustic guitar track than it does on a vocal, because the acoustical signature for the one track is so different from the other to begin with; or for that matter, and here's what I really don't know - you might have two very similar acoustic guitar tracks - and each might have jitter issues, but the sound of the jitter on each might be completely different sounding from the other (?) ...
Another factor might be the types/models/SR's of the clock you are be using... and, what it's clocking to ... like external DSP's or other digital devices that require a clock sync...
It might also be amplitude dependent? In that a track recorded hotter might have a different "kind" of jitter than another track that was recorded at lower levels... not that you'd have more jitter, it would just be a different sounding jitter.
I dunno... like I said, Bos and Ease have probably forgotten more about this stuff than I've ever learned...
My only first hand experience with obvious audible jitter was years ago; I was using a Yamaha O2R digital console and recording digitally to a rack of Tascam DA88's; As I recall, the console was the "master" clock, @ 44/20 and it connected to the first DA88 in the rack, which was the "slave", and then from that DA88, I connected FW sync cables to the other 2 decks so that they'd all run in sync, and with the last one being terminated with a "dummy plug". There was a BNC ( coaxial word clock) cable connecting the console to the first DA deck.
One day, I was working on a song that had breaks of silence, and out of nowhere, in those silent passages, I was hearing a "clicking " sound, coming off the master DA deck in the rack, and at a rate that was totally intermittent; there was no pattern to them - like 5 fast ones right in a row, then another two clicks 20 seconds later, and the clicks themselves sounded like very fast, short bursts of Time Code chatter, with a very high frequency being dominantly audible ( like 8k or something). The problem wasn't loud, or even noticeable throughout the song when there was audio content, but it was really noticeable during those silent passages. I tried everything, thinking it might be the digital tape, I put another tape in- the problem was still there. I tried making another of the DA88's the master deck, that didn't make any difference either. It drove me bug-crap crazy; I finally called TEAC Service, described the problem, and the tech told me he thought it might be "clock jitter" ... I'd never even heard the term before that. I finally resolved it by following his list of suggestions; re-initializing the console, "forcing" the console into 48 and then back to 44.1 - finally, it turned out to be the actual WC cable from the console to the decks - I connected a new one I had lying around, and the problem went away...
It was the first - and I think the last time that I ever had to deal with it, but there's no guarantee of that... I might have had jitter issues again after that, and they were either not as audible, or they didn't sound the same...
In any event, I don't think that jitter is something that is really much of a problem anymore, if you are using modern gear. I know a few guys - who, back in the early days of digital, dropped big money into those "pricey" MOTU converters - and they've mentioned that today's converters are much better, and cheaper - and that even today's entry level pre's/interfaces have better conversion in them than the high dollar dedicated clocking devices of the past.
Truthfully, I never really paid all that much attention to it back then, unless I could hear it, if it was audible, and obviously an issue. If the recordings were quiet, and sounded okay to me, then that's all that mattered to me.
Perhaps I should have paid more attention; maybe the projects I recorded in those days would have sounded better...
DonnyThompson, post: 435557, member: 46114 wrote: - you might ha
DonnyThompson, post: 435557, member: 46114 wrote: - you might have two very similar acoustic guitar tracks - and each might have jitter issues, but the sound of the jitter on each might be completely different sounding from the other (?) ...
That's what I'm wondering. Or even on the same track would the jitter exhibit different things?
Sounds like in your case it was the same sound just randomly timed hence maybe jittering? That's such a good name for a description of your case.
Lol why is the cable that last thing we always check!!???! ;)
I guess since we are having a tough time indentifying it, maybe it isn't such a problem to us mortals. I'm tendon to agree w you on that. It's certainly not making me want to buy any sort of external clock.
Although isn't one of the main things company's like BLA do is upgrade the clocking and analog sections to improve conversion. Is this a different form of clocking/clock?
I'm telling you man, the more I learn the less I know.
kmetal, post: 435559, member: 37533 wrote: I'm telling you man,
kmetal, post: 435559, member: 37533 wrote: I'm telling you man, the more I learn the less I know.
Good one!
Clock jitter does not have a sound of its own, so can be difficult to recognise purely on hearing tests. However, it's a parameter that is relatively straightforward to measure using test equipment, at least in terms of the distortion it causes.
The problem in identifying it in hearing tests is that signal distortion can be due a number of causes, and a poor quality sampling clock is not usually one of the immediately obvious ones. One clue is that distortion caused by sampling clock jitter is not usually a function of signal amplitude, so if there's no change in the signal quality when reducing the input amplitude by 6dB and increasing it by the same amount on the monitoring amplifier, then distortion added to the signal passing through the equipment could at least have a component due to clock quality.
Boswell, post: 435575, member: 29034 wrote: Clock jitter does no
Boswell, post: 435575, member: 29034 wrote: Clock jitter does not have a sound of its own, so can be difficult to recognise purely on hearing tests.
What did you think about the two videos here, specifically describing how he can hear clock jitter? I don't know if I could do that. Or at least say, yup... that's clock jitter.
audiokid, post: 435484, member: 1 wrote: And here is a video to add.thoughts?
In essence, what he can hear is a difference. Whether you can sa
In essence, what he can hear is a difference. Whether you can say the difference is due to clock jitter depends on whether you have ruled everything else out.
The concern over things like slow edges of external clocks feeding a clock input of a piece of gear that converts beween the digital domain and the analogue domain (in either direction) is these days largely unnecessary. This is because modern equipment still uses its internal clock that has good jitter properties, but uses the external clock input as a long-term frequency reference source to lock the internal clock's frequency to that of the external clock to keep them in sync. In this way, short-term effects such as jitter are not propogated into the internal clocking.
A lot of what he says in the second video about hearing the quality of high-frequency signals changing when your audio computer is doing other things at the same time as sampling audio is much more likely to be due to interference on the ground or on the power rails or coming down a USB cable rather than purely due to conversion clock jitter. Laptops are a major offender in this respect.
in a round about way, this is what I was thinking as well. Thank
in a round about way, this is what I was thinking as well. Thanks for your explanation.
Boswell, post: 435577, member: 29034 wrote: Laptops are a major offender in this respect.
I'm rushing to say this... then off for the day. I hope it makes more sense and good discussion.
For the guys using laptops, maybe we should open up a new thread sometime on how to get the best performance from a laptop and what to expect.
Being said, I noticed when mixerman was raving about how the 10M made huge (not subtle) improvements, If I recall, he also mentioned he was using a mac laptop and usb. My first thought was, well no wonder you are endorsing things I wouldn't even bother admitting helping your sound.
Imho, he was basically admitting how whacked his sound was to the world. Or how naive thinking this thing is improving the sound of the Orion, thus... others to trust this clock combo'd with the Orion 32 was incredible sounding.
Note: He is also coming from a dated Radar system to the Orion32. That alone would be an improvement on sound.
As rude as Eric (MM) was to me, I'm not trying to pick on him, but the way they are marketing it was a bit shady and misleading.
When I tested the Orion 32 via laptop and usb, it sucked so bad, I wouldn't even take a chance with clients. Now, this is also subjective to my DAW verses say, a less demanding platform, so I think that is also worth mentioning. There is only so much a laptop can handle, correct?
The stability I get from my internal PCIe interface and MADI or AES EBU is night and day. When you read the testimonials from a lot of these guys raving about the 10M, I look at what they are doing, their rats nest and interfacing and I shake my head. They are doing so much round trip bloat its a wonder they even get anything done.
There are several things in the video's that seem to be quite wr
There are several things in the video's that seem to be quite wrong to me. That may be because the presenter is trying to explain in layman's terms but if so, his analogies are not good. While it is true that the slope of a rising edge can be slow enough to cause timing errors, the majority of what constitutes "jitter" is not down to this. "Jitter" is the actual timing of the rising edge being in error and nothing to do with the rise time.
In the second video there is some completely off the wall speculation that he ascribes to jitter! How on earth can ascribe various audio artefacts to jitter purely by listening is completely beyond me.
I'm in the process of preparing some notes to post in a new thread to try to do two things. First, get the information in an easily found place and second to hopefully debunk a lot of the "noise" that is said about the sound of clock jitter. I also hope to promote a better understanding of the effects of clock jitter.
MrEase, post: 435581, member: 27842 wrote: "Jitter" is the actua
MrEase, post: 435581, member: 27842 wrote: "Jitter" is the actual timing of the rising edge being in error and nothing to do with the rise time.
perfect explanation.
MrEase, post: 435581, member: 27842 wrote: In the second video there is some completely off the wall speculation that he ascribes to jitter! How on earth can ascribe various audio artefacts to jitter purely by listening is completely beyond me.
+1
But, I admit, here are some people with skills or gifts that others would never believe so I always give the benefit of doubt.
MrEase, post: 435581, member: 27842 wrote: I'm in the process of preparing some notes to post in a new thread to try to do two things. First, get the information in an easily found place and second to hopefully debunk a lot of the "noise" that is said about the sound of clock jitter. I also hope to promote a better understanding of the effects of clock jitter.
Awesome,
http://recording.org/threads/what-does-clock-jitter-sound-like.59224/ :love:(y)
audiokid, post: 429996, member: 1 wrote: Without Prejudice: I wa
audiokid, post: 429996, member: 1 wrote: Without Prejudice:
I want to be very clear about this post. I appreciate good gear that help the mutual advancement towards better recording. I will try to be careful with my words.Fact or fiction? Hype or BS?
Demystifying Super Clocks. Who needs an external clock?
I used the 10m for over a year. I am a guy looking for any improvements at this point so when someone says, hey, this thing makes "Huge" (not subtle") improvements to your sound... I am all over it!
I understand that once you are up to a certain level of sound, the last 5% towards pristine sound is baby steps that come at a price$.I put this super clock to the test using it with the Orion 32 (including RME, Lavry and Prism converters).
After a year in my rack, I must be honest here and say, I never heard a bit of difference with or without the 10M.
I even doubted my ears so I had young ears (better than 18k hearing) spend time on this test and not one person heard value with the 10M. We did numerous advanced null tests, all which proved worthless. Iff the intent of this clock was to "improve" sound quality of your existing converters, it sure didn't do that for me.What could be wrong I ask so I called the rep and began searching forums for answers on this product.
My finding and gut feeling about this clock look like snake oil and I will tell you why.
When I called the rep in this video he exclaimed I needed better speakers or better treatment. He cared nothing about what I hear, or had any interest in the system I use, which btw ranks as a world class hybrid tracking and summing system.
I'm disturbed how this product is being marketed and shilled. It has me questioning the integrity of company marketing why top level engineers are promoting it to the mass as a sound improvement tool just by connecting it to your converters.
There are obviously well respected people claiming huge improvements so much so, that they are suggesting people doing electronic music get on the chain with this.
In fact... this was how the rep sold it to me too. He sent me a mastered track that was a before/after of electronic music :cautious:. The difference between both versions was HUGE! This was electronic music, ITB music so I am thinking.... WOW! this thing is amazing. I want one. :love:But... the videos, forum and direct marketing say a much different story.
As we see in the above video, here are world class guys claiming huge improvements in their sound using the 10M, yet I am using it, running a very sophisticated mixing and mastering system and hear no change. Why them, not me? How come the rep sent me a before after example and said...
Lets get a few scenarios out of the way.
- Could the reason I don't hear any change be because I use Sequoia and do not have added ADDA products messing, or reducing stability of my internal clock?
- Could it be because I do not round trip process back to the tracking DAW?
- Could it be that my system is simply telling the truth and exposing something?
class="xf-ul">
Internal vs external clockingThere are numerous top level testimonials that say internal clocking is choice.
Something worth mention. I notice the majority users claim this clock is improving their sound are either on Pro Tools or Logic.
I have read less trusting comments from people claiming huge improvements on PC systems.Being said, I really feel there is a placebo effect and/or support of purchase going on here more than anything though, and lets not forget that the rep also sent me an audio example that he did, claiming it was the difference between the 10M in and out.
To even put more questions out, I have asked for proof it in an public A/B and no one have stepped up. We just read the hype without hearing the comparisons online.
This product is only good for those having seriously bad clocking to begin with. Its certainly not for the guy with one converter in a simple system. Especially if all your mixing is ITB.
My feelings are, if a hybrid system is clocked properly to begin with, your clock should be just fine. This is suited for studio running all sorts of digital products in a rats nest going round and round.Thoughts?
I think until you AB your recordings in multiple playback situations you can’t make a “snakeoil” claim. After conversations with friend Paul Grundman (Grundman mastering - legend) he told me about how this unit is a game changer and after renting it for mixing and mastering I’ve found the 10m makes the portability of my mix/masters much better. Unless you’ve mixed and mastered daily for 20 + years I simply don’t think your ears are good enough to tell from studio monitors. You HAVE to listen on multiple real life playback systems doing AB tests if you Wana make any legitimate claim to the 10m and it’s value.
Clark, post: 464753, member: 51967 wrote: I think until you AB y
Clark, post: 464753, member: 51967 wrote: I think until you AB your recordings in multiple playback situations you can’t make a “snakeoil” claim. After conversations with friend Paul Grundman (Grundman mastering - legend) he told me about how this unit is a game changer and after renting it for mixing and mastering I’ve found the 10m makes the portability of my mix/masters much better. Unless you’ve mixed and mastered daily for 20 + years I simply don’t think your ears are good enough to tell from studio monitors. You HAVE to listen on multiple real life playback systems doing AB tests if you Wana make any legitimate claim to the 10m and it’s value.
Hi Clark, welcome to RO.
Thanks for your comment and opinion on the Antelope 10m.
.Do you have one?
audiokid, post: 464754, member: 1 wrote: Hi Clark, welcome to RO
audiokid, post: 464754, member: 1 wrote: Hi Clark, welcome to RO.
Thanks for your comment and opinion on the Antelope 10m.
.Do you have one?
Rented twice for final mix and Master of 2 different records. Find the benefits huge. Better floor to ceiling frequency clarity and removal of digital smear. And a more musical transient presentation. I’m a huge fan of the 10m.
Clark, post: 464755, member: 51967 wrote: Rented twice for final
Clark, post: 464755, member: 51967 wrote: Rented twice for final mix and Master of 2 different records. Find the benefits huge. Better floor to ceiling frequency clarity and removal of digital smear. And a more musical transient presentation. I’m a huge fan of the 10m.
Well,
Since you quoted me and suggested I don’t know what I’m talking about when it comes to the 10m , including Lacking experience... I know many professional, including myself that think the 10m is and was a complete ripoff. I have over 20,000 hours in programming and sound so I do know a bit more than you suggest. I have had one the the worlds finest mixing/mastering systems that was designed to hear precisely.
In case I may have old ears... I won’t mention names but we also used people with the ability to hear above 18k and no one heard any difference with it in or bypassed.
My unit was given to me by Marcel from Antelope and he gave it to me because he likely hoped I would promote it. I lost total respect for antelope after numerous tests and conversation that were misleading people over this total BS product.
Think what you like. If you heard a difference that would only be because your clocking was screwed to begin with. Or possibly because you needed some help in the clocking to keep a lot of gear clocked.
There is more I could say on my results but that’s old news as none of these clocks are necessary today.
Being said, This website is full of real engineers that have been around for decades.
We love good conversations on recording.org but clamp down on the bogus stuff. We take pride in keeping it real Around here. I personally never promote gear that doesn’t pass my personal tests of being useful for members. The 10m was the most disappointing product I’ve used in 45 years.
I look forward to many more conversations with you and your circle.
Let me throw another pebble into the puddle... Crystal driven c
Let me throw another pebble into the puddle...
Crystal driven clocks get stabler over time. An older clock is better than a new one.
Everything I've ever read or heard (in the audio sector) about clock jitter has been tuned by marketeers. Not exactly a lie, but lots of FUD.
As MrEase already stated "Nothing to worry about".
audiokid, post: 435419, member: 1 wrote: Remy [SPOILER=Remy]star
Dissaponting but not surprising. That's pretty lame man for sure. Guess some people can't be happy for others who may have great gear, or whatever they wish they had. Super lame.