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Old 08-06-2010, 11:48 PM   #81
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Quick question for those using the HDSPe AES... In my device manager, it is listed as RME HDSP AES-32 and NOT RME HDSPe AES... in other words, it's not showing the 'e' for PCI Express. Is this normal? Or do I need to get the PCIe drivers?
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Old 08-07-2010, 06:47 AM   #82
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Originally Posted by DeyBwah View Post
Quick question for those using the HDSPe AES... In my device manager, it is listed as RME HDSP AES-32 and NOT RME HDSPe AES... in other words, it's not showing the 'e' for PCI Express. Is this normal? Or do I need to get the PCIe drivers?
You're ok. There's only one driver that covers both.

Regards,

DB
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Old 08-07-2010, 06:48 AM   #83
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You'll have to post some pics of your new rig for us all to see Dey.
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Old 08-07-2010, 08:20 AM   #84
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From Lavry on this forum.

"Say you have a movie camera designed to take 100 picture frames per second, thus a frame every .01 second. You are pointing the camera at an ball moving from left to right at a constant speed, and you take a whole second of that motion, that is 100 picture frames. When you “play it back” with a movie projector that runs a picture each .01 second, everything would “be fine”, as intended.

But let us say that the camera is very “unsteady”. It takes a picture, then it waits .05 seconds, then it takes 5 frames within say .001sec, then a couple of frames at .01 second…. When you play it back, the projector “does not know” what happened at the camera, and the ball may like it is slowing at mid air for a brief time, then it zooms real fast some distance…. That is distortion due to timing error - time jitter. Note that if the camera was OK but the projector had jitter, that also would be a problem. So jitter counts at 2 places - at the camera (which is analogous to the AD), and at the projector (which is analogous to the DA). Jitter is important at the converters.

In one sense, jitter at the AD is more important, because once it takes place; it is in the signal forever. One can replace a bad DA with a good one, and that will eliminate the jitter issue of a poor DA, but what the AD does can not be undone.

Analogies can be misleading. In the case of movies, with enough still frames per second, the eye makes it looks like continues motion. In the case of conversion, it is the analog circuitry that takes the samples and makes them into a continues wave. But I chose the analogy of video, because audio and video (as well as many other applications) are fundamentally based on equal and precise time intervals. The time between each adjacent sample should be exactly the same, and if it is not, there is jitter, which will distort the outcome.

I said that jitter is important at the conversion. What about jitter in transferring say data from AD to a computer hard drive? The answer - it is not important, because we are just moving data from one place to another. You can move one frame every second, very slow indeed, or move a million frames a second, very fast. You can move half the data now, wait a while then move the rest of it… It does not matter, because you are not viewing it. But one you play it back you need the timing to be clocked precisely.

But some manufactures and sellers of clocks wanted to sell clocks, so they decided to convince the world that you need to clock everything. And with enough advertizing money, they where pretty successful doing just that. There are times when you need to use external clock box - when you want to have a lot of gear (AD channels) work together. But as long as you do not need external clock and you can use internal, use internal. It is not only cheaper, it is better!

What you need is the “best clock circuit you can make” that is very steady to be very near your AD circuit - short connection, good grounding… That is internal clock.

Say you take the same “best clock circuit you can make” and put it in another chassis. Will that be better? Not, it will be worse. You now have to deal with 2 chassis thus grounding issues. You have a cable that can pick interference, you have a cable termination imperfection, a cable driver, a cable receiver, and I did not even start… By the time your clock arrives from the clock box, it has so much jitter that it requires some “jitter cleaning circuitry” - typically a PLL circuit…. I pride myself for making very fine external clock circuitry, but no way can I make the external clock circuitry be as good as internal. Almost as good, yes, but never as good.

However, the clock BS’ers are still arguing that their external clock will improve the sound. There is some claim of a “proprietary clock signal” that will make things better. That is a crock if there ever was one! The clock box to the AD connection is a ONE WAY street. The clock “DOES NOT KNOW” what the AD is doing. What kind of a clock box signal is going to improve ALL the following an Ad's:

1. AD with a lot of jitter induce by 60Hz power line
2. AD with little jitter induced by 60Hz power line
3. AD with jitter induce from digital circuit noise
4. AD with jitter due to nearby radio transmitter
5. AD with jitter due to nearby power tools
6. AD with jitter induced from the digital audio data
7. AD with almost perfect timing
8. AD that is powered off…

This is analogous to a doctor that can cure all illness, doing so without any information about the patient…

One of the main offending marketing BS guys said that you can take a tone and have it sound better with jitter. You can alter a fixed tone with jitter, and can argue that you like it, or that you do not like it. But the alteration has to be deliberate for a specific constant tone (including fixed amplitude). You change the tone and the distortion changes... Jitter distortions are very complicated, and they are an INTERACTION between the clock timing AND THE MUSIC. The last I heard, music is not a constant fixed tone The simplest of jitter (random jitter) will increase your noise floor. More complex jitter makes for all sorts of undesired at frequencies that depend on the music, but at frequencies that are not musical harmonics, thus sound bad….

I first stated that internal clock is best a few years ago, and had to deal with a lot of attacks on a forum I was moderating. I insisted that the technical folks come in, instead of the marketing types, and sure enough, the technical types backed off after a short “fight” because they had no leg to stand on. A couple of years later, Digidesign wrote a paper about clocks, and they second me by saying that internal clocks are the best (when you can use internal clocks). I pointed that out and that brought about more attacks… The low jitter crock (I meant to say clock) goes on, and people are clocking with external clocks a lot of stuff they do not need to.

When your AD is using the internal clock for conversion, you are doing the best you can. The data sent forward to a computer, DAW or what not, is “after the conversion” so it does not need to be clocked with special care for jitter, and a “standard” link (say AES or SPDIF) is just fine for data transfer.

There are times when you need to use external clock, and when you need to, use external, when it is a needed trade off. For example, say you want to clock 2 or more AD chassis together... But other then that, internal is the better way to go!

Regards
Dan Lavry"
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Old 08-07-2010, 11:38 AM   #85
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Originally Posted by DBMusic View Post
You're ok. There's only one driver that covers both.

Regards,

DB
Thanks DB, just making sure.
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Old 08-07-2010, 11:40 AM   #86
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Originally Posted by Dannii View Post
You'll have to post some pics of your new rig for us all to see Dey.
That I do! I've been waiting on my lava lamps, plants, and tapestries for new pictures... I kid not! But yes, I'll definitely put em up on this thread soon.
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Old 08-07-2010, 12:02 PM   #87
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A few more questions for ya'll...

I'm listening to a session right now that I just ported from PT to REAPER using AATranslater. There are no plugs in this session, it's a fresh port. I'm noticing that when I turn up the playback buffer to anything above 512, I start hearing pops/crackles. The higher it goes, the more pops and crackles. Now this is a first for me... usually it was the opposite, the lower the playback buffer, the more pops. Can someone explain if this is normal, and if it is, why is it happening? (even if it isn't normal, if you have ideas I'm all ears)

I was reading the RME HDSPe manual, and it mentions many many times how to let the HDSPe be the master clock. In my scenario, I'd ALWAYS want to use the Lynx for clocking correct? Or in otherwords, the internal clock of the Aurora, opposed to clocking external to the HDSPe. That is my assumption, moreso after reading the post by NightScope.

I think I understand most of the button i/o on the front of the Aurora, but a couple, namely, the Meter and AES MODE buttons, I'm having trouble wrapping my head around.

For example, with Digital selected under Meter, if I press the TRIM/AES MODE button once, the DUAL WIRE IN lights up, and my sound only comes out of the left monitor. Pressing the button again, the DUAL WIRE OUT lights up, and sound comes out both monitors again. Pressing it again, lights up both, and sound comes out only the left monitor again. And finally pressing it again selects none, and sound comes out both speakers.

So what is the purpose of that switch?

And what about the Analog TRIM settings? When would I want to have +4dBu and -10dBV?

And finally... as for selecting Sample Rates. There are so many places I can set the sample rate.

For example:
-On the Aurora front i/o.
-In the HammerFall DSP settings by activating DDS. (this is for video projects mostly? Although it mentioned it could be use for creative control, ie. to change the pitch of the whole project)
-In REAPER under Audio>Device>Request Sample Rate. (btw, does checking this change the sample rate of the Aurora? What does it do exactly.. have always been a little hazy regarding this)
-Project Settings>Audio Settings>Project Sample Rate. There's a checkbox and a field, just like under Audio>Device.

So there are at least 4 places to change the sample rate and I'm not sure which takes precedence over the other.

There's one on the converter, on the soundcard, and two in the software(REAPER).

What's the standard practice? To adjust all sample rates to the session sample rate I'm guessing? I remember with the 002, there was no button to set the sample rate, it just matched to the session.

Finally, what about for normal computer use.. what sample rate should I use? Say if I'm listening to Pandora radio through my browser... I'm guessing I just have to match my sample rate to whatever the source is? So if I'm watching a DVD, I'd set it to 48k? And if I'm listening to mp3s.. then take it down to 44.1k?

I haven't even begun reading the section on TotalMix. Will do that now... probably have more questions too...
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Old 08-07-2010, 04:16 PM   #88
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Quote:
Originally Posted by nightscope View Post
From Lavry on this forum.

"Say you have a movie camera designed to take 100 picture frames per second, thus a frame every .01 second. You are pointing the camera at an ball moving from left to right at a constant speed, and you take a whole second of that motion, that is 100 picture frames. When you “play it back” with a movie projector that runs a picture each .01 second, everything would “be fine”, as intended.

But let us say that the camera is very “unsteady”. It takes a picture, then it waits .05 seconds, then it takes 5 frames within say .001sec, then a couple of frames at .01 second…. When you play it back, the projector “does not know” what happened at the camera, and the ball may like it is slowing at mid air for a brief time, then it zooms real fast some distance…. That is distortion due to timing error - time jitter. Note that if the camera was OK but the projector had jitter, that also would be a problem. So jitter counts at 2 places - at the camera (which is analogous to the AD), and at the projector (which is analogous to the DA). Jitter is important at the converters.

In one sense, jitter at the AD is more important, because once it takes place; it is in the signal forever. One can replace a bad DA with a good one, and that will eliminate the jitter issue of a poor DA, but what the AD does can not be undone.

Analogies can be misleading. In the case of movies, with enough still frames per second, the eye makes it looks like continues motion. In the case of conversion, it is the analog circuitry that takes the samples and makes them into a continues wave. But I chose the analogy of video, because audio and video (as well as many other applications) are fundamentally based on equal and precise time intervals. The time between each adjacent sample should be exactly the same, and if it is not, there is jitter, which will distort the outcome.

I said that jitter is important at the conversion. What about jitter in transferring say data from AD to a computer hard drive? The answer - it is not important, because we are just moving data from one place to another. You can move one frame every second, very slow indeed, or move a million frames a second, very fast. You can move half the data now, wait a while then move the rest of it… It does not matter, because you are not viewing it. But one you play it back you need the timing to be clocked precisely.

But some manufactures and sellers of clocks wanted to sell clocks, so they decided to convince the world that you need to clock everything. And with enough advertizing money, they where pretty successful doing just that. There are times when you need to use external clock box - when you want to have a lot of gear (AD channels) work together. But as long as you do not need external clock and you can use internal, use internal. It is not only cheaper, it is better!

What you need is the “best clock circuit you can make” that is very steady to be very near your AD circuit - short connection, good grounding… That is internal clock.

Say you take the same “best clock circuit you can make” and put it in another chassis. Will that be better? Not, it will be worse. You now have to deal with 2 chassis thus grounding issues. You have a cable that can pick interference, you have a cable termination imperfection, a cable driver, a cable receiver, and I did not even start… By the time your clock arrives from the clock box, it has so much jitter that it requires some “jitter cleaning circuitry” - typically a PLL circuit…. I pride myself for making very fine external clock circuitry, but no way can I make the external clock circuitry be as good as internal. Almost as good, yes, but never as good.

However, the clock BS’ers are still arguing that their external clock will improve the sound. There is some claim of a “proprietary clock signal” that will make things better. That is a crock if there ever was one! The clock box to the AD connection is a ONE WAY street. The clock “DOES NOT KNOW” what the AD is doing. What kind of a clock box signal is going to improve ALL the following an Ad's:

1. AD with a lot of jitter induce by 60Hz power line
2. AD with little jitter induced by 60Hz power line
3. AD with jitter induce from digital circuit noise
4. AD with jitter due to nearby radio transmitter
5. AD with jitter due to nearby power tools
6. AD with jitter induced from the digital audio data
7. AD with almost perfect timing
8. AD that is powered off…

This is analogous to a doctor that can cure all illness, doing so without any information about the patient…

One of the main offending marketing BS guys said that you can take a tone and have it sound better with jitter. You can alter a fixed tone with jitter, and can argue that you like it, or that you do not like it. But the alteration has to be deliberate for a specific constant tone (including fixed amplitude). You change the tone and the distortion changes... Jitter distortions are very complicated, and they are an INTERACTION between the clock timing AND THE MUSIC. The last I heard, music is not a constant fixed tone The simplest of jitter (random jitter) will increase your noise floor. More complex jitter makes for all sorts of undesired at frequencies that depend on the music, but at frequencies that are not musical harmonics, thus sound bad….

I first stated that internal clock is best a few years ago, and had to deal with a lot of attacks on a forum I was moderating. I insisted that the technical folks come in, instead of the marketing types, and sure enough, the technical types backed off after a short “fight” because they had no leg to stand on. A couple of years later, Digidesign wrote a paper about clocks, and they second me by saying that internal clocks are the best (when you can use internal clocks). I pointed that out and that brought about more attacks… The low jitter crock (I meant to say clock) goes on, and people are clocking with external clocks a lot of stuff they do not need to.

When your AD is using the internal clock for conversion, you are doing the best you can. The data sent forward to a computer, DAW or what not, is “after the conversion” so it does not need to be clocked with special care for jitter, and a “standard” link (say AES or SPDIF) is just fine for data transfer.

There are times when you need to use external clock, and when you need to, use external, when it is a needed trade off. For example, say you want to clock 2 or more AD chassis together... But other then that, internal is the better way to go!

Regards
Dan Lavry"
So perhaps people might take more notice of Dan lavry than me as that's exactly what i've been saying on this thread but to no avail


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Old 08-07-2010, 04:21 PM   #89
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Quote:
Originally Posted by DeyBwah View Post
A few more questions for ya'll...

I'm listening to a session right now that I just ported from PT to REAPER using AATranslater. There are no plugs in this session, it's a fresh port. I'm noticing that when I turn up the playback buffer to anything above 512, I start hearing pops/crackles. The higher it goes, the more pops and crackles. Now this is a first for me... usually it was the opposite, the lower the playback buffer, the more pops. Can someone explain if this is normal, and if it is, why is it happening? (even if it isn't normal, if you have ideas I'm all ears)

I was reading the RME HDSPe manual, and it mentions many many times how to let the HDSPe be the master clock. In my scenario, I'd ALWAYS want to use the Lynx for clocking correct? Or in otherwords, the internal clock of the Aurora, opposed to clocking external to the HDSPe. That is my assumption, moreso after reading the post by NightScope.

I think I understand most of the button i/o on the front of the Aurora, but a couple, namely, the Meter and AES MODE buttons, I'm having trouble wrapping my head around.

For example, with Digital selected under Meter, if I press the TRIM/AES MODE button once, the DUAL WIRE IN lights up, and my sound only comes out of the left monitor. Pressing the button again, the DUAL WIRE OUT lights up, and sound comes out both monitors again. Pressing it again, lights up both, and sound comes out only the left monitor again. And finally pressing it again selects none, and sound comes out both speakers.

So what is the purpose of that switch?

And what about the Analog TRIM settings? When would I want to have +4dBu and -10dBV?

And finally... as for selecting Sample Rates. There are so many places I can set the sample rate.

For example:
-On the Aurora front i/o.
-In the HammerFall DSP settings by activating DDS. (this is for video projects mostly? Although it mentioned it could be use for creative control, ie. to change the pitch of the whole project)
-In REAPER under Audio>Device>Request Sample Rate. (btw, does checking this change the sample rate of the Aurora? What does it do exactly.. have always been a little hazy regarding this)
-Project Settings>Audio Settings>Project Sample Rate. There's a checkbox and a field, just like under Audio>Device.

So there are at least 4 places to change the sample rate and I'm not sure which takes precedence over the other.

There's one on the converter, on the soundcard, and two in the software(REAPER).

What's the standard practice? To adjust all sample rates to the session sample rate I'm guessing? I remember with the 002, there was no button to set the sample rate, it just matched to the session.

Finally, what about for normal computer use.. what sample rate should I use? Say if I'm listening to Pandora radio through my browser... I'm guessing I just have to match my sample rate to whatever the source is? So if I'm watching a DVD, I'd set it to 48k? And if I'm listening to mp3s.. then take it down to 44.1k?

I haven't even begun reading the section on TotalMix. Will do that now... probably have more questions too...
What ever card is in your computer i.e. the ASIO driver chosen in Reaper should be your master,that way whatever sample rate you use it will change accordingly and your external AD/DA will switch.If you save your sample rate with your project it will open correctly and your asio diver will change the rate on your card,if it's different.



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Old 08-07-2010, 05:15 PM   #90
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So you're saying I should use the HDSPe as the master clock?
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Old 08-07-2010, 08:18 PM   #91
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Can I use the HDSPe as the master clock without the BNC cables? In otherwords, through the AES cables? I'm currently using the Aurora as the master clock through the AES cables..
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Old 08-07-2010, 08:30 PM   #92
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Looks like you don't need the BNC cables to use the RME as the master clock. I'll have to take some time to do an a/b comparison, but right off the bat, I didn't notice a blatant significant difference in sound between the two.
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Old 08-08-2010, 08:24 PM   #93
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Anyone know if you can have REAPER control the sample rate on the Aurora automatically to match the session's sample rate? I know I achieved this when running the RME card as the master clock. BTW, can we get some more opinions on which makes the better master clock between the Aurora and the HDSPe?
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Old 08-09-2010, 08:37 AM   #94
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So you're saying I should use the HDSPe as the master clock?
hell no, that clock sucks.
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Old 08-09-2010, 10:54 AM   #95
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Ya, I went back to the Aurora for the master clock duties. It's a bit of a drag having to remember to manually match the sample rate on the Aurora to the session sample rate, but at the same time, it's nice having a physical button on the hardware to switch sample rates quickly. (opposed to going through the HDSPe settings window)

So I'm up and running and so far everything seems to be pretty stable. I did notice a couple issues crop up, like the strange behavior I mentioned up a few posts where I get clicks and pops at higher playback buffer rates on a particular session. I haven't tested other sessions yet, if this continues, I'll post more on it.

But yes, other than the few quirky pop/crackle fits, I'm super stoked! I didn't have the opportunity to do extensive a/b testing because I sold my BLA 002R unit, but I didn't notice any difference in fidelity in favor of either/or. This is great, because the guys at BLA said their converters were superior to the Aurora by a long shot. Well, I sure didn't notice a degradation in my sound. Anyway, stability gained far out-weighs any quality difference that may or may not be present.
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Old 08-09-2010, 11:19 AM   #96
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because the guys at BLA said their converters were superior to the Aurora by a long shot.
i think theyre being a bit over-optimistic about their product.
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Old 08-09-2010, 11:28 AM   #97
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as far as the clock issue:

the clock doesnt need to know what the AD is doing. i have a battery powered clock in my kitchen and i have the clock on my computer that syncs to a central server. which one is more accurate? do either need to know anything about its users to be more or less accurate? they are simply providing a more or less accurate timing. the clock in a big ben is superior to the clock in behringer. turn it on, listen to the image get deeper and the bass get tighter. or. . . read what you like and do what you like, but people who do the a/b know what im saying.
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Old 08-09-2010, 11:46 AM   #98
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Originally Posted by DeyBwah View Post
Ya, I went back to the Aurora for the master clock duties. It's a bit of a drag having to remember to manually match the sample rate on the Aurora to the session sample rate, but at the same time, it's nice having a physical button on the hardware to switch sample rates quickly. (opposed to going through the HDSPe settings window)
Sure, but how often do you need to change your sample rate? I have on occasion when mixing someone else's project, but all mine are at the same sample rate. So this shouldn't be an issue.

I believe, btw, that Lynx recommends the Aurora to be the master clock. This would hold with the "closest to the converter" ideology too. That is where the magic (not really) is happening. The PCI-e card (and the system) just needs to know what the rate is...that's all.

Ok, now go! Make some recordings....tell use how your work flow/product has improved.

Regards,
David
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Old 08-09-2010, 11:47 AM   #99
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as far as the clock issue:

the clock doesnt need to know what the AD is doing. i have a battery powered clock in my kitchen and i have the clock on my computer that syncs to a central server. which one is more accurate? do either need to know anything about its users to be more or less accurate? they are simply providing a more or less accurate timing. the clock in a big ben is superior to the clock in behringer. turn it on, listen to the image get deeper and the bass get tighter. or. . . read what you like and do what you like, but people who do the a/b know what im saying.
Not sure what you're getting at Shawn.. are you saying I should clock to the HDSPe?
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Old 08-09-2010, 11:49 AM   #100
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Sure, but how often do you need to change your sample rate? I have on occasion when mixing someone else's project, but all mine are at the same sample rate. So this shouldn't be an issue.

I believe, btw, that Lynx recommends the Aurora to be the master clock. This would hold with the "closest to the converter" ideology too. That is where the magic (not really) is happening. The PCI-e card (and the system) just needs to know what the rate is...that's all.

Ok, now go! Make some recordings....tell use how your work flow/product has improved.

Regards,
David
Bahaha David!

Make some recordings I WILL! Can't express with words how nice it feels to be COMPLETELY free of AVIDesign PT!! Seriously, I celebrated with a glass of wine and a lamp shade! ...I'm Korean and therefore 1 glass of wine = 12 glasses of wine for a European. :P
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Old 08-09-2010, 05:46 PM   #101
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Not sure what you're getting at Shawn.. are you saying I should clock to the HDSPe?
absolutely not, the RME clock was the worst of the ones i tested. my comment was meant as a response to:

"However, the clock BS’ers are still arguing that their external clock will improve the sound. There is some claim of a “proprietary clock signal” that will make things better. That is a crock if there ever was one! The clock box to the AD connection is a ONE WAY street. The clock “DOES NOT KNOW” what the AD is doing. What kind of a clock box signal is going to improve ALL the following an Ad's:

1. AD with a lot of jitter induce by 60Hz power line
2. AD with little jitter induced by 60Hz power line
3. AD with jitter induce from digital circuit noise
4. AD with jitter due to nearby radio transmitter
5. AD with jitter due to nearby power tools
6. AD with jitter induced from the digital audio data
7. AD with almost perfect timing
8. AD that is powered off…

This is analogous to a doctor that can cure all illness, doing so without any information about the patient… "
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Old 08-09-2010, 09:11 PM   #102
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And the rebuttal?

As a person who has no problem of admitting ignorance to the intricacies of clocks in relation to other devices.. I'm taking notes and judging for myself. I find it interesting that people can often times have such opposing views of a topic. I wonder if there is a true 'right' answer sometimes or if it's more of a personal preference thing. Well, in this situation, I feel like there is a factual correct answer, but I was just speaking in regards to the non-unique situation that I'm sure a lot of forum goers experience when trying to 'figure' things out.

But yes, please continue! <scribbles in notepad>
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Old 08-09-2010, 09:21 PM   #103
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by the way, Deybwah, i didnt intend my statement to sound as though i was talking to you directly (". . . read what you like, do what you like") and disrespectfully.
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Old 08-09-2010, 11:17 PM   #104
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by the way, Deybwah, i didnt intend my statement to sound as though i was talking to you directly (". . . read what you like, do what you like") and disrespectfully.
Don't worry Shawn, I didn't see it that way at all. I just love researching and learning— the journey of acoustic sound is much more vivid being an artist AND engineer. I'm getting ready to make music my bread and butter and feel it's more possible with such a helpful and understanding community to reach out to.
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Old 08-10-2010, 12:14 AM   #105
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Not sure what you're routing needs are, but between Reaper, TotalMix, and 16 I/O points on the Aurora, you have some pretty creative options in front of you. Totalmix is uber-cool. Enjoy!!

Regards,

DB
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Old 08-10-2010, 01:58 AM   #106
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Don't worry Shawn, I didn't see it that way at all. I just love researching and learning— the journey of acoustic sound is much more vivid being an artist AND engineer. I'm getting ready to make music my bread and butter and feel it's more possible with such a helpful and understanding community to reach out to.
I'll give you some advice for what it's worth as you're geting ready to make your way.Don't worry so much about your gear,really,the difference betwen budget and high end theses days is a lot smaller than it used to be.listen to some of Pat Metheny's first digital records they were done on 16 bit recorders with technically a lot poorer specs than you have now,great sounding records though.Another example of good engineering/sound;James Taylor 'Hour Glass' that was recorded with a Yamaha O2R,not high end by any means but a great sounding record.

Last year my girlfriend who is a succesful singer and runs her own label wanted to do a live DVD,so we rehersed up a great little band and shot it in an artists studio. As i was MD I didn't want to be involved in the actual recording as I'd enough on my plate so the Film production company suggested someone.I spoke to the guy and he had amazing equipment and dropped all the right names,Neve mic pres, Royer ribbons,Neuman 47's,Lavry gold AD etc.

On the day though his choice of mic and postition was bad so when I got back to my studio to mix the sound was awful.I have recorded recently the same band,I didn't have that level of equipment.He chose to mic my Double bass with a Royer ribbon into an expensive pre ,OK fine,but we had foldback pointing at us so a ribbon was a poor choice as it's fixed figure of 8 so when I listened to the bass track it just sounded like a room mic.

This guy also claimed to be able to hear the difference between various DAW's etc and ran a succesful mastering studio as well.He may have had 'golden ears' but it didn't make up for the fact he made poor choices when micing up this band (drum overheads way to high with a very quiet drummer in a not great sounding room)

I have recently recorded the same band, I used a JM47 on the bass(a $100 chinese u87 copy about 15 years old now) into the pre on a steinberg MR816, a signal chain about 1/10 the price of the other,same situation though with live playing and a P.A system for monitoring.The diference was night and day,the bass sounds lovely

The room, the musicians, the songs will make much more difference to the end result of your recording than the mics/pres/AD/DA ever could.Yes, if you've got the budget buy high end, it's better made and will last longer,it will sound slightly 'better' too but unless you know what you're doing it won't make any difference

Enjoy your journey



MC
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Old 08-10-2010, 09:20 AM   #107
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And the rebuttal?

As a person who has no problem of admitting ignorance to the intricacies of clocks in relation to other devices.. I'm taking notes and judging for myself. I find it interesting that people can often times have such opposing views of a topic. I wonder if there is a true 'right' answer sometimes or if it's more of a personal preference thing. Well, in this situation, I feel like there is a factual correct answer, but I was just speaking in regards to the non-unique situation that I'm sure a lot of forum goers experience when trying to 'figure' things out.

But yes, please continue! <scribbles in notepad>
There is a science of audiology. There are researchers testing carefully to determine the acuity of human hearing. There are scientists evaluating the relationship between volume, distortion, frequency response, and human perception. There are scientists exploring the intensely subjective nature of human perception. There are clear guidelines for reducing the impact of the limits of our perceptions.

But there are many many folks who completely reject all this research, knowledge, and guidance.

When someone claims that they are able to hear minute subtle differences without double blind level matched same source comparisons, the answer is confirmation bias combined with bias blindness. The hilarious thing about bias blindness is that even when a person knows about it, it still works to prevent accurate perception.

It ain't easy being human.

Fran
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Old 08-10-2010, 02:52 PM   #108
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Originally Posted by DBMusic View Post
Not sure what you're routing needs are, but between Reaper, TotalMix, and 16 I/O points on the Aurora, you have some pretty creative options in front of you. Totalmix is uber-cool. Enjoy!!

Regards,

DB
I know DB, so many options... I don't even know what to do with it all!

Thanks for all your input in this thread, I appreciate your time and knowledge.
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Old 08-10-2010, 03:19 PM   #109
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Originally Posted by norbury brook View Post
I'll give you some advice for what it's worth as you're geting ready to make your way.Don't worry so much about your gear,really,the difference betwen budget and high end theses days is a lot smaller than it used to be.listen to some of Pat Metheny's first digital records they were done on 16 bit recorders with technically a lot poorer specs than you have now,great sounding records though.Another example of good engineering/sound;James Taylor 'Hour Glass' that was recorded with a Yamaha O2R,not high end by any means but a great sounding record.

Last year my girlfriend who is a succesful singer and runs her own label wanted to do a live DVD,so we rehersed up a great little band and shot it in an artists studio. As i was MD I didn't want to be involved in the actual recording as I'd enough on my plate so the Film production company suggested someone.I spoke to the guy and he had amazing equipment and dropped all the right names,Neve mic pres, Royer ribbons,Neuman 47's,Lavry gold AD etc.

On the day though his choice of mic and postition was bad so when I got back to my studio to mix the sound was awful.I have recorded recently the same band,I didn't have that level of equipment.He chose to mic my Double bass with a Royer ribbon into an expensive pre ,OK fine,but we had foldback pointing at us so a ribbon was a poor choice as it's fixed figure of 8 so when I listened to the bass track it just sounded like a room mic.

This guy also claimed to be able to hear the difference between various DAW's etc and ran a succesful mastering studio as well.He may have had 'golden ears' but it didn't make up for the fact he made poor choices when micing up this band (drum overheads way to high with a very quiet drummer in a not great sounding room)

I have recently recorded the same band, I used a JM47 on the bass(a $100 chinese u87 copy about 15 years old now) into the pre on a steinberg MR816, a signal chain about 1/10 the price of the other,same situation though with live playing and a P.A system for monitoring.The diference was night and day,the bass sounds lovely

The room, the musicians, the songs will make much more difference to the end result of your recording than the mics/pres/AD/DA ever could.Yes, if you've got the budget buy high end, it's better made and will last longer,it will sound slightly 'better' too but unless you know what you're doing it won't make any difference

Enjoy your journey



MC
+1000

I'm completely on board with everything you said there. It's so true that gear is such a small part of the end result when you have MAJOR variables that have infinitely more sway in the completed record. I do admit that I have forgotten that many times since I've started this quest to get a 'high-end' studio so I appreciate it every time I hear it.

I understood the big picture before I went into gear acquisition mode and for me, this road will be ending soon. I took on a 4 year mission to get into a high paying salary job so I can set up my studio and I'm at the last 5 months. Once the new year rolls around, I'll be out of the corporate grind, and therefore not have a 'choice' to even pursue gear on the level I have been.

I'm not sure if my mission is per say, realistic to others, but it is for me. I'm first-most an artist. And although I did luck out with a great technical proficiency, my heart is in capturing all the music in my head. But being a perfectionist.. I was not satisfied with my recordings knowing that I had crappy gear.. or in other words, my recordings could be higher fidelity.

And yes, my engineering skills(mic choice/placement) and mixing skills are no doubt my weakest link right now, but I have no doubt they will improve drastically come next year when I actually have time to enjoy the fruit of my labor. And that's what I meant by, maybe not realistic for others. ie. "Buying" a bunch of professional gear while lacking severely in "using" the gear professionally.

A lot of people think I'm crazy when I tell them I'm planning on leaving my job. I schooled my butt off in an intense Web Development program for 2 years, and have been working in the field for 2 years come 2011... and I plan on leaving. I don't blame them.. I can see how it may seem crazy. Especially in this economy. But I'm wired differently. I've dreamed of creating music since I was 6 and can't imagine sitting in a cubicle designing web-sites 50+ hours a week. I can't ever imagine that.

On a positive note, which I anticipated, I did get more than a great facility for my music, which in itself is pure awesome! I also learned how to create high-end web-sites, market effectively via the web, and research like a scholar.

So there it is, a very shortened summary of where I'm at. Sorry if I rambled a bit there!
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Old 08-10-2010, 03:34 PM   #110
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Originally Posted by Fran Guidry View Post
There is a science of audiology. There are researchers testing carefully to determine the acuity of human hearing. There are scientists evaluating the relationship between volume, distortion, frequency response, and human perception. There are scientists exploring the intensely subjective nature of human perception. There are clear guidelines for reducing the impact of the limits of our perceptions.

But there are many many folks who completely reject all this research, knowledge, and guidance.

When someone claims that they are able to hear minute subtle differences without double blind level matched same source comparisons, the answer is confirmation bias combined with bias blindness. The hilarious thing about bias blindness is that even when a person knows about it, it still works to prevent accurate perception.

It ain't easy being human.

Fran
It sure ain't Fran, and it's even harder being human in the GearSlutz forum—where confirmation bias + bias blindness + buyer's remorse REALLY comes into play!

I agree with the science of determining fidelity, but I also agree that there is still an element that science has a hard time measuring, human influence, which can present an entirely opposing reality to what science has determined. But nonetheless, it's real, because other people buy into it, and people are ultimately reality creators.

I try to strike a balance between the two as going to either side too deep can be counter-productive to my ultimate goal in capturing good music.

It ain't easy being human indeed!
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