Interesting to hear.
How hard/simple was the swap out?
Depends on your abilities of (de)soldering SMT parts, and the respective equipment availible. The PCBs are high quality, but only attempt this if you know exactly what you are doing.
I was doing that because PM was suggesting that folks aren't professional if they think there is a sweet spot in a converter. so I was letting him know that some of us do, but also agree with him that It might not be that important and that good gain staging is probably more important
Hmm...no, I didn't say that! I said in my experience, you rarely get time to experiment on a real-world session. It's the sort of thing that you might experiment with in downtime, or maybe on the odd occasion...like everything else that comes in under "experience" you then file this away in your brain and pull it out again when you use it.
In the same way that, reading on here and in magazines, you'd think that every session consists of long periods of trying every mic with every preamp to find the perfect combination for a source.....
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Let me help you re-word that psycho monkey so Greenhirmy agrees.
See below;
(My additions in CAPS - My apologies for taking creative 'license' with what you wrote...)
Quote:
Originally Posted by psycho_monkey
Hmm...no, I didn't say that! I said in my PROFESSIONAL experience, you rarely get PROFESSIONAL time to experiment on a PROFESSIONAL real-world session. It's the sort of thing that you might PROFESSIONALLY experiment with in PROFESSIONAL downtime, or maybe on the odd PROFESSIONAL occasion...like everything else that comes in under "PROFESSIONAL experience" you then file this away in your PROFESSIONAL brain and pull it out again when you use it PROFESSIONALLY.
In the same way that, reading on here and in PROFESSIONAL magazines, you'd think that every PROFESSIONAL session consists of long periods of trying every PROFESSIONAL mic with every PROFESSIONAL preamp to find the perfect PROFESSIONAL combination for a source.....
Sorry, I couldn't resist. I'm just being rambunctious this morning.
In my view I get the most pristine sound when I setup my hardware sound sources to output 90% of their total capacity and then adjust the level on my PrismSound Orpheus as hot as I can without making it clip. To me the discussion of capturing at lower levels never made sense, I want the signal as hot as possible going in, I can always bring it down ITB if I need to, but lost signal is lost signal. This approach though, I found out, requires really great way of connecting the instruments. In my case the unbalanced cables picked up a lot of noise, primarily from my DAW's fan, so I'm currently in the process of upgrading these, I'm probably getting some really nice dual mono DI box combined with some really hifi short unbalanced and balanced cables... But, 90% output towards the hottest level possible in the converter is the way I do it these days...
In my view I get the most pristine sound when I setup my hardware sound sources to output 90% of their total capacity and then adjust the level on my PrismSound Orpheus as hot as I can without making it clip. To me the discussion of capturing at lower levels never made sense, I want the signal as hot as possible going in, I can always bring it down ITB if I need to, but lost signal is lost signal. This approach though, I found out, requires really great way of connecting the instruments. In my case the unbalanced cables picked up a lot of noise, primarily from my DAW's fan, so I'm currently in the process of upgrading these, I'm probably getting some really nice dual mono DI box combined with some really hifi short unbalanced and balanced cables... But, 90% output towards the hottest level possible in the converter is the way I do it these days...
So you don't understand 24bit recording then...there is no "lost signal".
According to my research signal peaks at around -12dB is the best option for most converters, including the Aurora I presume. Personally, however, I tend to go a bit hotter than this.
According to my research signal peaks at around -12dB is the best option for most converters, including the Aurora I presume. Personally, however, I tend to go a bit hotter than this.
First, you need to decide what max level reference the unit needs to be set to +4 or -10, these two levels correspond to the 0dB levels in the relative analogue domains.
Second decide on the normal levels of the compressor whether you do this in the box or out of the box and what the normal operating levels are.
Working in the box, set the signal to peak at -8dB with a +4dB setting or the equivalent, this is called calibration on the equipment - matching one against the other.
But remember that the 0dB on any digital piece of equipment is the max.
Incidentally, this is of course dependent on the input signal whether balanced or unbalanced and remember that lynx lied about unbalance conformity - they have unbalanced issues probably around the input filter.
Personally, I would not touch the Lynx gear with a barge pole, if you can afford better, they have issues and are reluctant to assist in any way - having a constant argument over simple questions - they do not care.
I feel sorry for the purchasers of the Hilo and keep an eye on Ebay if you desperate to own one.
I'm in the don't care, just don't clip camp. You'll never go wrong with a conservative level into a modern high end A/D which is nice cause it's one thing I don't have to fret about. I also don't fret much over preamp selection so it's mics, mic choice, instrument and the player that matter the most.
Lynx Aurora 16, while potentially excellent sounding and fun to use, does not have the most headroom. Here's why:
Assuming a standard +/-15V power supply, the OP-275 op amps comprising the analog section can accept inputs of up to 10.5Vp before exceeding specified acceptable distortion. This equates to 7.425Vrms or +19.63dBu. Ok, so that's lower than the anywhere from +20 to +28dBm max output that quality mic preamps are capable of producing, but Lynx have probably implemented a differential receiver circuit having -6dB loss. This means analog (theoretically) won't actually clip until +25.63dBu (a conservative +24 is probably closer to reality), and even if I'm wrong, we can still watch our levels and make do.
However, there is another limitation. I haven't double checked lately, but I recall the stock (sans variable trim) Aurora A/D is calibrated for -20dBFS = 0dBu, as are many converters. This means a +20dBu input yields 0dBFS, also known as the onset of digital clipping. So, even if the power supply is a few volts higher for a few more dBu in analog domain, the digital output will still clip a few dB sooner, and it all clips before the majority of transformer balanced analog outboard and consoles that people like to track through.
Adding insult to injury is leaving all your software faders at 0 before bouncing to disk. For instance, (and I realize this is not a realistic example) let's say you have 16 tracks of random noise each idling at -12. The summed result is about 0. In other words, loud tracks combine to form an even louder 2-mix which clips your busses.
Now, if you're overdriving transformer-balanced preamps (to get thicker tones), then you can see how it's pretty easy to clip the Aurora A/D. In my experience, this will still sound forgiving on a couple tracks out of several, i.e. your kick and snare hit the red once in a while. However, if you push all of your tracks, then like me you may begin to hear a low level upper midrange haze across the mix. Some of this is the software mixbuss and some of it's the Lynx. This is called distortion, and it makes a lot of sense when considering the numbers explained above.
To answer the original poster's question, I find Aurora 16 sounds best when externally clocked and trying to print tracks which live around -18 in Pro Tools, depending on track count.
And all can be solved by lowering the output fader of the channel on the mixer you are trying to saturate. 24dBm max input is not bad at all, even if you are saturating your output transformer its not bad, you can always use an inline pad if you want even more saturation of the otput transformer, but IMO, its just easier to lower the fader and let the internal electronics of the mxier to melt and get that extra from the output transformer, hitting transformers hard doesnt work that well, I onced discussed this issue on prodigy-pro (now groupdiy), and most agreed that transformers sound best when driven moderatly low, or with some minor saturation, hard saturation of trannies eats up the low end and starts sounding ring modulated, you are better off hitting those trannies at a healthy level, and having the internal electronics melt.
So again, by lowering the output fader you wont have any problems if you want saturation.
I'm in the don't care, just don't clip camp. You'll never go wrong with a conservative level into a modern high end A/D which is nice cause it's one thing I don't have to fret about. I also don't fret much over preamp selection so it's mics, mic choice, instrument and the player that matter the most.
What is also significant is that the level of gain from the preamp will have a LOT more effect. eg someone hits the converters a bit hotter, they think it sounds better so they decide it's the converter stage doing it. In reality, they've turned up the gain to their preamp (or even just the output stage), which is what is actually causing the difference.
[QUOTE=psycho_monkey;8301723]Try it for yourself next time
I'm not sure if that was directed at me...
But I prefer to do more than just 'read' about the gear that I use everyday.
Especially on this site, where only about 10% of what is written can be taken with any authority.
I'm not sure if that was directed at me...
But I prefer to do more than just 'read' about the gear that I use everyday.
Especially on this site, where only about 10% of what is written can be taken with any authority.
not at all, it was directed at the Rev who thinks the way of forming an opinion on something is to listen to what everyone else thinks - I think WE are in agreement here!
What is also significant is that the level of gain from the preamp will have a LOT more effect. eg someone hits the converters a bit hotter, they think it sounds better so they decide it's the converter stage doing it. In reality, they've turned up the gain to their preamp (or even just the output stage), which is what is actually causing the difference.
EXACTLY, louder sounds better! SO MAKE SURE YOU GAIN STAGE PROPERLY.
not at all, it was directed at the Rev who thinks the way of forming an opinion on something is to listen to what everyone else thinks - I think WE are in agreement here!
Peaking -18 is way too low. RMS -18 is more like it. That usually means that serious transients (kik/Snare) can peak anywhere between -12 and -6.
I usually shoot for peaks around -10dBfs, but why is -18 "way too low" ? With something with the quality of a lynx converter, -18dBfs would probably have you capturing a dynamic range in excess of 100dB. I am not sure how that would create any real world problems.
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not at all, it was directed at the Rev who thinks the way of forming an opinion on something is to listen to what everyone else thinks - I think WE are in agreement here!
"Everyone else"?
More like the experts from Sound on Sound magazine, etc.
"Everyone else"?
More like the experts from Sound on Sound magazine, etc.
Sound on sound is mainly written by journalists. With the odd exception of course. I've never met a SOS writer on an actual session, and if they were all over the uk industry, I would have. "experts"? Not really....
Don't believe everything you read, and try things for yourself before forming an opinion.
I usually shoot for peaks around -10dBfs, but why is -18 "way too low" ? With something with the quality of a lynx converter, -18dBfs would probably have you capturing a dynamic range in excess of 100dB. I am not sure how that would create any real world problems.
I have tried many times to record peaks at -18.
It just turned into an exercise in futility. All my comp plugs had to be railed just to get 4 db of compassion. I was constantly looking for gain. It was just plain wrong ITB. Also It sounded worse. So I reverted back to recording with drum peaks in general-10, but sometimes -3 and all other instruments to fit accordingly. Pull back on the reigns where need be, don't go red, unless the plug requires it for whatever reason and mix away. Go in reasonable and come out reasonable. Always listen to the way it sounds.
You of all people should know that theory and practice are different.
I haven't double checked lately, but I recall the stock (sans variable trim) Aurora A/D is calibrated for -20dBFS = 0dBu, as are many converters. This means a +20dBu input yields 0dBFS, also known as the onset of digital clipping.
Is this true? I was planning on going VT to calibrate -20dbfs=0dbu but I would go stock if its already done.
Is it just me, or does anyone on a commercial session not have time to worry about converter "sweet spots" and so on?
I consider myself fairly conscientious - read that as "anal" if you like - but in 99% of commercial situations, you're under some form of time pressure. Of course you spend time on mic placement, moving things around, preamp drive, and if tracking to tape then definitely how hot you hit things. After all that, how much time is left to spend working out how to hit a converter's "sweet spot"?
Yes. Thank you. I'm sitting here, thinking to myself "None of my clients are going to be impressed by the converter sweet spot or how long it takes to find it before every tracking session."
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