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Post by terryrocks on Feb 15, 2017 14:55:57 GMT -6
I've gone many years without bothering to calibrate my studio for optimal performance.
i think it's hurting my productions.
i have a behringer cable tester with test tone capability
i also have test generators in both protools and logic
and multimeters, spl meter
i think i have all the necessary tools but I am having a difficult time finding a resource to help me work through the process.
here's one video i came across:
wondering what others do
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Post by gouge on Feb 15, 2017 15:46:51 GMT -6
I use a test tone on the console to calibrate all the outboard gear and busses. I do it regularly and always before a mix session.
That and manually zero out all the channels on the console.
I also recorded the test tone into a daw template and copied the track across 16 channels. That then gets used to find unity for converters and console channels.
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Post by terryrocks on Feb 15, 2017 15:58:41 GMT -6
My setup is:
16 outboard preamps (capi, hamptone, rnp, bla autuer, sytek) -> balanced patchbay -> motu 16a -> computer
I just sent a -20dBu 1khz tone from my cable tester into each of my motu inputs to make sure all channel meters in the box show me the correct reading.
Everything there seems correct.
I think i need to figure out the sweet spot for each type of preamp.
Once that's done, if every mono track I record averages close to -18dBu, I should be golden....right?
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Post by wiz on Feb 15, 2017 16:10:49 GMT -6
My setup is: 16 outboard preamps (capi, hamptone, rnp, bla autuer, sytek) -> balanced patchbay -> motu 16a -> computer I just sent a -20dBu 1khz tone from my cable tester into each of my motu inputs to make sure all channel meters in the box show me the correct reading. Everything there seems correct. I think i need to figure out the sweet spot for each type of preamp. Once that's done, if every mono track I record averages close to -18dBu, I should be golden....right? I use the 16A I have outboard gear... I use -20dBfs = 0VU So that translates to PEAKS when I am tracking of around no more than -12/-10dBFs. cheers Wiz |
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Post by swurveman on Feb 15, 2017 16:41:07 GMT -6
I know there's a lot of theory about gain staging at a lower dbfs level itb, but has their been a blind A/B test of mixes
where people overwhelmingly preferred the lower channel level mix, as opposed to a mix pulling down the master fader in a 32 float system?
Note: I would assume that in the 32 bit float blind A/B test that plugins with a volume control would be used so that the plugins would not be clipping.
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Post by wiz on Feb 15, 2017 17:04:13 GMT -6
I know there's a lot of theory about gain staging at a lower dbfs level itb, but has their been a blind A/B test of mixes where people overwhelmingly preferred the lower channel level mix, as opposed to a mix pulling down the master fader in a 32 float system? Note: I would assume that in the 32 bit float blind A/B test that plugins with a volume control would be used so that the plugins would not be clipping. its more about the analog portion of the process... not once its captured. You need to think in terms of how you run the analog gear... example... My U87 into my 1073 sounds great to me with 1073 gain at 50 and the trim bringing the output level down ( I do this to some extent on all my preamps, I have tested them at all different gain levels, different output fader levels, and found the sweet spot.) It goes into my STA Level and I capture the track via the MOTU 16A I might have a peak around -14. Most of the time it hovers around -18 -20.... Peak. You need to CAPTURE your recording, using mic position, and gain staging to achieve the SONICS you want, at a level that does not stress the ADC of your capture device. Gear works best at different points... yes there might be an exception that a AD sounds good clipping... but lets not go there.... Give yourself adequate head room on the ADC and the DAC of your gear, set the volume of your monitors to make it loud enough for you to mix at... Run, your mix digital path in the DAW using -18/-20dBFs as your 0Vu point and you will end up with mixes with adequate headroom for mastering. My summary 0VU = -20dBFs Motu 16A Peaks when tracking , generally no more than -12 (an errant snare hit might go higher) Mixing digital, same 0VU point, my monitors are set so that level is good for me to mix at without fatigue. Which means turning professional mastered music down by about 10dB at least when playing it through the system. This means, my final mix I send to mastering has peaks of round -10dBFs...... cheers Wiz |
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Post by swurveman on Feb 15, 2017 17:31:10 GMT -6
Thanks for your reply Wiz. I do what you do, which is the conventional wisdom. I'm still curious about blind A/B tests.
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Post by wiz on Feb 15, 2017 18:23:36 GMT -6
Thanks for your reply Wiz. I do what you do, which is the conventional wisdom. I'm still curious about blind A/B tests. Here is my two cents... once its captured..... short of plug ins that are level dependant...say saturation plug ins, compressors etc.... It doesn't matter 3/5 of f()*U all....8) You can have levels all over the joint once its in there....(to a point of course) but the difference between a mix peaking at -20 and -1 in the digital domain, after capture and subject to the caveats above...is nada. ( I am talking about within the DAW, not once you try and bring it back into the real world) Try it yourself.... if you are talking about having a mix on the stereo bus, that peaks at -10 vs -1 vs -15dBFs does it sound different once level adjusted... no cheers Wiz |
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Post by Guitar on Feb 16, 2017 17:48:56 GMT -6
What a fun thread! Thanks wiz and everyone else also.
With 24 bit conversion, you can be very conservative with your peak levels, and still be OK in the DAW mix.
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Post by ChaseUTB on Feb 16, 2017 23:34:15 GMT -6
I know there's a lot of theory about gain staging at a lower dbfs level itb, but has their been a blind A/B test of mixes where people overwhelmingly preferred the lower channel level mix, as opposed to a mix pulling down the master fader in a 32 float system? Note: I would assume that in the 32 bit float blind A/B test that plugins with a volume control would be used so that the plugins would not be clipping. 32 bit float is a bit of a fallacy imo. If you are ITB only and don't mind melting your DA to your monitors then feel free to run +12 over dbfs while 32 bit floating 🤓 It's all good til you hit an actual output and 32 bit float has to be converted to analog. Henceforth why " pulling down the fader" after floating above 0 dbfs is not an actual solution to the problem. Some new Converters claim 32 bit dynamic range however I have never heard of a DA converter capable of outputting / inputting 27 + dbu it will be clipped. Most HW can output this level easily. Also, seeing as that most work with HW on busses or in the mix period, this means everything will again have to be pulled down ( aka properly gain staged ) in order to maintain proper signal flow to and from the DAW... |
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Post by Deleted on Feb 17, 2017 10:36:23 GMT -6
I know there's a lot of theory about gain staging at a lower dbfs level itb, but has their been a blind A/B test of mixes where people overwhelmingly preferred the lower channel level mix, as opposed to a mix pulling down the master fader in a 32 float system? Note: I would assume that in the 32 bit float blind A/B test that plugins with a volume control would be used so that the plugins would not be clipping. 32 bit float is a bit of a fallacy imo. I can add a bit to that. Purely from a math standpoint, things in floating point will work just fine no matter how the level relates to 0dBFS. In a long and complicated workflow, you might be able to get away with all sorts of too-hot/too-cold signals as long as there's a final stage that somehow puts it in the proper range for D/A conversion. On signal capture, it won't really get you anything. It's still a matter of headroom, even though 24-bit conversion makes life a little less verklempt when you're setting levels. But a too-low signal when converted to float will still simply be encoding noise at the lowest levels. So those two most important points--capture and reproduction--still drive everything. But there's another thing that some of us must think about in plug-in world. For compressors, gates, distorters and so on, the relationship to 0dBFS is all-important. You have to have some sort of trust in the incoming signal level so that you know when and how a level-dependent algorithm should work. And there are occasions in which a badly-behaved plugin can cause havoc on down the line. In my own plugs I perform input monitoring, clearing out nonsense values and doing some input limiting. In most cases I'll allow the input to be as much as +24dBFS (although it shouldn't be) before I kick in hard limiting. You've got to draw the line somewhere ;-) All this means that we've still got to perform proper gain-staging in the digital world. The reasons are all different: it's not so much about hum and slamming busses as it is in the pure analog world. It's about allowing any dynamic-dependent algorithm to perform consistently from mix to mix. |
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