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Post by tasteliketape on Dec 23, 2015 23:21:17 GMT -6
I know this is discussed to death but a friend just came back from Nashville where he tracked his new project Went over to listen to ruff mix and the levels are hitting in the yellow especially bass and kik drum( not all instruments) Even some guitar . I track with my klanghelm meters set at -20 and try to stay close to that Which way is correct? I mean his project sounds great and I didn't get a chance to see what his levels db were but isn't that kinda hot? I retired at age61 this is my new tax write off coz sure as hell won't make money Now that's funny I don't care who you are lmao
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Post by mrholmes on Dec 24, 2015 1:36:58 GMT -6
Third post sums it up.... duc.avid.com/showthread.php?t=230766Quote: What noise? Your mic preamp that you are cranking up is going to have more noise in it than your protools system. Most mic preamps have a noise floor of about -90dB, protools is around -118dB to -120dB. Your mic and mic preamp will have AT least 30dB MORE noise in it than your DAW. So, if you are trying to record at louder levels the only thing you are doing is putting MORE NOISE into your tracks because you are turning your mic preamps up to get the signal as hot as possible without clipping. If you are recording into a DAW, you are NOT USING TAPE. There is no need to record as hot as possible since there is no tape noise/hiss to compete with. You are only going to make your recordings noisier by turning up the preamps. Calibrate your studio and everything will fall into place. The 192IO is set from the factory to -18dBfs (on the PT meter) = +4dBu = 1.228 Volts = 0 VU (on an Analog VU meter). This means, the old analog technique of "keep the needle right around 0 on the VU meter" translates to "Keep the signal right around -18dB on the Protools meter (which is just under halfway up the meter)". Some of the best tracking engineers I've seen, record everything with the faders set to "0" and change the mic preamp gain to place things proper ly within the mix WHILE TRACKING. Most people today don't really do this because pulling the fader down in PTHD doesn't really effect the sound of the track (until you get down around -90dB on the fader). Whereas on an analog console as soon as you start pulling the fader down you are changing the sound (since the fader is a voltage controlled amplifier/variable resistor). So, to answer the original poster's question... if you are going to record and mix completely in the box you are better off keeping the levels lower for better Signal to noise ratio and to keep intersample peaks from clipping plugins and such.
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Post by tasteliketape on Dec 24, 2015 1:53:56 GMT -6
thats more or less what i do except - 20 db and what i thought, but when i saw this project done by a reputable engineer and it sounds great i mean really good made me question what i thought i knew
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Post by Randge on Dec 24, 2015 2:44:42 GMT -6
thats more or less what i do except - 20 db and what i thought, but when i saw this project done by a reputable engineer and it sounds great i mean really good made me question what i thought i knew That depends on the finality he was shooting for. Was he thinking he was gonna have to mix it until someone told him different?
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Post by popmann on Dec 24, 2015 2:45:41 GMT -6
You should track at/around the calibration of your converters. This varies from manufacturer to manufacturer. My RME is -13=0....apogee are -18=0 by default, configurable to the video standard of -22.
At the end of the day, though, that is to be accurate. Many, like Apogee or Burl (in different ways) may have a pleasing sound driven into non linear operation....and "always in the yellow" is just as meaningless....are they full dynamic range or did they limit on the way in? What is "yellow" on the given meter? And is it momentary or constant? I describe level setting as having "regular peaks" and occasional peaks. Easiest to understand on a rock drum kit--the kick and snare on the beats should be regular peaks--and go to the calibration level, but he may dig in for a fill...or some accidental or intentional accent that goes far above. These are the occasional peaks. I find if I set my levels on a guitar to hit -13 regularly....there will be some -7 peaks here and there....all fine-one of the reasons you are aiming so low is how forgiving 24bit digital is....
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Post by tonycamphd on Dec 24, 2015 10:35:55 GMT -6
it's all negligible around 0 on any daw meter(usually -18dbfs), if you're tracking session is 32bit float (everyone should imo), you have some astronomical amount of HR before clipping.
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Post by Martin John Butler on Dec 24, 2015 10:52:03 GMT -6
Lets say I did track at those recommended levels, and when I have a good mix, I want to bring it up to broadcast level, (or at least a common CD level). How come my playback levels are still lower in comparison to my reference tracks, despite having enough compressor/limiter plug-in choices than I can use in two lifetimes? Am I just a shitty "mastering engineer", or am I missing something? (other than a true mastering engineer :-)
Cowboycoalminer's tracks are much closer to pro volume levels, yet his are both louder and less distorted than mine. When I bring my tracks up to his level, mine are too harsh.
Does it have something to do with using an analogue soundboard ?
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Post by jazznoise on Dec 24, 2015 11:07:09 GMT -6
-18dbFS @ 24 Bit. It's not even a question, there is literally no good reason to do otherwise unless your gain staging doesn't allow for it. Sometimes for vocals, lower numbers might have to be picked. This is why broadcast guys tend to split a feed with one recording at a quieter level, so there is always one track that's not clipping.
Sometimes for quiet sources I print ambience mics hotter to reduce the preamp noise by using a higher gain setting. But the peak/rms ratio is much lower, so it mitigates the risk.
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Post by popmann on Dec 24, 2015 11:49:26 GMT -6
Lets say I did track at those recommended levels, and when I have a good mix, I want to bring it up to broadcast level, (or at least a common CD level). How come my playback levels are still lower in comparison to my reference tracks, despite having enough compressor/limiter plug-in choices than I can use in two lifetimes? Am I just a shitty "mastering engineer", or am I missing something? (other than a true mastering engineer :-) Cowboycoalminer's tracks are much closer to pro volume levels, yet his are both louder and less distorted than mine. When I bring my tracks up to his level, mine are too harsh. Does it have something to do with using an analogue soundboard ? You're confusing absolute loudness and relative. You are indeed missing something pretty fundamental. Without a firm grasp of gain staging and signal flow you are pissing in the wind.
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Post by Bob Olhsson on Dec 24, 2015 12:27:36 GMT -6
Probably the biggest problem with most common digital audio converters is a sh!tty analog power supply. For that reason it's best to peak no higher than -10 and error on the low side. A lot of recording engineers don't understand this. Lack of dither along with over-stressed chips and power supplies is the main cause of what people don't like about the sound of digital audio.
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Post by Ward on Dec 25, 2015 7:25:12 GMT -6
-18 isn't -20, nor is it -10 or -3. It's -18dbfs. It's your 'zero'. Pure and simple.
but it's ok on bass and drums to be in the yellow... it happens when tracking progresses and we get more energetic. It isn't the end of the world, nor will it result in extra noise.
but please read this part:[/b] Reduce the waveform volume of tracks that are peaking out because they WILL overload your plugins. If your plugins start turning red, they're overloading. And this can cause as much noise as preamp and/or EQ and/ore compressor overload on the way in to your converters.
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Post by tasteliketape on Dec 25, 2015 10:22:21 GMT -6
just to clarify and yes I realize you will occasionally get overs but the session I was referring to was hitting in the yellow 70 to 80 %of the time yet sounded very very good an clean To me it wasn't the norm the bass which being a bass player myself I pay extra attention to was very hot an one of the cleanest bass tones I've heard Hey we can all always learn something at least hope I never stop learning
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Post by Martin John Butler on Dec 25, 2015 11:07:15 GMT -6
Ward, I don't believe i've ever seen a plug in "turn red", even though I've blasted them on occasion. Not saying you're wrong, it's just that I do hear too much edge in my mixes, likely from overdoing the plugs in the 2 bus, but it could be from what you're referring too. I assume this means plug-ins are modeled to overload like the hardware?
Bob said. "over stressed chips", I'm curious how that looks in my DAW, and what I can look and listen for to avoid this.
Popmann said, "You are indeed missing something pretty fundamental. Without a firm grasp of gain staging and signal flow you are pissing in the wind." Well, you got me there, I confess, you're on to something there. My grasp of gain staging is indeed quite weak, and I need to get this aspect together soon. Thanks for the push Popmann. :-)
If I tracked everything at lower levels, the left the 2 bus level at - 12 or 14 or 18 dbfs, whichever matches my Apollo, how much gain would it then take to reach a loudness level of say a Mark Knopfler or Lyle Lovette album? It seems I just can't get there without things getting edgy.
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Post by tonycamphd on Dec 25, 2015 11:48:36 GMT -6
just to clarify and yes I realize you will occasionally get overs but the session I was referring to was hitting in the yellow 70 to 80 %of the time yet sounded very very good an clean To me it wasn't the norm the bass which being a bass player myself I pay extra attention to was very hot an one of the cleanest bass tones I've heard Hey we can all always learn something at least hope I never stop learning what are the session parameters set at?
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Post by mobeach on Dec 25, 2015 13:00:11 GMT -6
For that reason it's best to peak no higher than -10 and error on the low side. Are you talking after mastering?
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Post by tasteliketape on Dec 25, 2015 13:33:45 GMT -6
Only thing I know about the session is it was 88/24 And an Apollo interface the silver one protools 11
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Post by Bob Olhsson on Dec 25, 2015 15:56:33 GMT -6
Distortion accumulates. You can literally record peaking to -20 with no noise penalty when you turn it back up. Stressed chips cause things to become more distorted down the line even when it wasn't very noticeable at first. I recommend just trying it.
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Post by popmann on Dec 25, 2015 20:48:20 GMT -6
A)he's not talking about "after mastering"...."after mastering" everything is present at or near full scale....and is again--UNrelated to tracking and mixing levels.
B) I want to clarify Bob's "stressed chips" to point out those are the ANALOG chips. The chip based line amps with the insufficient power supplies.
C) zero reason for running a session at 32bit float until mix time (assuming you're mixing/summing INSIDE the flaoting point mixer. Your mixer and plug ins are ALWAYS running 32bit float even if you pick 16bit for the session....your ADC is 24bit, period. Write that to 32bit float files and you're simply making the files bigger adding nothing. I stipulated "until mix" because there IS a theoretical advatange in an internal capture at native 32float format---giving THAT to the mastering engineer to start their job with. But, for tracking? Zippo. And in reality--I've had TDM based mastering houses kick the 32bit float files back and ask me for the 24bit (because TDM systems are not floating point systems).....and I've never actually heard a difference--but, I'm willing to do some things just based on theory. I should point out that if you do a bunch of DSP to the tracking--THAT is when you store it in floating point so that every DSP process doesn't have to go to 32 and back to 24 and back to 32, etc....but, technically, the new engines running at 64bit float might not even have a theoretical advantage unless you use 64bit float. But, for capture? 24bit outstrips the SNR of any line amp that's feeding it....and all the chips are 24bit anyway....so there's no point in writing 32bit float files during the tracking process.
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Post by Bob Olhsson on Dec 25, 2015 21:21:14 GMT -6
Analog chips and analog stages in converter chips.
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Post by jeromemason on Dec 25, 2015 22:03:33 GMT -6
Are you speaking on the recording stage or when you're mixing?
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Post by popmann on Dec 25, 2015 23:48:00 GMT -6
There's no difference. You track at the right level....it doesn't get considerably (absolute level) louder when you mix because you're not setting EQ and compressors to increase the PEAK level, right? You're altering the relative balance--not making stuff louder....unless--X needs to be louder relative to Y....
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Post by tonycamphd on Dec 26, 2015 0:20:50 GMT -6
rule of thumb, higher voltage, more headroom=better everything! 8) I'd also add that every session I start is 96k/32 float, and it finishes 32 float, I have more SSD hard drive space than i could ever use, so there is no reason today to do any less imo, I also have 64 gigs of memory on a Mac that geeks over 20k...so.. maybe I need to throw this bad boy in? accessories.dell.com/sna/productdetail.aspx?c=us&l=en&s=bsd&cs=04&sku=A8094805
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Post by LesC on Dec 26, 2015 9:08:25 GMT -6
rule of thumb, higher voltage, more headroom=better everything! 8) I'd also add that every session I start is 96k/32 float, and it finishes 32 float, I have more SSD hard drive space than i could ever use, so there is no reason today to do any less imo, I also have 64 gigs of memory on a Mac that geeks over 20k...so.. maybe I need to throw this bad boy in? accessories.dell.com/sna/productdetail.aspx?c=us&l=en&s=bsd&cs=04&sku=A8094805Maybe you should buy several of them and run an appropriate raid configuration, you can't be too careful these days. On a less jovial note, I've been running 96/32 on Cubase for a while with my Win7/64 system. Whenever I use any other music app, even just to play an mp3 or a wav file, it's randomly played back at 44.1 or 48k. Maybe I've set something wrong on my RME UFX, but I'm thinking of switching to 88.2/32 to alleviate the problem. Any suggestions? Be kind, it's Christmas time.
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Post by Ward on Dec 26, 2015 9:48:43 GMT -6
LesC, you aren't doing anything wrong... this is an OS preferences change and it can happen on a mac system too. By defauklt, MP3s are either 44.1 or 48 at input and once one is played, it instructs all software and hardware to adapt to its needs and thus changes it going forward and you have to change it back. At least, those have been my findings.
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Post by LesC on Dec 26, 2015 10:37:58 GMT -6
LesC, you aren't doing anything wrong... this is an OS preferences change and it can happen on a mac system too. By defauklt, MP3s are either 44.1 or 48 at input and once one is played, it instructs all software and hardware to adapt to its needs and thus changes it going forward and you have to change it back. At least, those have been my findings. Thank you, that would explain the apparent randomness. I think most of my music files are 44.1 so I'm going to try recording at 88.2 and hopefully minimize the problem. Or I'll just keep the Dangerous Source and use it's DAC for all non-Cubase audio, it always plays correctly.
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