adjust WHICH output to stop clipping

whymark

New member
just finished a rather raucus boogie-woogie .....2 mics on one piano....on playback the more "ambitious" parts jump over the "0" mark...OK so I back off the main ST OUT fader but leave the two mic IN faders as far up as I can get them without either of them approaching the dreaded Zone-of-Zero....now I have the main fader down to -30dB and the two mic faders at +6dB.....Is this the right way to go? or should I drop the two mic faders and raise the main fader?
 
Yo.......how about a little more detail on your setup? Are you using an audio interface with a DAW? Or maybe a stand alone Multitrack recorder? Or........???? Where is your ST OUT going to??? We need a clue.
 
Drop the track faders and raise the main fader - just make sure neither the track or mains are in the red zone.
 
I've got 2 mics on a grand piano...they go into a Steinberg UR44 interface (4 I/O plus phantom power) which connects to my computer (Cubase)....the "mixer" is a digital one built into Cubase....as I mentioned in my original post, I've pulled the main fader down so I can keep the two mic faders set higher...I've always thought that it's best to have max energy up to the very end of the process chain and use the main out fader to control clipping...is this correct?
 
just finished a rather raucus boogie-woogie .....2 mics on one piano....on playback the more "ambitious" parts jump over the "0" mark...OK so I back off the main ST OUT fader but leave the two mic IN faders as far up as I can get them without either of them approaching the dreaded Zone-of-Zero....now I have the main fader down to -30dB and the two mic faders at +6dB.....Is this the right way to go? or should I drop the two mic faders and raise the main fader?

It depends.
If the clipping occurred at the conveters then the damage is done, whether audible or not.

Balancing out your session faders is a little less critical but I aim to keep the master fader at unity and mix to suit it.
The only time I move the master fader is if the thing is 100% finished and I want to do fades or balance out song sections ever so slightly.

If a track fader clips at unity you recorded too hot.
If your master fader clips you're mixing your tracks together too hot, and possibly recorded too hot also.

Everything is fixed by going back a stage; Not forward.
 
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I've always thought that it's best to have max energy up to the very end of the process chain and use the main out fader to control clipping...is this correct?

No, you want it to be correct at all stages. Turning a clipped signal down doesn't unclip it.
 
I had a similar issue with a track I'm working on. I was getting clipping on the bass track and upon further review I realized that the original track was clipping when I recorded it. No matter how I mixed it I couldn't make it sound good so I re-tracked the bass with less gain on the input (DI into my US-1800) and it sounds so much better now.
 
1. Set the input of the interface so that the signal from the mics averages around -18dbfs or so on the input channels. Peaks can be upwards of -6dbfs. Leave the faders in the input mixer at unity. I have no idea why they are there, it would be stupid to move them.

2. Set the main output fader at unity.

3. Mix the song. If the master buss clips, you are probably mixing it too loud. The channel faders are there to mix the relative volume of each instrument, not the overall volume of the mix.

4. once the song is mixed, take the rendered mix file and process it to bring the overall volume up, if that is what you desire.
 
The best thing is to just ensure that you feed your mix with as much signal as possible to and from it, don't measure it in terms of dBFS, measure it in dBu.

When you track, you should track as hot as you possibly can using a converter with as great signal capacity as possible. A great mix signal does not come from great input track recording attenuation levels, it comes from great microphone setup for each sound source, where the converter is not in the way of that signal.

Once you have the mix full of great signal, don't get focused on the clipping, focus on the contents of the mix (how you shape the music) and that you do not kill it through signal attenuation, which happens very quickly because the Voltage RMS curvature is logarithmic. Most of the clipping noise is removed naturally during your automation work.

Train your ears to hear transient harshness rather than relying on clipping LED indicators, this is the only long term working solution because you might be re-locating between studios and are not fully aware of all the metering behavior at each studio.

The sound sources that clip the most are typically feeding the compressors slightly too hot, so when you track all sound sources hot and notice that the mix clips at some location, you can get a good sense of where the clipping is coming from simply by looking at the track signal levels.

If you have some analog hardware compressors you can use during tracking and you drive these with high quality power, then if you also set them up correctly and track them through a great converter fed by high quality power, you'll end up with great mixing friendly transients and don't need to work as hard with the transients during mixing. You can monitor this simply by looking at the waveforms, well tracked content has a more round/full waveform shape.

A general rule of thumb: Focus on the signal power and content quality, not on the clipping. Rely on your ears rather than your eyes and brain, but also train in harmonizing the three. Treat the bass element as if it was a mix in and of itself, it kind of is. It's incredible what a well mixed bass element can do to a mix.

The bottom line is that clipping should not be a major focus area during your work. I personally find that it's up to the master engineer to deal with it and that is successfully done when there is not distortion in the microphones during tracking and when the mastering engineer has full access to the whole mix project so that he can remove the clipping very precisely.

If you would have a look at my console, you would be surprised, you'll see crazy stuff. This is because I'm using my ears to make the music/sound more so than my brain and my eyes, I stopped looking at clipping indicators years ago. In the beginning of a producer's career you go with the flow, everything is locked down into bad music and sound because it has to be technically justified. But music and nature speaks its own language, it's about listening to that, using your own ears, then make up your mind yourself about how it truly works based on what your ears have produced. In other words, it's more important to learn to dial in great music, than to have a technically perfect path to it. Why is it like this for a human being? There are several explanations. The human brain is kind of good at analyzing products of things, however it is much less efficient at predicting products that it lacks experience of, it is not trained to think in geometrical patterns and transform those into great sound, although it is capable of it, the standard 10 numeric mathematical framework has confused it over centuries. You need to understand these things geometrically on a different mathematical framework in order to be able to cope with its multi-dimensional nature in a technically precise way. So the smartest thing to do about all of these technical ideas, like signal level, clipping etc. is just to get the experience first, then assign meaning to it and based on that put it into your production strategy to where it belongs. So what I do here is simply that I share my experience. In my case that has told me a great powerful high quality signal is worth focusing on, more so than clipping. Clipping and how to deal with it distracts when you have tons of bigger more musical issues in your productions you need to focus on.

When you track hot, you get much more detail from what the microphones have produced, including the distortion it inherently has created. There is great potential in becoming experienced about how you can feed more of each sound source into the mix with less amounts of distortion. You kind of become more surgical about the capturing of the sound. It's like enhancing the perception of each natural detail present in each sound source. This is why the quality gap between a real sound source and a sampled sound source becomes so dramatic in the hands of engineers that know how to get very surgical about the tracking process of real sound sources and why guys like Bob Clearmountain want to see more real sound sources returning to modern productions. The best word for this quality is resolution - the amount of information you have in each sound source in your production will have an impact on pretty much every aspect about the production. Clipping is bad, but make it musical first, then deal with whatever clipping noise is left.
 
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Ermahgherd... And most of the rest.

Thanks! I did not say use the wrong tracking setup, then capture beyond what your cheap pre-amp/converter is capable of. It was a standalone statement - when all other factors are efficient tracking as hot as possible will both yield the greatest sound and leave you with the most options left. Whatever harshness you have left is not due to the signal level in itself, it has to do with other factors that then need to be corrected. If you track as hot as possible it will also kind of give you the information about what does not work in tracking, much sooner. Because when you track hot it's like you come closer to the source, which means you get more of both the juicy details, however also more of the noise details. So for instance if you are recording a keyboard that has noise in it, you might not be aware of the noise when you track quietly, but when you track with a hot signal you will at some point become aware of the noise and be able to correct it. Tracking hot and using high quality monitoring is a great combo, it will make you able to profoundly adjust things during the tracking process and give the mix engineer a good signal to work with instead of one that will turn out noisy once he starts digging into it. Because a soft signal is soft, softer than what the sound source really is. It shadows the sound source. If I want a soft sound source, I want it to be soft across the whole velocity range, not only in a particular sweet spot.

At least in my experience I find a lot of myth making in the "track cold" idea, especially when it is approached from a dBFS perspective in some relative terms that engineers use to compare what the optimal signal level is, it cannot be compared like that. You have trim/volume faders you can use whenever a signal is too hot, but resolution or characteristics that are missing in the source can never be restored without re-tracking.

In my experience, most engineers end up with harsh mixes not because they tracked too hot, but usually because they used the wrong tracking setup, ran the compressors too hot and did not soften anything in the mix to add emotion. It also leads to wrong sound source choices and wrong arrangement decisions. Those are all separate things. That issue is divided by the tracking engineer making bad decisions in recording and the mixing engineer making bad decisions in mixing, not to be blamed on hot tracking. Although one more easily leads to the other, it is not enough to justify "cold tracking" in my point of view. ITB the more signal effects (excl. compressors/limiters) receive, the better the sound that come out of them. Generally also the more signal you feed out is going to give you better decisions end-to-end. Tracking hot should not be confused with harsh loudness, as you know loudness is something else, it has to do with where you focus the signal across the frequency spectrum and harshness is when you simply have too much distortion in the transients/in the signal.

Now, I know you are a mastering engineer and I know how awful it is to to receive a signal to master that is already crushed to death (I am a mastering engineer too). But that is why you should never (or at least avoid) locking the mix signal into a set of summed tracks. And that's a different issue too.

The way I have validated this has been both personally and by letting my friends who are heavy music listeners provide their opinions of what they find sounds the best. It is pretty much always the recording,mix,master that was tracked,mixed,mastered hot that comes out as the winner. The way they put it is like this: "it is fuller, it's like it has more body". When I then say is it not too powerful? They always agree, no we love that. My conclusion has been that people don't like a weak signal. It's like the mix loses its vitality.
 
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Thanks! I did not say use the wrong tracking setup, then capture beyond what your cheap pre-amp/converter is capable of. It was a standalone statement - when all other factors are efficient tracking as hot as possible will both yield the greatest sound and leave you with the most options left. Whatever harshness you have left is not due to the signal level in itself, it has to do with other factors that then need to be corrected. If you track as hot as possible it will also kind of give you the information about what does not work in tracking, much sooner. Because when you track hot it's like you come closer to the source, which means you get more of both the juicy details, however also more of the noise details. So for instance if you are recording a keyboard that has noise in it, you might not be aware of the noise when you track quietly, but when you track with a hot signal you will at some point become aware of the noise and be able to correct it. Tracking hot and using high quality monitoring is a great combo, it will make you able to profoundly adjust things during the tracking process and give the mix engineer a good signal to work with instead of one that will turn out noisy once he starts digging into it. Because a soft signal is soft, softer than what the sound source really is. It shadows the sound source.

At least in my experience I find a lot of myth making in the "track cold" idea, especially when it is approached from a dBFS perspective in some relative terms that engineers use to compare what the optimal signal level is, it cannot be compared like that. You have trim/volume faders you can use whenever a signal is too hot, but resolution or characteristics that are missing in the source can never be restored without re-tracking.

In my experience, most engineers end up with harsh mixes not because they tracked too hot, but usually because they used the wrong tracking setup, ran the compressors too hot and did not soften anything in the mix to add emotion. It also leads to wrong sound source choices and wrong arrangement decisions. Those are all separate things. That issue is divided by the tracking engineer making bad decisions in recording and the mixing engineer making bad decisions in mixing, not to be blamed on hot tracking. Although one more easily leads to the other, it is not enough to justify "cold tracking" in my point of view. ITB the more signal effects (excl. compressors/limiters) receive, the better the sound that come out of them. Generally also the more signal you feed out is going to give you better decisions end-to-end. Tracking hot should not be confused with harsh loudness, as you know loudness is something else, it has to do with where you focus the signal across the frequency spectrum and harshness is when you simply have too much distortion in the transients/in the signal.
ummm no. You simply record things at line level and not worry about the peak levels, assuming they don't clip. Purposely recording something with a low crest 10-15db hotter than line level is unnecessary, stupid, and will generally begin to overdrive the input stage.

'Resolution' doesn't come from bit depth, it comes from the sample rate. But even that is pointless when pushed beyond our ability to capture, playback or perceive the frequency range it can capture.
 
ummm no. You simply record things at line level and not worry about the peak levels, assuming they don't clip. Purposely recording something with a low crest 10-15db hotter than line level is unnecessary, stupid, and will generally begin to overdrive the input stage.

'Resolution' doesn't come from bit depth, it comes from the sample rate. But even that is pointless when pushed beyond our ability to capture, playback or perceive the frequency range it can capture.

With cheap preamps/converters you need to have a sweet spot thinking with the audio interface, but as soon as you start using higher end gear, what you have in the input stage is mostly a result of what you really have in the input. This is why newbies will never get pro level sound, they will keep adjusting the signal down to non-vitality instead of fixing the issues in their recording process and their gear. What allows it to happen is this keep it softer than it is, idea. Instead, they should push it to where it needs to be and find it is harsh, then throw that audio interface away, abandon their current tracking setups/approaches and go for a pro sound instead.

And no, resolution comes from both the sample rate and the bit depth. Both are equally important for resolution. It sits on a super accelerating curvature, with each step higher in bit depth or sample rate, comes a more profound sound quality increase. You can understand the combination as expressing a vortex.
 
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With cheap preamps/converters you need to have a sweet spot thinking with the audio interface, but as soon as you start using higher end gear, what you have in the input stage is mostly a result of what you really have in the input.
Even high end gear will start to saturate at some point. There is a reason people like pushing Neve preamps, it's because they saturate in a pleasant manner. Other preamps, not so much. But it still distortion.


This is why newbies will never get pro level sound, they will keep adjusting the signal down to non-vitality instead of fixing the issues in their recording process and their gear. What allows it to happen is this keep it softer than it is, idea. Instead, they should push it to where it needs to be and find it is harsh, then throw that audio interface away, abandon their current tracking setups/approaches and go for a pro sound instead.
Having an average signal level of -18dbfs is not compromising the signal in any way. It really only gives 3 bits of space for the transients. Pushing the preamps into saturation to get an extra bit or two of information is a silly trade-off.

And no, resolution comes from both the sample rate and the bit depth. Both are equally important for resolution. It sits on a super accelerating curvature, with each step higher in bit depth or sample rate, comes a more profound sound quality increase. You can understand the combination as expressing a vortex.
Sample rate does not affect quality beyond our ability to capture, reproduce and perceive the sound, since it only changes the possible frequency that can be recorded. If you are recording something that has no energy above 10khz, there will be no difference between a recorded sample rate of 44.1k or 96k...or 22.05k for that matter. Same goes for bit depth, which is one of the reasons 24 bit will be as far as converters will go.
 
Even high end gear will start to saturate at some point. There is a reason people like pushing Neve preamps, it's because they saturate in a pleasant manner. Other preamps, not so much. But it still distortion.


Having an average signal level of -18dbfs is not compromising the signal in any way. It really only gives 3 bits of space for the transients. Pushing the preamps into saturation to get an extra bit or two of information is a silly trade-off.

Sample rate does not affect quality beyond our ability to capture, reproduce and perceive the sound, since it only changes the possible frequency that can be recorded. If you are recording something that has no energy above 10khz, there will be no difference between a recorded sample rate of 44.1k or 96k...or 22.05k for that matter. Same goes for bit depth, which is one of the reasons 24 bit will be as far as converters will go.

The reason why Neve preamps are pushed hard is not due to their saturation, it's due to the lack of transient distortion compared to the distortions in cheaper pres at those signal levels, as well as taking advantage of the high signal levels themselves. The color/vitality comes from the components before it. It is simply put great amplification. Sound sources become vital at these levels when they are not distorting and when they mix with other sound sources they form stronger resonance. However, if you feed a non-harmonically distorted signal into a Neve preamp, there is no magic saturation in it that will harmonize that too. What comes in goes out, which is the job of a great amplifier - to stay clean at high amplification levels. But what a Neve preamp will do, beyond doing a really good amplification, is to tell the engineer when the microphone distorts. So tracking hot with a Neve is serving the engineer with the information about what the capture quality is and feeds the well captured and amplified signal to the next component in the signal chain. The combination of a great microphone setup, with a great Neve preamp, with a great monitoring setup, tracked hot, is a solution that is likely going to work. :listeningmusic:

When it comes to -18 dBFS it means +0 dBu on an +18 dBu converter and +6 dBu on a +24 dBu converter. The resolution of a sound source is directly proportional to the logarithmic Voltage RMS curvature of the dBu range, in other words you lose less and less sound quality as you reduce the signal level. I'm sure you know what only a +0.1 dB signal boost can do to your mix (in case you have a good audio interface, good monitors, good ears). This gives you a very good real world idea of how extremely precious the signal becomes at those signal levels and the potential production quality boost from getting clean transients at high signal levels and high visibility into that range.

When it comes to the aspects of sample rate, higher sample rate (when using high end converters) produces better results because it helps to combat some of the time dilation caused in the clocking. So even with the Nyquist–Shannon sampling theorem and the limitations of the ears, you have secondary reasons why you want as high sample rate as possible. The focus should be on getting as high level of clock accuracy as possible and then sample at as high sample rate as possible using that clocking technoloqy to get as little time dilation as possible and hence get as precise digital representation of the analog signal as possible (by also storing each sample with as high precision as possible). You do need a powerful DAW too as well as power stabilizers/conditioners, else you'll get the opposite effect. What happens is that the resolution simply increases, you'll end up with more of all the great qualities of a recording (given that all other aspects are efficient) - better low end, better mid range, better high end, better and wider stereo image, more 3D, more emotion/mix color, better sense of up/down and so on. When resolution increases it's like a flower that starts blooming, it starts blooming at the upper end of the dBu range and THAT is the way you should understand resolution, that's where the increase in resolution is the greatest. High end productions are like big beautiful flowers in bloom and that's beautiful. :thumbs up:

In practice, these kinds of things are kind of fundamental to understand before you move to what truly takes care of that potential - the microphone setup. So the best thing to do is to just accept the potential of great amplification at high signal levels, then move on to focusing on microphone selection and microphone setups, using your high quality amplification and monitoring as guidance in this process.

There are many ways to production beauty, in rock music I surely know one: high signal level of a high quality drum kit sound in the mix without clipping. That's a very beautiful flower... :D
 
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The reason why Neve preamps are pushed hard is not due to their saturation, it's due to the lack of transient distortion compared to the distortions in cheaper pres at those signal levels, as well as taking advantage of the high signal levels themselves. The color/vitality comes from the components before it. It is simply put great amplification. Sound sources become vital at these levels when they are not distorting and when they mix with other sound sources they form stronger resonance. However, if you feed a non-harmonically distorted signal into a Neve preamp, there is no magic saturation in it that will harmonize that too. What comes in goes out, which is the job of a great amplifier - to stay clean at high amplification levels. But what a Neve preamp will do, beyond doing a really good amplification, is to tell the engineer when the microphone distorts. So tracking hot with a Neve is serving the engineer with the information about what the capture quality is and feeds the well captured and amplified signal to the next component in the signal chain. The combination of a great microphone setup, with a great Neve preamp, with a great monitoring setup, tracked hot, is a solution that is likely going to work. :listeningmusic:
Not true. It's the saturation of the transformers that thicken it up. He has designed other preamps that are much more transparent, but they don't have the same 'magic' as the 1073. This is because the new ones don't saturate as much.

When it comes to -18 dBFS it means +0 dBu on an +18 dBu converter and +6 dBu on a +24 dBu converter.
Yes, it does depend on what your converters are calibrated to. But on average, from the factory they are mostly calibrated between -15dbfs and -20dbfs = 0dbVU.

The resolution of a sound source is directly proportional to the logarithmic Voltage RMS curvature of the dBu range, in other words you lose less and less sound quality as you reduce the signal level. I'm sure you know what only a +0.1 dB signal boost can do to your mix (in case you have a good audio interface, good monitors, good ears). This gives you a very good real world idea of how extremely precious the signal becomes at those signal levels and the potential production quality boost from getting clean transients at high signal levels and high visibility into that range.
Umm, no. A 0.1db boost of an entire mix will not make an appreciable difference. That sort of boost can make a difference of how a track sits in a mix, but boosting the entire mix that amount wont.

When it comes to the aspects of sample rate, higher sample rate (when using high end converters) produces better results because it helps to combat some of the time dilation caused in the clocking. So even with the Nyquist–Shannon sampling theorem and the limitations of the ears, you have secondary reasons why you want as high sample rate as possible. The focus should be on getting as high level of clock accuracy as possible and then sample at as high sample rate as possible using that clocking technoloqy to get as little time dilation as possible and hence get as precise digital representation of the analog signal as possible (by also storing each sample with as high precision as possible).
You can more accurately make a clock work at lower sample rates than at higher ones. In other words, the higher the sample rate, the less accurate the clock is. Even people who design high end converters will tell you that if you hear a difference between 48k and 96k, your converters are broken.

You do need a powerful DAW too as well as power stabilizers/conditioners, else you'll get the opposite effect. What happens is that the resolution simply increases, you'll end up with more of all the great qualities of a recording (given that all other aspects are efficient) - better low end, better mid range, better high end, better and wider stereo image, more 3D, more emotion/mix color and so on. When resolution increases it's like a flower that starts blooming, it starts blooming at the upper end of the dBu range and THAT is the way you should understand resolution, that's where the increase in resolution is the greatest. High end productions are like big beautiful flowers in bloom and that's beautiful. :thumbs up:
You should probably stop drinking the Kool-aid.
 
Not true. It's the saturation of the transformers that thicken it up. He has designed other preamps that are much more transparent, but they don't have the same 'magic' as the 1073. This is because the new ones don't saturate as much.

Yes, it does depend on what your converters are calibrated to. But on average, from the factory they are mostly calibrated between -15dbfs and -20dbfs = 0dbVU.

Umm, no. A 0.1db boost of an entire mix will not make an appreciable difference. That sort of boost can make a difference of how a track sits in a mix, but boosting the entire mix that amount wont.

You can more accurately make a clock work at lower sample rates than at higher ones. In other words, the higher the sample rate, the less accurate the clock is. Even people who design high end converters will tell you that if you hear a difference between 48k and 96k, your converters are broken.

You should probably stop drinking the Kool-aid.

When it comes to the +0.1 dB boost, I'm mostly talking about the pre-mastered version, as I mentioned the difference depends on what the signal level is to begin with, which is the whole point. If you cannot notice this difference, you either are used to much more dynamic content or it's something else, such as you haven't paid enough attention, you monitor on much lower volume or your monitor/converter solution does not reveil it in the same way as in my case. I have a very high end monitoring chain and I'm always amazed at what as little as a +0.1 dB boost can do to the mix. The master bus gain staging is one of the more important things I do in mastering, simply due to this very reason. It's about finding that -0.1dB below audible clipping, just much enough to use the full capacity and just little enough to leave the transients nice and clean. It can be tricky, but when you truly find it, that's when it gets sweet! :)

If you don't agree, that's fine, give it some time, maybe try it out. I have tried tracking cold, both with low end and high end converters and asked my friends about the result too. Initially you think it works, but soon you realize it doesn't. Now, there might be exceptions, like for instance if you record a 100% acoustic performance tracked really really well, but tracking hot will not cause any issues with those kinds of performances either when you know how to gain stage properly. But my conclusion has repeatedly been that when you track hot you know what you have and what you are doing. It's like it instantly gives you exactly the kind of information you need, so that you can make the right moves quickly and easily. The pre-mix process is a bit easier too because you can instantly hear where the potential is. For instance if you mix a pop ballad or something like that you might be interested in the quality of the electric piano, electric guitar and bass guitar. All of them might not have the desired mix characteristics, the electric piano might be quite hard sounding, the bass guitar might be a little flat, the electric guitar might be a little noisy. All of these qualities might go unnoticed when you receive softer versions of them, leading you in a direction when you assign a lot of mix signal to them and discover you ended up with a mix that is slightly hard sounding, a little flat and a little noisy, and you don't know why, so you target other sound sources like drums and vocals and it gets even worse. The mix sound you end up with has a lot to do with what the sound sources sound like at their full signal. When you fix that, then you'll more easily end up with great sounding mixes.
 
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