Possible Stupid Question Regarding Mix Levels

I think you need to get this sorted first. Can you use it to record? Or is it totally disconnected from Cubase. Does it show up as a device inside Cubase - I'm assuming that's where you are selecting the computer card. I suspect the headphone issue and the concern about levels will go away once you have it running properly.
 
The interface is a higher end focusrite. It has headphone/monitor outputs, but when I route cubase through it, I don't hear any sound, so I've been plugging the headphones into the PC sound card. I'll tinker with it more. It must be something to do with the drivers or the signal's route.
You've got to get this working. This situation is affecting your perception of how loud things are. You probably have the gain staging all messed up, because of the this work around.
 
I just tinkered with it and realized the focusrite "playback" in W7 was turned off, so it was using the default (headphone/speaker). I changed that, routed it through the daw, and now it's working. Now I need to test it once I get the channel levels right, which will take a bit of time now that I need to go back and redo them.

One thing I do notice is when the monitor is turned all the way up, there is some static noise.
 
You've got to get this working. This situation is affecting your perception of how loud things are. You probably have the gain staging all messed up, because of the this work around.

This ^^^ Go into your interface drivers and make sure they are installed correctly. Make sure you are installing ASIO drivers. Then go to Cubase and understand how you select the ASIO drivers and your interface. Cubase is a industry standard DAW and so should work with almost if not all ASIO complaint interfaces.
 
One thing I do notice is when the monitor is turned all the way up, there is some static noise.

Check your buffer settings. Again, understand how to setup your interface. ASIO drivers bypass Windows in the DAW.
 
Yes, I have ASIO4all

Double check the manufacture's website, but get the real ASIO drivers. ASIO for all is a simulated ASIO driver and is software trickery. Get the real ones, work much better and give you greater performance and control (since they are written for the interface).
 
Double check the manufacture's website, but get the real ASIO drivers. ASIO for all is a simulated ASIO driver and is software trickery. Get the real ones, work much better and give you greater performance and control (since they are written for the interface).

Strange, the focusrite asio driver has much more latency, though it is slightly less noisy than asio4all. The asio4all is only noisy at 9 or 10 (so at max volume). I think these headphones require an amp because it's still pretty low. They're Sennheiser 580D precisions -- does anyone know offhand if they require an amp? I'll google it.
 
I think I got everything working well. I used the focusrite's output and the method of trimming gain -10db, mixing at low levels, and using the channel faders to tweak. I saw this mentioned on youtube. Everything is sounding good. The mix is much better than yesterday when I had levels too high. Thanks guys.

The only remaining issue is the headphones have static now. Maybe this is just a raised noise floor and normal? It only happens on the highest output settings.
 
Strange, the focusrite asio driver has much more latency, though it is slightly less noisy than asio4all. The asio4all is only noisy at 9 or 10 (so at max volume).
The focusrite ASIO drivers are the ones specifically designed to do what you are trying to do. ASIO4All is a workaround for soundcards without ASIO drivers. Since you have a real interface, use the real drivers. Latency shouldn't be an issue when it is set up properly.

I think these headphones require an amp because it's still pretty low. They're Sennheiser 580D precisions -- does anyone know offhand if they require an amp? I'll google it.
Those headphones are pretty high impedance, so they will be quieter than most other headphones. An amp will help that.
 
Strange, the focusrite asio driver has much more latency, though it is slightly less noisy than asio4all. The asio4all is only noisy at 9 or 10 (so at max volume). I think these headphones require an amp because it's still pretty low. They're Sennheiser 580D precisions -- does anyone know offhand if they require an amp? I'll google it.

Unless you are live recording and monitoring, you won't know there is latency. However, once you need to work with the interface for real time recording, you can adjust the buffer settings to reduce this. For now, while you are just mixing, keep you buffer settings around 1024.

When you are ready to live record and monitor, we can work with you on that.
 
Unless you are live recording and monitoring, you won't know there is latency. However, once you need to work with the interface for real time recording, you can adjust the buffer settings to reduce this. For now, while you are just mixing, keep you buffer settings around 1024.

When you are ready to live record and monitor, we can work with you on that.

You're right. I tinkered with the control panel, and was able to get the focusrite down to 3ms and 4ms in and out, respectively. The Asio was 16ms, so this is better.

I mixed the song. It sounds great. Much better.

So with all that headroom I saved by mixing low, how do I now boost volume in the mastering process? Is it by using a limiter? Is there any other way? I really don't like the sound of compression on anything but vocals, and I already put compression on the vocal channel. I really don't want to squash the entire mix with a limiter just to gain volume.
 
So with all that headroom I saved by mixing low, how do I now boost volume in the mastering process? Is it by using a limiter? Is there any other way? I really don't like the sound of compression on anything but vocals, and I already put compression on the vocal channel. I really don't want to squash the entire mix with a limiter just to gain volume.

Using a mastering limiter is a simple way to get the level up. A limiter isn't like normal compression, you can use it to catch just the most extreme peaks. Usually limiting the mix a few dB won't hurt, but if you push it more you'll start to notice a change.
 
Exactly. Using a mastering limiter to just catch the extreme peaks will not even be heard.

What is the peak level on the master buss?
 
Exactly. Using a mastering limiter to just catch the extreme peaks will not even be heard.

What is the peak level on the master buss?

After everything (including a quick mastering), the peak levels are -10.45 left and -7.45 right. The maximum (RMS) left is -18.47 (left) and -18.40 (right). The averages are 26.76 (left) and 25.72 (right). Are these numbers okay?
 
That means that you can turn the volume up 7db without changing anything or clipping. If you use a limiter, you could easily get another 3db, just by catching the stray peak on the right in order to get the left up to 0dbfs. That's 10db of gain without really getting too far into the limiter.
 
That means that you can turn the volume up 7db without changing anything or clipping. If you use a limiter, you could easily get another 3db, just by catching the stray peak on the right in order to get the left up to 0dbfs. That's 10db of gain without really getting too far into the limiter.

Okay I will give it a try. It won't crush the dynamic range?

I sent the file to a buddy in the band. He said it sounds fantastic, but is kind of low on his PC...
 
You can turn it up 7db without changing the dynamic range at all or having to use a limiter.

Getting 3-5db into the limiter will technically reduce the dynamic range by 3-5db. But practically, it will not affect the songs dynamic range. You would only be knocking back the 3 or 4 snare hits that are louder than the rest. You won't be touching any of the other instruments, just the highest peaks on the percussion. Like I said, you won't notice anything different, except the volume level.
 
You can turn it up 7db without changing the dynamic range at all or having to use a limiter.

Getting 3-5db into the limiter will technically reduce the dynamic range by 3-5db. But practically, it will not affect the songs dynamic range. You would only be knocking back the 3 or 4 snare hits that are louder than the rest. You won't be touching any of the other instruments, just the highest peaks on the percussion. Like I said, you won't notice anything different, except the volume level.

Cool, thanks. I am going to compare the limited and unlimited one tomorrow.

One thing I don't understand: when I was reading up on the best way to gain stage, many people said to turn the trim down on each channel -10db and mix at low levels. Well, I did that, and now at the end of the process I have too low a signal. I am wondering if I would have been better off not doing that and using the faders. Because now I am using limiters to get back the volume I lost from that trim. Or so that is how it feels. Since I have no formal training I don't know if that's accurate.
 
The point about mixing on anything - not just digital is to maximise the s/n, so the gain staging thing is important - but if you are throwing signal away, and then have to boost the resulting file, then you got it wrong, because even a big dynamic range could have been bigger, if managed a bit better. You need to experiment with your system to see where it's limitations are. I mentioned that the actual level of each track doesn't have to be exactly the same, because in a multitrack CD, there is also a balance between the loudness of each track. If you have, in Cubase for example, just a few hot tracks, it's easy to have these summed in the mix bus and need the output level reducing to prevent the red light coming on. With lots of hot tracks it happens very easily. You can work with the channel faders lower, but with analogue and digital mixers this is a pain because you squash the useful operating range of fader travel into a lot smaller area - making delicate adjustment more difficult. If your recordings are too quiet, compared with others you have to inspect, then shove the master up a bit and do the mix again (presuming you recorded the fader movements?)

In the recording and mixing stage you are maximising signal to noise, and finding a working style that does it for you. My method is to have the nominal position of the faders at around the usual ¾ ish position for the key tracks, and push or pull for balance from that position. If you discover you are pulling back (or pushing) more and more tracks, then I reduce or increase them all, and do the balance again until I have a convenient working layout. If the track really makes use of the dynamic range available, then I will bring up the master and watch for headroom - which the cubase meter shows you quite easily. The resultant tracks are quite loud, and the relative balance between songs is restored later on. Once I have a .wav file, it's not good to have to process that file to make it louder. Quieter is easy!
 
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