Pico's studio

  • Thread starter Thread starter picostudios
  • Start date Start date
I do notice something about my recordings. When ever I record my drum tracks, it seems to have more on the top side of the wave form than the bottom. Like it's uneven. I am unsure if that is natural or maybe something wrong with my equipment.
 
picostudios said:
I do notice something about my recordings. When ever I record my drum tracks, it seems to have more on the top side of the wave form than the bottom. Like it's uneven. I am unsure if that is natural or maybe something wrong with my equipment.

Well, if you look at the top waveforms (the white background window), what you describe is exactly what i see as well. The waveform as a whole is very slightly shifted up, as it clips on the top (positive side) but not the bottom (negative side). Remember that audio goes from zero, to positive, crosses over zero, to negative, and repeats.

What equipment are you using?

I ask because some manufacturers are "lazy" and save a few dollars on their meter circuits.

VU meters, whether the old fashioned "needle" kind or a LED bargraph, are essentially voltmeters. The measure the average voltage (or peak voltage) of the audio signal that is fed to it, but can only display one half of the waveform (positive or negative), but not both. An additional little bit of electronics is necessary to inverse, and sum the other half of the waveform together, so you have an average reading of the positive side of the waveform, as well as the negative. Often times manufacturers leave out this little extra electronics because it costs money.

The non "Eurodesk 8000 series" Behringers are like this, as are the Tascam TM-D1000's, for example. They look at one half of the waveform, so you are "blind" to the other half.

I have several TM-D1000 mixers, and if I were to feed a +.9V signal to channel 1, which is balanced, I'll see an indication on the VU meter. If I feed it a -.9V signal to the same channel, the VU meter LED's go out to indicate ZERO SIGNAL.

Scary, huh?

That might explain the clipping, you can't see it on your meters.

Another possibility is that you have your individual channels slightly too high. Remember a mixing console is nothing more than an analog addition machine. You're adding all the channels together, so if the individual levels are too high, the sum of them all together will be above the headroom limit of your mixer.

On analog mixing consoles you could bump 0db on the meters and even go above that just a tad , because of the nature of analog tubes and eventually analog semiconductors. Digital mixing consoles are much less forgiving. Tap 0db and you hard clip, each and every time.

This is because the analog signal is converted to a digital numerical equivilent, and there are only so many numerical equivilents.

On a 24 bit mixing console, the audio signal is converted into a numerical value, from 0 to 16,777,216. Since audio has both positive and negative voltages, this numerical range is usually expressed as - 8,388,607 to 8,388,608, half below zero to represent negative portions of the waveform, half above zero to represent positive portions of the waveform.

However, when the digital circuits sample the analog waveform, it equates say, +15V as 8,388,608, and -15V as -8,388,607. If your signal is 15.1V, even briefly, the digital circuit will record the numerical value of 8,388,608, because that is as large a number as it can store. Same thing if you apply +17V, you get the same value because its the largest. Thats what causes digital clipping.

Digital is very unforgiving if you exceed the design of the A/D converters. That's why earlier I mentioned the -.1db marker as a good point to mix to, instead of 0db. Gives you just a few bits of margin whereas you can avoid clipping a little easier.

Hope that helps.

Frederic
 
Last edited:
well, I have a m audio delta 1010 8 channel sound card. I use a mackie 1604VLZ mixer to plug all my mics into.

I noticed that it only really happens with drum tracks and not guitar or vocal tracks. Maybe it's just the way it is I guess...
 
Someone posted in the mp3 forum and had the same problem only more severe. The positive half of the waveform was severly clipped but the negative half was not. I bet it's a software plugin that people are using screwing shit up.
 
frederic said:
Another possibility is that you have your individual channels slightly too high. Remember a mixing console is nothing more than an analog addition machine. You're adding all the channels together, so if the individual levels are too high, the sum of them all together will be above the headroom limit of your mixer.


What fredric is talking about is bus headroom. Bruce (Blue Bear) has a very good series of articles on his website about mixing, and he goes into concerns about bus headroom, and talks specifically about Mackie mixers, if you want to learn more about the concept.


Light

"Cowards can never be moral."
M.K. Gandhi
 
Ok well I was read his article. Heh, ok, so my mackie 1604VLZ mixer has the main mix meters. I know it goes from -30 to 0 Db. 0dB is the most it can go right? But it still has more it goes all the way up to 28 dB until it hits the red. So does this mean that 0dB on the mixer isnt the highest level?
 
I gotta say, I'm been impressed with how Pico is taking this criticism. Good attitude, man. With that going for you, you'll only get better and better at this.



Good luck dude.

Nice sig too, btw.
 
picostudios said:
Ok well I was read his article. Heh, ok, so my mackie 1604VLZ mixer has the main mix meters. I know it goes from -30 to 0 Db. 0dB is the most it can go right? But it still has more it goes all the way up to 28 dB until it hits the red. So does this mean that 0dB on the mixer isnt the highest level?

"Zero dB" on the Mackie is UNITY. Nothing being taken away from the signal, and nothing added.

The "speed limit" if you will of digital is indeed 0 dB. Set all your channels in the Delta to 0dB and what have you got on your main stereo buss in the Delta? CLIPPING!!!!!!!

The 0-to- +28dB range on the Mackie mains are your headroom. Try an experiment. Set your levels on 8 channels of the Mackie to unity. So when you use the "level set" function each channel is peaking at 0dB.

Now what do those mains read when you play those 8 tracks simultaneously???
 
picostudios said:
Ok well I was read his article. Heh, ok, so my mackie 1604VLZ mixer has the main mix meters. I know it goes from -30 to 0 Db. 0dB is the most it can go right? But it still has more it goes all the way up to 28 dB until it hits the red. So does this mean that 0dB on the mixer isnt the highest level?


There are many different scales in audio, and they are all measured in dB. Digital audio is measured in dBFS (decibels Full Scale), in which 0 dBFS represents all bits being used (for a 16 bit signal, 1111 1111 1111 1111). So you can not, in dBFS go above O dBFS.

Then there is dB SPL which is a measure of actual acoustic volume, in which 0 dB SPL is silence, and you can go just about infinitely high, though at about 128 dB SPL your ears start to bleed (seriously, not metaphorically).

Then there are a couple of different scales used for analog. The basic difference is what they use as their reference voltage, but in all of them, 0 dB is unity gain, or nothing added, nothing removed. Your Mackie has a little legend above the meters which says 0 dB = 0 dBu, which is equal to 1mW at 600 ohms (you don't need to know this, and I had to look it up in order to write that sentence). It is just important to notice that there are different references, and each one has a different label. The other common one is dBV.

As a teacher of mine once said, the great thing about standards is that there are so many of them.

At any rate, the meters on your Mackie are referencing something different from what your DAW is talking about. One of the best books you will ever read on the technical subject in audio is Yamaha's "Sound Reinforcement Handbook." Whether or not you ever want to do live sound, I highly recommend that you buy a copy of this book, and read every word in it. You will, I think, learn a lot.


Light

"Cowards can never be moral."
M.K. Gandhi
 
UH Blue Bear is right. If you can't even keep fundemental problems such as clipping out of your recordings, you're not doing a "damn good job" and you shouldn't be charging people $10 an hour to do things wrong. Recording engineers are hired mainly for their ears, knowledge, and skills - not the equipment that they own. I'd pay a real engineer to record me on a 4-track before I'd pay some guy who is suddenly a self proclaimed recording engineer because he bought a mixer and Cool Edit.

Also, your website is like totally butts. Not compatible with all browsers?! Non-resizable pop-up windows for every sub-section?! Why the hell would you do that?
 
The Seifer said:
Also, your website is like totally butts. Not compatible with all browsers?! Non-resizable pop-up windows for every sub-section?! Why the hell would you do that?



Yea, man. You need to learn web design from Seifer. ;) :D
 
The Seifer said:
Also, your website is like totally butts. Not compatible with all browsers?! Non-resizable pop-up windows for every sub-section?! Why the hell would you do that?

Ok, we've already covered the website issue.

I just had some questions. I know about clipping. I was just seeing if there wa something I missed. I'm gettin real tired of this negative comment overload. I'll just go ahead and leave this thread already.
 
picostudios said:
On my M-Audio delta 1010 meters, I always make sure it doesnt clip when I record. As far as the mackie meters go, I always make sure it doesn't hit the "red".

Well, the 1604VLZ is an analog mixer, and the line inputs are rated to +22db before the channels clip, which indicates to me it has a 36V split power supply (+/-18V,), which after summing you can probably go to +3db or +6db on the mains without clipping.

But...

The Delta 1010 cannot. 0db is its mathematical maximum and looking at your mp3's, you're definately hitting it. See the printscreen above as evidence :)

I am going to take a wild guess here, and its truely wild. I'm thinking your 1010 meters are just "off".

When I bought a few tascam digital mixers, I had the same problem. Clipping here, clipping there, but NEVER did the digital meters on the mixer light the 0db red LED indicating clipping.

Tascam, through an annoying menu, allows you to change the scale of the meter, so I "adjusted" it. Now when the red LED of the 8-bar meter flickers, thats -1db, not 0db, indicating to me that I'm not clipping, but darn close. I'm glad the tascam had that feature, but *I* come from the analog world... you know.... large format console with an analog meter per channel wired up to a pair of 24-track Otari's, also with individual meters. Analog board and analog tape sound "warmer" when you hit the 0db point and cross up just a little bit... so instead of adjusting my way of thinking, I simply adjusted the equipment because I got lucky in that my meters could be adjusted this way.

Maybe the 1010 can have the same adjustments?

If not, keep the mix a "hair" below 0db and all your clipping willl go away. But be gentle here because your ears hear things logarithmically rather than linearly, so every +3db doubles the volume, and every -3db cut, reduces the volume in half. Knock the levels down enough to reduce clipping, but thats it.

Now, to make things "loud" again, you need to do two things.

First, fill the audio spectrum 20-20Khz
Second, compress the final stereo mix.

The latter is straight forward, you and I have talked about it a few times, and I know you're playing with settings and getting the hang of of your compressor.

But lets talk about audio spectrum for a minute. You're ears can hear the range of 20hz to 20Khz, and are interesting in that they have little hairs inside that are different lengths, and vibrate at different frequencies, and that individual electrical signals generated are sent to your brain, and mushed together. Kinda like a huge, natural mixing console :)

If only a few hairs within your ear are stimulated, meaning that you're listening to one frequency, say, 5Khz, your brain will be confused and not necessarily hear the "volume" correctly. This happens to machinists and large equipment operators all the time, actually. The particular machine they operate generates a specific frequency of "noise", and after a while of operating that machine, they can't even hear it anymore. This is because the brain cuts it out, sorta a "limiting" function, to use an audio term.

The next time you mow your lawn, you'll notice how painfully loud the mower is when you first start mowing, and by the time your'e done mowing, its not that loud anymore. But when you shut it off, you'll probably either hear ringing in your ears, or a slight hissing for a short while. This is your brain trying to cancel out the noise your ears hear, by making internal noise. When the mower stops, your brain keeps going for a while. Neat, huh?

Anyway, do this experiment. Take the mp3 (mike, lance etc #1) and put it into winamp, and enable the boring spectrum analyzer, and play it all the way through, and watch the multi-colored bargraph. You'll see that you have audio data in the 40,60hz range, very little in the 80,120,160 hz range, a lot of midrange, and some highs. The highs might be limited because of the conversation from wave to mp3, so really look at the 80,120,160hz range and see that the little bars don't dance much.

Now load my mp3, and look at the spectrum. Notice that is "fuller". All of the spectrum bars move, and move quite a bit. This "fullness" adds to the volume in a musical way, and your brain perceives it as "louder". Part of this is careful placement of sounds on my part, part of it is the compression.

I'm sure you're sick of listening to my techno mp3, but I'm trying to help you here. Filling up the musical spectrum adds to the loudness perceived by the listener, and doesn't necessarily add to the clipping if the levels are watched, because its more of a "brain trick" than anything else.

BTW, this particular mp3 has been played in clubs (in its full length of course), whereas the raver's have felt the cartilage between their ribs pulsate with the bass drum. You'd have to agree thats "loud". And no clipping.
 
Ok, I am going to show you an mp3 from a local home studio here as well. Tell me what you think about it.



Give me your opinion on that. That is my competition. I know he totally whoops my ass, but he has better equipment that I do, and more years experience. Someday I'll get there.
 
picostudios said:
Ok, I am going to show you an mp3 from a local home studio here as well. Tell me what you think about it.



Give me your opinion on that. That is my competition. I know he totally whoops my ass, but he has better equipment that I do, and more years experience. Someday I'll get there.

Pico, can I be honest with you? The equipment you have is not awful by any means. I'm not a mackie fan because of the tonal aspects, but thats a subjective opinion not an indication that your mixer sucks. They offer excellent value for the money they ask, always have. In fact, in one pro studio I owned, our console in studio "A" was a Mackie 32x8 with four 24E sidecars. The 1010 is a nice unit too actually, I know several people who use them. In fact, next time I see one of them I'll ask them about the meter thing. Maybe we can shed some light on this.

And I don't mean to pound you so hard either, maybe its just my writing style, so please, don't ever take anything I say as a person slight towards you. Its not meant that way. I am genuinely trying to help.

Remember, I've been doing this a very long time, but also I do remember the day I tore the shrinkwrap off my first Tascam 244, my first mixer/recorder. I can absolutely assure you my first recordings were BY FAR hideous.

Yes, hideous. I just don't mention that often :-D

Anyway, in regards to the mp3 you posted... across the 3:54 minute recording, there are 4 points of clipping. Very minor, I actually only heard one clip with my ears, the rest I saw on my scope.

I also didn't really hear the kick drum consistantly throughout the recording, it got buried from time to time, but that might be more about style, and was deliberate. This type of music is not my forte, I'll be straight forward about that.

And while loud and one can hear slight amounts of overall compression, the frequency spectrum isn't as full as it could be. There a small hole in the 160hz range. The harmonics of the rhythm guitar slightly overlap with the cymbals, I think I might have done something about that if possible. I did like how crisp the snare/toms were recorded - absolutely no "ring tails".
 
Hey frederic, could you please give the same straight foward commentary on this recording I made in my bedroom:



I'm learning to mix as well and my use of eq and compression is something that nobody comments on in the clinic, plus I'm interested to know if there is clipping or distortion that my ears can't hear yet.
 
Wow, I think it sounds good. I like the drum machine. Needs some lyrics!
 
Back
Top