Decibel Types (dB's), Metering, and Standard Operating Levels for Recording Studio Equipment

Meters, meters everywhere, and dB this, dB that, dBxyz......but what does it all mean? 0 dBVU = +4dBu = -16dBFS....huh?  Sure you can record a great album without fully understanding all this stuff, but you are more likely to keep yourself out of trouble while hooking equipment up and make things sound better if you have a deeper knowledge of these things.

NOT JUST ANOTHER dB - DECIBEL TYPES
First, it is important to understand that the term 'dB' just by itself doesn't mean a whole lot.  Even if I say '47 dB' there is still a lot of different things that could mean.  It is essential to understand that the decibel (dB) is just a way to express a ratio between two different values.  The ratio is a logarithmic scale so that we can discuss the difference between really, really small values, and really, really large values quickly, easily, and with manageable numbers.  However, we always have to have a reference value as our starting point for comparison.  So, to be specific and have a better idea what we are talking about, dB should be followed by the the qualifier to tell us what we are referencing to, ie. dBu, dBSPL, dBFS, etc.

The use of the decibel is not exclusive to audio production.  There are all sorts of other disciplines that use the decibel such as optics and video, radar, and other areas of physics that use the concept of decibels referenced to all sorts of things.....dBZ, dBsm, dBd, etc.  I am going to talk about the few dB scales that I believe are most important to have a basic understanding of for working in a recording studio.....dBu, dBV, dBFS, dBVU, dBSPL, and dBm.

Common types of dB seen in audio production. 
  
dBu (u means it is referenced to .775 volts)  +4 dBu is the standard operating level for professional audio equipment.

dBV (V means it is referenced to 1 volt)  -10 dBV is the standard operating level of consumer equipment such as home stereo receivers, cd, tape, dvd, tape players etc.

It is worth noting that -10 dBV and +4 dBu are NOT 14 decibels apart! This because they are referenced to a different value.

I want to take a moment here to talk a little more about the idea of standard operating level.  If we want to be able to hook up all sorts of different devices together, there needs to be some expectation that these devices will be able to interact with each other with similar input and output levels.  If this were not true then an output from one device might be too much and distort the next device in the signal chain, or be too little and have a really weak noisy signal.  First we will look at the example of the classic home stereo component system below with a FM receiver, tape player, CD, equalizer, etc.  They are all connected together using RCA cables, and each device has output and input levels of -10 dBV so that everyone gets along nicely.

The same idea holds true for studio recording equipment.  If I want to go from the output of a preamp to the input of an eq, then into a compressor, and finally into an A-to-D convertor, each of these devices need to be designed so that they interconnect and expect roughly the same voltage levels.  For professional equipment, however, the level is +4 dBu.  Most high-end equipment operates solely at +4dBu, some more mid-grade prosumer equipment is switchable between +4 dBu and -10dBV for project studio-type setups that might have to interact with both.  See below.

dBVU (VU stand for Volume Unit) This is used with VU meters like the one pictured below.  The most common use of a VU meter is to set input or output levels.  In this situation, the needle starts all the way to the left, and as you increase signal the needle moves to the right.  Once the needle is at 0 VU that means the device is getting an input level (or output depending setting) that is right at its designated standard operating level for that device (sometimes this referred to as unity gain). So if you set the output of a channel strip so the VU meter reads 0 VU, you have optimally set the level, and that corresponding voltage at the outputs would be +4dBu.   

dBm (m means it is referenced to 1 milliwatt) Because the reference level is in watts, it means we are dealing with units of power.  This is used often times in conjunction with amplifiers, so is more common to have to work with dBm in live sound situations as opposed to studio.  

dBSPL (SPL is referenced to the threshold of hearing) SPL stands for Sound Pressure Level. Again, this is used more in live sound situations than in the studio.  This has to do with actual acoustic wave pressure in air.  When you go to a rock concert it is probably about 120 dBSPL. Conversational speech or normal office background noise would be about 50-60 dBSPL. Your ears are flattest (from a frequency response perspective) at 85 dBSPL, so that is a good volume to mix at so long as you are taking frequent breaks.  

dBFS (FS stands for Decibels Full Scale) dBFS is used with digital systems.  You will often see peak program meters (PPM meter) that use dBFS in digital audio workstations on or digital pieces of gear like analog to digital convertors.   

Note that at the very top of the meter is 0 dB, and all the values below 0 are negative.  This is because in digital systems there is an absolute digital ceiling at which sounds can't get any louder.  For example, if we were working with 8 bit digital audio the loudest sound we can represent is 11111111.  All bits are turned on and there is no way to represent a sound any louder than this.  Any attempt to reproduce a sound louder than this will be digital clipping which is NOT pleasant sounding.  11111111 corresponds to 0 dBFS.  We start at the loudest possible sound, and count down from there using negatives in most digital systems.  

Another important concept with PPM meters is that they are very quick and responsive.  This is important in a digital system because we need to know, even if only for a brief moment, if we clip our signal.  So with a snare hit the meter will do a very short fast spike that follows the transient information of the snare hit and then drop off right away.  The responsiveness of a PPM meter is very different to that of a VU meter which is much slower, and does not react to sharp transient information as well.  VU meters give more of an average level of the signal over time, which more closely corresponds to how are ears work.  Both types of meters have their time and place.  This characteristic of a meter's responsiveness is often called ballistics.  

THE 'BLACKBOX' MODEL FOR LEVELS IN AN AUDIO DEVICE

Before we look more closely at how these different ideas are incorporated in to a system, I think it is important to understand some basic concepts and terminology that are fairly universal to all audio equipment.  

We have already established the idea of standard operating level as sort of the ideal nominal level that each piece of gear is expecting to see.  ('A' in the diagram below) All gear obviously has an upper ceiling at which point if signals get any louder we end up with distortion and ultimately 'clipping'.  ('B' in the diagram below) (Clipping = BAD!) On the opposite side to clipping, there is a level at which if we turn things down enough the inherent noise level of the of the electronics in the device are actually louder than the signal we are trying to process.  This is called the 'noise floor'.  ('C' in the diagram below)  The distance from the standard operating level until we reach clipping is called 'headroom'. ('D' in the diagram below)  The distance from the standard operating level down to the noise floor is call the 'signal-to-noise' ratio.  ('E' in the diagram below) 

It is significant to note here that the standard operating level is the same for all equipment designed to work in a studio.  The elements that are likely to vary are the headroom and signal-to-noise ratio.  As a general rule, higher quality professional equipment has a greater headroom before clipping, and a lower noise floor which also offers a greater signal to noise ratio.     

HOW DIFFERENT dB SCALES INTERACT IN A SYSTEM

If we set the level on our preamp so that the VU meter reads 0VU that means that we have optimized our signal level in the device, and from a voltage standpoint there is +4dBu at the output.  Now the next piece of equipment in the signal chain will receive that signal with plenty of headroom, and a strong signal far from the noise floor.  We can go from device to device with this being true, until we get to an A-to-D convertor.  At which point the input level is still an analog voltage of +4 dBu; however, once we start to be concerned with the audio signal as digital information inside that box instead of an analog voltage, that same level now corresponds to -16dBFS.  Remember that digital systems count down from 0 dBFS with 0 being the highest possible value.  So this means that there is typically 16 decibels of headroom from standard operating level in a piece of digital gear. Hopefully this image below will help clear a few things up. 

   For the most part you can just remember for professional studio equipment standard operating level is +4dBu = 0 dBVU = -16dBFS.  :)

Everything you wanted to know about analog & digital audio recording cables/connections.

Knowing how to hook things up is the first step to being able to use it.  Here is an overview of most of the cables you are going to run into in the studio and other general audio cables.

ANALOG CONNECTIONS

Balanced vs. Unbalanced

Unbalanced Connections - carry a positive signal and a ground, sometimes referred to as a shield.  Unbalanced connections are much more susceptible to noise and interference from the external environment than balanced connections, but the circuitry is much cheaper and simpler to build than that in balanced systems.

Balanced Connections - carry a positive signal (hot), an inverted negative signal (cold), and a ground (shield).  When a piece of equipment has balanced inputs/outputs it inverts the negative signal so that any noise picked up by the cable from long cable runs, other electronic equipment in the area, lights, etc., will be cancelled out when the negative signal is re-inverted at the receiving device.

Connection Types

XLR
Balanced connection
Use to primarily to connect microphones to mic preamps / mixers / interfaces. Also used as an interconnect for professional level equipment such as compressors, eqs, preamps, effects devices, etc.

TRS
TRS = Tip/Ring/Sleeve
Balanced connection
Used mostly as an interconnect for mid to pro level equipment. (Preamp / compressor / eq / efx) The configuration is identical to an XLR cable.....positive, negative, ground.

D-sub / DB-25

Balanced connection

Carries 8 separate balanced connections over one cable with 25 pins.  Used to connect professional multichannel equipment such as recording consoles, patchbays, AD/DA convertors, ProTools HD interfaces, etc.

Often there is a DB25 on one side and 8 XLR or TRS on the other.  This is to connect to a patchbay and then breakout to all the individual equipment.

Bantam / TT (Patch cables)
Balanced connection
TRS - smaller than 1/4" larger than 1/8" headphone jack.
Used on professional studio patchbays.

RCA
Unbalanced connection
Used mostly to connect consumer equipment such as home stereo receivers, CD / tape / record players.  Consumer equipment functions at at a lower 'standard operating level' than professional recording equipment. (-10dBv for consumer equipment as opposed to +4dBu for professional equipment.)

1/4 TS (Instrument cable)
TS = Tip/Sleeve
Unbalanced connection
Used mostly as an instrument cable.   Primarily to connect a guitar to an amplifier, keyboard to an amp or direct box, etc.
1/4 TS cables are also used as interconnects on entry/mid-level (or prosumer) equipment.  (Art, presonus, entry level focusrite, for example)

TS vs TRS cables up close.

Insert (Send and Receive Cable)
Unbalanced connection
An insert cable has a TRS connector on one end, and two TS connectors on the other end.  Insert cables are mainly used in conjunction with 'inserts' on mixers, but maybe some audio interfaces.  They are also referred to as send and receive cables because they send the signal out to the INPUT of a device, and then back from the OUTPUT of that device over one cable.  Don't be confused by the TRS on one side of the insert cable.  An insert cable carries two SEPARATE unbalanced signals, not a single balanced signal like a regular TRS cable. The TIP carries one signal to one of the TS plugs, and the RING carries another signal to the other TS plug.  Equipment that utilizes this type of cable often labels whether the tip or the ring is used for input or output.

Speakon
Used for live sound and stage use.  Connects the power amplifier to the speakers.  The connectors lock in place so there is no worry of them coming undone.  They can carry much higher current, and because the connections are covered there is no risk of getting shocked.  Speakon can actually have 2, 4, or 8 connections in the one cable, so you can run a biamplified system on one cable.




DIGITAL CONNECTIONS

Analog cables carry a continuous analog signal that represents the frequency and amplitude of the waveform, just like a microphone or analog tape.  Digital cables transfer binary encoded digital information of the audio. (1s and 0s)  Everything above a designated voltage is representative of a '1' and everything below a designated voltage is designated a '0'.

Digital Connections and 'Protocols' (Digital format types - meaning how the information is packaged and transferred.)

S/PDIF Coaxial
Connector: RCA
Carries 2 channels of digital audio over one cable. (stereo left/right)
Can transfer 44.1 or 48k sample rates (sometimes referred to as 1x rates).  It is possible to use regular analog RCA cables for spdif coaxial, but the impedance rating of the cable is different so should be avoided for long runs.  Short runs will not notice a difference.

The TOSLINK optical cable is used for the following 3 different digital protocols: S/PDIF Optical, S/MUX, and ADAT Lightpipe.







S/PDIF Optical
Connector: Toslink
Carries 2 channels of digital audio over one cable. (stereo left/right)
Usually limited to transfer 44.1 or 48k sample rates (1x rates)
Instead of a voltage representing the 1 or 0, a light beam does.  Light on = 1, light off = 0.

S/MUX
Connector: Toslink
Carries up to 4 channels of digital audio over one cable.
Can transfer 44.1/48k or 88.2/96k sample rates (1x or 2x rates)
Often seen in multichannel preamps that have convertors and digital outs.

ADAT Lightpipe
Connector: Toslink
Carries up to 8 channels of digital audio over one cable.
Can transfer 44.1/48k sample rates (1x rates)
Often seen in multichannel preamps that have convertors and digital outs, interfaces, ad/da convertors, and digital consoles.

AES (AES3, AES/EBU)
Connector: XLR
Carries 2 channels of digital audio over one cable.
Can transfer 44.1/48k or 88.2/96k or 176.4 or 192k sample rates (1x, 2x, or 4x rates).  Professional digital interconnect for high resolution audio.  It is possible to use regular mic cables for AES, but the impedance rating of the cable is different so should be avoided for long runs.  Short runs will not notice a difference.

AES (AES3, AES/EBU)

Connector: DB-25

Carries 8 channels of digital audio over one cable.

The same info is transfered as the AES xlr variation above, the multiple pin connector just allows for more channels, similar to the analog example at the top of the page.

MADI Optical
Connector: SC type
Carries up to 64 channels on one cable
Can transfer 44.1/48k or 88.2/96k sample rates.  (1x or 2x rates)
More recent than the other digital formats, but becoming very prominent.

MADI Coaxial
Connector: BNC
Carries up to 64 channels on one cable
Can transfer 44.1/48k or 88.2/96k sample rates.  (1x or 2x rates)
More recent than the other digital formats, but becoming very prominent.  MADI optical is more common than MADI coaxial.

Word clock
Connector: BNC
Carries ZERO channels of audio!! Wordclock is used to transmit clocking information to all the different digital devices in a system.  There can only be one 'master clock' signal providing the digital clock pulse that all digital equipment references to ensure each sample pulse is locked between each of the devices.  All other digital gear not set as 'master' is 'slaved' to reference the incoming clock signal.  A consistent clock signal helps eliminate digital clicks and pops, and reduce jitter.

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Try not to be confused by the fact that several different types of connections are made with the same type of cable/connector in both the analog and digital world.  That doesn't mean you can hook up an analog XLR out to an AES digital in because they share a connector type, or a wordclock out to a MADI in, or a Lightpipe out to a S/PDIF in, etc.  It is just all the more reason you need to know all of these things exist, and when each is being used or needed.

Good luck!

Quick-Unique Sounds & Transitions

Can't figure out how to transition from one part of your song to the next? Or, maybe you just need a new unique sound to inspire your creativity.  Keep reading.....

Most of the time we do an audio crossfade between two regions in our DAW's we are probably just crossing a tiny little segment between the same sounds, and usually trying to make it as unnoticeable as possible.

But.... have you ever done a one or two measure long crossfade between two DRAMATICALLY, COMPLETELY different sounds?  It doesn't always give you magical result, but sometimes you can end up with some really fun, unexpected results to push your production to the next level.  ....and the best part is you are going to separate yourself from everyone else that just uses off-the-shelf preset sounds.

Below I have a screenshot and an audio example where I used this concept in an electronic song several years ago.  There are technically three big crossfades happening about the same time in this example: a lead line with a distorted verb, and then a choir sound fades in.  My example is electronic, but you could really use this in any genre if you get creative.

mp3   wma

One more note, if you use mostly software virtual instruments, bounce them out (render to audio) so that you get more interesting textures and sounds. 

So go try something! Get creative! Get wacky!  ....and, let me know if you come up with something cool. (Please feel free to keep it to yourself if it's not cool. :)

Reading Music on the Grid: Counting Ticks in Pro Tools & Logic

Reading music is a fundamental component of learning and performing on a musical instrument.  In the same way, if you want to edit music (well) you should understand how to read traditional music on a staff as well as on the grid in your editing software.  I believe knowing how to edit well is one of the most important skills you can develop to set your recordings and productions apart from the crowd, so this will be the first of many posts related to this topic.

The way that Pro Tools and Logic incorporate their grid system/ticks are different, so I will be going over both.

What are 'ticks'?
Ticks are the extra-small subdivisions within a quarter-note in a DAW.  This stems from the idea of ppq or ppqn (parts per quarter note) from the early days o f MIDI.  Initially, early MIDI gear recognized 24 ppq.  Nowadays, most DAW's by default are divided into 960 ppq, or ticks.  (At some point in the future I will do a post on tricks and benefits to setting up tracks to reference ticks vs. samples in Pro Tools.)

Reading Music on a Staff
Below is a fairly simple drum beat that I am going to use for the example showing how to read ticks in Pro Tools and Logic.  But first, a quick tutorial on how to read drum music in case you have never seen it before.

The top line is the high hat playing  1 & 2 & 3 & 4 &

The middle line is the snare playing 2, the A of 2, and 4

The bottom line is the kick playing  1, 3, the & of 3, and the E of 4.

(If you don't know how to read rhythms go find some basic youtube videos to teach you!! It's important!)

Reading Music on a MIDI Editor
Here is the same beat viewed on a MIDI editor (this Logic but others would look very similar.)

For the purposes of being easy to read, I made the kick red, the snare yellow, and the high hat green.  However, those colors actually correspond to the velocity of the notes being played...... red kick the loudest, yellow snare medium, and green high hat the quietest.  (It is common for MIDI programmed drums for the kick to be C1, the snare to be D2, High hat to be A#2.)

Reading Music on the Timeline
Here is the same beat on the Logic timeline.  You can see the heavy kick on 1, 3, &, E. The moderate amplitude snare on 2, A, 4.  And the light high hat on 1&2&3&4&.

If you look closely above at the audio region you can see the green arrow/cursor on the kick drum beat on the E of 4.  In Logic this cursor location is identified by 1 / 4 / 2 / 1 as shown below:

The first "1" indicates the measure.

The "4" indicates the fourth beat of the measure.

The "2" indicates the sub-beat. (sub-beat: 4, E, &, A)

The last "1" indicates ticks. (How far off the sub-beat, in this case right on the sub-beat "E".)

In Logic there are 240 ticks per sub-beat.  There are 4 sub-beats per quarter note.  240 times 4 = 960

If instead the cursor were on the kick drum hit on the & of 3, the cursor location in Logic would read 1/3/3/1.  Go back and look at the above example until that makes sense.  

So how does this differ in Pro Tools?

Below is the same beat in Pro Tools.  You can see the audio version right above the MIDI version.

The bars and beats counter in Pro Tools is divided into three sections as apposed to the four in Logic.  If we go back to our previous example to show where the counter is on the E of beat 4, in Pro Tools it would show 1 / 4 / 240 as shown below.

The "1" indicates the measure.

The "4" indicates the fourth beat of the measure.

The "240" indicates the ticks. 

Right on beat 4 would be 1/4/000

The E of beat 4 would be 1/4/240

The & of beat 4 would be 1/4/480

The A of beat 4 would be 1/4/720

Adding 1 tick to 1/4/959 would yield 2/1/000.

There are 960 tick per quarter note, just like in Logic, it is just implemented differently.  

Our other example from above, the kick hit on the & of 3, would be indicated by 1/3/480.

It is important to have a basic understanding of these concepts to do common editing and programing.  Once you get the idea you can do some simple math to figure out how to do 32nd notes, triplets, shuffle feels, etc.    

1st Post: The most important concept of all.

So, as I was trying to figure out where I wanted to start with all of this blogging business, I realized that instead of any technical mumbo-jumbo I would make my first post about something more important than all that.... a topic that I am more passionate about than any of that stuff......hard work. My students hear me rant about it constantly, music is probably the the most competitive of all disciplines, and in order to be useful to someone, to be legitimately good, be it a studio musician, live performer, producer, engineer, or whatever, it takes LOTS of hard work. Period. There are three different simple, and similar, ideas that have really motivated me (and others) over the years that I want to share here.

1. "Very good things can happen if you actually care about your life moving forward." 

This was something I heard someone say once that I wrote down, made it my desktop, and my screensaver. It was someone who I admired greatly, who was a very successful musician/producer, and had accomplished more before the age of 35 than I probably will in my whole life. Before meeting him I remember thinking "How did he accomplish all of that?", and this was one of the very first things I heard him say. I can't tell you how many hundreds of times, literally, that saying popped in my head and it got my off the couch to go practice drums or work in the studio. Now one of my most fundamental mantras in life is just "Keep moving forward!"

2. 10,000 Hours. Live it.
This has become a popular concept in the last several years. You have probably heard about this. All sorts of studies have been done that show regardless of the discipline, whether talking about creative or non creative fields, it generally takes about 10,000 hours to become an expert at most things. If you do some math that amounts to about 40 hours a week for 5 years, or 20 hours a week for 10 years, or 10 hours a week for 20 years. You get the idea. If you are not careful, and have an unhealthy affinity for video games, Netflix, your couch, or any number of other things, it could easily take your whole life to get to 10,000, if at all. (This is a big pet-peeve of mine......  Person X:"I want to be a producer." ME:"What did you do last weekend?" Person X:"I binge watched three series on Netflix."////  Person Y:"I want to be a studio musician." ME:"What did you do last weekend?" Person Y:"I beat Halo 3 again.") I've actually started advocating that my audio production students shoot for 20,000 hours to account for inflation (joke), and the fact that the field they want to work in is much more competitive than most (not a joke).

3. "It's amazing what can happen when preparation intersects with opportunity!"
This is another quote that I took from someone that I greatly respect and that has accomplished truly great things. He explained it as an idea of there being a line of preparation and a line of opportunity that run parallel to each other throughout our lives. These lines cross occasionally in everybody's life and it is import to do everything we can to prepare (practice, rehearse, etc) for those rare great opportunity's when they arrive. He got this idea from his mentor who also happened to be a successful and influencial person.  See a pattern?

As you can see these all boil down to the same thing. Work Hard! There is a very, very real and inseparable correlation between being excellent at something and the amount of time put into becoming excellent at it. Sometimes I hear "So-and-so is just a natural. yada, yada.." ... for the most part, though, I think people understand it takes hard work to be successful. Of course. However, here is where I think the HUGE break down is, DISCONNECTIA MASSIVIA, NO COMPRENDE......... Most of the population, the average person, those who are just aspiring instead of doing, have a hard time comprehending the actual amount of sacrifice it takes to become truly excellent.  The sacrifice of time first and foremost, but probably also friends, and family, and creature comforts, and all sorts of stuff really. The definition of "hard work" to someone who has put in the hours and is successful is very different than the definition of "hard work" to the average aspiring musician, engineer, or anything else. To illustrate this I recently did some math where I calculated out the amount of hours my typical students would have towards their 10,000 by the time they graduate with a 4-year degree if they only work on the projects that are assigned, and not work on projects of their own. That amounts to about 500-750 hours after 4 years. Some of them would perceive that as really "hard work" and may even think that they are going to be experts by the time they graduate, but as you can see this is still a LONG ways from that "magic" 10,000.  Do you wanna be an expert, and be ready when opportunity crosses your path? Do some math.  Calculate when you want to arrive at your goal, and how many hours a week you need to practice to get there.

This ended up up being way longer than I anticipated. If you are still reading this........WHY??????? Get up and go practice something productive!!!! Keep moving forward! WORK HARD!!!