The Different EQ bands and What they mean (part 2)

Dyn3 7-band EQ (Avid Pro Tools free plugin)

If you look toward the bottom of the EQ pictured above, you will notice 5 different bands: 1. LF, low frequency, red; 2. LMF, low-mid frequency, orange; 3. MF, mid frequency, yellow; 4. HMF, high-mid frequency, green; 5. HF, high frequency, blue.

In today’s blog I will talk about these five bands. I want to start with band 1 and 5. These are typically used and referred to as “shelves.” Band 1, low frequencies, is the low shelf, and band 5, high frequencies, is the high shelf.

But these two bands each have two different settings. The small left icon, next to the LF and HF, is called a bell-type EQ. It kind of looks like -o-. This will either boost or cut a section of frequencies set by you with the frequency knob. The ‘Q’ knob will determine how wide or narrow the bell curve will be. A low Q setting will give you a wide band of frequencies, and a high Q will render a narrow band of frequencies. A good rule of thumb is wide when boosting and narrow when cutting.

The typical use for this is to, say, boost the lower frequencies to bring out a kick drum or synth bass. On the high end, with the HF knob, we can boost upper ‘air’ frequencies to make guitars or vocals stand out or sound brighter. Of course, we can also cut in these frequency ranges as well.

The other icon setting is called a ‘shelf.’ This is the more common use for these two bands. Typically we use a boost here (low or high). When boosted, it looks just like a “shelf.” If on the low shelf, we set the frequency knob to 125 Hz, then everything from 125 on down (to 20 Hz) is boosted the same amount. On the high shelf, we might add a shelf for vocals starting at 6 kHz. In this case everything from 6 k up will have a boost. Of course, we can also cut using a shelf, but this happens less often then a boost.

The Q factor is a bit more complicated and will have to be reserved for another post.

Bands 2, 3 and 4 allow for bell curve settings only. These are the same as the bell curves on bands 1 and 5. These are used for low-mid, mid, and high-mid frequencies. There are only three knobs: Frequency, Gain and Q. Frequency, of course, sets the frequency that you want to work with. Gain is volume (loudness) and can be plus (positive) or minus (negative). We might say boost 2 kHz 2 dB (2 dB) which is a positive gain. Or cut 1200 Hz 3 dB (-3 dB) which would be a negative gain.

As stated above, Q determines the amount of frequencies being altered by the EQ.

As always, I hope this helps!

And, HEY! Make it a Great day!

Tim

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Compression Dos & Don’ts

To wrap things up regarding compressors, I will offer 3 Dos and Don’ts as my final word for now. These are things to always keep in mind when working with compressors. Some may have been previously stated in an earlier blog post.

DO          Avoid using extreme settings to begin with, if you are just trying to control the dynamics.

DON’T   Add compression to every channel by default. Start off with minimal compression, and carefully choose where to add compressors.

DO         Experiment with different types of compressors – hardware and software. There can be differences in how they sound. Compressors can and do sometimes sound different from one another.

DON’T  Forget to bypass the compressor occasionally while setting to check the results.

DO         Remember to balance the output gain so the level doesn’t change when engaged and bypassed. This way you can accurately compare before and after. Also, typically compression is added AFTER the mix has been balanced. So you don’t want to alter levels with either compression or EQ.

DON’T  Be afraid to experiment. Some of the greatest sounds in the history of recorded music came from misused and abused compressors.

Compressors – What is the Knee and What does it do?

As you get better with compressors, you will start playing with other knobs and features. One of these is the knee. The knee refers to when and how the ratio starts to change when the compressor starts to take effect. A ‘hard knee’ means the compression becomes immediately active as soon as the input signal hits the threshold. A ‘soft knee’ means the compression becomes audible more gradually. A ‘soft knee’ also means that gentle compression starts happening further below the threshold. Another way to say this is it starts acting before the signal actuall reaches the threshold setting.

Both hard- and soft-knee compression have their uses; two examples: if you want to squash a signal’s transients quickly, you’ll want hard knee compression. If you want to use a compressor to gently glue a mix together by tightening up transients, you’ll want a soft-knee compressor.

Lastly, if you have a compressor, like the Dyn3 Compressor/limiter which comes free with Pro Tools, look at the picture of the knee. It actually looks like a human knee!

As always – I hope this helps!

And…. HEY! Make it a great day!

Tim

Creating a Click Track (in Pro Tools)

Using a click track during recording is, of course, imperative. We can’t do our work if we can’t play to the beat! Luckily for us Pro Tools makes it easy to set up a click track! Just simply go to Track on the menu bar, scroll all the way to the bottom and choose Create Click Track. Pro Tools will create an auxiliary track and automatically put a metronome plugin on the track. The metronome will automatically set to the tempo (bpm) of the song.

The metronome is customizable. You can change the sounds used for beat 1 and all other beats (2, 3, 4, etc.). The volume for beat 1 can be set and the volume for all other beats can be set to something different. I usually have mine set so that beat 1 is louder than other beats and is a different sound. That makes it easy to find the downbeat while tracking.


When I set up my templates, I already have the click track set up and ready to go! Easy! And since I organize and colorize my tracks, for me, the click track is all the way to the left in the Mix window and a bright lime green. I always know where it is in the session, no matter working in the mix or edit window.

Lastly, you can save a preset of the type of click you like. On the click plug-in, select the drop-down arrow next to Preset, select Save As, and name it! That’s it!

Peace! And HEY! Make it a great day!

Tim

6 Recording Myths – Busted!

It is hard to learn how to record and mix music today. With so much information available on the web, sometimes it is hard to know if the information is true or not – whether it can be trusted or not. Here are six myths that are not true! Ask anyone who really knows his stuff and is experienced and successful.

Myth 1 – You can’t use ribbon mics on loud sources

This myth is a good one to start with because like the best myths, there’s just enough of a grain of truth to it to keep it going. It’s true that the actual ribbon element can be more fragile than the diaphragm of a moving coil or condenser microphone. It’s also true that in the early days of ribbon mics, those classic RCA mics from the 1940s would fail readily if you tried to use them on a screaming guitar amp or a kick drum. However, that hasn’t been true for decades. These days, arguably the most venerated guitar cabinet mic, the Royer R-121,  is a ribbon mic. Ribbon mics these days can easily withstand extremely high Sound Pressure Levels (SPL) and can be used on any source. Some ribbon mics such as the Shure KSM313/NE utilize a ribbon made of Roswellite, a substance created using carbon nanofilm technology that is virtually unbreakable and can endure levels up to 146dB SPL.

Myth 2 – Always record as hot as you can

This is another myth that has roots in the early days of recording to tape. Back when your recordings had to stay above the noise floor of the tape, tracking too quietly could render your recording noisy and unusable. Not only that, but recording engineers realized that for rock music, slamming your recording levels produced a very pleasing tape compression and “heat” that could make things sound great. With digital recording, however, both of these are no longer true. With 144dB of dynamic range (24-bit recording) you can even record at -40dB and have 100dB of dynamic range. Early analog-to-digital converters (from decades ago) did sound better when recording near the top of their range but that is no longer the case. In fact, with digital recording, overloading your recording levels is decidedly unpleasant, resulting in a digital distortion when clipping that is ugly and abrasive.

Myth 3 – External digital clocking improves the sound of your audio interface

If you’re interconnecting a lot of digital gear you may want to use a master digital clock. Get the best clock you can afford, and make sure everything is connected properly via Word Clock cables. In many cases, the master clock won’t have a drastic influence on the sound; the uniform clocking simply makes everything work together without digital pops and ticks. Just taking your audio interface and hooking it up to an external clock isn’t going to improve the sound quality of its digital-to-analog and analog-to-digital converters unless the clock in your interface is really poor. If you really want to improve your recorded sound, get the best mics, preamps, and audio interface you can. Only buy an external digital clock after you’ve made sure the rest of your audio chain is the best it can be.

Myth 4 – Egg cartons or mattress foam are good acoustic treatments

No, not even close! And despite what you may read on the internet, they don’t sound-proof anything. Materials such as drywall, insulation, and acoustic foam can be great acoustic treatment materials. With these materials and proper construction and application methods, you can effectively tackle the two general aspects of studio construction: isolation and acoustics. First, if you’re concerned with keeping sound from getting in or out of your recording space, you’ll need to tackle isolation. This is best done with some form of mass-air-mass construction. A wall with drywall and insulation, empty space, then another identical wall with drywall and insulation will provide a great start. For controlling the acoustics inside your space, you’ll need a combination of absorption and diffusion. There are myriad ways and a long list of proper materials to implement this — egg cartons and mattress foam are NOT on the list!

Myth 5 – External hardware always sounds better than digital plug-ins

In the early days of digital, this may have been true, but definitely not today. Sure, there are hardware compressors, equalizers, and effects processors with a certain mojo that sound amazing. But there are also digital software processors that sound incredible and offer a level of precision and recall that you’ll never get with external hardware. There’s a reason that nearly every pro studio has a ton of high-quality plug-ins even if they already have and use great outboard gear. You may like the sound of a piece of hardware, but you may like, or even prefer, the sound of a digital processor. The days of digital being second best are far behind us.

Myth 6 – There’s a “correct” way to record

It might seem counter-intuitive after all these “wrong” myths to proclaim that there’s no “right” way. But it’s true! One way of doing things may not get you the results you’re after, but then there are multiple ways that will. The name of the game is experimentation! Never stop experimenting and searching to find techniques that work for you, your music, your musicians, your studio. If you wonder if something will work, even if it seems patently false, give it a go! At worst you’ll need to redo it. At best you may add another unique tool to your toolbox. And that’s what recording is all about!

These are truths that all of us can learn from. I hope this helps musicians and engineers alike get better at their craft!

Peace – and HEY! make it a great day!

T

Calculating File Sizes (How much hard drive space does it take to record a song?)

So . . .  you want to record a song and you’re running out of space on the computer or external hard drive? Wondering if you have enough room? Here’s how to figure out if you do have enough space:

The sample rate and bit depth of the audio you record are directly related to the size of the resulting files. In fact, you can calculate file sizes using these two parameters:

— Sample Rate x Bit Depth = Bits per second

Or, stated another way:

— Sample Rate x Bit Depth x 60 = Bits per minute

In the binary world of computers, 8 bits make a byte; 1, 024 bytes make a kilobyte (KB); and 1,024 KB make a megabyte (MB). Therefore, this equation can be restated as follows:

— (Sample Rate x Bit Depth x 60) / (8 bits per byte x 1,024 bytes per kilobyte x 1, 024 kilobytes per —  megabyte) = Megabytes (MB) per Minute

Reducing terms gives us the following:

— Sample Rate x Bit Depth / 139, 810 = MB per Minute

A lot of folks are recording these days at 44.1/ 24. That’s a sample rate of 44,100 with a bit depth of 24 bits. Here is the calculation:

— 44,100 x 24 / 139,810 = 7.57 MB per minute.

Here is a basic chart of different sample rates and bit depths:

44.1/16 bit  =  5.04 MB/minute
44.1/24 bit  =  7.57 MB/minute
48/  16 bit   =  5.49 MB/minute
48/  24 bit   =  8.24 MB/minute
88.2/16 bit  = 10.09 MB/minute
88.2/24 bit  = 15.14 MB/minute
96/  16 bit   = 10.99 MB/minute
96/  24 bit   = 16.48 MB/minute

If you figure a normal song of 3 1/2 minutes recorded at 44.1 sample rate and 24 bit, you can plan on it taking roughly 26.50 MB of disk space. I am starting to run a lot of my sessions now at 96/24 bit. So a 3 1/2 minute song is costing me 57.68 MB of hard drive space per song.

Considering that terabyte hard drives are now running close to $50 these days, all this math stuff is not nearly as important as it was just a few years ago. But I know a lot of guys who still aren’t purchasing a whole lot of TB hard drives! It’s still useful information if it’s needed in a crunch!

Hope this helps!
HEY!! Make it a great day!!

T

Organization Pt. 3 (patchbays)

Patchbays

Having a patchbay helps to optimize your signal routing and organization. Even the most modest of studio setups can benefit from the simple addition of a patchbay. Almost any configuration of cable connection – xlr, 1/4″ TRS jack, Cat5, etc. can be connected to a patchbay. And you can make your connections without ever having to leave your seat!

If all your equipment I/O (input/output) is connected to the patchbay and it is labeled well, it will save you time by not having to go around the back of your gear to connect things. It will save wear and tear on the connections of equipment. It also centralizes the grounding of gear and reduces potential ground loop noise problems.

I use two main types of patchbays in my studio: xlr and 1/4″. The [Hosa] xlr patchbay is configurable. That is, the two types of xlr – male and female – can be configured to fit one’s needs. You can have the front panel all xlr male, all xlr female, or a mixture. Conversely, you can set the back of the patchbay as well.

The [Neutrik] 1/4″ patchbay has a two-row topology and is typically set up with an out-over-in signal flow, or a downward signal flow direction. For instance, outputs on the top row and corresponding inputs below. Or, returns on top and sends on bottom. TRS patch panels have configurable setups called “normalled” and “half-normalled.” In patchbays, a normalled configuration is an internal connection from the top row of jacks, to the bottom row. Normalling allows connections that are normally in effect to exist without the need for inserting a patch cable in the front of the bay. For example, the stereo outs of a mixer are generally connected to the inputs on a stereo mixdown deck. By connecting the mixer’s outputs to the top back row of a normalled patchbay’s jacks, and the mixdown deck to the bottom back row, a connection is made internally in the bay, and does not require extra patch cables. 

When a jack is inserted into the lower plug, however, the normalled connection is broken. This provides a convenient way to route signals to multiple destinations. For example, the output of a mixer that is normalled to the input of a DAT on the patch bay can also be simultaneously routed to another patch point. To do this one would simply run a patch cable from the patch point that is the output of the mixer (an upper jack) to the patch point that is the input to the other device (a lower jack). This connection will break the normal of what would normally be feeding that other device in favor of the mixer signal that has been patched in. Signal will now be routed to the DAT and the other device. Another application might be to insert an EQ after a preamp but before the converters. Simply route the output of the EQ (an upper jack) to the normal input jack (a lower jack) for that mic preamp.

As a final note – label, label, label!! In a previous post I mentioned owning a digital label maker. This is when it comes in handy! Also, write all the routing options down on paper first to help figure out organization.

Having a patchbay will simplify your studio life and make routing an easy task!

Peace –

and HEY! make it a great day!!

Tim

Organization (in your Production Studio) Pt. 2

Group Cabling

Keep cables grouped together by type (audio, MIDI, mains, etc.). This makes it easier to find faulty cables, patch equipment quicker and easier, and help reduce cable-borne noise problems. Putting a little distance between different types will reduce the potential for EMI (electromagnetic interference) problems, because mains cables can induce a 50Hz hum on audio cables. Always try to keep power cords of any kind separate from audio cables. If they do have to cross, try to do it at a 90° angle to minimize hum induction.

Cable Wraps

Using cable ties of some sort are definitely in order! Use Velcro or easy-release plastic ties. DO NOT use the infamous rock-n-roll duct tape! It will leave sticky residue once removed!

Label

Label Everything! I used to use the little string tags (and if I was to be honest, there are still some being used). But I did finally invest in a digital label maker. Good Investment! I now have 2 different tape sizes and label everything – patch bays, monitor control, hard drives, cables, wires, boxes, etc. You get the point!

‘Faulty’ Box

Get a large plastic box and label it “Faulty”, “Not Working”, “Needs Repair”, “Bad” or something like this. You will end up with equipment that needs repair, faulty wires or cables, stomp boxes not working correctly, or just components that you can use later for recycling of parts. Almost everything in our studio can be repaired or used for parts. This helps keep unusable equipment from getting mixed up with the good. How many times have you reached for that cable that doesn’t work or needs a wiggle when you use it! Throw it in the faulty box! When you have some time, get into the box and pick a project! Or better yet, sell the lot on eBay under ‘needing repair’ and get them out of your hair.

I now have initiated all of this in my own studio. I am much better organized, I have better workflow, waste less time and can focus more on my tracking or mixing. I hope this helps you too!

As always-
Make it a great day!

Tim

Timebase in Pro Tools

In Pro Tools, material (audio or midi) on a track is associated with a type of Time Scale. All track types can be set to either sample-based (for the Sample Time Scale) or tick-based (for the Bars/Beats Time Scale). Different tracks can be set to different timebases as needed.

Audio tracks are sample-based by default. This means that audio clips have absolute locations on the timeline and are tied to specific sample locations. If you change the tempo or meter the audio will not move. This is helpful, for instance, if you import an audio clip and want to build other audio or midi tracks around it and end up changing tempos or meters a few times. You don’t want to affect the original clip. 

However, MIDI and instrument tracks are tick-based by default. This means that midi clips are fixed to bar and beat positions and move relative to the sample timeline when tempo and meter changes. So if you change the tempo, the midi will either speed up or slow down accordingly. 

A good tip to keep in mind is Elastic Audio-enabled tracks can be switched to tick-based in order to automatically follow tempo changes in your session and conform to the session’s tempo map. 

And lastly, you select whether a track is sample-based or tick-based when you create it, but you can change timebases later as needed. 

Hope this helps!
Peace – and HEY make it a great day!
Tim

Powering Up Your Music Production System in the Proper Order

Did you know that it is important to power up your system and equipment in a certain order? In the early days, for me, I didn’t know that!

Because systems are typically composed of both hardware and software, preparing your system for use might involve more than simply turning your computer on and launching your DAW of choice. The larger the system, the more important it becomes to follow a specific startup sequence. Starting components out of sequence could cause a component to not be recognized, prevent the software from launching, or cause unexpected behavior.

The recommended sequence is as follows:

1. Make sure all your equipment, including the computer, is off.
2. Turn on any external hard drives that use external power (wait about 10 seconds for them to spin up to speed).
3. Turn on any MIDI interfaces and MIDI devices (including any MIDI control surfaces) and synchronization peripherals.
4. Turn on your audio interface. Wait at least 15 seconds for the audio interface to initialize.
5. Start your computer.
6. Turn on your audio monitoring system, if applicable.
   
   If your audio interface gets it’s power from the computer, it doesn’t need to be powered up in advance.
   
   That’s it! When you get in the habit of always starting your recording or mixing sessions this way, it will ensure that everything works properly as it should!
   
   Till next time – Peace!
   And, HEY, make it a great day!
   
   T