Happy New Year from Sound Schematic!
Heres a shout out to my friends Bad Bad Meow, whose EP’s I recorded and mixed at Engine Studios back in June. If high energy acoustic punk is your scene, you will definitely dig these guys. The music will be released over the course of the next month. Keep up with them here:
Here is a quick sample of Beatle Bass feedback, recorded for an upcoming album by local Chicagoans Constantine and the Emperors. As you can tell, the physical performance is just as important as the sound.
Check back soon for the final product, and please excuse the low quality iPhone video.
Search this link on Youtube
<https://www.youtube.com/watch?v=5yphW2D1C30>
Claude Debussy – man of great flute and piano duets
Recently, I had the opportunity to record a flute and piano duet. This was my first foray into recording true classical music, and I would like to share my experiences.
For this session, the two performers were playing live in the same room. While I have experience recording pianos, I have never worked with a flute. There was definitely some trial and error in finding the proper mic placement for this instrument. Trying to get a healthy balance between the airiness and body of the flute proved to be quite a challenge. After going through various flute mic’ing techniques, I ended up with the microphone (Neumann u87) elevated two feet or so behind the performer, aiming at the center of the instrument. To me, this seemed to sound the most natural.
There are so many piano recording techniques. Since this session did not allow for extra time to experiment with the subtleties of various microphone placements, I went with something that was familiar. For the baby grand piano, I choose to use a 90 degree near coincident stereo pair (NOS) up top and a large diaphragm condenser under the piano, as close to the soundboard as possible. The result is a full and balanced piano sound. The under mic picks up a lot of sustain and adds more low end to the stereo pair. I used Neumann km-54s as the NOS stereo pair and a Neumann u87 underneath.
Signal Chain:
Flute –> u87 –> Hardy M1 –> ProTools
Piano (top) –> km-54 NOS stereo pair –> Trident 80B channels –> ProTools
Piano (bottom) –> u87 –> Hardy M1 –> ProTools
Overall, I would say I am very satisfied with the way things turned out. I think for the type of music being recorded, the final sound is fitting. The mic choices and placements seem to work well in a classical context. Since both performers were in the same room, there was a bit of instrument bleed in each microphone. In the setting of these live recordings, this actually turned out in a favorable mannor. Having the two instruments together helped add a bit of natural reverb and also gave each instrument a place in the room. After doing more research, I think I would like to try a ribbon mic on flute the next time I have the chance to work with one.
Out of the three pieces recorded, one was solo flute, one was a flute/piano duet, and the last was a duet for piano and bass flute. I used the same flute mic’ing technique outlined above on the bass flute. This was my first time ever hearing one of these – what a cool instrument! As far as I know, the bass flute is the same as a regular flute but scaled down one octave.
Here is an excerpt of one of the pieces – this one features the bass flute.
I recently did a session with the Ohio based group Manic Sandwich. After two long days of tracking, we had enough material for a six song EP. When time came to mix the project, the band was clear on one thing: they wanted as much slap back as they could get!
I messed around with various tape delay units and plugins, but ultimately decided this mix would be a great chance to print some real tape slap back, using a tape machine. Since this mix was going to be a console mix anyway, why not generate some delay out of the box as well? The mixing was done on a Trident 80B console, and I used an 8 track Otari mx5050 for the slap. I tried my best to document what I was doing for a quick “analog tape slap back tutorial” here!
This type of effect is commonly used on vocals. I did this on a few tracks, and was quite pleased with the results. However, I found one song where it worked really well on drums. This is the set up I will be reviewing here.
To start with, I bussed the snare, toms, over heads, and hi hat to an unused stereo output in protools. I decided not to include the kick drum, as the extra low end would muddy up the mix. On the patch bay, this stereo send was routed to the tape machine input. From the tape machine output, the signal was routed through a Vintech Dual 72 and into the line inputs of the Trident board. So, here we have the signal chain:
ProTools drum mix –> Otari MX 5050 –> Vintech Dual 72 –> Trident 80B –> ProTools stereo input
note: the Dual 72 was used for extra coloration/saturation of the drum signal from the tape. The Trident channels were used for EQ’ing.
The three heads: erase, record, and playback
On the typical professional tape recorder, there are three heads: erase, record, and playback. When recording a signal while monitoring off the playback head, there is a brief time difference between the actual recording and the listening. This is a result of the physical distance between the record and playback head. Generally, you wouldn’t notice this during normal tracking, only if you had a consistant playback source.
So when looking at my signal chain above, I used the first two inputs on the Otari to record the drums onto tape. As you can see in the picture below, these two channels were set to “Input”, and are record ready. However, I am monitoring off the playback (repro) head. By changing the tape speed, I am effectively changing the distance between the two heads. For example, a fast tape speed (15 ips) would result in a shorter delay effect, while at a slower tape speed (7.5 ips) the slap back effect would be more pronounced. This machine also has a varispeed control for fine tuning.
It is important to remember that this only works when you are actually recording the signal onto tape. Otherwise, there would be no delay – just a straight doubling of your input source. With this kind of set up, you are fairly limited in your echo time options. In some scenarios, using a plugin is much easier, because the proper tempo can be set right away. Using a tape machine requires more fine tuning and can be a bit of a headache. In my own experiences, I have had the best results using this effect on slower songs that are open, giving the effect room to breathe.
Although I am not sure who actually discovered this, I am of the opinion that tape delay was originally discovered on accident. Some engineer in the 50’s was messing around with a tape machine and realized that the delay effect produced sounds really cool. The rest, of course, is rock n roll history.
To check out the song I have been referring to in this tutorial, click the link below (this is a premastered version of the mix).
There have been a lot of outstanding music releases this year. What follows are a few of my favorites. In no particular order…
What can I say? Tom Waits is the man. Every song on this release is concise and strong – no filler material here. If you havent heard it yet you must.
Although slower and not as pop-oriented as her previous work, I really like the direction this is heading in. The recordings are pristine, and every song features an eclectic cast of instruments.
A mixture of free jazz, electronic manipulation, and killer musicianship. It is somewhat bizarre, but I cannot stop listening to it. “A modern classic of the future”.
This album hits hard. Although the subject matter is fairly dark throughout, the songs are extremely well produced and recorded. It is much more focused then the groups previous releases.
I am a big Ennio Morricone fan, so this is right up my alley. It’s great to hear spaghetti western stylings made more accessible.
A throw back to your great grandfathers dusty 78 collection, these tunes make me want to drink whiskey and dance the night away.
Happy holidays! Let me know if you agree/disagree with me on these choices.
When starting a session, its never a bad idea to be a bit over-prepared. Overcoming minor roadblocks and hitches, especially early on, will get things off to a good start. The following items have come in extremely handy for me numerous times (particularly when working at a studio that is unfamiliar).
1. Harddrive – Kind of obvious, but having a reliable backup drive is crucial. My professor in college always told us in regards to the digital domain “If it doesn’t exist in two places, it doesn’t exist at all”. Taken with a grain of salt, but you never know when the session drive is going to crap out or files get misplaced etc. Also comes in handy if you are mixing a project from home or working in different locations.
2. Pencil and Paper – Keeping good notes will help you out in the long run. Many artists will spout out ideas and have no idea what they said an hour later, so its good to have a reference to come back to. Jotting down mixing ideas, documenting external device settings, or just listing potential overdubs is stuff to stay on top of.
3. Headphones – any respectable studio will have decent headphones. However, its convenient to have an extra pair you are familiar with. This will be an advantage when you are doing critical EQing, auditioning microphones, or simply getting the headphone mix together for the band.
4. Drum Key – These are extremely easy to misplace. Instead of spending a half hour looking for it in the studio equipment room, keep one in your bag.
5. Guitar related items – Although technically the respobsibility of the performers, it is good to have extras. Tuner (with batteries), spare set of strings, 9v adapter and picks. Simple things like this can stall a session out of nowhere, so being able to replace them on the fly can save the day.
6. Flashlight – Comes in very useful when placing mics on guitar cabinets or crawling around behind consoles and outboard gear racks.
7. Blank CDs/DVDs – At the end of the session give the artist a rough mix of the days work.
Bonus – Laptop – This is not as important as the others, but can also come in handy. Possible uses are playing back mixes, auditioning sounds, and finding out where to eat lunch.
Seattle natives “Telekinesis” recently came through Chicago on tour. Part of their itinerary included a live in-store performance at Logan Square’s own “Saki Records”. I got the opportunity to help out with the live recording of this set. Cool!
Head on over to Epitonic to check out the set for yourself – Telekinesis Saki Session
For those interested, the set up was real simple. We simply took the stereo feed from the sound board and combined it with a wide stereo pair in the room. Blending the two together gives a convincing live sound: plenty of presence on all the instruments but enough room to put it in perspective.
Check out Saki records here: Saki
When used properly, compression can greatly add to the overall feel of a song. Recorded tracks are clearer, louder, and sit better within the context of the recording as a whole. However, misuse can result in distortion, lack of dynamic range, and an unexciting recording. Lets take a look at how a compressor works.
A compressor reduces input levels that exceed a selected threshold by a specified amount. This reduced dynamic range signal can then be boosted in level at the output, thereby allowing the softer signals to be raised above other program or background sounds. (Huber, Modern Recording Techniques)
Now that we got the textbook definition out of the way, lets look at this tool in real terms. As a dynamics processor, a compressor will “squeeze” together the dynamics (loudness/softness) of an inputed source. Basically, you are taking a sound and gently pushing it into a denser waveform. In effect, this will tame louder signals while making softer sections more audible. Any part of the signal entering the compressor above a certain level, or threshold, will be proportionately reduced to a lesser volume.
Since the loudest portions of the source material will now be turned down, it is possible to boost the entire level of the signal. In other words, since the dynamics have been reduced as a whole, the signal can now be amplified. The loud signals are still prominent, but the softer signals are much more present as well.
Most dynamic processors and compressors have a similar set of parameters. Here they are with a basic explanation of how to use the controls on a compressor:
Input Gain – how much signal is fed to the compressor
Threshold – this is the cut off level that initiates gain reduction. Signals that enter above this level will be attenuated according to the ratio, while signals that enter below will remain untouched. Some units may not have a threshold knob. With these, the input gain level controls the threshold – the higher the input gain, the lower the threshold level.
Ratio – the amount of attenuation. For example, a 3:1 ratio means that for every 3-dB of input signal over the threshold level, only 1-dB will be output. Generally speaking, ratios between 2:1 and 4:1 are considered “light” compression, while anything over 10:1 can be considered “heavy compression or “limiting”. A ratio of infinite:1 (“all buttons in” on an 1176) means only one 1-dB of signals above the threshold will pass to the output stage, no matter how high the input signal. Think of ratio as an input/output gain reduction ratio.
Attack – this controls how quickly the compressor responds to incoming signals that exceed the threshold level. For example, a fast attack time is ideal for sounds that have sharp and quick peaks – like a snare drum or hi-hat. In contrast, instruments such as acoustic guitar or bass have a longer sustain time, and may work better with slower attack settings. With a really fast attack, it is possible to actually hear the compressor kick in, causing a “pumping” sound.
Release – pretty much the opposite of the attack setting, release controls how long the compressor holds on to (stop compressing) a signal after it has fallen below the threshold range. As with attack, you have to experiment with this control to a find the most transparent setting. A quick release time may cause too fast of changes in dynamics, while too long of a release time could steal too many dynamics from the track.
Output/Make-up Gain – sets the level at which the reduced signal enters the mix. Use this to literally “make up” for the amount of gain reduction that has happened in the previous stages.
It is a well known fact: over compression will suck the life out of your signal. Transient peaks and some changes in dynamics breathe variety and flavor into your tracks. Think of compression as a tool to control dynamic range of a signal, but not eliminate it completely. This is why it is important to understand the above list of features. With this being said, trust your ears and remember that using compression is a compromise between your original and processed signal.
At first glance, dithering may seem like an extremely complicated mathematical equation. It is beyond the scope of this article to delve into scrutinizing detail concerning dither. However, as a necessary part of digital recording, it is important for audio engineers to have a basic grasp on dither.
Anyone who has spent some time in the realm of digital audio has probably encountered dither in some form or another. You may have noticed it as a part of limiter plug ins or other mastering related effects. Here’s the low down on what dither will actually do for you:
As mentioned above, dithering is absolutely necessary in digital recording. Whereas analog recording theoretically has an infinite resolution, digital audio works in discrete but quantifiable steps. When working with high bit rates in the digital world, you eventually need to down convert. The classic example of this is converting a session recorded at 24 bits to 16 bits for a CD. Whenever a high resolution signal is reduced in resolution, quantization errors will be introduced into the signal. The signal is now truncated – this means artifacts and digital distortion (the bad kind). This is where dithering comes in.
So what exactly is dither? Believe it or not, it is a very low level random noise. This white noise is undetectable under the music or whatever is happening on the track. Introducing this into the signal will improve the resolution of the conversion process down to the lowest bit level. It will also greatly reduce distortion and peaks in a way that aids the final signals performance. Although it sounds counter-intuitive, adding white noise to a truncated digital audio signal will produce a much more pleasing and natural result.
I have always been taught to apply dither at the final stage of production, and to only do it once. Typically, this will be during the mastering process when the master CD is made for duplication.
For an in depth look at dither, I suggest the following reading: http://www.users.qwest.net/~volt42/cadenzarecording/DitherExplained.pdf
Since microphones are the literally the first element in the recording chain, they can make or break any recording. Understanding and knowing how to use different microphones is essential knowledge for any serious recording engineer. Although you may not have immediate access to a wide selection microphones, having a working knowledge of general sonic characteristics will come in handy when you are in a situation to use mics that are new to you.
A closer look at the three most common studio recording microphones:
This microphone works on an electromagnetic principle, and features a stiff, thick diaphragm wrapped in a coil of wire. Housed in a magnetic structure, the coils are caused to vibrate by changes in sound pressure – disrupting the magnetic field.
Dynamic mics are extremely tough. Not only can they handle phyisical abuse, but they can also stand up to the loudest sound sources. These mics are many times the go-to choices for kick drums, snares, guitar amps, and bass amps (not to mention live sound as a whole). They are considered to produce a “clean” and “uncolored” but “non-sensitive” sound.
Famous Examples: Shure SM 57/58, ElectroVoice RE-20, AKG d-12, Sennheiser MD421
Also operating on an electromagnetic principle, ribbon microphones feature a very thin aluminium diaphragm placed between two strong magnets. Acoustic pressure from the front or back causes vibrations based on the acoustic waveform. Compared directly to the moving coil of a dynamic mic, the output signal is quite low. This necessitates the use of a dedicated transformer to boost the signal. Ribbon mics are known to respond well to low impedance pre-amps.
Since the aluminum diaphragm is so thin, this is a very fragile design. Way too often, I hear stories of blown ribbons because of super loud sound sources or hot plugging (non grounded application of phantom power). Don’t let this happen to you! Make sure you handle these mics with great care. That being said, ribbon mics were the go-to studio microphones in the late 40s, 50s and early 60s. Many legendary early rock ‘n’ roll records were recorded with nothing but ribbons. Since the ribbon is open to both the front and back of the mic, it picks up sound in a bi-directional (figure 8 ) fashion. Generally, these are described as producing “dark”, “smooth” and “warm” sounds. Experiment with them on guitar amps (moderate levels, of course!), string/reed instruments, and as drum room mics.
Famous Examples: RCA 44/77, Royer 121, Coles 4038
A word on modern ribbon mics: Advances in technology have resulted in much smaller and more efficient ribbon design. With the use of rare earth magnets and filters in the grill, new designs are much less prone to wind damage and can handle louder sound sources. This design results in a hyper-cardiod pick up pattern as opposed to the traditional bi-directional. The Beyer M160 is a classic example of this new design.
Unlike the previous two microphone designs, a condensor microphone uses an electrostatic capsule. Basically, two plates are involved. One is stationary (the back), while the other one reacts to changes in sound pressure. In order to properly function, a constant voltage is necessary to polarize the plates. Differences in sound pressure result in voltage differences between these plates. Now acting as a capacitor (storing voltage), the signal is run to a resistor and then amplified. The two most common approaches for amplification are “field effect transistor” (FET) or vacuum tube.
Capsule sizes often vary among condensor mics. You will commonly hear of “large diaphragm” and “small diaphragm” mics. Since this design requires an initial current to polarize the plates and then to amplify the signal, it is necessary to power these microphones. For solid state mics (FET), this is easily accomplished with phantom power, which introduces a constant 48 volt signal. Otherwise, microphones with a tube circuit have their own separate power supplies. Condensor mics are quite versatile, and are ideal studio mics. They are more sensitive than dynamic mics, and respond well to acoustic instruments, vocals, and drum overheads. As a whole their sonic characteristics could be described as “warm, “full” and “bright”.
Famous Examples: Neumann 67 (tube), Neumann 87 (solid state), AKG 414, Blue “Bottle” series
Hopefully this rough guide will help you understand some basic differences between these three mic classifications. Remember, these are merely guidelines and suggestions.There are many other factors to consider (room, playing style, pickup pattern, etc). There are no hard rules when it comes to recording! Use your ears, and don’t forget to experiment.
As probably the most common for of signal processing, EQ is an essential engineering tool. For new comers to audio, the concept of “EQing” can be a bit overwhelming. As with most things in our industry, you get better at it over time (as your ears develop).
So what is equalization? Basically, it is a tool that lets us control the relative amplitude of particular frequencies found in the audible bandwidth. To put it more simply, EQ allows us to boost or attenuate different tones in a recorded sound. There are many good reasons to use EQ, but there are also just as many not to. In many cases, it boils down to a stylistic thing – some engineers favor equalization during tracking, some do not. For me, starting with a strong performance, good mic placement and a good musician are key. EQ only comes into play if necessary.
Some reasons to uses EQ:
– flatten frequency response of a particular mic
– correct specific problems in a recording
– alter sounds for creative reasons
– blend together contrasting sounds
Reasons not to use EQ:
– easy to get carried away, reducing decent sounds to something worse
– no effects can fix a bad recording
Based around certain parameters, this EQ lets you alter certain frequencies with variable bandwidth and gain. Although models vary, typical designs revolve around variable center frequencies.
This type of EQ is common on many popular recording consoles, as well as being the standard EQ in many major DAW’s. One of the great features of parametric equalization is precise (surgical) cuts or boosts in any frequency without effecting other frequencies. In fact they often overlap.
ProTools Parametric EQ (notice the overlapping frequencies)
Although perhaps not as common on consoles anymore, this type of EQ is very popular in live sound. A graphic EQ allows you to cut or boost levels on equally spaced frequencies (ie. octave). These are usually identifiable by a slew of vertical sliders which theoretically give an overall readout of the frequency response curve.
Graphics are generally found in live sound for feedback control. If using enough bands, it is easy to notch out undesirable frequencies (ex: 60 hz electrical hum). Many popular computer applications, such as iTunes, use a digital graphic EQ for the user to shape the sound.
31 Band Graphic EQ
Which EQ is the right one for the job? That depends on the occasion and personal preference. Generally speaking, parametric EQ has a higher learning curve and can cause more extreme changes in the sound. A graphic EQ is more visual and easier to understand right away. Again, a good practice for audio younglings would be to focus on getting good sounds before you try to change them with equalization or other effects. There are no real short cuts here – it just takes a little time.
One of the first points addressed in any session is the click track. Many musicians believe that using a metronome during recording immediately results in a stiff track with no breathing room. Engineers often favor click tracks because it makes editing and overdubbing much easier. Click tracks are an invaluable tool, but sometimes they are not right.
As mentioned earlier, a common mentality among musicians (especially younger ones) is that using a metronome will result in robotic performance with no emotion or energy. This is the wrong mindset. The truth is that no performance will ever be “perfectly” on time. The purpose of the metronome is simply to provide a rhythmic boundary for the group and prevent the drums from varying in tempo. Any good drummer will be able to work within these bounds and still “groove”.
Recordings done with a click track should end up sounding just as natural as any other recording. If the drummer can groove and the band has a good pocket, this group will have no problem following a metronome. If a drummer refuses to use a click track simply based on the argument that it will “sound robotic”, I interpret this to mean they do no practice enough. For me, I believe a drummers job is to keep the time – if you can’t do this you should probably start practicing! I would guess that a huge majority of the songs played on the radio in the past couple decades has been recorded with a click track – enough reason for most people to realize that click track recordings are the way to go.
However, a click track is not always a good idea. Some examples include songs with multiple time signatures, live recordings with no overdubs, orchestral recordings, inexperienced groups who simply cant perform to a click, and bands so good they do not need one (unlikely). A click track does not mean the performance will automatically be awesome. In most scenarios, I feel it definitely does help glue together a performance, but there are some cases in which a metronome will negatively effect the performance. The most common is when a drummer is speeding up, realizes they have lost the tempo, then slow down to catch it again. This wavering of the tempo results in a super unprofessional sounding recording, and should be avoided. Another example of this could be if a band is significantly altering their performance to accommodate a click track.
If any type of overdubbing is part of the session, it is foolish to not use a click track. This is especially true for songs that have breaks where instruments (drums in particular) drop out. Having a click guide for this type of stuff is like night and day. On this note, it is important to not force a click on a band just because it makes it easier to “protools” a mix later on. The decision to use a click should solely take into account improvements in the music – of course it doesn’t hurt to mention ease of overdubbing in most scenarios.
Generally speaking, I think almost all bands will benefit from click track recording. When used properly, it will increase the tight-factor of any song while doing wonders for intensity and build of a track. As we talked about, however, it is not always right. I like to consider these points for any session I am doing and then decide whether or not a click track is the right choice for the session.
