Last weekend whilst having a hunt round the BPM show at the NEC we came upon our friends at Steinberg and their wonderful little CMC series controllers. You can pick and choose from a number of cheap and cheerful units to make up your very own dream control panel. With that in mind and a view to populate his own x-mas wants list even further our Tom sat down with Andrew from Steinberg and had him take us through the options available.
Many thanks to all those that made it down for our open day last Saturday making it a great success. The theatre area was set up and demos, training sessions and Q&A’s took place throughout the day with Ableton production and remixing sessions being popular and the showmanship of Dj Rasp entertaining all between the seminars themselves.
If you missed out this time watch the site for futher announcements as we have another being planned before the end of the year.
Building A Silent PC Solution in the NoFan Set A40 Case
One of the main considerations for any audiocentric build has traditionally been the overall noise of the final system. If you get to design a studio from the ground up, you find yourself able to rack up or remove the computer hardware into a separate area away from your recording section of the studio. For a lot of users through especially those working in a small project studio environment this may not be viable and you may have to compramises in order to make the overall setup work. In this situation you may still need to have the ability to edit and record in the studio space where your setting up mic’s and instruments so the last thing you want to be able to hear in your final recordings is noise from the computer doing the processing work.
Whilst all of our systems are designed with this in mind and components are carefully chosen to ensure as little background noise is created as possible, what if we could go further than that? Ideally we want to be removing as many moving parts from inside a computer as we can to ensure you end up with the lowest noise footprint possible.
Enter the 3XS NF26 and the SET-A40 case.
NoFan are a new company setup by the original designers from Zalman, who have left to start up a new company developing unique designs and innovating in the world of PC silence. The SET-A40 bundle we base this build around includes a case, cooler and 400w PSU which are all designed to run passively with no fans for required for cooling and allowing use for CPUs rated at upto 95w TDP.
To sum up the design and idea behind the system I’ll add in here what the company themselves have to say about this product: ” Nofan’s bundle comprises their revolutionary CR-100A IcePipe Fanless Cooler, a fanless 400W power supply and a specialized convection case to accommodate the CPU cooler and any other components that are required to build the perfect silent computer, with zero dust build up.”
So without further ado let’s take a look at it.
Well it’s a case box. Indications on it that we should expect 0 db(a) of noise from the system and indications of the components inside. It’s at this stage that you’ll get the first indication of the crazy cooling system from the artwork on the side but more of this later.
Once we break it out of the box we get to take at look at the front panel. It has a couple of exposed 5.25 bays and a 3.5 for your card readers. The are the normal selection of ports and jacks on the front and all in all, so far, so ordinary.
With the side panel off we see once more fairly typical case design but lurking in there are some out of the ordinary bits and pieces. First of all the large brown box taking up most of the free space is certainly in need of further examination…
Having opened the box the first thoughts through many peoples minds are pretty much “What is that?!!?!?”. In office we discussed the lot from hamster wheel right through to salad spinner. In actual fact this is the very heart of the machine.
I present to you the NoFan fanless cooler.
Based around a liquid pipe design it certainly is sizable but at the same time suprisingly light. Underneath we can see a nicely polished base with the company logo etched into it with a couple of heatpipes as well as the support arms designed to carry heat away from the heatsink base itself.
It’s certainly a nice tidy design and no sharp edges to it which will make builders use to the Zalman flower designs of old breathe a sigh of relief!
So how does this monster heatsink attach to the motherboard? With surprising ease in fact and it’s clear a lot of thought has gone into this design. 4 mounting poles are attached via screw mounts into a special backplate and that is more or less that which for a heatsink of this size is once again quite surprising.
A side on shot of the system with the motherboard mounted inside. You’ll notice at this point the rather odd mounting position of the psu. As this is also passive and generating heat NoFan have decided to mount it at the front rather than the rear to keep the heat evenly spread throughout the system.
Once again the thought that has gone into this design becomes apparent when you attempt to mount the heatsink. Some designs can be very hard to mount but with this you can just drop the cooler into the case and then you line up the screw holes…
Screw in the thumbscrews and the job is pretty much done.
Quick easy and far less hassle than a lot of other designs.
And there you have it. The system is assembled and ready to be fired up for the first time.
So what do we think of it here in Scan? It’s a bit of a niche item but it does the job very well if it fits your requirements. Annoyingly some mechanical parts are still required to complete the build but you can work around these as well. We’ve set ours up with an SSD for the O.S. drive which will keep the performance up and the noise down but you’ll still need something to hold your project data and a larger mechanical is still the only real option. For our demonstration unit we set it up using a caddie that allows you to fit both a laptop hard drive and a slimline slot loading Dvd drive. The laptop harddrives are generally quiet solutions and by using the caddie it allows the drive to be swappable allowing for quick backups or the moving of projects between machines. Also by using a slot loading optical drive in this solution we get a reduced noise level as slot loaders tend to pinch the optical disk on both sides rather than a single sided spindle lift you see in tray designs so less rattle from your dvd’s.
On the all important performance side it work fantastically too. We initially tried it with a normal 95w 2600 CPU and ran it on Prime 95 for around 6 hours with the CPU running around the 85 degree mark. Whilst that is still within Intels limits we like to build our systems with a bit more overhead as studios as we all know tend to get a little bit toasty once all those lovely toys are turned on! So we broke out the 2600S low powered edition which uses around 30% less power than it’s bigger brother. We clocked up the CPU to around 3.3Ghz which is only just slightly slower than the 3.4Ghz rated 2600 regular and did the same Prime 95 test once more. This time we got an average of around 70 digress over the same time frame which is far below our own in house threshold and once you factor in that the machine will never be run in the real world at this sort of level for more than a few minutes at a time it promises a long and stable life for the machines usable duration… most impressive!
Whilst you can never fully remove every mechanical component from your build, by using these options you will minimize the noise levels of what’s left and the result will be an extremely quiet solution ideal for in studio usage.
This system is available on the Scan 3XS site on the audio system configurator : 3XS NF26 Silent P.C.
The NoFan Set A40 is also available as a barebones bundle : NoFan Set – A40
One of the key choices faced by eveyone when starting out making music is which sequencer software to learn in order to be able to produce your own recordings. As the heart of any modern studio the sequencer will allow you to record, edit and even master your music in your own preferred recording space and with the power and features available in even the most humble of software studios these days you can get astounding results, that even just a couple of decades ago were unthinkable by anyone working outside of a large studio environment. The problem with having to make the choice at this stage is that your most probably at your most unprepared for what is essentially a choice that will shape your work flow dynamic for years to come, so in this situation just how do you decide when its likely your not even sure what you need?
To keep things simple for this article we’ll break them down into a couple of groups and we’ll start with the traditional sequencers Cubase & Sonar. Both of these solutions have long heritages with Steinberg’s Cubase first appearing on the Atari ST in the late 80’s and Calkwalk’s Sonar appeared a few years prior to that under it’s original brand name of Cakewalk, meaning that both of these solutions are regarded as long established industry standards with Cubase being the popular choice in Europe and Sonar the leader in the USA market.
Designed initially as midi sequencing tools used to record and edit playback data controlling synths and other external hardware, it was with the advent of the Steinberg introducing the V.S.T (Virtual Studio Technology) standard in the mid 90’s which over time has become the dominate format over Cakewalks own DXi plugin standard (Sonar also supports VST) that we’ve seen sequencers grow from their humble beginnings to the all encompassing studio in box solutions we see now. When choosing between these two software packages you’ll see that most of the features found in either one will tend to be available in the other sooner or later. The has over the years been a history of them pushing each other on when developing new features and improvements which has resulted in great feature rich solutions being developers for users working with either client.
Over time we’ve seen these sequencers also introduce timeline based real time audio editing and manipulation which was previously was the greatest strength of the other classic recording software ProTools. Once again originally developed in the late 90’s but this time as a replacement for the classic multitrack tape recorders found in every recording studio up until this point Pro Tools was developed as a medium to allow loss less digital recording in an environment where the audio could be manipulated and processed without degradation associated with working analogue or even digital tape formats. ProTools was regarded as a game changer as it could speed up the mixing and mastering process and allow all sorts of editing tricks to be applied that were previously only be dreamed of by the average razorblade wielding tape based editors of old.
ProTools however in the early days by design was developed to only work with dedicated hardware solutions (audio interfaces) which whilst ensured a high quality audio recording environment also put this far outside the price range of the average home studio recordist. Over time however the platform has opened up with ProTools HD remaining at the highend we saw the introduction of the LE revision and with a wider range of features such as full VST support although still required special hardware (the MBox range) to support and run it. Recently we’ve seen this evolve into the ProTools 9 release which like its counterparts Cubase & Sonar will now run on any sound card and hardware configuration it joins them as a fully featured elder statesman of sequencers.
So that’s the old guard covered what about the newer solutions?
Over the last decade or so we’ve seen any number of newer software packages appear and whilst some are designed in the same fashion as the older sequencers with midi being a primary concern with the most notable being the superb Reaper client, we’ve seen a number of software houses approach the process with new ideas and tailor their software more towards those of us who work fully inside the box rather than make music with external hardware.
The one package that can probably lay claim to making the most impact on how we think about arranging and working with sequenced music in recent times is Ableton. Originally developed as a live performance tool that would give the ability to remix and edit loops and audio on the fly in the early days we saw ground breaking DJ sets where the artist would load up all of their self written tracks as component parts and perform by mixing and matching components of their music blended together allowing for a unique performance each and every time. As artists got use to doing this live and discovered just how quick and easy it was to work with they started to use it more and more as a studio tool rather than just a live performance instrument and the Ableton development team have picked up on this and continued to develop it into the one stop solution no matter if your working in the studio or performing out on the road.
Ableton’s session view is a great alternative to the more traditional arrangement window setup.
Other notable packages include Sony Acid, FL Studio and Reason which all continue to go from strength to strength. Both FL Studio and Sony Acid started out as a loop based sequencers and have evolved to play host to a lot of the features of the larger more established packages and offer support for the popular plugin standards. Reason on the other hand is a popular all in one package which restricts it’s users by not supporting VST/DXi and other none native formats but rather maintains its own synth and sampler selection as part of the package. Whilst this can be seen as a negative by users wishing to dip into the wider waters of plug ins, it does have the notable advantage of focusing the user and by keeping those choices more limited which can actually help speed up workflow as anyone who’s ever faced a screen full of synths wondering which would be most suitable tor the idea in their head will tell you. Perhaps because of this a number of artists have mentioned that they prefer to write within this environment as they find themselves being at their most productive working this way, although they may still find themselves having to transfer projects over to other software solutions to complete the tracks at mixdown stage if they want to take advantage of tools not available inside of Reason to mix or master the project.
Hopefully this brief rundown has given you a few ideas of where you wish to look and our only other advice would be to get hands on. All sequencers initially require a bit of time to get to grips with, but as you pick up the concepts your ability to get your ideas down as you want them will get quicker and quicker as you learn more and more. Obviously with so many options some of these will prove better for you than others so we highly recommend you trail each package that appeals to how you wish to work.
Thankfully the majority of software firms offer trails of their sequencers giving you a few weeks to spend time with each one before you decide upon that initial outlay, so you should take advantage of this and give each one that stands out a try in order to make sure you make the right choice along the road to making music for yourself.
Every year we find with computer systems as with so many other products it seems that the is always something bigger, better and faster becoming available. The question is how do we validate those claims and work out which solution will fit which user whilst offering the best performance at any given price point?
Here in Scan we use a number of different tests and where gamers concern themselves with performance indicators like 3DMark and video people concentrate on Cinebench for audio the stand out test used by retailers and reviewers alike is DAWBench for audio computer system benchmarking. DAWBench’s working methodology is a rather large subject in itself and something we will be covering in later articles in much depth but here we can give a quick overview covering how it relates to audio computer system performance.
The DAWBench tests revolve around running as many instances of a given effect or audio source as possible until the CPU overloads and audio corruption is generated in the signal path. The most common variation of this test is the RXC compressor test which has been in use now for a number of years and has plenty of results generated overtime making it ideal for us to look at how performance has grown from generation to generation of audio computer systems.
The test itself is fairly simple to carry out and can be run in a number of popular sequencers including (but not limited to) Cubase, Reaper, Sonar and Protools. The template for the test can be downloaded from the DAWBench website which consists of 4 tracks of audio parts and 40 channels of sine waves. On each of these sine wave parts 8 RXC compressors are included already set up but not yet activated and it is these you switch on one at a time in order to put the system under more and more load. Whilst testing the sine wave channels that you are working with are turned down but the accumulated compressors continue to up the load on the system and you monitor the situation by means of the looping audio tracks playing through your speakers. As you reach the point where the processing ability of the system reaches its maximum handling ability the audio you hear will start to distort and break up and it’s at this point where you have to turn off a few compressor instances taking it back to the point where the audio is clean and unbroken, which when you have the audio this point you then make a note of the total number of RXC compressor instances achieved and that is your score at the buffer setting in question.
A quick real world explanation of buffer latency for those not familiar with it is this. A low buffer setting means that your input devices can communicate quickly with the CPU inside of the audio computer system and the data can be processed quickly and for real time interaction this is crucial. Something you can try yourself is setting the buffer latency in your sound card control panel firstly to it’s lowest figure normally around the 32/48/64 level and playing a note on your midi controller which you will find is very responsive at these settings. If however you raise the latency settings up to around the 1024 level or higher and now trigger your midi controller you’ll notice a definite amount of lag between the key press and the sound coming out of the speakers.
So why would we want to run an interface at 1024 or higher settings?
As you bring down the buffer figure to improve response times your placing more and more load upon the CPU as a smaller buffer is forced to talk to the CPU more often which means more wasted cycles as it switches from other jobs to accommodate the data being processed. Whilst an artist performing or recording in real time will want the very lowest settings to enable the fastest fold back of audio to enable them to perform their best, a mix engineer may wish to run with these buffers set far higher to free up plenty more CPU headroom to enable high quality inline processing VSTi’s the performance to carry out their tasks without overloading the processor which as we’ve seen before would cause poor results in the final mixdown.
Too keep the playing field level the results below have been tested with Windows 7 64bit and in all these tests we have used a firewire M-audio Profire 1814 interface to ensure the results are not skewed by using various interfaces with different driver solutions. The are better cards that will give better results at super low latencies, with the RME range for instance going down to buffer settings of 48 on the USB/Firewire solutions and even 32 on the internal models. The M-Audio unit however has great drivers for the price point and we feel that giving fair figures using an interface at an accessible pricepoint gives a fair reflection of performance available to the average user and those who are in the position to invest in more premium units should find themselves with additional performance gains. We will be comparing various interfaces in the future here on the blog and the are benchmarks being produced in the DAWBench forums which also good further reading for those of you looking for new card solutions in the meantime.
So what does the chart above show us?
The are a number of audio computer systems being tested on there from over the last few years and it shows the continued growth of performance as newer hardware has been released. The stock i7 2600 proved to be a great performer when stacked up against the previous high end Intel systems even coming close to the hexcore flagship chips from that generation. What we also see is that once you take a 2600k and overclock it as we do here the performance available is greater than the 990x for a great deal less cost wise although it has to be noted that the X58 platform has more available bandwidth which can help increase performance in some real world instances where the user is working with vast sample libraries, the results we see here are a good indicator of how the machines will run for a more typical user.
Also worth noting in the performance results above is the i5 2500 result as we use it in our entry level value systems currently. The performance is roughly half of the overclocked 2600k system and in real world terms the cost of the system is roughly half as well meaning that whilst neither unit offers better value for money than the other in the cost vs performance stakes, in instances where your recording requirements are not quite as great the value spec still offers plenty of power to get you going and achieve completion on smaller projects even if it doesn’t offer the additional cooling and silencing features we have as standard on the high end solutions. It’s also worth noting that the i5 2500 scores close to the last generation i7 930 which shows how much performance improved between the last generation and the current one.
Our high end laptop solution in all but the very lowest latency situations also proves to be pretty much on par with the last X58 based i7 930 processor which itself still offers enough power to the user to get the job done in all but the most demanding situations which means that the age of the full desktop replacement laptop is very much with us making it as easy to edit, mix and produce fully formed mixes on the road as it is to perform every night with the very same units.
Hopefully that helps explain how we rate audio computer systems in house for performance testing and will help you decide upon your own next system. We run these tests on each new range we release so keep an eye out for further articles showing testing results as new hardware reaches the market.
Microphone diaphragm sizes
Any microphone with a diaphragm larger than (and potentially including) 3/4″ is considered to be a Large Diaphragm microphone. In general, Large Diaphragm microphones tend to have a “big” sound that engineers find especially pleasing where a little more character might be advantageous, such as is the case with most vocals. Large diaphragms are generally more sensitive than small diaphragm or medium diaphragm mics because of the increased surface area. A common myth is that large diaphragm mics capture more low frequencies than small diaphragm mics. Sometimes their colouration may make it sound like this is the case, but a properly designed small diaphragm mic is more likely to be accurate throughout a wide range of frequencies, whereas the coloration of a large diaphragm mic can tend to enhance certain desirable characteristics in a sound, which sometimes amounts to more apparent bass or low end.
The definition of Medium Diaphragm is a potentially controversial subject. Historically there have been large diaphragm and small diaphragm mics, but more recently the medium size has begun carving out its own category, though not everyone agrees on the precise upper and lower limits. Most professionals and manufacturers agree that any microphone with a diaphragm near 5/8″ to 3/4″ can be characterized as a Medium Diaphragm microphone. Generally speaking, Medium Diaphragm microphones tend to do a good job of accurately catching transients and high frequency content (as a small diaphragm would) while delivering a slightly fuller, round and potentially warmer sound (as a large diaphragm might).
While there are no final standards regarding a diaphragm size that defines Small Diaphragm, most professionals and manufacturers agree that any diaphragm smaller than 5/8″ would be considered a Small Diaphragm. Generally speaking, Small Diaphragm microphones tend to do a good job of capturing high frequency content and transients. They will tend to have a bit more “air” to their sound and often have less coloration than medium or large diaphragm microphones. Most of this is due to the reduced mass of the smaller diaphragm, which allows it to more closely follow any air disturbances it is subjected to.
SCAN GUIDE TO MIDI CONTROLLERS:
WHAT IS A MIDI CONTROLLER KEYBOARD?
Basically a MIDI Controller Keyboard is way of communicating with a Synth or a sampler or a Computer running a VST instrument or other software based Sound generator.
It is possible to play music via a computer by simply entering data into the DAW via the QWERTY key board, but it’s not very musical.
Keyboard controllers are usually based on a standard piano keyboard. Pressing down a key allows a Note On/Note Off message to be sent to a receiving device, a sampler maybe, telling it exactly which note to sound. At the same time a velocity message is transmitted, showing how hard the key was struck.
Compared to a real piano, most keyboard controllers have small keys and provide a playing range of just a few octaves.
They usually have no internal sounds of their own. Many units also come with various sliders, knobs and pads which can be assigned to control other MIDI functions linked to filters and Oscillators.
The Controller usually connects to the computer via a USB port, doing away with the need for a MIDI interface. Most software can recognise the USB as a MIDI Device.
OTHER MIDI CONTROLLERS…….
As well as devices based on a Piano keyboard, there are many controllers available which are based around a series of pads, or sliders and knobs, as well as dedicated controllers for software packages like Ableton Live.
They basically all work the same way, sending MIDI controller data to the computer or synth, and allowing the user to manipulate the sound in a much more tangible and intuitive way.
PERCUSSION MIDI CONTROLLERS:
These are specialised ‘Drum Kits’ or individual drum like elements that allow the user to play sampled or computer generated percussion sounds in an authentic way.
32 bit systems are limited to 4gb of memory in theory (in reality its between 3-3.5gb that windows can actually use). While this might sound a lot, every time you open up a plugin or virtual instrument, it uses memory.
When you start looking at sample based instruments, such as orchestral libraries these can easily load gigs of sounds into memory.
64 Bit systems can run 32 bit programs, but each application can only use 4gb of memory.
This is currently a popular choice, as most DAW’s come with 32 and 64 bit versions that can be installed at the same time.
64 Bit issues (and how to get round them)
64 Bit Sequencers cannot use 32 bit plugins or instruments.
Whilst many manufacturers are now producing 64bit versions of thier plugins and instruments, if you do switch to a 64 bit DAW, you will probably be left with plugins that you cannot use.
Many DAW’s, such as Steinberg Cubase 6 have built in “bridges” that try to make them work, but they only seem to work for some plugins.
Cubase’s bridge mode also limits you to 4gb of memory for all of the bridged plugins.
The best soulution to this that we have found is a piece of software called Jbridge ( €14.99)
Jbridge is about 95% compatible, and has a number of options to get problems plugins to work. Jbridge lets you use 4gb of memory per plugin.
The second issue is that rewire will not work in 64 bit daw’s.
“Rewire” channels are virtual midi and audio connections to and from your daw to (predominantly) Propellerhead Reason or Ableton Live programs.
A work-around to this issue is a plugin called Rewire VST (€19.00)
This provides one stereo and six mono audio channels into your DAW (plus midi control).
Whilst this is no way near the 64 possible connections that rewire normally offers, it does mean that you can run a handful of reason or ableton instruments alongside your 64bit DAW.
Types of microphones
|*||What’s a USB Microphone?
A USB mic contains all the elements of a traditional microphone: capsule, diaphragm, etc. Where it differs from other microphones is its inclusion of two additional circuits: an onboard preamp and an analog-to-digital (A/D) converter. The preamp makes it unnecessary for the USB mic to be connected to a mixer or external mic preamp. The A/D converter changes the mic’s output from analog (voltage) to digital (data), so it can be plugged directly into a computer and read by recording software. Plug in your mic, launch your DAW and start recording.
The condenser microphone is a very simple mechanical system, with almost no moving parts compared with other microphone designs. It is also one of the oldest microphone types, dating back to the early 1900’s. It is simply a thin stretched conductive diaphragm held close to a metal disk called a backplate. This arrangement basically produces a capacitor, and is given its electric charge by an external voltage source. This source is often phantom power, but in many cases condenser mics have dedicated power supply units. When sound pressure acts on the diaphragm it vibrates slightly in response to the waveform. This causes the capacitance to vary in a like manner, which causes a variance in its output voltage. This voltage variation is the signal output of the microphone. There are many different types of condenser microphones, but they are all based on these basic principles.
A dynamic mic is one in which audio signal is generated by the motion of a conductor within a magnetic field. In most dynamic mics, a very thin, light, diaphragm moves in response to sound pressure. The diaphragm’s motion causes a voice coil that is suspended in a magnetic field to move, generating a small electric current. Generally less expensive than condenser mics (although very high quality dynamics can be quite expensive), dynamics feature quite robust construction, can often handle very high SPLs (Sound Pressure Levels), and do not require an external power source to operate. Because of the mechanical nature of their operation, dynamic mics are commonly less sensitive to transients, and may not reproduce quite the high frequency “detail” other types of mics can produce. Dynamic mics are very common in live applications. In the studio, dynamics are often used to record electric guitars, drums and more.
A type of velocity microphone. A velocity microphone responds to the velocity of air molecules passing it rather than the Sound Pressure Level, which is what most other microphones respond to. In many cases this functional difference isn’t important, but it can certainly be an issue on a windy day. Very old ribbon mics could be destroyed from the air velocity created just by carrying them across a room; today’s ribbon mics can handle the rigors of daily studio use. A ribbon mic works by loosely suspending a small element (usually a corrugated strip of metal) in a strong magnetic field. This “ribbon” is moved by the action of air molecules and when it moves it cuts across the magnetic lines of flux causing a signal to be generated. Naturally ribbon mics have a figure 8 pick up pattern. You can think of it like a window blind; it is easily moved by wind blowing at it, but usually doesn’t move when wind blows across it from left to right. Ribbon mics were the first commercially successful directional microphones.
WHAT IS EQ?
EQ stands for Equalizer or Equalizing. An equalizer is a device which allows someone to equal out the tonal characteristics of a sound. They were originally conceived to help get a flat response down telephone lines, and to make up for the deficiencies in low end HiFi equipment.
These days however, EQ’s are used much more creatively to boost certain frequencies and alter the relative balance of others in order to produce tonal effects within a recording.
The Equalizer has the ability to cut or boost the amplitude in specific frequency ranges by using filter circuits. (A filter is an electronic device designed to reduce a signal’s energy at a specific frequency.)
Good EQ’ing is a skill which is learnt over time and can be the most important tool in a recording or mix engineers bag.
At the recording stage, good use of EQ can make instruments or voices ‘cut through’ better leaving less to do at mix down, and at mix down good EQ’ing can be used in place of volume boosting to improve a tracks definition within the mix.
Here are a list of frequencies that are useful to remember when recording or mixing down.
BASS GUITAR, CELLO, UPRIGHT BASS
To reduce the “boom” of the bass, BOOST at 50 Hz.
To “un-bury” overtones, BOOST at 50 Hz.
To increase the bass line in a final mix, BOOST at 50 Hz.
To increase loud rock bass lines, BOOST at 50 Hz.
To add a harder bass sound to lowest frequency instruments, BOOST at 100 Hz.
To add more “power” to lowest frequency instruments, BOOST at 100 Hz.
For increased clarity of bass guitar, cello, or upright bass; BOOST at 800 Hz
For increased clarity of bass guitar, cello, or upright bass; BOOST at 1.5 kHz.
For punchier bass guitar, cello, or upright bass; BOOST at 800 Hz or at 1.5 kHz.
For more “pluck” of bass, BOOST at 3 kHz.
For more “finger sound” on bass, BOOST at 5 kHz or at 7 kHz.
To add more fullness to kick, floor tom, and bass drum, BOOST at 50 Hz
To add fullness to snare, BOOST at 100 Hz.
To add fullness to snare for a harder sound, BOOST at 200 Hz.
To reduce gong sound of cymbals, CUT at 200 Hz.
To reduce “cardboard” sound of lowpitched drums and toms, CUT at 400 Hz.
To reduce ambiance on cymbals, CUT at 400 Hz.
For greater attack on low frequency drums, BOOST at 5 kHz.
For greater attack on floor or rack toms, BOOST at 5 kHz.
To add attack on low frequency drums a more metallic sound, BOOST at 7 kHz.
To give snares a more metallic sound, BOOST at 7 kHz.
To add attack to percussion instruments, BOOST at 7 kHz.
For added hardness on cymbals, BOOST at 10 kHz.
To brighten cymbals, BOOST at 15 kHz.
ECHO AND REVERB
To prevent reverb or echo from muddying the mix, CUT at 100, 125, 150, 200 or even 300
To create a “cave sound”, BOOST at 100 Hz.
To emphasize reverb or echo, BOOST at 3 kHz or 5 kHz
To get “Elvis” style echo, BOOST using a broad parametric EQ “bell” centered at 300 or
To reduce muddiness of mid-range instruments, CUT at 200 Hz.
To add clarity to bass lines, especially when speakers are at low volume, BOOST at 400
To bring a part forward, BOOST using a broad parametric EQ “bell” centered at 2 kHz.
To send a part backward, CUT using a broad parametric EQ “bell” centered at 2 kHz.
To make background parts more distant, CUT at 5 kHz.
GUITAR AND STRINGS
To add fullness to guitars, BOOST at 100 Hz.
To remove boom on guitars, CUT at 100 Hz.
To add clarity to guitars, CUT at 100 Hz.
To add fullness to guitar for a harder sound, BOOST at 200 Hz.
To make “cheap” sounding guitars sound less cheap, CUT at 800 Hz.
To remove dullness of guitars, BOOST at 1.5 kHz.
For more attack of electric or acoustic guitar, BOOST at 3 kHz.
To disguise out-of-tune guitars, CUT at 3 kHz.
To accentuate the attack of acoustic guitar, BOOST at 5 kHz.
To add brightness to guitars, especially rock guitars, BOOST at 5 kHz.
To soften “thin” guitar, CUT at 5 kHz.
To add sharpness to rock guitar or acoustic guitar, BOOST at 7 kHz.
To brighten string instruments, BOOST at 15 kHz.
To add “light brightness” in acoustic guitar, BOOST at 10 kHz.
HORNS AND WOODWINDS
To add warmth to horns, BOOST at 100 Hz.
To reduce shrillness of horns, CUT at 5 kHz or at 7 kHz.
To brighten flutes and woodwinds, BOOST at 15 kHz.
KEYBOARDS AND PIANO
To add warmth to piano, BOOST at 100 Hz.
For more attack on low piano parts, BOOST at 3 kHz.
To accentuate the attack of piano, BOOST at 5 kHz.
To add sharpness to synthesizers and piano, BOOST at 7 kHz.
To make sampled synthesizer sound more real, BOOST at 15 kHz.
To add “light brightness” for a piano, BOOST at 10 kHz.
To add fullness to vocals, BOOST at 200 Hz.
To reduce muddiness of vocals, CUT at 200 Hz.
For more clarity or hardness on voice, BOOST at 3 kHz.
Reduce to breathiness, or to reduce “soft sounds” of background vocals, CUT at 3 kHz
To disguise out-of-tune vocals, CUT at 3 kHz.
For greater vocal presence, BOOST at 5 kHz.
To brighten a dull singer, BOOST at 7 kHz, or send them to college.
To brighten vocals, BOOST at 10 kHz.
To reduce sibilance, which is the “s” sound, on singers, CUT at 7 kHz or at 10 kHz.
To brighten vocals by emphasizing breath sound, BOOST at 15 kHz.