Audio Computer System Benchmarking

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.

Dawbench Homepage

Microphone Diaphragm Sizes

Microphone diaphragm sizes

Large Diaphragm
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.

Medium Diaphragm
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).

Small Diaphragm
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


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.

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.

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.

64 Bit Computing for Windows Musicians

This is an important decision that you need to make when choosing your new pc, not only for your operating system, but also for your DAW software.

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.

J Bridge working with Kontakt 3
J Bridge working with Kontakt 3


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

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.
* Condenser Microphone
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.
* Dynamic Microphone
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.
* Ribbon Microphone
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 E.Q.?

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.

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.
Page 3
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
500 Hz.
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.
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.
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.
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.
Page 4
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.

Ways to promote your music online

  1. 1. Social networks
    My Space and its brethren are still a good way to get your music up and ‘out there’.  Nowadays there are any number of dedicated music upload sites, and each has its own merits, however, the aim is the same… get your music heard!!!

2. Websites
Websites have never been cheaper or more easy to maintain and update so ensure you have a good focused presence online as it also looks a lot more professional. Check out some of the ‘template’ based builder sites to get yourself up and running quickly with the added advantage that the are a number of them now doing easy to set up digital download shops to help you sell through your music online.
3. Keep your profile current
Regular profile/website updates will also keep things interesting for returning fans. News stories about what you’ve been up to or where you played last, plus upcoming gigs, all this is the life blood of your profile.
4. Biography
Biographies are a great way to let people know about who you are and why you make the music you do, but DONT MAKE THEM BORING!!!!
Honestly, there is nothing worse than reading about the first puppy your Granny bought you at the age of 7, and how your Mum never let you eat Chocolate fingers……………
5.  Good photographs
Good quality images will enhance your profile better than anything. Local student photographers are usually up for taking some profile pics for the price of a good lunch/few drinks, and it can greatly enhance your Online profile
6. Free downloads
If you ain’t signed to a label or publishing deal, you need to do as much as you can to get people to hear your music, and the odd free giveaway track can help a lot especially if you can target the multitude of blogs covering new music, which will help spread your name further.
7. Reply to mail.
If someone is nice enough to get in touch with some praise, or leave a nice comment on your you tube clip, try and get a personal reply back to them. It’s a nice thing to do and people will remember you for it.
8. Don’t Spam people!
Seriously, nothing puts people off more than you abusing your Facebook or Twitter feed with useless information about your gig at the “Old Bull and Bush, Peckham” when they’re in Liverpool. Keep information local and targeted. Got any email addresses from the last time you played here, by all means tell them you’re here again

9. Get your songs on iTunes
Nothing says professionalism like having your songs available to buy on the world’s largest online music store. Websites such as CD Baby and Bandcamp can get your tracks online for a small charge – you can then link to your songs in the store from your website/profile.

Understanding microphone polar patterns

Understanding microphone polar patterns
A microphone polar (pickup) pattern. Characterized by strong sensitivity to audio from the front of the mic, good sensitivity on the sides (at 90 degrees, 6 dB less than the front), and good rejection of sound from the rear, the Cardioid pattern can almost be visualized as a “heart-shaped” pattern (hence its name). The ability to reject sound from the rear makes Cardioid patterns very useful in multi-miking situations, and where it is not desirable to capture a large amount of room ambience. Popular in both studio and live use (where rear rejection cuts down on feedback and ambient noise), Cardioid mics are used for a very high percentage of microphone applications.Keep in mind that like all non-omnidirectional mics, Cardioid mics will exhibit pronounced proximity effect.

A polar pattern name used to describe the pickup pattern of some microphones. The Supercardioid pattern is very similar to, and often confused with, the Hypercardioid pattern. The Supercardioid pattern is slightly less directional than the Hypercardioid pattern, but the rear lobe of sensitivity is also much smaller in the Supercardioid .
A polar pattern name typically used to describe microphone pick up characteristics. Hypercardioid patterns are similar to Cardioid and Supercardioid patterns in that the primary sensitivity is in the front of the microphone. They differ, however, in that the point of least sensitivity is at the 150 – 160 and 200 – 210 degree positions (as opposed to directly behind the microphone in a Cardioid pattern). Hypercardioid microphones are thus considered even more directional than Cardioid and Supercardioid microphones. Hypercardioid microphones are frequently used in situations where maximum isolation is desired between sound sources.
Literally, from all directions. In audio, microphones are said to be omnidirectional if they can detect sound equally from all directions. An Omnidirectional microphone will not exhibit a pronounced proximity effect.
A microphone polar pattern in which the mic is (nearly) equally sensitive to sounds picked up from front and back, but not sensitive to sounds on the sides. This produces a pattern that looks like a figure 8 on paper, where the microphone is at the point of crossover on the 8. The pattern is also known as bi-directional.

Recording vocals


The main reason for having a vocal filter, is to try and record a ‘dry’ signal.
If your studio doesn’t have the best acoustic treatments it could have, you may find one of these filters invaluable. They are basically a curved composite wall, which sits behind any microphone by means of a variable position stand clamp assembly which ships with the product.
The Vocal Filter helps prevent any reflected sound reaching the back and sides of the mic. Its shape and size have been carefully tested to maximize absorption while keeping ‘coloration’ down to a minimum, and leaving the microphone’s polar pattern unaffected.

Session singers are trained how to work with a microphone, so that when they are singing words that have plosives within them, they will turn their heads slightly away from the mic, or when singing a loud part, they will move slightly away from the mic to reduce the power of the vocal, but still, sometimes a ‘Pop’ will find its way through.
Whats happening is that plosives are letters or parts of words that require a push of wind from the lips, (words beginning with ‘P’ are usually culprits). The Pop screen allows the voice to pass through the mesh, but the ‘Push’ of air on a plosive is broken up and as a result the ‘Pop’ disappears.
The screen needs to be set up a couple of inches from the mic itself, and needs to be attached to the mic stand so it sits between the singers lips and the microphone.
Back in the day, home studios used to make their own Pop shields with a wire coathanger and a pair of ladies tights, but the price of the professional product nowadays is so reasonable, there seems little point.

Beginners guide to recording Acoustic guitar

One of the commonest instruments used in popular music, the Acoustic guitar can also be one of the most difficult to record for a home or budget studio.
In this piece we look at some simple rules and concepts that should ensure you get the best recording possible.
The first thing to remember is that if you get a good initial recording, it saves the amount of processing you have to do at the mix down stage, and this is a GOOD THING!
Acoustic guitars are sometimes referred to by engineers as ‘Jangle Boxes’, and with good reason.
As a rule they send out frequencies from all over the place, and the sound a listener hears when they stand in front of a live acoustic player is a combination of all those different frequencies at once.
So one of the first things to consider is the guitar itself. Make sure it sounds as good as it can before you even begin to record it. Does it need new strings? Is it properly in tune? Does it sound good? Is there any chance of borrowing a better one for the recording. Sounds simple, but it’s very important, if the guitar doesn’t sound right before you start, its unlikely to get any better later on.
Once you have the best sounding instrument you can get, take a look around you.
What is the room doing to the sound?
Is it a big wide expansive room that allows natural reverb and an airy ‘live’ feel, or is it a small bedroom filled with soft duvets and pillows and curtains that will swallow all the signal and leave you with a flat, dead sound going into the mic? Again, these are hugely important considerations, If you capture the sound of a Cathedral echo as part of your recording, its VERY difficult to get rid of it.

Generally speaking the best microphone for the job will be a condenser mic. They are generally much better at handling the higher end frequencies that an acoustic guitar puts out.
The next decision is whether to use a Mic with an OMNI pattern or a CARDIOID pattern. The OMNI pattern will be much better at picking up both the guitar and the sound of the room, whereas the Cardioid will take much more of the guitar, and is therefore a good choice for a smaller room.
There have been millions of pages written about Mic placement, and in truth its one of those esoteric arts that each engineer will have their own theories about. However, there are a couple of simple rules that apply to almost every situation.
Most of us are familiar with seeing the image of a guitar player on stage, with a microphone placed very close to the soundhole, but miking on stage is a different art to miking in the studio.
In the studio, you have the luxury of trying different mic positions, capturing a mixture of the guitar and the room, and we recommend you take some time to do this. If you place the mic too far away from the guitar you risk recording too much room noise, you lose definition and run the risk of increased noise from the mic preamp as you raise the gain. Too near, and you lose the feel of the room.
Generally  you will find a sweet spot in every location, where the blend between the two is best, but as a start, we would recommend aiming the microphone at the point where the Guitar neck joins the body. As a general rule, the closer you get to the soundhole, the warmer the sound, and as you move nearer the neck you get a more treble based signal. As regards the distance, this will depend on all sorts of factors including how hard the guitarist is playing, whether he’s using a pick or fingers etc. but start around a foot away and listen to how the sound varies as you move closer or further away.
Finally, remember that guitarists are a varied bunch, one will sit beautifully still and play accurately with the minimum amount of fuss, while another will flail about wildly, creating creaks from the stool, hands squeaking up and down the strings, and it’s your job to capture the performance complete with all its nuances.

Plywood is cheap and has a shiny side. A couple of sheets 5’ X 3’ are a useful thing to have. You can put them in front of a guitarist to allow the sound to reflect off the shiny side, and this will have a big affect on your recorded sound.
Nashville tuning, is a trick used by some session players to create a really bright and jangly sound. Simply replace the bottom three strings of the guitar (the thick ones!) with another set of top strings. You can then tune the ‘new’ bottom three strings a full octave above where they should be and create a very Jingly sound indeed.