Unit 2: Audio Engineering Principles - Dream Studio Setup

The application of my home recording facility is to have a resonably diverse setup, in which I could record a full drum kit at most, but be geared to overdub recordings and not record full bands at once, but have the option to record electronic music and synthesizer based stuff as well, so really a hybrid of two specific setups to meet my needs. This would give me complete creative flexibility over the genre of music I wanted to produce or collaborate on, as it could fuse genres like rock and electronic together, or instead do one or the other. This makes sense for me as I like all sorts of music, and often want to try different things, so I want to create a setup with no real limitations so I'm never stifled creatively.

Equipment List:
The Key equipment for my studio is as follows:

• iMac & Logic Pro 9 - Standard model iMac computer which is capable of running hosting audio projects with its capacity and hosts Logic Pro 9 which is the sequencer I will be using. I'm using Logic as I need a hybrid of audio and MIDI suitability, and though Garage Band maybe better for audio I feel that with the MIDI data I want to use to send to a synth or make sequence parts that Logic will be the best option as I know it has good options to work with either from experience using it.
• Allen & Heath WZ4 - I'm using this analog desk to set levels and get the signal from mic level to line level before sending it to the interface. This is because I like the process of setting levels and am reassured that my procedure to setup for a recording is correct as I know all the functions of this desk. It's also useful as it has auxiliaries and a headphone output so I could send a click track from my Mac, through the interface in to the desk and use the aux and one of the spare channels to get the click to a drummer to play to. Every channel of this desk has phantom power, which is what's needed to provide power for condenser microphones. This is very useful as my RME doesn't have this feature, so it means I can arrange my condensers in any position on the channel list and use any channel of my desk without encountering problems. As my setup is for overdubbing recorded parts, with the foresight to have a setup that one person could utilise themselves, the drums is the only reason I'd need a click track, as everything after that would be played to the drums, and if I was beginning with a software instrument or kit this would be played in MIDI and aligned to the grid so the click wouldn't be neccessary. The desk has 16 channels with direct outs, of which I'd only be using 7 at most which means the line out of the desk can use the 7 of the 8 available line ins on my interface with an eighth for the signal from the liquid channel.
• RME Fireface UCX - This interface has enough audio I/O for my needs, and is highly regarded for good signal conversion which is the key function of an interface. It also has balanced line outputs, so I will go balanced jack to XLR to send the recorded audio to my monitors to listen back and mix. Using balanced cables means less impedance due to more wires for the signal to travel down and low noise with no risk of interference like unbalanced cables. Another key feature of the interface is a SPDIF input, which I'll use for my separate liquid channel pre-amp. This is because it's a digital processor, so I can send my analog signal to the liquid channel where it's converted and processed, and then rather than having to convert it back and lose quality potentially, I can keep it digital and put it through the SPDIF then into the sequencer.
• Neumann KH120A - These monitors were featured in a Sound on Sound article on active monitors and were highly regarded which lead me to believe they'd be worth considering for my studio. Firstly as active monitors they generate their own power from mains power rather than needed more equipment to provide it with power at a certain level which costs more, and also they're smaller than average but still "pack a punch" as SOS mention in their article, which is great for a home setup as space maybe invaluable, so if they perform well as near field monitors and are small they're ideal for the setup I need. They can also be bracketed to the wall which is another interesting option and space saver, and gives me options when considering how to arrange my desk, mac and monitors to create a comfortable and practical work space. They have a balanced XLR input which is quieter than jack's meaning there's no noise interference when mixing and listening back. They have switches to tailor the frequency response to the room properties so you aren't hearing any accentuated frequencies by your rooms properties and the placement in the room. They suggest how to set these according to your room which is great if you don't know how to calculate your room properties. There's a monitor volume switch too which again should ensure that the volume isn't too loud for the space and overwhelming when trying to mix. There frequency response is 54Hz to 20,000KHz which almost covers the range of human hearing, except in the very low bass which would in most cases be lower than any frequency in a mix, but also there's a subwoofer pair to be bought separately that go with these speakers if deemed necessary. All in all I think these are the best of both worlds for my setup with their small size and large fq response.
• Focusrite Liquid Channel - This is my chosen dedicated vocal processor for getting a nice sounding vocal rather than using the sequencers reverbs, compressors etc. It's a digital processor so there's no noise added as a bi-product of the processing and it has many preset reverbs and compressor emulations to act like vintage analog models to achieve the sound I'm looking for on any particular track.
• The most I will need to record in one take will be a drum kit, for which I'd need a mic for the kick, snare and two overheads, and I may double up my vocal mic as a fifth option in front of the kit to get some ambient sound or a picture of the whole kit. So my microphone list is as follows, with brief justification for my choices:-
• AKG D112 - To record my kick drum/bass amp or bass synth through an amp if necessary. It can record frequencies as low as 20Hz so gives detailed bass, is robust and can withstand high SPL's, finally it's an industry standard mic for this purpose and is a feature in the college studio so I know first hand that it works well for the purpose.
• Shure SM57 - For my snare, and guitar amps. Dynamic mic with 40-15KHz range and tailored frequency response curve for clean reproduction of acoustic instruments and amplified sounds. It's another industry standard mic I've had experience with at college so I know it works well.
• Neumann KM184 Pair - For drum overheads, and acoustic instruments like acoustic guitar, percussion etc. The stereo pair make great overheads as I've used them in college on recordings, and they'll be useful for other instruments too. These microphones are very quiet and don't produce any audible noise from their internal parts so pick up subtleties and detail of acoustic instruments well, and have a rise at 9Khz point in the frequency response which adds warmth to mid range frequencies and accentuates detail there. They don't have to withstand hight sound pressure levels so this won't be an issue, and otherwise they're great for multiple recording purposes in my studio.
• Rode NT1-A - This will be my microphone for recording vocals, as it a fixed carded pattern, large diaphragm condenser which will record clear vocals, and as long as it's used in a deadened space (with dense domestic objects in the home setup) should provide great results to then be digitally processed according to the style of vocal and the genre of music, for example in a Psychedelic Rock track the vocal may have any mix of reverb, phase, delay or vocoder-esque modulation on so the key will be to get the good dry sound first to then add these effects to best affect afterwards.
• Finally I would need about 8-10 XLR to XLR cables for these, some of which are backups in case of faulty ones, and at least 4 tall microphones stands for use on overheads, vocals, snare drum 

Other Instruments & Synthesizers/ Outboard Processing:

As I said I want a studio with no limit to the style or genre I could work in I would therefore want an assortment of analog synthesisers, electric keyboards and outboard processing to both create and process sounds. For all of these I would could use a patch bay which would act as a port for all of the connections into my desk but in one place away from it so cabling is tidy, however as I have a small desk, and interface with limited ins and outs and plan to overdub recordings, this wouldn't be necessary and instead I'd plug directly into the desk as and when I needed to use each piece of equipment. As the liquid channel is my main outboard processor for vocals, the other crucial outboard piece is a reverb unit which for example I could send all my drums to and add a reverb to make them sound unified and played in one particular space. As synths I would like the Korg Volca series as they give me percussion, baselines and melodies to create a complete electronic spectrum of sounds and rhythms which could create an electronic track alone. Also vintage electric keyboards would be excellent like Fender Rhodes or Farfisa organ, which I would play through an amplifier mic'ed up with SM57 and the large diaphragm condenser as a room mic. All these connections are analog, so would go XLR to my desk and then onto my interface to be converted to digital and via USB into the sequencer. Finally my MIDI keyboard is how I will control software instruments and MIDI data, and quickly get ideas for parts into the sequencer as I record.

Connections/Signal Flow Diagram:
This diagram explains all the connections between my equipment, what cables are required to make them and whether signal is analog or digital at each stage of the setup. Firstly, the analog desk is where all my analog signals from microphones are first connected. This is where I set levels to get the best sound to noise ratio, and provide the condenser microphones with phantom power. At tho stage all the signal are analog and aren't converted at any point in the desk before going through the line outputs at line level. It then enters the interface via jack at the same level, is converted to digital data and all sent down the USB as data to the sequencer in the mac. My seperate vocal processing side has analog signal enter the focus rite where it's converted to digital to be processed and instead of then converting it back to send it to the rme to convert again which would degrade the signal power and clarity, I go out digitally the AES output to a Hosa digital interface which then gives me a SPDIF output which the Focusrite doesn't have, to go via RCA cables into the RME, already as a digital signal which doesn't need converting.

Digital Conversion:
Digital conversion is a process that involves taking an analog signal and making a digital recreation of it, the accuracy of which is defined by the sample rate and bit depth of the conversion and indeed the quality of equipment doing the conversion. As mentioned in task 1, the bit depth and sample rate are what define the quality of the conversion, as a higher sample rate means more samples of the analog wave are taken per second than would be at a lower rate which equates to greater detail, and for each sample the bit depth is how many volumes the sample could be interpreted at, so greater bit depth means a more accurate representation of the sound, but also more headroom with levels and more stages for low level noise meaning even quiet sounds have detail and accuracy. My RME interface converts at 24bit and 96KHZ, which is an optimum level for conversion, it could go higher but the difference would be inaudible, they key is that it converts at 24bit and therefore has far more stages or volumes than 16bit conversion, and the rate is more than double the red book standard, but is exactly double 48KHz which is DVD audio quality, as the Nyquist theory suggests this rate should double.


Balanced and Unbalanced cabling:
An unbalanced cable or TS (standing for Tip Sleeve) is a cable containing one core signal carrying wire, with a casing around it which acts as the return carrier. The conductive metal surrounding the core  can carry a signal which could interfer with the wanted signal going through the core, which may alter its sound. There's also a problem with low level noise in unbalanced cables which comes about in the same way, where the signal from the return path affects the main signal, it leaves a quiet version of itself on the main signal which sounds like a low level noise. Where balanced cables differ is that they have 2 cores for the same signal to travel down, with one phase inverted, both surrounded in the protective screen but this importantly isn't part of the signal path as in unbalanced, so it can't add the low level noise in with the main signal. The phase inversion at the input to the cable means that whatever interference effects the hot signal will be cancelled out by the opposite in the cold signal, so when the inverted signal is inverted back to normal and the two are re-joined at the output there is now noise or interference to the full signal. So the balanced (TRS) cables ultimately result in a better signal quality than unbalanced and are therefore crucial in the studio setup, for example it is really important for cables going to monitors from the interface as low level noise could have a negative affect when mixing.

Data Storage & Retrieval:
There are many options for storage of projects to backup the main drive. Some of these are external and use hardware and other are more modern solutions like cloud storage. In my studio I will backup all my recordings, files and software to an external hard drive and store this away from my studio space so it cannot be stolen along with my mac, as then it would be useless. I would also investigate putting my files onto apple cloud storage, which is a cloud server meaning that you upload your files to it via the internet, and retrieve them by logging in over the internet and downloading them. This is very secure, and cannot be stolen or damaged, but there's no guarantee it will be 100% safe as apple could have server problems of their own with the storage. The external hard drive will be connected to my mac via USB to put files onto it, and take files off it. My mac has no disc drive so DVD and CD storage isn't a viable option.

Operating Levels/Signal Types:
Operating levels are electrical current levels used to power recording equipment and configure the signal gauges (VU meters) of old analog recording desks. Devices have nominal levels they're intended to operate at, and in the case of desks with VU meters, the signal level that calibrates the meter in dB is used to make sure the needle is at 0dB, so when setting levels its guaranteed they're getting a good sound to noise ratio with enough headroom. With digital or A/D conversion equipment the operating level is relative to the level of signal it can process, while keeping headroom intact, and the difference between professional and domestic equipment operating levels (+4dBu compared with -10dBu) is an audible difference of 12dB with therefore a relative signal difference of the same amount. This means if I we're daisy chaining between equipment intended for domestic levels to professional ones, I'd see a drop in signal level and output level of this size, or risk overdriving my domestic level equipment as compensation. In my setup, my RME interface inputs are changable levels of +4dBu and -10dBu, aiming to work compatibly with either, this does mean that at each level the dBfs (decibels full scale) output varies so although the RME has the advantage of working at either level, it doesn't resolve the problem of drastic signal changes altogether. This is all in relation to line level signal, the loudest or strongest level signal of mic, instrument and line levels. At mic and instrument, the noticeable signal difference means they'd have to boosted to line before beginning conversion in the RME, which is where the channel strips of the desk come into use, as they take a microphone level signal via XLR of a microphone recording a voice for example, and in the channel the signal is boosted and can be processed with EQ settings etc. before being outputted at the new line level, having been boosted in the desk so they've a loud enough input level for my RME. The maximum input level of my RME is +19dBu, so unless I took a signal of +4dBu and massively added gain to beef it up, I wouldn't have any problems with operating levels, or signal types.

Audio Formats:
There are many types of audio formats, but all come either in the category of lossy or lossless audio. Lossy audio is any format where the audio is compressed to reduce the file size meaning it takes up less space in storage, and the downside of this is that some audio detail is lost, but it is done in such a way that it isn't so obvious to the listener that certain information is missing. As lossless then is the opposite of lossy and is any format which is completely uncompressed with no information removed. An example of a Lossy formats is MP3 and a lossless is AIFF.