Guide to Setting up your Studio Part 1http://www.focusrite.com/answerbase/en/article.php?id=064

This article is a tutorial-style guide to setting up your system correctly, and eliminating any problems to get the best results from your studio. The two most popular home setups are likely to be similar arrangements to the following:

Fig. 1: Setup 1

Fig.1 shows vocals or instruments being processed through a Focusrite channel strip of some kind and being fed to a desk, which is possibly receiving signals from a keyboard or other devices. Additional Focusrite units may be in use, as inserts, with the desk. The desk then sends to monitors and a recording medium of some kind, most likely a hard disk- or analogue/digital tape-based format. The desk and recording medium may be substituted by a digital multitrack in some cases. Fig.2 is probably the most common configuration, and illustrates a computer-based setup running Pro Tools/Logic or another audio platform. The vocals or instruments are being processed through the Focusrite unit (and possibly being converted by the A/D card) and then going to an interface (Mbox/Digi 002) connecting to a computer. The interface(s) (Audio and MIDI) may be a soundcard within a computer as opposed to an external device. Additional Focusrite units may be in use, as inserts, with the interface. The computer might be sending MIDI information to other equipment (through the same or a separate interface) which is then also feeding the interface. Depending on the front-end Focusrite unit in use, audio may be returning from the interface to the Focusrite device in order to make use of Zero Latency Monitoring.

Fig. 2: Setup 2

The main consideration with either setup is whether audio is to be transmitted in analogue or digital format. ANALOGUE: If analogue is chosen, then the obvious problems are to do with signal degradation, interference and noise. Where does noise and interference come from in an analogue system? Providing the components and circuitry within a studio's equipment produce no unwanted sounds or distortion, the problematic noises on an audio signal are normally caused by inadequate wiring. The main reason for this is that wires are essentially antennas, and are therefore likely to pick up external fields caused by RF sources, electrostatic events or magnetic fields. This effect is reduced by shielding cables with braiding or foil. One of the most irritating interfering currents in larger studios (with long cable runs) comes from the mains supply. These power lines radiate a lot, but fortunately the 50Hz pickup is minimal with small cables, and so shouldn't cause significant problems in most home studios. However, a common annoyance in a variety of audio applications is caused by the use of differing ground sources, which create hum. The cable's shielding that serves to protect against high frequency interference can compound this hum. All systems need to have a ground reference (0V), which is used to compare with the input signal to amplify and process the difference. This reference, generally connected to the chassis or grounding pin of the mains plug, will have all voltages inside the unit measured against it. Problems occur when two pieces of equipment are linked because their references, that are also joined, must then agree. With the cable shield often picking up hum and/or carrying current that results in a voltage drop (unavoidable if it's part of the signal path - as with unbalanced interconnects), there can be a difference in value between the two references. An input circuit may interpret this variation as the signal and amplify it. You may think this would be insignificant, but although the voltage may be small, the impedance is tiny and this, combined with Ohm's law, can make the current high. The hum that results, despite being at 50Hz, is not easily filtered due to the upper harmonics (100Hz, 150Hz...) generated. More complex studio arrangements (especially post-production houses) can develop problems as a result of multiple ground paths combining to produce a loop antenna, which picks up 50Hz noise. These "ground loops" can be difficult to solve, as the internal structure of equipment is not always easily worked out. Balanced cabling was developed in professional audio applications to be immune to these unavoidable occurrences. Probably the most common source of confusion is on the subject of connecting balanced and unbalanced equipment. Many users seem unsure of compatibility and, if compatible, the method of wiring two devices together. So here's a rough guide: What do the terms 'Balanced' and 'Unbalanced' mean? The terms 'Balanced' and 'Unbalanced' (or 'single-ended') are used to describe the kind of electrical interface (and hence cabling required) between devices. The balanced circuit has its electrical midpoint grounded whereas one side is grounded in unbalanced. Unbalanced cables only have two contacts and the 'common' conductor (the shield) is used as a return path for the signal. The chassis/ground references of two pieces of equipment are likely to have

different voltages (see above), causing a ground noise current to be carried by the shield and hence be added to the audio signal. This produces undesirable noise or hum. (This isn't generally much of a problem in home studio arrangements with shorter cable runs and less equipment/power lines...) A balanced cable, however, uses two dedicated conductors for the forward (+) and return (-) paths of the signal. The voltages of the two conductors, at any point in the balanced cable, are equal in amplitude and opposite in phase (180 deg out of phase). Any external interference, causing identical noise/distortion in the two signal carrying conductors, will be ignored at the balanced input stage by a differential amplifier, which only processes the difference between signals. With a 'twisted pair' cable, the two conductors are intertwined and wrapped with the shield to further prevent distortion from RF sources, electrostatic discharge or magnetic fields. The two chassis are then linked together using the shield, which may be an antenna for attracting hum but is not a signal-carrying conductor. So, providing the internal grounding of each bit of equipment is properly designed, no undesirable noise is created.

Fig. 3: Balanced cable

Fig. 4: Unbalanced cable

Ideally, the perfect system should have balanced interconnects between all devices, with cable shields tied to the metal chassis on entry at both ends of the cable. This will guarantee hum-free results and the best possible protection from radio frequency interference and noise. Can Balanced and Unbalanced equipment be connected? Yes, but it isn't ideal and must be done carefully as "blindly" connecting unbalanced with fully balanced can lead to audible problems. As most Focusrite outputs are balanced, on XLR or TRS jack, the following guidelines relate to these instances (balanced output to unbalanced input). The main issues regarding the correct cabling procedures are to do with the nature of the balanced output; whether it's transformer or electronic. With an electronic output, the connecting technique is less important but a transformer output will almost certainly require some cable modification to work. Sending the forward signal (+/live) from pin 2 of the XLR, or the tip of the TRS jack, to the tip of the phono plug or mono jack is always necessary. Connecting the return signal (-) from pin 3 of the XLR, or the ring of the TRS jack, to the shield of the phono plug or mono jack, will give a signal if the output is electronically balanced (Fig 5). However, this signal will be reduced by 6dB. If it's a transformer output there'll be no signal at all.

Fig. 5: Balanced (not transformer) to unbalanced

Fig. 6: Balanced (transformer) to unbalanced

As most outputs are cross-coupled electronic or transformer, the best configuration is Fig. 6. Connecting the shield of the TRS jack, or pin 1 of the XLR, to either the ring of the jack or pin 3 of the XLR (as shown) will result in a signal from the transformer output that is reduced by 6dB, and no loss of level from the electronic cross-coupled output. All TRS Jack balanced outputs of Focusrite gear (which are electronically balanced and cross-coupled) can be connected to an unbalanced jack/phono input of a desk or interface, using a standard mono jack to mono jack/phono cable. This may cause the unbalanced input to be overloaded as the balanced signal output from the

Focusrite will be at line level. To overcome this, either : reduce the output level disconnect pin 1 from pin 3 use a -10 unbalanced jack output where available If connecting an unbalanced output to a balanced input, the connections are more straightforward. Simply join the tip of the phono plug or mono jack to pin 2 of the XLR, or the tip of the TRS jack, and then the shield of the unbalanced connector to pin 3 or the ring, as follows:

Fig. 7: Unbalanced to balanced

If the input happens to be transformer balanced, then either the ring of the TRS jack or pin 3 of the XLR should be connected to the shield of the cable.

Guide to Setting up your Studio Part 2

http://www.focusrite.com/answerbase/en/article.php?id=080

Impedance Issues Some users seem concerned about whether the impedance differences between various inputs and outputs will cause any problems with their signal path. With regards to connecting unbalanced outputs to balanced inputs, there can be some problems due to reduced level. This occurs when an unbalanced unit with high impedance connects to a balanced unit with low impedance. This is the reason why a separate high impedance Instrument input is supplied on many Focusrite products. On the whole though, this is not going to be a major concern of users of Focusrite units, most of which have high or selectable input impedances - and different (mic/line/inst) inputs for each signal. The main consideration in relation to impedance for the Focusrite user is to do with microphone output impedance and pre-amp input impedance combination. Having a low input impedance compared to the microphone output impedance, makes the level lower and emphasises the frequency related variations of the microphone. Matching the impedance reduces the level and SNR by 6dB. Having a high input impedance (normally the desired case) produces the greatest level and overall flattest response. With pre-amps offering selectable impedance, ISA 428/430 MKII and the TwinTrak Pro, the higher impedance values will give an increased amount of top-end, to add an extra clarity and 'shimmer' to recordings. DIGITAL: In order to fully understand problems and regulations surrounding digital audio, it is necessary to have knowledge of the different formats and their transfer capabilities. Digital Audio Formats AES/EBU (Audio Engineering Society/European Broadcasting Union) is a professional format, transmitted with XLR interconnects. Data is sent in blocks of multiple frames, each one having a separate subframe for the left and right channel. Every subframe has sync information (one of three specific 4-bit patterns), signifying that a block or left/right subframe is beginning. These sync words help a decoder establish correct positioning in the data stream, and provide timing information if being used as a clock source for synchronising to other units. Up to 20 bits of audio data are sent per subframe, although the four bits of auxiliary data are now often used to allow 24-bit resolution. 2 channels of digital audio are transmitted on a single XLR cable, with the ability to function at all current sample rates up to 192kHz. SPDIF (Sony and Philips Digital InterFace) is a domestic digital audio format, transmitted with unbalanced RCA (phono) connectors and coaxial cable, or via TOSLINK (optical). The data format is essentially the same as

AES/EBU, however the slightly different usage of just several bits in each data block can make the two interfaces incompatible; some equipment can be very fussy about the data it receives, mainly SPDIF receiving AES. Assuming this isn't the case however (AES device receiving SPDIF), the electrical interfaces can be connected from one to the other (AES/EBU balanced signal at around 5V; SPDIF unbalanced at around 0.5V) fairly simply. For more details, see Connecting SPDIF to AES section below. 2 channels of digital audio are transmitted per single cable, with the ability to function at all current sample rates up to 192kHz. ADAT is a multi-channel interface, used in professional and domestic applications, that is capable of transmitting up to eight channels of digital audio along a single wire. The interfacing cable is optical and known as " lightpipe" or " TOSLINK". ADAT can cope with sample rates up to 96kHz and 24-bit resolution audio data. Connecting SPDIF and AES As mentioned above, the difference in formats is small. One bit at the start of each data block defines what the format is, professional or consumer, and this also decides how the digital audio stream will be read by the decoder. The majority of the code for both formats is identical. However, there is an overlap where 4 bits at the start of a data block used for de-emphasis in AES/EBU, are used for SCMS (copy protection) in SPDIF. This means that SPDIF devices sometimes won't read AES format, but the other way round works more often than not. The only difference then is to do with the nature of the electrical interfaces, going from an unbalanced RCA phono connector to a balanced XLR, and the level of the signal. Some interfaces which only have SPDIF i/o's will sometimes happily connect to AES devices. One example of this was discovered when we at Focusrite linked an Mbox with The Liquid Channel; in this case, the SPDIF signal from the Mbox could be used as a clock source for The Liquid Channel, which in turn successfully sent a digital (AES) signal into the SPDIF input of the Mbox. Simply connecting the units together with a pair of XLR to phono cables should be sufficient in this instance. However, it's impossible to guarantee whether any specific setup will work without testing it first so if unsure, contact the Manufacturer of your equipment. In addition, even if one setup works, an identical arrangement in another location may not. The best and most secure solution is to buy a converter, which can be picked up from Neutrik for around 30. This will allow any consumer or professional digital interfaces to be linked together; a Liquid Channel could be connected to a soundcard with only SPDIF inputs, and a TrackMaster with only SPDIF outputs could send a digital signal to an interface with only AES inputs. The A/D card for the TrakMaster, VM Pro, TT Pro and Penta can output professional format code through a phono ouput, thereby allowing full compatibility with an AES interface. A 'jumper' as circled in blue on this picture of the A/D card defines what format the code will be in. Leaving the jumper on, makes the format professional. Taking it off, makes it consumer. If sending audio digitally between devices in a studio arrangement, the main area that causes problems and confusion is surrounding Clocking Regulations. Here's a guide: When using multiple pieces of digital audio gear, it is necessary to have one clock source for all, so that every audio data stream has identical time divisions. Failure to set up a single clock source can cause audio information to be read at incorrect speeds, leading to the occurrence of audible clicks and glitches. To avoid this, units must be synchronised or 'locked' to a common reference, which is known as word clock. (This can also be a video reference, used mostly in post-production studios.) An external generator can send a clock to all units if numerous digital devices are in use or, in smaller applications, the clock from a single digital audio data stream can be used to synchronise to the receiving device. In this latter case, the sending device is the 'master' and the receiver is the 'slave'. Any units that have digital outputs can become the 'master', without having to have a word clock output, as all digital formats (ADAT, SPDIF and AES/EBU) have timing information in the data stream. So, when should your device be the 'master'? In applications where there is one sending device; if a VoiceMaster Pro is processing a mic/line signal, converting it into a digital format and sending it to a recording platform (e.g. Pro Tools); that sending device can be the 'master'. This is achieved by setting the receiving device to synchronise to its digital input (in Pro Tools, opening the 'session setup' window will allow this configuration to be selected). You must ensure that the correct digital format (ADAT, SPDIF or AES) has been chosen to 'lock' to, and that the bit depth and sample rates are identical. When using this arrangement, the device's internal crystals will be generating the system clock (at whatever sample rate is selected) so there should be no BNC (coaxial) cable connected to the word clock input of the A/D card. When a cable is attached to the EXT WCLK INPUT then the internal clock will be bypassed and the unit will attempt to lock to the incoming signal. And when should you 'slave' to an external source?

When there are two or more devices sending digital audio data; if an OctoPre and MixMaster are both transmitting digital audio streams to a recording platform (e.g. Pro Tools); they must be synchronised with each other by 'locking' to an external device's clock. This can come from a separate word clock generator or the word clock output of a digital device (DAT machine or digital mixing console), and must be connected to the word clock input of both units via a BNC cable. With the OctoPre, in addition to connecting a cable to the EXT WCLK INPUT, the EXT SYNC button on the rear panel must also be pressed for a 'lock' with an external device to be reached. Once again, all devices should be set to identical sample rates and bit depths. The TT Pro is different because it features D/A in addition to A/D conversion, so specific applications will require specific word clock solutions: Additional clocking issues with the TT Pro The TT Pro has more flexibility since it has a built in D/A as well as the optional A/D card. Hence the digital input and digital output may need to be synchronised together. If just using the D/A converter and sending analogue audio, no clocking issues will arise, as the TT Pro will synchronise to the embedded word clock in the SPDIF signal it receives using a phase locked loop. However, if wishing to lock to an additional unit (an A/D converter with a basic word clock spec. for example), a reference can be sent from the TT Pro's word clock output to an additional device. The TT Pro D/A converter does not generate word clock when no digital signal is being received. When using the A/D card alone, the internal crystals provide the desired sample rate unless an external clock source is fed to the WRD CLCK INPUT, in which case the device will lock to the external 'master'. If the A/D card and D/A converter are both in use, the ideal clocking solution in terms of signal performance and noise issues is to link the word clock output of the D/A converter to the word clock input of the A/D card. This is done by connecting a short BNC cable between the two. If using the TT Pro with a recording platform such as Pro Tools (where a digital signal is being sent from the TT Pro to the computer audio interface and the digital output of the computer interface is being fed into the TT Pro for latency free monitoring), the software must be set to lock to its internal clock not the SPDIF signal it is receiving from the TT Pro. If the software is set to lock to the SPDIF input, which is often the default when the SPDIF input is selected for recording, the interface will become confused and high amplitude noise may occur. However, when using an additional item of equipment (synth/DAT machine...etc.) to feed the digital input of the TT Pro, whilst the digital output of the TT Pro feeds the computer interface (with Pro Tools for example), the software should be set to lock to the SPDIF input. ADDITIONAL CONSIDERATIONS How to set up Latency Free Monitoring The Latency Free Monitoring feature on various Platinum products enables delay problems, caused by attempting to listen to a signal being recorded after it has passed through a computer's sound card, to be eliminated. These delays can be fairly substantial due to the digital conversion and processing taking place, making it difficult to sing or play in time with the pre-recorded tracks. By sending the output of the soundcard to the Platinum device, a mix of the pre-recorded material and the mono or stereo signal being recorded can be listened to on headphones or external monitors. Here's how to correctly set up the system: Connecting the outputs of the DAW (from a soundcard for example) to the balanced TRS jack monitor inputs on the rear panel of the Platinum device (TT Pro/VM Pro) sends the signal direct to the monitor outputs, and to an adjustable 'headphone mix' in the latency free monitoring section (playback). The amount of this pre-recorded mix and the mic/line/inst. input(s) to the TT Pro (i.e. whatever's being recorded) can be mixed using the crossfadestyle dial labelled 'headphone mix'. Rotating the dial fully anticlockwise, to 'INPUT', will mean that just the signal being recorded using the TT Pro will be heard. Turning the dial towards 'PLAYBACK' will increase the amount of the pre-recorded signal (from the monitor inputs) on the headphone mix. In addition, the amount of FX (i.e. reverb from a separate device - using the FX send and return) on the headphone signal can be controlled with the 'FX Level' dial, from wet (maximum FX) to dry (no FX); this features on the TT Pro and VM Pro. On the TT Pro, this headphone mix is also fed to two balanced TRS jack outputs on the rear panel, for the option of sending to an external headphone distribution amplifier. If the 'headphones to monitor' switch is engaged, the headphone mix will be sent to the monitor outputs, if not, simply whatever signal feeds the monitor inputs will be going direct to the monitor outputs. As the VM Pro has no "headphones to monitor" switch, the monitors can only be fed the signal coming from the monitor inputs; plugging in the headphones in order to set-up a zero latency headphone mix will mute the monitor outputs. Metering and Decibel levels

There are two main metering methods in audio, one that displays the average level of the signal and one that shows the peak level. The former is called a VU (Volume Unit) meter, and was primarily used when recording to analogue tape. As a result, the VU Meter's values are designed to help achieve an 'optimal' level to tape, displaying a scale from -20 to +3dB where the last 3dB are in red. 0dB on the VU meter corresponds to 4dBu. Just to clarify, dBu is a voltage reference point equal to 0.775Vrms. This reference was originally labelled dBv (lower case) but was often confused with dBV (uppercase) so it was changed to dBu (for un-terminated). dBV is another voltage reference point equal to 1Vrms, which some engineers find easier than 0.775Vrms and is used on the -10dBV unbalanced outputs on the Platinum range. It's worth mentioning that dBu can cause confusion sometimes when given as the scale on a threshold for a compressor (as on the MixMaster for example), as users may be used to having values from 'full scale' (maximum level, 0dBFS, digital clipping point) downwards (-40dBor so). In dBu, this will be marked on the dial as around +20dB down to -20dB. ALL ISA AND RED RANGE UNITS HAVE VU METERING. The other type of metering was introduced because VU meters aren't ideal for displaying the peak level, something that's crucial in situations where overloading must be avoided, when broadcasting or converting to digital for example. The PPM (Peak Programme Meter) scale varies depending on the country. In the UK, there are no decibel markings, instead there are indications from 0 to 7. Most marks have 4dB between them apart from 0 and 1, which have 6dB, and 7 and 8 (8 not displayed), which have 2dB. On a UK PPM meter, 4 relates to 0dBu. The scale used for digital conversion and more commonly with processors today (Platinum products for example), is the dBFS ('full scale') range. This provides 0dBFS as the threshold for digital clipping. Above this, the signal cannot be given a coded value and will clip, causing the audio to distort. An O/L LED is normally provided to indicate when the limit has been exceeded. Focusrite meters generally range from 0 down to -20 or -30dBFS. MOST PLATINUM PRODUCTS HAVE METERING WITH A dBFS SCALE (ALL EXCEPT TONEFACTORY, VOICEMASTER AND COMPOUNDER, WHICH HAVE VU). Simultaneous use of Analogue and Digital outputs When using Focusrite products, the analogue and digital outputs can be used simultaneously. HOWEVER, using simultaneous digital outputs on the ISA 430 MkI and Platinum MixMaster digital cards is not recommended as it will damage the transmitter IC. REMAINING FAQs International Transfer and Fuse changing VM Pro and OctoPre (only) have two different product versions, a 110/120V version and a 220/230V version. Hence a 110/120V unit can be used in US/Japan, with the correct primary tap connected, but cannot be used in Europe (for example) unless a step-up transfomer is purchased and used. There is no cost-effective way to modify VM Pro's or OctoPre's to allow 110/120V units to run on 220-240v mains supply or vice versa. All other Focusrite Platinum units except these two models can operate as 110/120/220/230V units, and the user can switch the voltage simply by rotating the external fuse holder. This can easily be done by using a very small screwdriver to lift the fuse holder away from the rear panel, and choosing the required voltage text on the fuse holder cover to be facing the right way (aligned with the arrow). When "120v" is positioned so that the arrow points to it, the unit is set to 120v and suitable for use in the USA. The user will of course need to obtain a mains cable of the correct type to match local plug format, and the fuse value must be correctly set BEFORE switching the unit on! The Red3 and Blue 230 require a special transformer for operation at 220V AC and a very special transformer for operation at 100V AC.

Guide to Setting up your Studio Part 3http://www.focusrite.com/answerbase/en/article.php?id=081 Fig. 8: Table of fuses required for Focusrite units Focusrite device Red 1 Red 2 Red 3 Red 6 Red 7 Red 8 Blue 230 Blue 315 Blue 330 ISA215 ISA110LE ISA430 ISA220 ISA428 ISA430 MKI & MKII ISA828 VoiceMaster ToneFactory Compounder MixMaster Penta TrakMaster OctoPre VoiceMaster Pro TwinTrak Pro Liquid 4Pre Liquid ChannelPlatinum A/D Card 'Jumpers'

Fuse 230/220V AC T250mA anti-surge (Time T250mA anti-surge (Time T250mA anti-surge (Time T250mA anti-surge (Time T250mA anti-surge (Time T250mA anti-surge (Time T250mA anti-surge (Time T250mA anti-surge (Time T250mA anti-surge (Time T250mA anti-surge (Time T160mA anti-surge (Time T500mA anti-surge (Time T500mA anti-surge (Time T500mA anti-surge (Time T500mA anti-surge (Time T500mA anti-surge (Time T160mA anti-surge (Time T160mA anti-surge (Time T160mA anti-surge (Time T315mA anti-surge (Time T160mA anti-surge (Time T160mA anti-surge (Time T315mA anti-surge (Time T315mA anti-surge (Time T315mA anti-surge (Time T500mA anti-surge (Time T500mA anti-surge (Time

lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag) lag)

Fuse 115/100V AC T500mA anti-surge (Time lag) T500mA anti-surge (Time lag) T500mA anti-surge (Time lag) T500mA anti-surge (Time lag) T500mA anti-surge (Time lag) T500mA anti-surge (Time lag) T500mA anti-surge (Time lag) T500mA anti-surge (Time lag) T500mA anti-surge (Time lag) T500mA anti-surge (Time lag) T315mA anti-surge (Time lag) T1.6A anti-surge (Time lag) T1.6A anti-surge (Time lag) T1A anti-surge (Time lag) T1.6A anti-surge (Time lag) T1.A anti-surge (Time lag) T315mA anti-surge (Time lag) T315mA anti-surge (Time lag) T315mA anti-surge (Time lag) T630mA anti-surge (Time lag) T315mA anti-surge (Time lag) T315mA anti-surge (Time lag) T630mA anti-surge (Time lag) T630mA anti-surge (Time lag) T630mA anti-surge (Time lag) T1.A anti-surge (Time lag) T1.A anti-surge (Time lag)

The optional A/D card for installing into Platinum units has several 'jumpers' (small rectangular plastic clips), which have differing effects on the action of the converter when removed. On the diagram below, the jumper circled in BLUE changes the digital output between professional and consumer format. (This is only a feature on A/D cards for the Penta, TrakMaster, VM Pro and TT Pro - the MixMaster and OctoPre have designated outputs for AES/EBU.) Leaving it on results in a professional format digital output that can be connected to an AES/EBU input if the correct cable is used (RCA (phono) to XLR). Taking it off makes the digital output a standard consumer SPDIF format. The four jumpers circled in RED change the level of 0dBFS, the 'full scale' level at which digital clipping occurs. These jumpers should be left on if the card is in use with a Penta but removed if in use with any other Platinum unit (TrakMaster/VM Pro/TT Pro). If left on with any of the three latter units, the clipping level will be 6dB down, which is unlikely to cause significant problems but may result in minor distortion if the signal is 'hot'.

Fig. 9: Platinum optional A/D card jumper specification

Jumper J1 changes the termination Z of the Ext word clock input from 75ohms to HiZ. This is should be done when 'daisy chaining' several devices together.Pinout for OctoPre

To enable correct wiring of the Analogue and Digital outputs of the OctoPre, should you be making or modifying your own cables, here are the diagrams indicating the specifications and pinouts for both:

Fig. 10: Pinout for OctoPre's Digital o/p's

Fig. 11: Pinout for OctoPre's TASCAM 8 channel Balanced Analogue o/p