User manual FOCUSRITE VOICEMASTERPRO

ENGLISH CONTENTS CONTENTS................................................................................................................. 2 IMPORTANT SAFETY INSTRUCTIONS .............................................................. 2 INTRODUCTION...................................................................................................... 3 GETTING TO KNOW THE UNIT........................................................................... 3 REAR PANEL CONNECTIONS .............................................................................. 3 GETTING STARTED ................................................................................................. 4 FACILITIES AND CONTROLS ................................................................................ 5 DISCRETE CLASS A PRE-AMP........................................................................... 5 OPTICAL EXPANDER.......................................................................................... 6 VINTAGE HARMONICS...................................................................................... 6 OPTICAL COMPRESSOR.................................................................................... 7 TUBE SOUND........................................................................................................ 8 VOICE OPTIMISED EQ ........................................................................................ 9 DE-ESSER................................................................................................................ 9 OUTPUT SECTION ............................................................................................ 10 LATENCY-FREE MONITORING .................................................................... 10 DIGITAL OUTPUT OPTION............................................................................. 11 OBTAINING GOOD QUALITY SOUND ............................................................. 12 CORRECTING PROBLEMS................................................................................... 12 A BEGINNER'S GUIDE TO COMPRESSION...................................................... 13 A BEGINNER'S GUIDE TO EQUALISATION..................................................... 15 FREQUENTLY ASKED QUESTIONS ................................................................... 15 TROUBLESHOOTING............................................................................................ 17 CONTACTING US ................................................................................................... 17 SPECIFICATIONS..................................................................................................... 82 FOCUSRITE DISTRIBUTOR LIST ....................................................................... 86 RACK VENTILATION: AS THE VOICEMASTER PRO IS A CLASS A DEVICE, PLEASE ENSURE IT IS PLACED TOWARDS THE BOTTOM OF YOUR EQUIPMENT RACK, WITH SUFFICIENT SPACE ABOVE AND BELOW FOR VENTILATION. This unit is supplied pre-configured to operate only at the voltage indicated on the rear panel. Ensure correct mains voltage is available and the correct fuse value is fitted before connecting to the mains supply. To avoid the risk of fire, replace the mains fuse only with the correct value fuse, as marked on the rear panel. The internal power supply unit contains no user serviceable parts. Refer all servicing to a qualified service engineer, through the appropriate Focusrite dealer. · IMPORTANT SAFETY INSTRUCTIONS Please read all of these instructions and save them for future reference. Follow all warnings and instructions marked on the unit. · · · · · Do not obstruct air vents in the rear panel. Do not insert objects through any apertures. Do not use a damaged or frayed power cord. Unplug the unit before cleaning. Clean with a damp cloth only. Do not spill liquid on the unit. Ensure adequate airflow around the unit to prevent overheating. As this is a Class A unit, we recommend leaving a blank 1U panel above the unit to aid ventilation. Unplug the unit and refer servicing to qualified service personnel under the following conditions: If the power cord or plug is damaged; if liquid has entered the unit; if the unit has been dropped or the case damaged; if the unit does not operate normally or exhibits a distinct change in performance. Adjust only those controls that are covered by the operating instructions. Do not defeat the safety purpose of the polarised or grounding-type plug. A polarised plug has two blades with one wider than the other. A grounding type plug has two blades and a third grounding prong. The wider blade or the third prong is provided for your safety. When the plug provided does not fit into your outlet, consult an electrician for replacement of the obsolete outlet. WARNING: THIS UNIT MUST BE EARTHED BY THE POWER CORD. UNDER NO CIRCUMSTANCES SHOULD THE MAINS EARTH BE DISCONNECTED FROM THE MAINS LEAD. 2 ENGLISH INTRODUCTION The VoiceMaster Pro is a combined, high-performance microphone pre-amplifier, dynamics processor and equaliser. Although it has been specifically designed to enable the user to set up a great vocal sound, the VoiceMaster Pro is flexible enough to also be used when recording and mixing down a range of other instruments, such as guitars or drums. When recording, do not assume you must route your signal through a mixing desk: simply connect a microphone to the VoiceMaster Pro and connect the output of the VoiceMaster Pro directly into your sound card or recording device. This form of direct recording will ensure you record the cleanest signal at the highest quality, since it removes the possibility of noise being added to the signal when routing through a mixer. There are seven separate signal-processing sections in the VoiceMaster Pro: · Discrete Class A Pre-amp · Optical Expander · Vintage Harmonics · Optical Compressor · Tube Sound · Voice Optimised EQ · De-Esser To ensure the cleanest signal path to your recording medium, each section can be individually switched out of the audio path (`hard bypassed') when not in use. There is also a global PROCESS BYPASS control. GETTING TO KNOW THE UNIT When you are getting to know the unit, use it with a sound source with which you are familiar. For example, you could run a favourite CD through the unit, as working with a familiar track makes interpretation of the results easier. (Note, however, that tracks are already compressed for CD, so you may find it hard to hear the results of using the Optical Compressor.) If this is the case, try using samples instead, or record your own track uncompressed and then play it back through the VoiceMaster Pro. The easiest way to understand the creative power of the VoiceMaster Pro, particularly if you are not familiar with each of its separate parts, is to switch in each individual section, and try each control in turn. Finally try them all together to hear the creative power of the VoiceMaster Pro! REAR PANEL CONNECTIONS The VoiceMaster Pro features MIC INPUT XLR and LINE INPUT TRS connectors on the rear panel, with the MIC INPUT duplicated on the front fascia together with a 3 ENGLISH quarter inch jack INSTRUMENT INPUT. The INSERT SEND and INSERT RETURN quarter inch jack connectors on the rear panel allow an additional processor to be inserted into the signal chain between the pre-amp and the other processing sections of the unit. Outputs are provided on both XLR (balanced, +4 dBu) and quarter inch jack (unbalanced, -10 dBV), and there is also a PRE DE-ESSER output XLR (balanced, +4 dBu). Additionally, the VoiceMaster Pro includes a mono, unbalanced FX SEND and balanced stereo FX RETURN with quarter inch jack connectors, to allow an effects unit to be monitored, together with balanced stereo EXT MONITOR INPUT and MONITOR OUTPUT connections on quarter inch TRS jacks. See the LATENCYFREE MONITORING and the EXT MONITOR INPUT/MONITOR OUTPUT sections on pages 10-11 for further details. Finally, the ADC EXT INPUT (balanced, quarter inch TRS jack) allows an external signal to be routed to the second channel of the optional digital converter card. This allows the digital converter to be used for stereo mastering applications, as a standalone A/D converter, or simply to provide a second (line level) record input. For information on the optional digital output board, see page 11. GETTING STARTED 1. Ensure that nothing other than the mains supply is connected to your VoiceMaster Pro, then switch it on via the POWER switch on the right hand side of the unit. If your unit is permanently connected to a patchbay, ensure audio is not being fed to any connected speakers to avoid any turn-on speaker pops. 2. Connect the appropriate OUTPUT (either +4 dBu balanced XLR or ­10 dBV unbalanced jack) on the rear panel of the VoiceMaster Pro to your recorder or audio interface. If using the digital output option, connect the digital output to the digital input of your recorder or audio interface. See page 11 for more information on the VoiceMaster Pro digital output option. 3. Connect the MONITOR inputs and outputs to your external kit/monitor speakers. 4. Ensure that each processing section is switched out (IN switch disengaged and unlit), and that the PROCESS BYPASS switch is also disengaged (out). 5. Connect your input source as required. A microphone can be plugged into the XLR MIC INPUT on either the front fascia or the rear panel. If you wish to connect a line-level source (to use the VoiceMaster Pro's dynamics processing when mixing down, for example) connect this to the TRS LINE INPUT on the rear panel. Alternatively, you may connect an electric guitar or bass to the INSTRUMENT INPUT via the unbalanced quarter inch jack input on the front fascia. 6. Check that the correct input is selected in the DISCRETE CLASS A PRE-AMP section. If recording a line level source connected to the rear panel LINE INPUT, ensure the LINE switch is engaged. If a microphone is connected to either of the MIC INPUTS, or an electric guitar or bass is connected to the INSTRUMENT INPUT, ensure the LINE switch is disengaged. 7. Make sure the phase reverse (Ø) and HPF ( ) switches are disengaged and that the INPUT GAIN control is fully counter-clockwise. Set the OUTPUT FADER to the `0' position. 8. If using a condenser microphone that requires phantom power, engage the +48V switch. If you are unsure whether your microphone requires this phantom power, refer to its user guide. Phantom power can damage some microphones, especially ribbon microphones. 9. Increase the INPUT GAIN control, checking the input level meter LEDs and ensuring the red O/L LED does not illuminate, except occasionally and briefly when the loudest signal is present. 10. If using a microphone, ensure that the microphone placement is at its best. Before you start recording, alter the microphone placement until you get as close as possible to the sound you want. Note that moving the microphone may have an effect on the level of the signal entering the VoiceMaster Pro, requiring an alteration to the INPUT GAIN setting. 11. Add additional processing as required using the various signal-processing blocks, which may be individually switched in or out. For more information on the specific functions of the various sections, refer to the following section, FACILITIES AND CONTROLS. 4 ENGLISH FACILITIES AND CONTROLS - Turns the unit on. We recommend that the unit be powered up before connecting to any equipment that it is feeding, to avoid clicks or thumps which may harm output devices. It is also a good idea to allow the unit to stabilise for a couple of minutes before use to ensure that the internal circuitry is properly initialised. POWER (switch) - This provides +48V of phantom power for condenser microphones (affecting the MIC INPUT only.) If you are unsure whether your microphone requires phantom power, refer to its user guide before connecting, as it is possible to damage some microphones (most notably ribbon microphones) by providing them with phantom power. +48V (switch) DISCRETE CLASS A PRE-AMP - This is a high impedance 1/4" jack input that allows you to connect an electric guitar or bass guitar to the unit without loading the pickups, and without the need for a DI box. If both the mic and instrument inputs are connected, the instrument input will override the mic input. INSTRUMENT INPUT This part of the unit is a pre-amplifier, used to amplify the incoming signal being fed to the MIC INPUT or INSTRUMENT INPUT to a suitable level before any further processing is applied. - This allows the phase of the input signal to be reversed e.g. to correct phase problems when incorrect wiring polarity has occurred. Ø (Phase Reverse switch) - When engaged (in), this switch selects the rear panel LINE INPUT, and an LED is illuminated in the switch cap to indicate that the LINE INPUT is active. If this is disengaged (out), the MIC INPUT and INSTRUMENT INPUT are active. LINE (switch) (HPF knob and switch) - This is a high-pass filter, which removes unwanted low frequencies such as stage rumble via microphone stands, or 'proximity effect' (where low frequencies are over-emphasised when using certain types of microphone at close range). The knob sets the cut-off frequency (from 30 to 400 Hz, 18 dB per octave), and the switch must be engaged (in) for the control to function. - This is used to set the optimum input signal level. Connect an input signal to the unit, ensuring that the INPUT GAIN control is set fully counter-clockwise, and increase the INPUT GAIN control whilst observing the LED signal meter. The red O/L (overload) LED may light occasionally, but only if the input signal gets particularly loud. If the O/L LED stays on continuously for any period, or you hear the unit distort during loud peaks, you should reduce the INPUT GAIN. INPUT GAIN (knob) - This is an XLR connector that allows you to connect a microphone to the unit. There is also a MIC INPUT XLR on the rear panel, but only one may be used at a time; do not connect both MIC INPUTs simultaneously. If using the VoiceMaster Pro's mic pre and feeding the output into a mixing console, bypass the console's own mic pre and connect to the channel's line input. This will mean the superior VoiceMaster Pro mic pre is used to route signal to its destination, e.g. a recording device, avoiding unwanted distortion and colouration from an inferior mic pre. Always avoid routing the VoiceMaster Pro's mic pre into a second mic pre, as this will produce greatly inferior results. MIC INPUT Note that the meter is calibrated to read 0 dBfs at the top of the meter- this has been set up to enable simple metering when recording to digital media. The best level to set for recording depends on your recording medium. If recording to an analogue medium like tape, where extra headroom is required, a level of ­18 dBfs will give a suitable +4 dBu equivalent output. If recording to digital media, you may wish to record at a higher level, peaking at e.g. ­4 to ­6 dBfs. Confused? Visit www.sospubs.co.uk/sos/may00/articles/digital.htm for further illumination. With the MIC INPUT selected, the INPUT GAIN control provides 0 dB (fully counter-clockwise) to +60 dB (fully clockwise) of gain. With the INSTRUMENT INPUT selected, the INPUT GAIN control provides +4 dB to +34 dB of gain. With 5 ENGLISH the LINE INPUT selected, the gain is adjustable from ­10 dB to +10 dB. Setting the LEVEL control to the 12 o'clock position will not alter the gain of a line level input signal. - The INSERT connectors on the rear panel allow an additional external signal processor to be inserted into the signal chain between the pre-amp and the other processing sections of the VoiceMaster Pro. Connect the INSERT SEND to the line input of the external processor, and connect the processor's line output to the INSERT RETURN. Like all inserts, the signal returned to the INSERT RETURN jack will rejoin the signal path at the same point from which it was sent, after the benefit of external processing. INSERT (rear panel connections) - Determines the signal level at which noise reduction begins. The higher the threshold, the more low-level noise is reduced. The range is ­60 dB to ­20 dB THRESHOLD (knob) - Determines the time taken for the gain reduction to return to normal once the signal exceeds the threshold. The range is 0.25 seconds to 4 seconds. RELEASE (knob) - Shows how much noise reduction is being applied, and should light progressively during quiet passages. During louder sections, the lights should go out. By watching the meter when a vocal passage is starting and finishing, you can check that the OPTICAL EXPANDER is not affecting the vocals ­ if the meter stays lit during the vocal, reduce the THRESHOLD control. GAIN REDUCTION (LED meter) OPTICAL EXPANDER The OPTICAL EXPANDER reduces the volume of quiet sections in the performance, by reducing the gain of the signal when it falls below the threshold set by the user. This is a similar principle to a noise gate, but rather than muting the signal altogether, an expander simply `turns the volume down'. Use it to get rid of background noise, either while recording, (for example, getting rid of bleed from headphones into the microphone,) or while mixing down, (for example, getting rid of tape hiss). The expander has a gentle noise-reducing effect: you can set it so that it reduces background noise without affecting the beginning and end of vocal passages. VINTAGE HARMONICS An all-new enhancement tool, this section simulates the original-tape based enhancement method for vocals used by many famous engineers and producers during the 1970s. A historical footnote: many tape machines came with DolbyTM noise reduction units which compressed the signal into the quiet region of the tape track. During playback the DolbyTM would then expand the signal back to its original dynamic range. Some enterprising producers and engineers found that if vocals were recorded with the DolbyTM unit switched on, (compression applied relative to frequency,) but were then played back with DolbyTM unit switched off, (with no expansion,) a pleasing emphasis was added to the source material. The Focusrite VINTAGE HARMONICS section reproduces this effect during record by compressing (relative to frequency) all signals below the threshold point for both mid- and high- frequency bands. The original technique relied upon the skill of the engineer to vary the input level to tape to get the desired effect from the DolbyTM unit, which only had a single fixed threshold point. The Focusrite VINTAGE HARMONICS however, has the added benefit of variable threshold for both bands, plus a depth switch so the effect can be simply tailored for any voice. The VINTAGE HARMONICS section splits the audio into three bands: Low Band (below 100 Hz this band is never affected or controlled by the Vintage Harmonics - buy a Focusrite Compounder if you need to hear bass as you have never heard it before!), Mid Band, and High Band. How the VINTAGE HARMONICS threshold control works This control is calibrated from ­40 dB to ­10 dB (fully clockwise). When the audio passing through the circuit is above the threshold level (set by the knob) the audio is unaffected and it remains flat. When the audio signal falls BELOW the threshold set on the front panel knob, the audio band (either Mid or High) is compressed by a ratio of - Switches the OPTICAL EXPANDER into the signal path. When engaged, the red LED in the switch cap is lit. IN (switch) 6 ENGLISH 2:1, so that frequency band appears to stay at a more constant volume and does not fall off in level. The overall effect is that the dynamic range of the selected band (Mid or High) is compressed to a higher average level without affecting the peaks of the original signal. Therefore, for example, a vocal can be made louder and set at a more constant level without aggressively compressing the original signal. The side benefit is that the two bands can be adjusted independently, to create different tonal mixes of the original signal, so the user can create subtly enhanced sounds or massively effected sounds very easily. - This controls the relative depth of the boost of the harmonic frequencies. Engaging the switch (in) causes more obvious harmonic enhancement, disengaging the switch (out) reduces makes the enhancement more subtle. DEPTH (switch) - When engaged (in), inserts the VINTAGE HARMONICS section after the OPTICAL COMPRESSOR section in the signal path. When disengaged (out), VINTAGE HARMONICS occur pre-compressor. Placing the VINTAGE HARMONICS post-compressor means that you are able to stop the affected harmonics changing the way the compressor responds. If you prefer to have the VINTAGE HARMONICS affect the way the compressor responds, disengage the POST COMP switch, placing the VINTAGE HARMONICS section before the OPTICAL COMPRESSOR. POST COMP (switch) OPTICAL COMPRESSOR The OPTICAL COMPRESSOR acts like an automatic volume control, turning down the volume of a signal if it gets too loud. This reduces variation between loud and quiet passages, as it automatically reduces the gain when the signal exceeds a given volume, defined as the threshold. Using the OPTICAL COMPRESSOR helps to `even out' a performance, stopping a vocal from clipping and/or disappearing in the mix. - Switches the VINTAGE HARMONICS into the signal path. When engaged, the red LED in the switch cap is lit. IN (switch) - This allows boosting of the mid frequency signals. The amount of boost is controlled by the MID BAND THRESHOLD knob in relation to the level of the audio passing through the circuit. A low threshold will result in a larger Mid Frequency Harmonic boost. The circuit peaks at 3k. MID BAND THRESHOLD (knob) - This allows boosting of the high frequency signals. The amount of boost is controlled by the HIGH BAND THRESHOLD knob, in relation to the level of the audio passing through the circuit. A low threshold will result in a larger High Frequency Harmonic boost. The circuit peaks at 18k. HIGH BAND THRESHOLD (knob) - This shows the Mid (left meter) and High (right meter) bands' relative boosts, as controlled by the threshold knobs. GAIN BOOST (LED meters) - Switches the OPTICAL COMPRESSOR into the signal path. When engaged, the red LED in the switch cap is lit. IN (switch) 7 ENGLISH - Determines when the OPTICAL COMPRESSOR starts to compress the signal - the lower the threshold, the more the signal is compressed. The signal is only compressed when it exceeds the threshold, so quieter passages maintain their natural dynamic range, whilst loud passages (that exceed the threshold) are compressed. THRESHOLD (knob) · · · - Sets the output volume of the compressed signal. Since compressing a signal makes it quieter, use the MAKEUP GAIN control to restore the signal to its original volume. Compare the volume of the original and the compressed signal by using the IN switch to switch the OPTICAL COMPRESSOR on and off. MAKEUP GAIN (knob) VH EQ COMP - disengage the VINTAGE HARMONICS section's POST COMP switch, engage the OPTICAL COMPRESSOR's POST EQ switch. COMP VH EQ - engage the VINTAGE HARMONICS section's POST COMP switch, disengage the OPTICAL COMPRESSOR's POST EQ switch. EQ COMP VH - engage both the VINTAGE HARMONICS section's POST COMP switch ANDd the OPTICAL COMPRESSOR's POST EQ switch. TUBE SOUND - Determines the time taken for the gain reduction to return to normal once the signal drops below the threshold. The faster the release, the louder the signal appears to be. RELEASE (knob) The TUBE SOUND processor simulates the sound of valve (tube) and tape distortion. The TUBE SOUND circuit is a FET-based circuit, and operates in 3 stages. As you turn the pot clockwise you add firstly 2nd order harmonics, then 2nd + 3rd order, and finally 2nd + 3rd + 5th order harmonics (fully clockwise). - Displays the amount of gain `lost' due to compression. Since compression reduces the volume of the signal, the meter drops as compression is applied: for example, a 9 dB drop shows as -9 on the meter. GAIN REDUCTION (LED meter) - When engaged (in), selects a higher compression ratio, which gives a very flat, compressed sound. Do not use the HARD RATIO switch if you want to maintain most of the original dynamics. HARD RATIO (switch) - When engaged (in), selects a slower attack time, which allows more of the transient peaks of the signal through the compressor. This can help retain a sense of the original signal's dynamics when compressing heavily. For example, this can be useful to allow compression of a snare drum without losing the initial `crack' of the drum stick striking the snare skin. SLOW ATTACK (switch) - When engaged (in), inserts the OPTICAL COMPRESSOR after the VOICE OPTIMISED EQ section in the signal path. When disengaged (out), compression occurs pre-EQ. Placing the compressor after the EQ means that you are able to affect the way the compressor responds by making changes to EQ settings. If you prefer to have the compressor act independently from the EQ, disengage the switch, placing the compressor section before the EQ section. POST EQ (switch) - Switches the TUBE SOUND section into the signal path. When engaged, the red LED in the switch cap is lit. IN (switch) Note that the user has control of the position of both the OPTICAL COMPRESSOR versus EQ section AND the VINTAGE HARMONICS versus OPTICAL COMPRESSOR section. Hence four different signal flow arrangements are possible: · VH COMP EQ ­ disengage both the VINTAGE HARMONICS section's POST COMP switch AND the OPTICAL COMPRESSOR's POST EQ switch. - Determines which frequencies are affected. In the BRIGHT position (fully clockwise), the whole signal is saturated; as you rotate the control counter-clockwise towards the MELLOW position, it introduces a low pass filter, so that only the frequencies below the cut-off frequency (5 kHz) are affected by the TUBE SOUND processing. TONE (knob) 8 ENGLISH - Determines how much saturation is applied. As you rotate the drive control clockwise from COOL to WARM, you progressively increase the amount of overdrive applied to the signal, adding harmonic-rich peak compression, and creating a more `rounded' tone. As the signal level increases, so more harmonics are created. The DRIVE LED provides a visual indication of the amount of distortion being applied, by changing colour from blue, (no distortion,) through green, to red (high distortion). DRIVE (knob and LED) - Boosts or cuts the `edge' and high mid of the voice. This circuit uses a bell-shaped EQ curve centred on 1.3 kHz. MID (knob) - This control allows you to reduce the volume of the frequencies that make a vocal sound coarse or harsh. This circuit uses a bell-shaped EQ curve centred on 3.9 kHz, which allows 0 dB (fully clockwise) to 10 dB (fully counter-clockwise) of cut to be applied. ABSENCE (knob) VOICE OPTIMISED EQ The VOICE OPTIMISED EQ is a sophisticated tone control that boosts or cuts selected frequency bands and so modifies the tonal quality of the input signal. It can be used correctively (to fix problems with the original sound) or creatively (to enhance a signal and help a track stand out in a mix). - Boosts or cuts low frequencies in the signal. The frequencies that are affected are determined by the TUNING control. Cutting frequencies with the WARMTH control affects a narrower range of frequencies than when boosting. The Q value is 0.7 for boost (knob pointing to the right) and 2.5 for attenuation (knob pointing to the left.) WARMTH (knob) - Determines which frequencies are affected by the WARMTH control ­ in general lower frequencies affect male voices and higher frequencies affect female voices. The range of frequencies which can be affected in this band range from 120 Hz (knob fully counter-clockwise) to 600 Hz (knob fully clockwise). TUNING (knob) DE-ESSER The DE-ESSER lets you remove excessive sibilance from a vocal performance. (A sibilant sound is one in which the "ess" sound is over-emphasised.) The VoiceMaster Pro uses the phase cancellation de-essing circuit from Focusrite's flagship ISA 430 to neatly eliminate a very narrow band of frequencies (user-definable, see `CUT FREQUENCY' below,) centred around the sibilant frequency. - Switches the VOICE OPTIMISED EQ into the signal path. When engaged, the red LED in the switch cap is lit. IN (switch) - Boosts or cuts the high frequencies in the signal. Adjust the Breath control to accentuate or reduce the breathy part of a vocal. The SHIFT switch determines the frequencies that are affected. BREATH (knob) - Determines the shelving frequency for the BREATH EQ band. With the SHIFT switch disengaged (out) the BREATH control affects frequencies above 16kHz; with the switch engaged (in) it affects frequencies above 10kHz. SHIFT (switch) 9 ENGLISH - Switches the DE-ESSER into the signal path. When engaged, the red LED in the switch cap is lit. IN (switch) - Determines how much gain reduction is applied to the selected frequency (as determined by the CUT FREQUENCY control, described below). The lower the threshold, the more de-essing is applied. THRESHOLD (knob) FADER, always start quietly and increase the output level until you reach the correct level ­ do not start with the output level set high, as it may damage the next unit in the chain. Be careful that peak signals do not exceed 0 dBfs if the internal A/D is used; the output level should be set so that it peaks close to ­2 dBfs, allowing a small safety margin. Always check the receiving device to ensure that it is not registering overload. If inserting the VoiceMaster Pro into a channel of a mixing console, set the OUTPUT FADER at 0, and perform any output level adjustments using the console's fader. - The inclusion of a custom VU meter in the VoiceMaster Pro's output section allows the user to accurately monitor the levels being sent to external analogue or digital equipment from the VoiceMaster Pro's analogue and digital outputs. The peak-reading meter displays levels from ­24 dBfs to 0 dBfs. An overload LED shows when levels are excessive- if this lights, reduce the level of signal being fed to the outputs using the controls in the EQ and compression section, or using the output level control. PEAK LEVEL METER - Displays the amount of gain reduction applied to the selected frequency, in dB. Range is ­2 dB to ­12 dB. GAIN REDUCTION (LED meter) - Determines the frequency to be removed. Adjust the CUT FREQUENCY control to select the sibilant frequency to be removed. Range is 2.2 kHz to 10 kHz (fully clockwise). CUT FREQUENCY (knob) - Allows you to monitor the DE-ESSER sidechain. When engaged (in), the sidechain signal is fed to the output of the VoiceMaster Pro. This allows you to use the CUT FREQUENCY control to solo the `essy' frequency range of the signal very easily ­ simply adjust the CUT FREQUENCY until the sidechain signal is as `essy' as possible, then disengage the LISTEN switch and adjust the threshold (whilst observing the GAIN REDUCTION meter) to apply the DE-ESSER process as required. LISTEN (switch) - This switch allows you to globally bypass all the processing sections in the VoiceMaster Pro, and is useful for comparing the level/sound of processed and unprocessed signals. PROCESS BYPASS (switch) LATENCY-FREE MONITORING When using the DE-ESSER, ensure that you do not set the threshold too low, or you will affect too much of the vocal. When you have the threshold set correctly, the effect of the DE-ESSER should not sound obvious until you compare the affected signal with the original signal. (Toggle the IN button to compare them.) OUTPUT LEVEL The LATENCY-FREE MONITORING section is used in conjunction with the HEADPHONE socket to provide flexible monitoring of the signal being recorded. Latency is a major problem when recording to e.g. a digital system via a sound card. If the signal to be monitored has to pass through the digital recorder and then be relayed back for external monitoring, significant delays may occur, making it difficult or impossible to sing, speak or play in time with the other tracks already recorded. The VoiceMaster Pro's LATENCY-FREE MONITORING section allows the user to monitor in stereo directly from the monitoring section, before passing through the digital recording system. Thus latency is eliminated and the recording artist can listen to other tracks already recorded, whilst speaking, singing or playing to those tracks in perfect time. - The OUTPUT FADER is used to set the correct output level from the VoiceMaster Pro to suit the input level of the next unit in the chain (such as a PC sound card or DAT/CD recorder). When setting the OUTPUT OUTPUT FADER (knob) 10 ENGLISH - This controls the level of the FX RETURN input on the rear panel. For example, reverb could be applied from an external processor using the (mono) FX SEND and (stereo) FX RETURN connectors, so that a vocalist can hear his/her vocal with reverb whilst recording. FX LEVEL (knob) In addition to the analogue outputs, a high quality 24 bit, 128x over-sampled digital output may be fitted as an option, which can operate at sample frequencies of 44.1, 48, 88.2 or 96 kHz. All of the following functions are available on the rear panel when this option is fitted: - This quarter inch jack (balanced, 0 dBfs to +22 dBfs) line level input on the rear panel allows an additional signal to be routed through the `spare' channel of the stereo digital output. This input always routes external signal to the right hand channel of the A/D converter. For example, two VoiceMaster Pro units could be used simultaneously, with a single digital output option. The first VoiceMaster Pro unit would feed the left channel of the A/D card installed. The second VoiceMaster Pro unit output would be connected to the ADC EXT INPUT, and would feed the right channel of the same A/D card, allowing two channels of A/D conversion. ADC EXT INPUT (rear panel) - This controls the mix sent to the HEADPHONE output, and allows you to mix between VOICE & FX (a combination of the signal sent to the main output plus the FX RETURN signal, as controlled by the FX LEVEL knob) and MONITOR (the signal from the EXT MONITOR INPUTS). HEADPHONE MIX (knob) HEADPHONE LEVEL (knob and TRS jack socket) - This knob controls the level sent to the stereo HEADPHONE jack. - These inputs are provided on balanced (+4 dBu) quarter inch TRS jacks. They allow routing of your main stereo mix outputs (e.g. from a digital recording system) to the VoiceMaster Pro's LATENCY-FREE MONITORING section. This means that you can monitor both the stereo mix already recorded, AND the processed signal from your VoiceMaster Pro at the same time. (So you can leave your VoiceMaster Pro permanently rigged up in your recording system for tracking, whilst also allowing you to monitor the output from your main DAW (e.g. finished stereo mix with VoiceMaster Pro-processed, recorded vocals.) See the LATENCY-FREE MONITORING section above. EXT MONITOR INPUTS (rear fascia) - This 24 bit output is S/PDIF format on an RCA phono connector. If 16 bit resolution is required, the receiving device should dither the 24 bit signal to achieve 16 bit performance. S/PDIF OUTPUT - Two switches give a choice of four sample frequencies as marked on the rear panel. The left-hand switch selects between 44.1kHz (switch in) and 48kHz (switch out), and the right hand switch doubles the selected frequency, providing for 88.2 and 96kHz sample frequencies. SAMPLE FREQUENCY (switch) - These outputs are provided on balanced, (+4 dBu) quarter inch TRS jacks. They allow routing of your main stereo mix from e.g. a digital recording system (DAW etc) to a pair of monitor speakers. Note that these monitor outputs are separate from the headphone bus/latency-free monitoring section, so the outputs relay only signal that is fed from the Ext. Monitor Inputs. (The headphone bus allows monitoring of the VoiceMaster Pro-processed signal when tracking.) MONITOR OUTPUTS (rear fascia) - If an external Wordclock source is fed to the BNC connector, the VoiceMaster Pro will attempt to synchronise to it. When the unit is correctly locked to the external clock source the ADC LOCK LED (on the front panel) will light to indicate correct operation. (The ADC LOCK LED should light continuously. If this flickers it indicates bad jitter on the synchronising signal, requiring investigation of the Wordclock-generating device.) EXT WORD CLOCK INPUT DIGITAL OUTPUT OPTION Fitting Instructions See the separate A/D option owner's manual for instructions on how to fit the A/D option. 11 ENGLISH OBTAINING GOOD QUALITY SOUND MICROPHONE POSITIONING give the correct sound in context. If necessary at this time, you may also need to increase the amount of BREATH. FLAT Recording vocals requires a different technique to that used when mixing live vocals, where the vocalist usually sings with the microphone touching his or her lips. In a studio recording situation it is usually desirable for the vocalist to be at least 50 cm away from the microphone. If this affects the vocal performance, (or if the vocal sounds weak), allow the vocalist to move closer to the microphone, but use a pop shield. It may also be necessary to use the VoiceMaster Pro's HIGH PASS FILTER ( ) to remove excessive bass tip-up, caused by the vocalist singing too close to the microphone (the so-called 'proximity effect'). USING COMPRESSION Increase the MID control in the VOICE OPTIMISED EQ. You may also need to add some BREATH, and/or maybe some WARMTH (if the result has too much top end). However, beware of overdoing the effect by adding too much. HARSH Use the ABSENCE control in the VOICE OPTIMISED EQ. This creates a natural dip in the harsh frequencies. You may also want to add some WARMTH, and if necessary, remove some MID or BREATH. LOST IN THE MIX If the vocalist is having difficulty staying a constant distance from the microphone, the recorded performance will get softer and louder as the distance from the microphone varies. To even out variations in level, use the OPTICAL COMPRESSOR to compress the signal. USE OF EFFECTS PEDALS Increase the MID control in the VOICE OPTIMISED EQ. Avoid using too much WARMTH on the vocal, as you will be boosting frequencies in the same frequency range as many of the instruments on the track. SIBILANT When using the INSTRUMENT INPUT, connect any effects pedals in-line, before the VoiceMaster Pro's INSTRUMENT INPUT. The output (whether analogue or digital) from the VoiceMaster Pro should be connected directly to your recorder. SPOKEN WORD If sibilant components (unpleasant "s" sounds) are standing out, use the DE-ESSER to make them sound more natural, as described in FACILITIES AND CONTROLS. MIX LACKS CHARACTER When recording the spoken word, use the WARMTH control in the VOICE OPTIMISED EQ to maximise depth, resonance and power. This is the sort of vocal sound favoured by many radio broadcasters. BACKING VOCALS When mixing down, don't be afraid to be outrageous. In pop, for example, the vocal is invariably heavily compressed, and often equalised. In the TUBE SOUND section, try using the DRIVE control (with the TONE control in its BRIGHT position) to give one channel an analogue sound. Or reduce the TONE control to create a low fidelity effect. Also try using the VINTAGE HARMONICS section to give the vocal a classic 1970s enhancement effect (see page 6 for more details). REVERB OR DELAY PROBLEMS Backing vocals are normally heavily compressed, since you want them to have a uniform presence without volume variations. Engage the HARD RATIO switch in the OPTICAL COMPRESSOR and adjust the THRESHOLD so that GAIN REDUCTION meter shows between 9 and 15 dBs of compression. To avoid the backing vocals becoming too fat and overpowering, use the WARMTH control in the VOICE OPTIMISED EQ to reduce the amount of bass in the vocals. CORRECTING PROBLEMS MUDDY Sometimes, reverb or delay can sound too lively and tends to "zing" ­ this is caused by sibilance in the voice. If you find this with the vocals you have recorded, you can try using the DE-ESSER to fix the problem, by heavily de-essing the signal that will be sent to the external effects unit. Then, at the recorder, mix the dry signal (from the PRE DE-ESSER output XLR socket on the back of the VoiceMaster Pro) with the wet signal from the effects unit. FIXING BLEED (`SPILL') Use the WARMTH control in the VOICE OPTIMISED EQ to remove some of the low frequencies. Solo the track, set the WARMTH control on full cut, and adjust the TUNING control until the vocal sounds more balanced. Then listen to the vocal in the context of the mix, and adjust the amount of cut on the WARMTH control to If there is noticeable bleed from other instruments off the vocalist's headphones, remove it when mixing down using the OPTICAL EXPANDER. Note that the more you compress a track, the more noticeable any bleed will become. 12 ENGLISH A BEGINNER'S GUIDE TO COMPRESSION Compressors are probably the most widely used signal processors in the audio industry. A compressor can be thought of as an automatic volume control. Once the volume of the signal exceeds a certain level (called the 'threshold'), the compressor reduces the gain (in other words, 'turns the volume down'), causing the signal to be less loud than it would otherwise have been. Let's take an example with some real numbers. If the threshold is set to -10 dB and the ratio is set to 4:1, any signal whose level exceeds -10 dB needs to rise in level by 4 dB for the output of the compressor to rise by 1 dB. Therefore an input signal with a peak at -6 dB (which is 4 dB above the threshold) would emerge from the compressor with a peak at -9 dB (1 dB above the threshold). Signal levels below the threshold are unaffected, so if the signal in the above example varied between -20 dB and -6 dB before entering the compressor, it will vary between -20 and -9 dB after being compressed. Its dynamic range (the difference between the quietest and loudest parts of the signal in dB) is reduced from 14 dB to 11 dB. Compression results in any variations in the volume of the signal (in other words, the signal's dynamic range) being reduced - the amount of this reduction is determined by the threshold (the level above which the gain is reduced) and the ratio (the amount by which the gain is reduced.) Higher ratios are referred to as hard ratios; lower ratios are called soft ratios. Because compression causes a reduction in volume level of loud signals, gain must be applied after the compressor to bring the overall volume level back up, so that the maximum volume before the compressor is the same as that after the compressor. This is called 'make-up gain', and is necessary so that the maximum level of the signal is always the same, for correct level matching with any further processing or other equipment. Once 'make-up gain' has been applied, the part of the signal that was lower than the threshold volume (and hence not compressed) will now be louder than it was before the compressor. This will cause any compressed instrument to sound louder. One use for this phenomenon is to give guitars more sustain. In most pop music, the backing instruments (such as drums, bass guitars, rhythm guitars etc) tend to be compressed heavily (using a fairly hard ratio and low threshold), so that they remain at a consistent volume level throughout the track. This will provide a solid backing, without occasional drum hits or bass notes poking through (or disappearing from) the mix untidily. A soft ratio tends to be used on instruments such as lead guitars or vocals that 'sit' on top of the mix. In this situation it is often desirable to preserve more of the dynamics of the original performance, to retain more expression. A reduction in variation of volume level is still required (for the reasons mentioned above), but not to the same extent. The amount by which the compressor reduces the gain is determined by the 'ratio'. The ratio is conventionally expressed as a numerical value, e.g. '4:1', which represents the amount by which the gain is reduced when the volume of the signal rises above the threshold. 13 ENGLISH The other controls included on most compressors are attack and release. Attack determines the speed at which the compressor starts to reduce the gain once the threshold has been exceeded. Think of it as the time taken to turn the volume down. Very short attack times mean the compressor `kicks in' very quickly ­ short attack times are typically used for vocals in order to keep the levels under strict control. Longer attack times mean more of the original signal's attack dynamics are preserved ­ this is a good way of keeping percussive and guitar sounds exciting and punchy. Release determines the speed at which the compressor stops acting once the signal drops below the threshold. Think of it as the time taken to turn the volume back up. Short release times mean the compressor very quickly returns the signal to its normal level. This can produce a `pumping' sound, where the changes in volume are very audible. Depending on the style of music, this can be undesirable, or a useful creative effect. Longer release times may mean that parts of the signal below the threshold end up being compressed, or that the gain doesn't have a chance to return to normal before the next `above threshold' sound ­ remember that the compressor works on the whole signal. See the diagram below: 14 ENGLISH A BEGINNER'S GUIDE TO EQUALISATION Equalisers are also widely used in the audio industry, and are effectively just tone controllers, though a bit more involved than those found on most hi-fi systems. They allow you to cut or boost certain frequencies or frequency bands within the audio signal. There are two main applications for using equalisation, or EQ (as it's more commonly known). The first is `creative' use. This involves enhancing a sound that is already present in some desirable way. Typical examples might involve boosting lower frequencies to give more depth, or boosting the high frequencies to give more of a `sparkle' to a sound. Because the precise frequencies that give these qualities will vary from instrument to instrument, it is sometimes necessary to be able to adjust the point at which frequencies will be cut or boosted by the EQ, as well as the amount of cut or boost. The other main application of EQ is `corrective' use. This involves using EQ to remove or reduce the level of unwanted frequencies. Here are a few examples of `corrective' use of EQ: Cutting low frequencies to reduce `proximity effect', where low frequencies have been over-emphasised as a result of close miking with certain types of microphone. Cutting the frequencies that may cause a vocal to sound boxy, nasal or harsh. Cutting the frequencies that may cause a drum to ring undesirably. Parametric EQ (such as the WARMTH control) allows the user to focus in on a specific band of frequencies in order to cut or boost them. This is particularly useful for `corrective' applications of EQ as the offending frequency may be honed in on, and its gain reduced. It is also useful for `creative' applications, for example giving warmth or presence to a vocal. Check out www.focusrite.com for links to more information on the subjects of compression and EQ. FREQUENTLY ASKED QUESTIONS Q. Is the VoiceMaster Pro only suitable for recording vocals in musical applications? A. No, the VoiceMaster Pro is suitable for recording many other sound sources too, such as electric guitars and bass (via its INSTRUMENT INPUT). It is also ideal for applications such as recording the spoken word, live sound applications, radio and TV broadcast, dubbing, post production and so on. Q. Is the VoiceMaster Pro a Class A device, and why is that important? A. Yes, the VoiceMaster Pro is a Class A device. Why? Class A is a type of amplifier design in which you have standing DC current running through your amplifier circuits all the time. As the signal comes along you vary what you're taking from that, rather than switching between supplying a positive current for one half of the waveform and a negative current for the other half. This results in the ability to represent audio in a linear (distortion free) manner all the way through the circuit. Cheaper processors use IC amplifiers which run close to Class B and don't have the same standing DC current, which means the transistors inside the chips switch off and on, inevitably resulting in less linear performance. Q. My VoiceMaster Pro gets quite hot when in use. Should I be worried about this? A. No. This is a result of the high quality Class A circuitry inside your VoiceMaster Pro, which has been designed to dissipate heat. As a precaution, it is wise to rack hotter units lower down your rack than cooler units. If space is available, fit a blank panel between units to allow extra ventilation. Q. What is the difference between +4 dBu and ­10 dBV? A. These are different signal operating levels. +4 dBu usually refers to professional equipment and ­10 dBV usually refers to semi-professional or consumer equipment. It is important to make sure that any two or more devices connected to each other are operating at the same signal level. If the +4 dBu output of a device feeds the ­10 dBV input of another device, this may cause the second device to overload. Alternatively, if the ­10 dBV output of a device feeds the +4 dBu input of another device, the second device may receive a signal level which is too low (i.e. too quiet). -10 dBV devices are usually connected using a mono 1/4" jack. This is known as an `unbalanced' connection. +4 dBu devices are usually connected using a TRS (stereo) 1/4" jack, or XLR. This is known as a `balanced' connection. Q. Should I use balanced connectors with my VoiceMaster Pro? A. Yes, where possible. The line level analogue input is balanced, operating at +4 dBu. The VoiceMaster Pro provides both balanced (+4 dBu) and unbalanced (-10 dBV) output connectors. See the 'Rear panel connections' section on page 3 for more information on connecting the analogue line level inputs and outputs. 15 ENGLISH Q. Does the VoiceMaster Pro have the same kind of spectacular bandwidth that has given the Red and ISA range units their reputation for 'open-ended' sound? A. Yes. The audio bandwidth of the VoiceMaster Pro is 10 Hz to 200 kHz! Q. Can I take my VoiceMaster Pro with me when I travel internationally? A. It depends. There are three versions of the VoiceMaster Pro mains transformer. One is suitable for use in North America, one in Japan (both with mains voltages in the 100-120V range). The third version is designed for use in the UK and Europe, with mains voltages in the 200-240V range. If you buy a VoiceMaster Pro in a particular territory, it will be configured for ONLY that territory's mains voltage range. For example, if you're travelling from the USA to the UK, you CANNOT use your US model VoiceMaster Pro. But if the mains voltage in the country you're visiting is in the same range, you can use the VoiceMaster Pro with no problems - so taking a VoiceMaster Pro from Germany to France, for example, would be fine. Q. Is there an optional digital input card? A. No, because all the processing in the VoiceMaster Pro is entirely analogue - so even if there were a digital input, the digital signal would have to immediately pass through a D/A converter to allow processing! Q. Why is the 24 bit 96kHz specification important? A. An A/D converter works by sampling the audio waveform at regular points in time, and then quantising those values into a binary number, which relates to the number of bits specified. The quantised signal must then be passed through a D/A converter before it becomes audible. In simple terms, the D/A essentially 'joins the dots' plotted by the A/D converter when the signal was first converted to digital. The number of dots to join, combined with how little those dots have been moved, determines how accurate the final signal will be compared to the original. The greater the sample rate and bit rate, the more accurate the whole digital process is. So 24 bit/96 kHz performance will ensure more accurate digital transfer of your audio information compared to 16 bit/44.1kHz standards. This is especially important if further digital signal processing is to be applied to the signal once converted to digital, as any mathematical operations taking place on the data (for example as a result of a gain change, or dynamic effect process) may result in quantisation and rounding errors. The higher the resolution of the digital data, the smaller the audible effect of these errors. Q. Can I retrofit a digital board to an analogue VoiceMaster Pro at a later date? A. Yes, and you can do it yourself - it can easily be retro-fitted by the customer without any soldering etc, just a few screws to undo, and one clip-connector to join to the main PCB. Q. What is Wordclock? A. Whenever multiple digital audio devices are connected together digitally, all the devices must be Wordclock synchronised to avoid data transfer problems. All devices must send and receive their data at the same sample rate (e.g. 44.1kHz) but they must also have their internal clocks running in sync. This ensures that all units send, receive and process their data streams simultaneously. Failure to achieve this will mean a drastic reduction in audio quality, and other unwanted audible artefacts, such as pops and clicks, may occur. At a sample rate of 44.1kHz for example, there are 44,100 spaces every second that need to have samples inserted. If there is a slight drift in one of the clocks, some of those samples will be `missed'/will move forward one place, which results in distortion. To avoid such problems, every digital system needs to employ Wordclock. One unit should be designated the `Wordclock master', and all others should be designated `Wordclock slaves'. Setting this up is often simple, since most digital transfer formats include embedded Wordclock data (e.g. S/PDIF, AES/EBU, ADAT). Where this is not the case (e.g. TDIF), Wordclock can be provided via a separate Wordclock connection. Note that timecode synchronisation (e.g. SMPTE) is different to Wordlock synchronisation, but equally important. Timecode enables recording and playback devices to run in sync with one another, and carries a regular series of absolute time values (hrs:mins:secs:frames). The two timing systems are quite independent. 16 ENGLISH TROUBLESHOOTING No LEDs illuminate · Is the POWER switched on? · Is the correct mains voltage being used for your unit? If not, the fuse may blow, requiring the correct fuse to be refitted. No · · · · output when using the MIC INPUT Is the power switched on? Is the LINE switch on the front panel switched out? Is the INPUT GAIN set correctly? (See 'Facilities and Controls' section for details.) For microphones that require phantom power, is the +48V switch engaged? (If you are unsure whether your microphone requires phantom power, check the user guide for your microphone.) output when using the LINE INPUT Is the power switched on? Is the LINE switch on the front panel switched in? Is the INPUT GAIN set correctly? (See 'Facilities and Controls' section for details.) output when using the INSTRUMENT INPUT Is the power switched on? Is the LINE switch on the front panel switched out? Is the INPUT GAIN set correctly? (See 'Facilities and Controls' section for details.) The OPTICAL COMPRESSOR is not working · Is the compressor section's IN switch engaged? · Is the INPUT GAIN set correctly? If set too low, the signal level may not be high enough to activate the compressor. · Is the THRESHOLD control set correctly? (If set too high, the input level may not reach the threshold at which compression starts.) · Is the PROCESS BYPASS switch engaged? The VOICE-OPTIMISED EQ is not working · Is the EQ section's IN switch engaged? · Has some cut or boost been applied? · Is the PROCESS BYPASS switch engaged? No · · · Wordclock lock Is your external Wordclock source transmitting Wordclock? Is the sample frequency set to match that of the Wordclock-transmitting device? Is a Wordclock cable connected if required? (See `What is Wordclock?', page 16.) No · · · No · · · No output from the digital output option · Is the sample frequency set correctly? · Is the receiving device set to receive at 24 bit? CONTACTING US If have any questions about your VoiceMaster Pro, or are continuing to have difficulty, you can email us for help at tech@focusrite.com. Alternatively, telephone us on +44 (0)1494 462246, or contact your local distributor (see listing at the back of this manual). 17 DEUTSCH INHALT INHALT ...................................................................................................................... 18 WICHTIGE SICHERHEITSINFORMATIONEN ................................................. 18 EINLEITUNG ............................................................................................................ 19 SICH MIT DEM GERÄT VERTRAUT MACHEN............................................... 19 ANSCHLÜSSE RÜCKSEITE.................................................................................... 19 MIT DEM BETRIEB BEGINNEN........................................................................... 20 FUNKTIONEN UND BEDIENUNG...................................................................... 21 DISCRETE CLASS A PRE-AMP......................................................................... 21 OPTICAL EXPANDER........................................................................................ 22 VINTAGE HARMONICS.................................................................................... 22 OPTICAL COMPRESSOR.................................................................................. 23 TUBE SOUND...................................................................................................... 24 STIMMENOPTIMIERTER EQ .......................................................................... 25 DE-ESSER.............................................................................................................. 25 AUSGANGSSEKTION ......................................................................................... 26 LATENZFREIES MONITORING...................................................................... 26 OPTIONALER DIGITALER AUSGANG .......................................................... 27 BESTE SOUNDERGEBNISSE ................................................................................. 27 BEHEBEN VON PROBLEMEN .............................................................................. 28 EINFÜHRUNG KOMPRESSION........................................................................... 29 EINFÜHRUNG EQ................................................................................................... 31 HÄUFIG GESTELLTE FRAGEN ............................................................................. 31 PROBLEMLÖSUNG ................................................................................................. 33 KONTAKT ................................................................................................................. 33 WARNUNG: DIESES GERÄT MUSS DURCH DAS KALTGERÄTEKABEL GEERDET WERDEN. UNTER KEINEN UMSTÄNDEN SOLLTE DIE NETZERDUNG VON DER NETZLEITUNG GETRENNT WERDEN. Dieses Gerät wird für die an der Rückseite angegebene Spannung geliefert. Gehen Sie sicher, dass die korrekte Netzspannung vorhanden ist und der korrekte Sicherungswert angepasst ist, bevor Sie das Gerät an die Stromversorgung anschließen. Um das Brandrisiko zu verringern, ersetzen Sie die Netzsicherung nur durch eine Sicherung mit dem richtigen Wert, der an der Rückseite angegeben ist. Die interne Stromversorgungseinheit besitzt keine Teile, die vom Benutzer gewartet werden können. Überlassen Sie jegliche Wartung einem qualifizierten Wartungstechniker durch den entsprechenden Focusrite-Händler. RACK VENTILATION: DA DER VOICEMASTER PRO EIN CLASS-A GERÄT IST, GEHEN SIE SICHER, DASS ES WEITER UNTEN IN IHREM EQUIPMENTRACK MONTIERT IST, MIT GENÜGEND PLATZ FÜR DIE LUFTZUFUHR NACH OBEN UND UNTEN. WICHTIGE SICHERHEITSINFORMATIONEN Bitte lesen Sie alle diese Anweisungen und bewahren Sie zum späteren Nachschlagen auf. Befolgen Sie alle Warnungen und Anweisungen, die auf dem Gerät angegeben sind. · Verdecken Sie nicht die Lufteinlässe an der Rückseite. Führen Sie keine Objekte durch eine der Öffnungen ein. · Verwenden Sie kein beschädigtes oder sprödes Kaltgerätekabel. · Trennen Sie das Gerät vor der Säuberung von der Stromversorgung. Nur mit einem feuchten Tuch säubern. Verschütten Sie keine Flüssigkeit auf der Einheit. · Sorgen Sie für genügend Luftzirkulation um die Einheit herum, um Überhitzung zu vermeiden. Da dies eine Class-A Gerät ist, empfehlen wir, dass Sie eine HE (evtl. mit Blindpanel) über der Einheit für die Luftzirkulation frei lassen. · Trennen Sie das Gerät von der Stromversorgung und überlassen Sie die Wartung qualifiziertem Servicepersonal unter den folgenden Umständen: Wenn das Kaltgerätekabel oder der Stecker beschädigt ist; wenn Flüssigkeit in das Gerät eingedrungen ist; wenn das Gerät heruntergefallen ist oder das Gehäuse beschädigt wurde; wenn das Gerät nicht normal funktioniert oder Veränderungen in einer Funktion zeigt. Benutzen Sie nur die Regler, die in der Bedienungsanleitung erklärt werden · Ignorieren Sie nicht den Sinn und Zweck von Schutzleiter-Steckern. Ein Erdungsstecker besitzt zwei stromführende Kontakte und einen dritten Erdungskontakt. Dieser dient ihrer Sicherheit. 18 DEUTSCH EINLEITUNG Der VoiceMaster Pro ist ein kombinierter Hochleistungsmikrofonvorverstärker, Dynamikprozessor und Equalizer. Auch wenn er speziell für Vocals entwickelt wurde, ist er doch flexibel genug, um auch hervorragende Aufnahmen und Abmischungen einer Reihe anderer Instrumente, wie z.B. Gitarre und Schlagzeug, zu ermöglichen. Bei Aufnahmen müssen Sie das Signal nicht durch ein Mischpult führen: verbinden Sie Ihr Mikrofon einfach mit dem VoiceMaster Pro und verbinden Sie diesen direkt mit ihrer Soundkarte oder mit dem Aufnahmegerät. Diese Form der direkten Aufzeichnung sichert Ihnen das klarste Signal bei höchster Qualität; die Möglichkeit, dass Nebengeräusche durch ein Mischpult hinzugefügt werden, sind dadurch ausgeschlossen. Es gibt sieben separate Signalprozessor-Sektionen im Voicemaster Pro: · Discrete Class A Vorverstärker · Optical Expander · Vintage Harmonics · Optical Compressor · Tube Sound (Röhrenemulation) · Stimmenoptimierter EQ · De-Esser Um den saubersten Signalweg zu ihrem Aufzeichnungsmedium zu gewährleisten, kann jede Sektion individuell aus dem Audioweg genommen werden (,,hard bypassed"), wenn sie nicht benutzt wird. Außerdem gibt es eine globale Prozessor BypassSchaltung. SICH MIT DEM GERÄT VERTRAUT MACHEN Wenn Sie sich mit dem Gerät vertraut machen, verwenden Sie am besten eine Ihnen bekannte Soundquelle. Z.B. könnten Sie eine favorisierte CD über das Gerät laufen lassen; das Arbeiten mit einem bekannten Track macht die Interpretation der Ergebnisse einfacher. (Bedenken Sie allerdings, dass die Tracks für die CD bereits komprimiert sind, also ist es kaum möglich, die Auswirkungen des Optical Compressors zu hören. Wenn dies der Fall ist, versuchen Sie stattdessen Samples zu verwenden oder nehmen Sie eigene Signale unkomprimiert auf und hören Sie diese über den VoiceMaster Pro ab.) Die einfachste Art, die kreative Leistung des VoiceMaster Pro zu hören, besonders, wenn Sie nicht mit jeder einzelnen Prozessorsektion vertraut sind, ist, jede Sektion einzeln anzuschalten und jeden Regler auszuprobieren. Probieren Sie sie schließlich alle Sektionen zusammen aus und überzeugen Sie sich von der kreativen Power des VoiceMaster Pro! ANSCHLÜSSE RÜCKSEITE 19 DEUTSCH Der VoiceMaster Pro besitzt Anschlüsse für Mikrofoneingang (XLR) und Line-Input (Klinke) an der Rückseite, der Mikrofoneingang ist zusätzlich auf der Vorderseite neben einem Instrumenteneingang (Klinke) vorhanden. Die INSERT SEND und INSERT RETURN Klinken-Eingänge auf der Rückseite erlauben, einen externen Prozessor in die Signalkette zwischen Vorverstärker und Prozessorsektion des Gerätes einzuschleifen.Ausgänge liegen sowohl als XLR (symmetrisch, +4dBu) und Klinke (asymmetrisch, -10dBV) vor, außerdem gibt es einen Ausgang vor dem De-Esser (PreDe-Esser, XLR, symmetrisch, +4dBu). Zusätzlich besitzt der VoiceMaster Pro einen asymmetrischen Mono-FX-SEND und symmetrische Stereo-FX-RETURN Klinkenanschlüsse, um einen externen, eingeschliffenen Effektprozessor, zusammen mit den symmetrischen Stereo EXT MONITOR INPUT- und MONITOR OUTPUT Klinkenanschlüssen, abzuhören. Siehe auch LATENZFREIES MONITORING. Schließlich kann ein externes Signal via des Eingangs ADC EXT INPUT (symmetrisch, Klinke) zum zweiten Kanal der optionalen digitalen A/DKonverterkarte gesendet werden. Dies erlaubt den A/D-Wandler für Stereo-Mastering Anwendungen zu verwenden, als zusätzlichen Stand-Alone-Wandler oder um einfach einen zweiten (Linepegel-) Recording-Eingang zu bieten. Informationen über die optionale Digitalausgangskarte siehe OPTIONALER DIGITALER AUSGANG. MIT DEM BETRIEB BEGINNEN 1. Stellen Sie sicher, dass nichts außer der Netzstromversorgung mit Ihrem VoiceMaster Pro verbunden ist, dann schalten Sie ihn mit dem Power-Schalter an der rechten Seite des Gerätes ein. Wenn Ihr Gerät permanent an ein Patchfeld angeschlossen ist, stellen Sie sicher, dass kein Audiosignal an angeschlossene Lautsprecher geführt wird, um deren Zerstörung durch Pegelspitzen beim Einschalten zu verhindern. 2. Verbinden Sie den entsprechenden Ausgang an der Rückseite des VoiceMaster Pro mit ihrem Aufnahmegerät oder Audiointerface. Wenn Sie den optionalen Digitalausgang verwenden, verbinden Sie entsprechend den digitalen Ausgang mit dem digitalen Eingang Ihres Recorders oder Ihrer Audioschnittstelle. Siehe auch OPTIONALER DIGITALER AUSGANG. 3. Verbinden Sie die Monitor-Ein- und Ausgänge mit Ihrem externen Setup/Monitor-Lautsprechern. 4. Gehen Sie Sicher, dass jede Prozessorsektion ausgeschaltet ist (der IN-Schalter ist jeweils nicht gedrückt und unbeleuchtet), und dass der PROCESS BYPASS Schalter ebenfalls nicht gedrückt ist. 9. Drehen Sie die INPUT GAIN Regler auf, übewachen Sie die EingangspegelAnzeige-LEDs und stellen Sie sicher, dass die rote Übersteuerungsanzeige O/L LED lediglich leicht und kurz bei den lautesten Pegelspitzen aufleuchtet. 10. Wenn Sie ein Mikrofon verwenden, stellen Sie sicher, dass dessen Platzierung optimal ist. Bevor Sie mit den Aufnahmen beginnen, ändern Sie die Position des Mikrofons solange, bis Sie so nah wie möglich an der Soundquelle sind. Beachten Sie, dass das Ändern der Mikrofonposition Pegelveränderungen nach sich ziehen kann, die am Eingangsregler des VoiceMaster Pro korrigiert werden müssen. 11. Wenn gewünscht fügen Sie zusätzliches Prozessing hinzu, indem Sie die verschiedenen Signal-Proczssing-Blocks benutzen, die individuell ein- und ausgeschaltet werden können. Für mehr Informationen über die speziellen Funktionen der verschiedenen Sektionen ziehen Sie den folgenden Abschnitt zurate. 5. Schließen Sie Ihre Eingangsquelle wie benötigt an. Ein Mikrofon kann entweder in den XLR-Mikrofoneingang an der Vor- oder Rückseite eingesteckt werden. Wenn Sie eine Line-Quelle anschließen wollen (um z.B. die Prozessorfunktionen des VoiceMaster Pro beim Mischen zu nutzen), verbinden Sie diese mit dem KlinkenLine-Eingang an der Rückseite. Alternativ können Sie eine elektrische Gitarre oder einen Bass über ein asymmetrisches Klinkenkabel mit dem INSTRUMENT INPUT an der Vorderseite verbinden. 6. Überprüfen Sie, ob der korrekte Input in der DISCRETE CLASS A PRE-AMP Sektion gewählt wurde. Wenn Sie eine Line-Quelle, die an der Rückseite am LINE INPUT angeschlossen ist, aufnehmen wollen, stellen Sie sicher, dass der LINESchalter gedrückt ist. Wenn ein Mikrofon an einem der Mikrofoneingänge oder eine elektrische Gitarre/Bass mit dem INSTRUMENT INPUT angeschlossen ist, muss der LINE-Schalter ausgeschaltet sein. 7. Stellen Sie sicher, dass die Phasenumkehrung (Ø) und der HPF-Schalter ausgeschaltet sind und das der INPUT GAIN Regler ganz links steht. Stellen Sie den OUTPUT FADER auf die ,,0"-Position. 8. Wenn Sie ein Kondensatormikrofon benutzen, das Phantomspeisung benötigt, aktivieren Sie den +48V-Schalter. Wenn Sie nicht sicher sind, ob Ihr Mikrofon diese Phantomspeisung benötigt, ziehen Sie bitte dessen Bedienungsanleitung zurate. Phantomspeisung kann dafür nicht geeignete Mikrofone beschädigen. 20 DEUTSCH FUNKTIONEN UND BEDIENUNG ­ Schaltet das Gerät ein. Wir empfehlen, das Gerät einzuschalten, bevor Sie es mit folgendem Equipment verbinden, um EinschaltPegelspitzen, die Beschädigungen verursachen können, nicht an die folgenden Geräte weiterzugeben. Es ist ratsam, das Gerät einige Minuten vor der Benutzung ,,warmlaufen" zu lassen, um sicher zu gehen, dass alle inneren Schaltkreise stabilisiert sind. POWER (Schalter) ­ Dieser bietet +48V Phantomspeisung für Kondensatormikrofone (nur für die Mikrofoneingänge). Wenn Sie unsicher sind, ob Ihr Mikrofon Phantomspeisung benötigt, ziehen Sie dessen Bedienungsanleitung vor dem Anschalten zurate. Einige Mikros können durch Phantomspeisung beschädigt werden. +48V (Schalter) DISCRETE CLASS A PRE-AMP ­ Dies ist ein hochohmiger Klinkeneingang um eine elektrische Gitarre oder einen Bass anzuschließen, ohne eine DI-Box zu benötigen. Wenn ein Instrument und ein Mikrofon angeschlossen sind, überlagert der Instrumenteneingang den Mikrofoneingang. INSTRUMENT INPUT Dieser Teil des Gerätes ist ein Vorverstärker, der dazu verwendet wird, das Eingangssignal am Mikrofon- oder Instrumenteneingang auf Linepegel zu verstärken, bevor es weiteren Prozessingsektionen zugeführt wird. ­ Dies erlaubt, die Phase eines Eingangssignals umzukehren, um Phasenprobleme zu korrigieren wenn falsche Polarität vorliegt oder z.B. bei gegengerichteten Mikrofonanordnungen (z.B. Snare oben/unten). Ø (Phasenumkehrschalter) ­ Wenn dieser Schalter eingeschaltet ist, ist der LINE INPUT an der Rückseite aktiv und die LED des Schalters leuchtet. Wenn dieser ausgeschaltet ist, sind der MIC INPUT und der INSTRUMENT Eingang aktiv. LINE (Schalter) (HPF Regler und Schalter) ­ Dies ist ein Hochpass-Filter, der ungewollte, tiefe Fequenzen entfernt, wie z.B. Vibrationen der Bühne, die über Mikrofonständer übertragen werden, oder den Nahbesprechungseffekt (niedrige Frequenzen werden übermäßig betont, wenn bestimmte Mikrofone auf sehr kurze Entfernung verwendet werden). Der Regler setzt die Grenzfrequenz (30 bis 400 Hz, 18 dB pro Oktave), der Schalter muss auf ,,An" stehen, um die Funktion zu aktivieren. ­ Dies ist ein XLR-Anschluss, der den Anschluss eines Mikrofones an das Gerät ermöglicht. Auf der Rückseite gibt es ebenfalls einen Mikrofoneingang, allerdings sollte nur einer zur gleichen Zeit benutzt werden; schliessen Sie nicht beide Mikro-Eingänge gleichzeitig an. Wenn Sie den Vorverstärker des VoiceMaster benutzen und den Ausgang an ein Mischpult-Kanal anschliessen, verwenden Sie den Line-Eingang des Kanalzuges! Der bessere Mikrofonvorverstärker des VoiceMaster Pro wird so verwendet, um das Signal ohne ungewollte Störungen eines schlechteren Vorverstärkers an sein Ziel zu senden. Vermeiden Sie es immer, den Ausgang des VoiceMaster Pro mit einem weiteren Mikro-Vorverstärker zu verbinden, da dies sehr viel schlechtere Resultate zur Folge hat. MIC INPUT ­ Dieser wird benutzt, um den optimalen Pegel des Eingangssignales zu erreichen. Schliessen Sie ein Eingangssignal an (der INPUT GAIN Regler sollte ganz gegen den Uhrzeigersinn gedreht sein) und drehen den INPUT GAIN Regler weiter auf, während Sie die LED-Anzeige beobachten. Die rote O/L (Overload) LED darf nur ganz selten und kurz während der höchsten Signalspitzen aufleuchten. Wenn die Übersteuerungs-LED für längere Zeit leuchtet oder Sie Verzerrungen des Signals bei Pegelspitzen bemerken, sollten Sie den INPUT GAIN verringern. INPUT GAIN (Regler) Beachten Sie, dass die LED-Kette so kalibriert ist, 0dBfs als Maximalwert anzuzeigen, um das Einpegeln bei Digital-Aufnahmen zu erleichtern. Welcher der beste Pegel für Ihre Aufnahmen ist, hängt von ihrem Aufnahmemedium ab. Wenn Sie auf einem analogen Medium aufnehmen, wo zusätzlicher Headroom benötigt wird, ergibt ein Pegel von ­18dBfs den äquivalenten Ausgangspegel von +4 dBu. Bei DigitalAufnahmen ist u.U. ein maximal Pegel von ­6 - -4 dBfs erwünscht. 21