Mac audio mpx 2500 instrukcja

The core which is coloured brown must be connected to the terminal marked L or coloured red. If you need to change the plug and if you are qualified to do so, refer to the table below. WARNING: If the ground is defeated, certain fault conditions in the unit or in the system to which it is connected can result in full line voltage between chassis and earth ground.

Severe injury or death can then result if the chassis and earth ground are touched simultaneously. Sicherungen nur durch solche, gleicher Stromstarke und gleichen Abschaitverhaltens ersetzen. Sicherungen nie uberbrucken. Jedwede Beschadigung des Netzkabels vermeiden. Netzkabel nicht knicken Oder quetschen. Beim Abziehen des Netzkabels den Stecker und nicht das Kabel enfassen. Beschadigte Netzkabel sofort auswechseln.

Gerat und Netzkabel keinen ubertriebenen mechanischen Beaspruchungen aussetzen. Urn Beruhrung gefahrlicher elektrischer Spannungen zu vermeiden, dart das Gerat nicht gebffnet warden.

Im Fall von Betriebsstbrun- gen dart das Gerat nur Von befugten Servicestellen instandgesetzt werden. Im Gerat befinden sich keine, durch den Benutzer reparierbare Telle. Eindringen von Feuchtigkeit und Flussigkeiten in das Gerat vermeiden. Bel Betriebsstorungen bzw. Safety Instructions French On s'assurera toujours que la tension et la nature du courant utilise correspondent bien a ceux indiques sur la plaque de I'appareil. N'utiliser que des fusibles de meme intensity et du meme principe de mise hors circuit que les fusibles d'origine. Ne jamais shunter les fusibles.

Eviter tout ce qui risque d'endommager le cable seceur. On ne devra ni le plier, ni I'aplatir. Lorsqu'on debranche I'appareil, tirer la fiche et non le cable. Si un cable est endommage, le remplacer immediatement. Ne jamais exposer I'appareil ou le cable a une contrainte mecanique excessive. Pour eviter tout contact averc une tension electrique dangereuse, on n'oouvrira jamais I'appareil. En cas de dysfonctionnement, I'appareil ne peut etre repare que dans un atelier autorise.

Aucun element de cet appareil ne peut etre repare par I'utilisateur. Pour eviter les risques de decharge electrique et d'incendie, proteger I'appareil de I'humidite. Eviter toute penetration d'humidite ou fr liquide dans I'appareil. En cas de dysfonctionnement ou si un liquide ou tout autre objet a penetre dans I'appareil couper aussitot I'appareil de son alimentation et s'adresser a un point de service apresvente autorise. Safety Instructions Spanish Flacer funcionar el aparato solo con la tension y clase de corriente sehaladas en la placa indicadora de caracten'sticas.

Reemplazar los fusibles solo por otros de la misma intensidad de corriente y sistema de desconexibn. No poner nunca los fusibles en puente. Proteger el cable de alimentacibn contra toda clase de dahos. No doblar o apretar el cable. Al desenchufar, asir el enchufe y no el cable. Sustituir inmediatamente cables dahados. No someter el aparato y el cable de alimentacibn a esfuerzo mecanico excesivo. Para evitar el contacto con tensiones electricas peligrosas, el aparato no debe abrirse. En caso de producirse fallos de funcionamiento, debe ser reparado solo por talleres de servicio autorizados.

En el aparato no se encuentra ninguna pieza que pudiera ser reparada por el usuario. Para evitar descargas electricas e incendios, el aparato debe protegerse contra la humedad, impidiendo que penetren esta o llquidos en el mismo. En caso de producirse fallas de funcionamiento como consecuencia de la penetracibn de llquidos u otros objetos en el aparato, hay que desconectarlo inmediatamente de la red y ponerse en contacto con un taller de servicio autorizado. Safety instructions italian Far funzionare I'apparecchio solo con la tensione e II tipo di corrente indicati sulla targa riportante I dati sulle prestazioni.

Sostituire I dispositivi di protezione valvole, fusibili ecc. Non cavallottare mai I dispositivi di protezione. Evitare qualsiasi danno al cavo di collegamento alia rete. Non piegare o schiacciare II cavo. Per staccare II cavo, tirare la presa e mai II cavo. Sostituire subito I cavi danneggiati. Non esporre I'apparecchio e II cavo ad esagerate sollecitazioni meccaniche.

Per evitare II contatto con le tension! In caso di anomalie di funzionamento I'apparecchio deve venir riparato solo da centri di servizio autorizzati. NeN'apparecchio non si trovano parti che possano essere riparate dall'utente. Per evitare scosse elettriche o incendi, I'apparecchio va protetto daN'umidita. Evitare che umidita o liquidi entrino neH'apparecchio. In caso di anomalie di funzionamento rispettivamente dopo la penetrazione di liquidi o oggetti neH'apparecchio, staccare immediatamente I'apparecchio dalla rete e contattare un centre di servizio qualificato.

Manual The Operating Manual contains instructions to verify the proper operation of this unit and initialization of certain options. You will find these operations are most conveniently performed on the bench before you install the unit in the rack. Please review the Manual, especially the installation section, before unpacking the unit.

Trial Period Precautions If your unit has been provided on a trial basis: You should observe the following precautions to avoid reconditioning charges in case you later wish to return the unit to your dealer. It is not wise to ship in other than the factory carton. Set the unit on soft, clean surfaces. Support the unit when tighten- ing the screws so that the threads do not scrape the paint inside the slotted holes. Packing When you pack the unit for shipping: 1 Tighten all screws on any barrier strip s so the screws do not fall out from vibration.

If you are returning a unit for repair, do not enclose any of the above items. Further advice on proper packing and shipping is included in the Manual see Table of Contents. Trouble If you have problems with installation or operation: 1 Check everything you have done so far against the instructions in the Manual. The number is 1 If it is not instaiied and used as directed by this manuai, it may cause interference to radio communication. This equipment compiies with the iimits for a Ciass A computing device, as specified by FCC Ruies, Part 15, subject J, which are designed to provide reasonabie protection against such interference when this type of equipment is operated in a commerciai environment.

Operation of this equipment in a residentiai area is iikeiy to cause interference. Simply walking across a rug can gen- erate a static charge of 20, volts. This is the spark or shock you may have felt when touching a doorknob or some other conductive surface. Static damage will not be covered under warranty. There are many common sources of static. Most involve some type of friction between two dissimilar materials. Some examples are combing your hair, sliding across a seat cover or rolling a cart across the floor. Since the threshold of human perception for a static discharge is volts, you will not even notice many damaging discharges.

Basic damage prevention consists of minimizing generation, discharging any accumulated static charge on your body or workstation, and preventing that discharge from being sent to or through an electronic component. You should use a static grounding strap grounded through a protective resistor and a static safe workbench with a conductive surface. This will prevent any buildup of damaging static. Other patents pending. Orban and Optimod are registered trademarks. All trademarks are property of their respective companies. This manual is part number Published October A reference to a numbered step or a page number except in the Index is a live hyper- link; click on it to go immediately to that reference.

If the bookmarks are not visible, click the "Bookmarks" tab on the left side of the Acrobat Reader window. This manual has a table of contents and index. To search for a specific word or phrase, you can also use the Adobe Acrobat Reader's text search function. Because all processing is performed by high-speed mathematical calculations within Freescale digital signal processing chips, the processing has cleanliness, quality, and stability over time and tempera- ture that is unmatched by analog processors.

The can be used either as a full-featured multiband FM audio processor in- cluding stereo encoder or as an extremely high-quality stand-alone stereo encoder operating at 64 kHz to kHz sample rates and offering lowpass filtering, over- shoot limiting, composite limiting, and an ITU multiplex power controller. When used in the latter mode, the must be driven usually via an STL by a full- featured FM audio processor like Orban's that incorporates preemphasis- aware HF limiting and peak control.

Thousands of these processors are on the air all over the world. Because OPTIMOD-FM incorporates several audio processing innovations exclusive to Orban products, you should not assume that it should be oper- ated in the same way as less sophisticated processors. If you do, you may get disappointing results. A small investment of your time now will yield large dividends in audio quality. Section 2 explains how to install it and set it up.

Sections 4 through 6 provide reference information. OPTIMOD-FM was designed to deliver a high quality sound while simultaneously in- creasing the average modulation of the channel substantially beyond that achiev- able by "recording studio"-style compressors and limiters. Because such processing can exaggerate flaws in the source material, it is very important that the source audio be as clean as possible. No other audio process- ing is necessary or desirable. Navigation is by dedicated buttons, soft buttons whose functions are context-sensitive , and a large rotary knob.

It can be configured to interface ideally with any commonly found transmission system in the world. Its pre-emphasis control is seldom audibly apparent, producing a clean, open sound with subjective brightness matching the original program. Both digital in- put and digital output are equipped with sample-rate converters and can oper- ate at 32 kHz, The pre-emphasis status and output levels are separately adjustable for the analog and digital outputs. Robust line drivers enable them to drive feet of RG coaxial cable without audible performance degradation. One input can be re-jumpered to provide a 19 kHz pilot reference out- put.

Rear-panel accessible PC-board-mounted trim pots allow the user to adjust the sensitivities of the two SCA inputs, allowing both inputs to accommodate subcarrier generators with output levels as low as mV p-p. This prevents overshoots in uncompressed digital links operating at a 32 kHz-sample rate and prevents interference to the pilot tone and RDS or RBDS subcarrier. Because this limiter closes a feedback loop around the audio processing, it allows the user to adjust the processor's subjective setup controls freely without violating BS limits, regardless of program material.

The multiplex power limiter acts on all outputs not just the composite output. It reduces clipper drive when it re- duces power, simultaneously reducing clipping distortion. To prevent audible gain pumping, a user-adjustable gain offset control allows the user to minimize the amount of gain control that the controller performs. This facilitates using the in single-frequency network applications. Only one processing structure can be on-air at a time. A special Two- Band preset creates a no-compromise "Protect" function that is functionally similar to the "Protect" structures in earlier Orban digital processors.

The Opti- mum Five-Band and the Two-Band structures can be switched via a mute-free crossfade; switching to or from the Ultra-Low Latency Five-Band structure causes the audio to mute momentarily. The can Increase the density and loudness of the program material by multiband compression, limiting, and clipping. This means that you can use an , , or to develop presets for , provided you do not use fea- tures in the other processors not supported by the If you try to import a preset that uses features unsupported by , the will interpret that pre- set as best it can by using the available features — see To Import Archived , , and Presets into Your on page for details.

You can configure PC Remote to switch between many s via a convenient organizer that supports giving any an alias and grouping multiple s into folders. The clock can be set automatically from an Internet timeserver. This combines look-ahead and band-limited clipping techniques to control STL-induced overshoots while minimizing artifacts. It can operate in either "Half- Cosine Interpolation" mode or conventional hard clipper mode. The "Half- Cosine" mode provides better separation and preservation of stereo imaging, while the "Hard" mode provides brighter sound because it creates waveforms that are closer to square waves.

Both modes provide excellent spectral protec- tion of the pilot tone and subcarrier regions. To ensure accurate peak control, the limiter operates at kHz sample rate. This allows you to compensate for overshoots in the signal path upstream from the , preventing excessive re- duction of the multiplex power. Factory Presets The Factory Presets are our "factory recommended settings" for various program formats or types. The description indicates the processing structure and the type of processing. The factory presets are listed and described starting on page You can change the settings of a Factory Preset, but you must then store those settings as a User Preset, which you are free to name as you wish.

The Factory Preset remains unchanged. To select "audio processor mode" or "stand-alone stereo encoder mode," recall a Factory or User Preset that uses this mode. The will automatically re-load DSP code to switch modes. This reload will cause all outputs to mute for about two sec- onds.

At the composite outputs, the stereo pilot tone will mute but any external subcarriers applied to the 's SCA inputs will not. An audio mute will also occur when switching to and from a "UL" ultra-low-latency preset because this also re- quires DSP code to be reloaded. User Presets User Presets permit you to change a Factory Preset to suit your requirements and then store those changes.

You can store more than User Presets, limited only by available memory in your which will vary depending on the version of your 's software. You can give your preset a name up to 18 characters long. User Presets cannot be created from scratch. You must always start by recalling a Factory Preset. Make the changes and then store your modified preset as a User Pre- set.

You can also recall a previously created user preset, modify it, and save it again, either overwriting the old version or saving under a new name. In all cases, the original Factory Preset remains for you to return to if you wish. The only way you can choose the structure of a factory preset is to edit it from a Factory preset having that structure or to edit it from an older User Preset having the desired structure.

You cannot change the structure that an existing User Preset uses. They both have sample rate converters to allow operation at 32, Both analog and digital outputs are active continuously. The 's output sample rate can be locked either to the 's internal crystal clock or to the sample rate present at its AES3 input. The can apply J. The 's provisions for J. The level, de-emphasis, and other parameters of these outputs are set in System Setup and are the same regardless of whether the is operating in its audio processor or stand-alone stereo encoder modes.

Input impedance is greater than lOkO; balanced and floating. The left and right analog outputs are on XLR-type male connectors on the rear panel. Output impedance is ; balanced and floating. The outputs can drive or higher impedances, balanced or unbalanced. Level control of the analog inputs and outputs is accomplished via software control through System Setup see step 3 on page and step 8 on page or through PC Remote. See the footnote on page and refer to Figure on page Independent level control of each output is available via software see step 6 on page The subcarriers are mixed into each composite output and their level is not affected by the compos- ite level control for that output.

Subcarrier inputs sum into the composite baseband outputs. Thus both inputs accommodate subcarrier genera- tors with output levels as low as mV p-p. The correct peak level of the stereo program applied to the stereo encoder some- times depends on the number of subcarriers in use. Some regulatory authorities re- quire that total baseband peak modulation be maintained within specified limits. You de- fine the amount of reduction in percent using the procedure in step 21 on page 2- See page for information on programming the remote control.

A jumper J6 on the circuit board can reconfigure the SCA 2 input to provide the stereo pilot tone only, which can provide a pilot reference for an RDS subcarrier gen- erator. The remote control of overshoot compensation and SCA modulation see page is not latching. You must supply a continuous current to the programmed remote input to hold the gain at its compensated level. When the chip detects such an error, it automatically switches the input to Analog. You can reconfigure the functions of the inputs and outputs via System Setup.

For example, if you are not using the stereo encoder, the three inputs ordinarily dedi- cated to controlling the state of the stereo encoder can instead be re-configured to call three additional presets. See page for information on programming the remote control interface. These computer interfaces support remote control and metering, and allow downloading software upgrades. The program displays all of the 's LCD meters on the computer screen to aid remote adjustment.

RS Serial Port PC Remote can communicate at up to 1 1 5 kbps via modem or direct connection between the computer and the through their RS serial ports. See Networking and Remote Control on page for more information. It accepts a lx 5V p-p squarewave wordclock signal at 32, You can configure the to lock its 19 kHz pilot tone and output sample frequency to this input. The sample frequency at the 's digital output does not have to be the same as the reference frequency to be locked to it.

If the output frequency is different, the output sample frequency will be the product of a quotient of integers times the ref- erence frequency. If the reference frequency is 48 kHz and the output frequency is set to It is precisely and absolutely high-frequency-controlled and peak-controlled to prevent over- modulation, and is filtered at 15 kHz to protect the 19 kHz pilot and prevent distor- tion caused by aliasing-related non-linear crosstalk. If this signal is fed directly into a stereo encoder, peak modulation levels on the air will be precisely controlled.

Peak modulation will increase, but average modulation will not. The modulation level must therefore be reduced to accommodate the larger peaks. Re- duced average modulation level will cause reduced loudness and a poorer signal-to- noise ratio at the receiver. Landline equalizers, transformers, and 15 kHz low-pass filters and pre-emphasis net- works in stereo encoders typically introduce frequency response errors and non- constant group delay.

There are three criteria for preservation of peak levels through the audio system: 1 The system group delay must be essentially constant throughout the frequency range containing significant energy ,Hz. If low-pass filters are present, this may require the use of delay equalization. An all-pole de- emphasis network like the classic series resistor feeding a grounded capacitor is not appropriate. However, this network could be corrected by adding a second resistor between ground and the capacitor, which would introduce a zero.

Low-pass filters including anti-aliasing filters in digital links , high-pass filters, trans- formers, distribution amplifiers, and long transmission lines can all cause the above criteria to be violated, and must be tested and qualified. It is clear that the above criteria for optimal control of peak modulation levels are most easily met when the audio processor directly feeds the stereo encoder.

In the , no circuit elements that might distort the shape of the waveform are interposed between the audio processor and the stereo encoder. We therefore recommend using the with its built-in stereo encoder whenever practical. If this is impossible, the next best arrangement is to feed the 's AES3 digital output through an all- digital, uncompressed path to the transmitter's exciter.

Use the 's left and right analog audio outputs in situations where the stereo en- coder and exciter are under the jurisdiction of an independent transmission author- ity, and where the programming agency's jurisdiction ends at the interface between the audio facility and the link connecting the audio facility to the transmitter.

This situation is not ideal because artifacts that cannot be controlled by the audio processor can be introduced by the link to the transmitter, by transmitter peak limiters, or by the external stereo encoder. If the transmitter is not accessible: All audio processing must be done at the studio and you must tolerate any damage that occurs later. If you can obtain a broadband kHz phase-linear link to the transmitter, and the transmitter authority will accept the delivery of a baseband en- coded signal, use the 's internal stereo encoder at the studio location to feed the STL.

Then feed the output of the STL receiver directly into the transmitter's ex- citer with no intervening processing. If an uncompressed AES3 digital link is available to the transmitter, this is also an ex- cellent means of transmission, although it will not pass the effects of the 's composite processor if you are using it. However, if the digital link employs lossy compression, it will disturb peak levels. If only an audio link is available, use the 's left and right audio outputs and feed the audio, without pre-emphasis, directly into the link.

To ensure maximum quality, all equipment in the signal path after the studio should be carefully aligned and qualified to meet the appropriate stan- dards for bandwidth, distortion, group delay and gain stability, and such equipment should be re-qualified at reasonable intervals. If the transmitter is accessible: You can achieve the most accurate control of modulation peaks by locating OPTIMOD-FM at the transmitter site and then using its stereo encoder to drive the transmitter.

However, many composite base- band STLs do not control peaks perfectly because of bounce see page , and lo- cating OPTIMOD-FM at the transmitter site where it can control peaks just prior to the transmitter's RF exciter is thus likely to maximize loudness. However, the link should have low noise, the flattest possible frequency response from ,Hz, and low non-linear distortion. We strongly recommend that you use the 's internal stereo encoder to feed the output of the encoder directly to the baseband input of the exciter through less than feet 30 meters of coaxial cable.

See Figure on page This separation is comfortably above the separation approximately 20 dB that starts to cause perceptible changes in the stereo image. An exception is Orban's stereo encoder, which does not add overshoot and contains its own overshoot limiter and composite limiter equivalent to the one in the when operated in its stand-alone stereo encoder mode.

Adkins and Robert D. Audio Engineering Society, vol. Subjects listened to Hz tones, broadband noise, and stereophonic program ma- terial through earphones and adjusted the channel separation, via a manual control, until the degradation of the spatial effect became detectable. Mean channel separa- tions ranged from 10 to The results are discussed in terms of exist- ing data on detectable stereo separation and on the discrimination of interaural time differences.

Its instruction manual contains complete information on its installation and applica- tion. If a separate stereo encoder must be used, feed the encoder directly from the 's left and right analog or preferably digital outputs. If possible, bypass the pre- emphasis network and the input low-pass filters in the encoder so that they cannot introduce spurious peaks.

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STLs are used in three fundamentally different ways. They can either 1 pass un- processed audio for application to the 's input, 2 they can pass the 's peak-controlled analog or digital left and right audio outputs, or 3 they can pass the 's peak-controlled composite stereo baseband output. The three applica- tions have different performance requirements. In general, a link that passes un- processed audio should have very low noise and low non-linear distortion, but its transient response is not important. A link that passes processed audio does not need as low a noise floor as a link passing unprocessed audio.

However, its transient response is critical. We will elaborate below. Digital Links Digital links may pass audio as straightforward PCM encoding, or they may apply lossy data reduction processing to the signal to reduce the number of bits per sec- ond required for transmission through the digital link. Such processing will almost invariably distort peak levels, and such links must therefore be carefully qualified before you use them to carry the peak-controlled output of the to the trans- mitter.

While the desired program material may psychoacoustically mask this noise, it is nevertheless large enough to affect peak levels severely. For any lossy compression system the higher the data rate, the less the peak levels will be corrupted by added noise, so use the highest data rate practical in your system. The data rate should be at least of "contribution quality" — the higher, the better. Because the uses multiband limiting, it can dynamically change the frequency response of the channel. This can violate the psychoacoustic masking assumptions made in designing the lossy data reduction algorithm.

Therefore, you need to leave "headroom" in the algorithm so that the 's multiband processing will not un- mask quantization noise. This is also true of any lossy data reduction applied in the studio such as hard disk digital delivery systems. Some links may use straightforward PCM pulse-code modulation without lossy data reduction. If you connect to these through an AES3 digital interface, these can be very transparent provided they do not truncate the digital words produced by the devices driving their inputs. Because the 's output is tightly band-limited to 15 kHz, it can be passed without additional overshoot by equally well by any link with 32 kHz or higher sample frequency.

Currently available sample rate converters use phase-linear filters which have con- stant group delay at all frequencies. If they do not remove spectral energy from the original signal, the sample rate conversion, whether upward or downward, will not add overshoot to the signal.

This is not true of systems that are not strictly band- limited to 15 kHz, where downward sample rate conversion will remove spectral en- ergy and will therefore introduce overshoot. If the link does not have an AES3 input, you must drive its analog input from the 's analog output. This is less desirable because the link's analog input circuitry may not meet all requirements for passing processed audio without overshoot. It uses a block-companded floating-point representation of the signal with J. Older technology converters including some older NICAM encoders may exhibit quantization distortion unless they have been correctly dithered.

Additionally, they can exhibit rapid changes in group delay around cut-off because their analog filters are ordinarily not group-delay equalized. The installing engineer should be aware of all of these potential problems when designing a transmission system. The digital input and output accommodate sample rates of 32 kHz, Composite Baseband Microwave STLs The composite baseband microwave STL carries the standard pilot-tone stereo base- band, and therefore receives the output of a stereo encoder located at the studio site.

Thus, no stereo encoder is needed at the transmitter. In general, a composite microwave STL provides good audio quality, as long as there is a line-of-sight transmission path from studio to transmitter of less than 10 miles 16 km. If not, RF signal-to-noise ratio, multipath distortion, and diffraction effects can cause serious quality problems. Uncompressed digital composite baseband microwave STLs, if properly designed, have excellent performance and we recommend them highly.

They are particularly desirable in a installation because they allow you to use the 's composite limiter to increase on-air loudness. However, the fact that they are digital does not eliminate the requirement that they have low frequency response that is less than 3 dB down at 0. Any such STL should be qualified to ensure that it meets this specification.

However, problems include gain- and phase- matching of the left and right channels, overloads induced by pre-emphasis, and re- quirements that the audio applied to the microwave transmitters be processed to prevent over-modulation of the microwave system. Lack of transparency in the path will cause overshoot. Unless carefully designed, dual microwave STLs can introduce non-constant group delay in the audio spectrum, distorting peak levels when used to pass processed audio. Nevertheless, in a system using a microwave STL, the is sometimes located at the studio and any over- shoots induced by the link are tolerated or removed by the transmitter's protection limiter if any.

The can only be located at the transmitter if the signal-to-noise ratio of the STL is good enough to pass unprocessed audio. Of these, the and are currently manufac- tured as of this writing and are the preferred choices because their AGCs are identi- cal to the AGC in the If the is located at the transmitter and fed unprocessed audio from a micro- wave STL, it may be useful to use a companding-type noise reduction system like dbx Type 2 or Dolby SR around the link.

This will minimize any audible noise buildup caused by compression within the Many such links have been designed to be easily configured at the factory for composite operation, where an entire FM stereo base- band is passed. The requirements for maintaining stereo separation in composite operation are similar to the requirements for high waveform fidelity with low over- shoot. Nevertheless, in a dual-microwave system, the is usually located at the main FM transmitter and is driven by the microwave receivers.

One of Orban's studio level control systems, such as the ST, protects the microwave transmitters at the stu- dio from overload. These units also perform the gain riding function ordinarily exe- cuted by the AGC section of the 's processing and optimize the signal-to-noise ratio obtainable from the dual-microwave link. If the STL microwave uses pre-emphasis, its input pre-emphasis filter will probably introduce overshoots that will increase peak modulation without any increases in average modulation.

If the studio level control system is capable of producing a pre- emphasized output, we strongly recommend that the microwave STL's pre-emphasis be defeated and pre-emphasis performed in the studio level control system. This frees the system from potential overshoot. The Orban ST can be readily con- figured to produce a pre-emphasized output. Further, it is common for a microwave STL to bounce because of a large infrasonic peak in its frequency response caused by an under-damped automatic frequency control AFC phase-locked loop.

This bounce can increase the STL's peak carrier de- viation by as much as 2dB, reducing average modulation. Many commercial STLs have this problem. Some consultants presently offer modifications to minimize or eliminate this prob- lem. If your exciter or STL has this problem, you may contact Orban Customer Service for the latest information on such services.

Whether landlines should be used or not depends upon the quality of the lines lo- cally available, and upon the availability of other alternatives. Due to line equalizer characteristics and phase shifts, even the best landlines tend to veil audio quality slightly. They will certainly be the weakest link in a FM broadcast chain. Slight frequency response irregularities and non-constant group delay characteristics will alter the peak-to-average ratio, and will thus reduce the effectiveness of any peak limiting performed prior to their inputs.

This is usually practical. There are similar requirements for FM exciters. Nevertheless, in some installations some overshoot is inevitable. If this is a problem in your installation, the 's remote control feature offers the means to reduce the peak level of the 's audio output as necessary. In addition, source material is often supplied through a lossy data reduction algorithm, whether from satellite or over landlines. All such algorithms operate by increasing the quantization noise in discrete fre- quency bands.

If not psychoacoustically masked by the program material, this noise may be perceived as distortion, "gurgling," or other interference.

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Psychoacoustic calculations are used to ensure that the added noise is masked by the desired pro- gram material and not heard. Cascading several stages of such processing can raise the added quantization noise above the threshold of masking, such that it is heard. In addition, at least one other mechanism can cause the noise to become audible at the radio. This can cause noise that would otherwise have been masked to become unmasked because the psychoacoustic masking conditions under which the masking thresholds were originally computed have changed. Accordingly, if you use lossy data reduction in the studio, you should use the highest data rate possible.

This maximizes the headroom between the added noise and the threshold where it will be heard. About Transmission Levels and Metering Meters Studio engineers and transmission engineers consider audio levels and their meas- urements differently, so they typically use different methods of metering to monitor these levels. The VU meter is an average-responding meter measuring the approxi- mate RMS level with a ms rise time and decay time; the VU indication usually under-indicates the true peak level by 8 to 14 dB. The PPM has an attack time of 10ms, slow enough to cause the meter to ignore narrow peaks and under-indicate the true peak level by 5 dB or more.

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The absolute peak-sensing meter or LED indica- tor shows the true peak level. It has an instantaneous attack time, and a release time slow enough to allow the engineer to read the peak level easily. Studio Line-up Levels and Headroom The studio engineer is primarily concerned with calibrating the equipment to pro- vide the required input level for proper operation of each device, and so that all de- vices operate with the same input and output levels.

This facilitates patching devices in and out without recalibration. For line-up, the studio engineer uses a calibration tone at a studio standard level, commonly called line-up level, reference level, or operating level. So the studio standardizes on a maximum program indication on the meter that is lower than the clipping level, so those peaks that the meter does not indicate will not be clipped.

Line-up level is usually at this same maximum meter indication. Older facilities with h-8 dBu standard level and equipment that clips at h or h dBu will experi- ence noticeable clipping on some program material. Line-up level is usually PPM4, which corresponds to h-4 dBu. Instantaneous peaks will reach h dBu or more on voice. Instantaneous peaks will reach -i-l 1 dBu or more. This peak overload level is defined differently, system to system. In AM modulation, it is negative carrier pinch-off.

In digital, it is the largest pos- sible digital word. For metering, the transmission engineer uses an oscilloscope, absolute peak-sensing meter, calibrated peak-sensing LED indicator, or a modulation meter. A modulation meter usually has two components — a semi-peak reading meter like a PPM , and a peak-indicating light, which is calibrated to turn on whenever the instantaneous peak modulation exceeds the overmodulation threshold. Left and right input level is shown on a VU-type scale 0 to dB , while the metering indicates absolute instantaneous peak much faster than a standard PPM or VU me- ter.

The input meter is scaled so that 0 dB corresponds to the absolute maximum peak level that the can accept h- 27 dBu. Composite modulation is indicated in percent- age modulation, absolute instantaneous peak indicating. Note that if the 's subcarrier inputs are used, the meter will not indicate the subcarriers' effect on composite modulation because the subcarriers are mixed into the composite signal in the analog domain, after it is metered.

Therefore, you must mentally add the subcarriers to the meter indication, or refer to an external, cali- brated modulation monitor. Built-in Calibrated Line-up Tones To facilitate matching the output level of the to the transmission system that it is driving, the contains an adjustable test tone oscillator that produces sine waves at 's analog or digital left, right and composite outputs. The frequency and modulation level of the line-up tones can be adjusted from the front panel as described in Section 3.

The pilot tone stereo system has an interleaving property, which means that the ste- reo composite modulation is approximately equal to the higher of the left or right channels. Because the pilot tone is phase-synchronous with the stereo subcarrier, the composite modulation will actually increase about 2. Thus, as the frequency of the Test Tone is changed, the level at the 's line output will fol- low the selected de-emphasis curve. At Hz, switching the de-emphasis out or in will have negligible effect on the level appearing at the 's left and right audio outputs.

You can adjust the frequency and modulation level of the built-in line-up tone. You can use the front panel, the PC Control software, or the opto-isolated remote con- trol interface ports to activate the Test Tone. It will also pass program material, with no gain reduction or pro- tection against overmodulation. Monitoring on Loudspeakers and Headphones In live operations, highly processed audio often causes a problem with the DJ or presenter's headphones. Some talent moving from an analog processing chain will require a learning period to become accustomed to the voice coloration caused by "bone-conduction" comb filtering.

This is caused by the delayed headphone sound's mixing with the live voice sound and introducing notches in the spectrum that the talent hears as a "hollow" sound when he or she talks. All digital processors induce this coloration to a greater or lesser extent. Fortunately, it does not cause confusion or hesitation in the talent's performance unless the delay is above the psychoacoustic "echo fusion" Haas threshold of approximately 20 ms and the tal- ent starts to hear slap echo in addition to frequency response colorations. The normal delay through the is about 1 5 ms.

A 1 5 ms delay is comfortable for most talent because they do not hear echoes of their own voices in their head- phones. Further, the offers a second, ultra-low-latency multiband structure with a delay of about 5 ms. Although this does not offer the same favorable trade- off between loudness, presence, and low distortion as the optimum multiband struc- ture, it is available for use in situations where a given individual cannot tolerate the 1 5 ms delay of the optimum structure. However, management should carefully con- sider whether compromising the sound of the radio station for its entire audience is an acceptable price for indulging a given personality's demands.

Because of the availability of both optimum and low latency structures, customers can confidently replace an older, low-delay processor with the with no studio wiring changes. Moreover, off-air cueing of remote talent is routine. A better solution to the monitoring conundrum is this: The 's analog outputs can be switched to provide a low-delay monitoring feed while still keeping the op- timum multiband structure on-air by using the digital or composite output to drive the transmitter. The monitor feed has no peak limiting and thus cannot drive a transmitter, but its 5 ms delay is likely to be more comfortable to talent than the 15 ms delay of the optimum processing chain because of less acoustic comb filtering.

If the talent relies principally on headphones to determine whether the station is on the air, simple loss-of-carrier and loss-of-audio alarms should be added to the system when the 's monitor output is used. The can be interfaced to such alarms through any of its eight its GPI remote control inputs, cutting off the low-delay au- dio to the talent's phones when an audio or carrier failure occurs. Run EAS tones and data through the Note that processing may not allow the full modulation level as required by EAS standards.

It may therefore be necessary to temporarily defeat the 's processing during the broadcast of EAS tones and data. See "Test Modes," on page for more in- formation. Place the in Bypass mode locally. You can set the bypass gain with the Bypass Gain control located to the right of the Mode control. B Begin EAS broadcast. This will restore the processing preset in use prior to the Test Mode. Place the in Bypass mode by remote control. C Place the in bypass mode by remote control. You may also choose to insert EAS broadcast tones and data directly into the transmitter, thus bypassing the for the duration of the EAS tones and data broadcast.

PC access is permitted only with a valid user-defined passcode. PC remote control can be ended from the front panel; this feature effectively pre- vents simultaneous remote and local control. See Security and Passcode Programming starting on page for more detail. We will carefully re- view your suggestions for improvements to either the product or the manual. Please email us at custserv orban. IMPORTANT: This warranty does not cover damage resulting from accident, misuse or abuse, lack of reasonable care, the affixing of any attachment not provided with the product, loss of parts, or connecting the product to any but the specified recep- tacles.

This warranty is void unless service or repairs are performed by an authorized service center. No responsibility is assumed for any special, incidental, or consequen- tial damages. However, the limitation of any right or remedy shall not be effective where such is prohibited or restricted by law. Simply take or ship your Orban products prepaid to our service department. Be sure to include a copy of your sales slip as proof of purchase date.

We will not repair transit damage under the no-charge terms of this warranty. Orban will pay return shipping. See Technical Support on page No other warranty, written or oral, is authorized for Orban Products. This warranty gives you specific legal rights and you may have other rights that vary from state to state.

Some states do not allow the exclusion of limitations of inciden- tal or consequential damages or limitations on how long an implied warranty lasts, so the above exclusions and limitations may not apply to you. This war- ranty does not cover damage resulting from misuse or abuse, or lack of reasonable care, or inadequate repairs performed by unauthorized service centers. Shipment of the defective item for repair under this warranty will be at the customer's own risk and expense. This warranty is valid for the original purchaser only.

This offer applies only to new Orban products purchased from an authorized Orban Dealer. To accept the extended five-year warranty, please sign and date below, and fax this copy along with a copy of your original invoice showing date of purchase to Installation consists of: 1 unpacking and inspecting the , 2 mounting the in a rack, 3 connecting inputs, outputs and power, 4 optional connecting of remote control leads and 5 optional connecting of computer interface control leads. When you have finished installing the , proceed to "Quick Setup," on page 2- DO NOT connect power to the unit yet!

Unpack and inspect. A If you note obvious physical damage, contact the carrier immediately to make a damage claim. If you should ever have to ship the e. C Complete the Registration Card and return it to Orban. Please fill in the Registration Card and send it to us today. The Registration Card is located after the cover page. Customer names and information are confidential and are not sold to anyone. Install the appropriate power cord. AC power passes through an lEC-standard mains connector and an RF fil- ter designed to meet the standards of all international safety authorities.

If you need to change the plug to meet your country's standard and you are qualified to do so, see Figure Otherwise, purchase a new mains cord with the correct line plug attached. Mount the in a rack. There should be a good ground connection between the rack and the chas- sis — check this with an ohmmeter to verify that the resistance is less than 0. Mounting the unit over large heat-producing devices such as a vacuum-tube power amplifier may shorten component life and is not recommended. Equipment life will be extended if the unit is mounted away from sources of vi- bration, such as large blowers and is operated as cool as possible.

Connect inputs and outputs. See the hookup and grounding information on the following pages. Connect remote control interface, optional For a full listing of 's extensive remote control provisions, refer to Remote Control Interface Programming on page Optically-isolated remote control connections are terminated in a type DB male connector located on the rear panel. It is wired according to Figure The - terminals can be connected together and then connected to ground at pin 17 to create a Remote Common.

A current- limited -IVDC source is available on pin TALLY 1 TALLY 2 Connect tally outputs optional See the schematic on page In stereo encoder mode, the supports two hardware tally outputs, which are NPN open-collector and operate with respect to pin 1 common. Therefore, the voltage applied to the load such as a relay or opto-isolator must be positive. You can use the 12 VDC source on pin 25 to drive the high side of the load, tak- ing into account the fact that the voltage on pin 25 is current limited by a Q resistor. The tally outputs are protected against reverse polarity.

To avoid damaging the , limit the current into a tally output to 30 mA. DO NOT connect a tally output directly to a low-impedance voltage source! The tally outputs are not protected against this abuse and the output transistors are likely to burn out. Note that the tally outputs have no special RFI protection. Therefore, it is wise to use shielded cable to make connections to them. See step 12 on page for instructions on using the tally outputs.

Connect to a computer You can connect to a computer via the 's serial connector or via an Ethernet network. See Networking on page Procedures and instructions for connecting to a PC are subject to development and change. We advise you to download the latest version of this manual in pdf format from ftp. You can use Adobe's. If you do not have the. A Remote Interface Connector allows you to connect the to your existing transmitter remote control or other simple contact-closure control devices.

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See Remote Control Interface Programming on page The remote control ac- cepts a DB connector. Analog Inputs and Outputs are provided to support left and right audio signals through XLR-type connectors. Two Composite Baseband Outputs are provided, each with independent output level control. Each output uses a BNC connector. Each input uses a BNC connector. Typically, the reference signal comes, directly or indirectly, from a GPS-derived frequency standard or a rubidium frequency stan- dard.

You can lock the 's DSP clock to this reference, which in turn locks the 19 kHz pilot tone to the reference. This is useful in single frequency networks to pre- vent beating between various transmitters' pilot tones in areas of mutual interfer- ence. Input and Output Connections Cable We recommend using two-conductor foil-shielded cable such as Belden or equivalent for the audio input and output connections because signal current flows through the two conductors only. The shield does not carry signal and is used only for shielding. For this application, the dBm scale on volt- meters can be read as if it were calibrated in dBu.

The input is EMI suppressed. Connect the black wire to the pin on the XLR-type connector 3 or 2 that is considered Low by the standards of your organization. In high RF fields like a transmitter site , also connect the shield to pin 1 of the male XLR-type connector at the input. The source impedance is 50Q.

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The outputs are EMI suppressed. Connect the Low pin of the XLR-type connector 3 or 2, depending on your organization's standards to circuit ground; take the High output from the re- maining pin. No special precautions are required even though one side of the output is grounded. Instead, connect the shield to the input destination. Con- nect the red or white wire to the pin on the XLR-type connector 2 or 3 that is considered High by the standards of your organization.

The program input and output are both equipped with sample rate converters and can operate at 32, Per the AES3 standard, each digital input or output line carries both the left and right stereo channels. The connection is Q balanced. The AES3 standard specifies a maximum cable length of meters. While almost any balanced, shielded cable will work for relatively short runs 5 meters or less , longer runs require used of llOn balanced cable like Belden B, B plenum rated , multi-pair xF, xF, or 78xxA.

Single-pair category 5, 5e, and 6 Ethernet cable will also work well if you do not require shielding. The AES3id standard is best for very long cable runs up to meters. This specifies 75Q unbalanced coaxial cable, terminated in BNC connec- tors. The digital input clip level is fixed at 0 dB relative to the maximum digital word.

The maximum digital input will make the input meters dis- play 0 dB. The reference level is adjustable using the Dl Ref control. The is a "multirate" system and its internal sample rate is 32 kHz and multiples thereof up to kHz. The output is strictly band-limited to 16 kHz. Therefore, the output can pass through a 32 kHz uncom- pressed link with bit-for-bit transparency. Because sample rate conversion is a phase-linear process that does not add bandwidth, the 's output signal will continue to be compatible with 32 kHz links even if it under- goes intermediate sample rate conversions for example, 32 kHz to 48 kHz to 32 kHz.

Composite Output and Subcarrier Input There are two composite outputs. They carry the encoded stereo signal, the stereo pilot tone, and any subcarriers that may have been applied to the 's subcarrier inputs. Each output's level is independently adjustable from dBu to h- As shipped, the link is on pins 3 and 4, yielding 0 Q impedance. To reset a given output to 75Q, place the link on pins 1 and 2 of its associated jumper.

See the schematic on page and the parts locator diagram on page Each output can drive up to 75n in parallel with 0. Connect the 's composite output to the exciter input with up to feet Longer runs of coax may increase problems with noise, hum, and RF pickup at the exciter. In general, the least troublesome installations place the close to the exciter and limit the length of the composite cable to less than 6 feet 1.

We do not recommend terminating the exciter input by 50n or 75n unless this is unavoidable. The frequencies in the stereo baseband are low by comparison to RF and video, and the characteristic impedance of coaxial cable is not constant at very low frequencies. Therefore, the transmission system will usually have more accurate amplitude and phase response and thus, better stereo separation if the coax is driven by a very low impedance source and is terminated by greater than Ikn at the exciter end.

This also eases thermal stresses on the output amplifier in the stereo encoder, and can thus extend equipment life. Ground loops can occur if your exciter's composite input is unbalanced, although you can usually configure system grounding to break them for Load Capacitance pF Figure Separation vs. If you encounter intractable ground loop or other noise problems between the and your exciter, we suggest purchasing the Orban CIT25 Composite Isolation Transformer.

Designed to be installed adjacent to each exciter, the CIT25 provides rigorous ground loop isolation between the composite output and the ex- citer's input and presents the with a balanced, floating load. If the CIT25 is used, the exciter must present a IkQ or greater load to the transformer for proper transformer operation. The subcarrier SCA inputs are provided for convenience in summing subcarriers into the baseband prior to their presentation to the FM exciter. The subcarrier inputs will accept any subcarrier or combinations of sub- carriers above 23 kHz.

To access J6, remove the 's top cover according to the instructions in step 1 on page The schematic showing J6 is on page Connect your subcarrier generator s to the 's subcarrier input s with coaxial cable terminated with BNC connectors. The subcarrier inputs have greater than Q load impedance and are unbalanced. You can use the 1 9K Ref control in Setup to determine whether the 1 9 kHz pilot ref- erence output will be in-phase 0 DEG with the pilot tone present in the composite output or will lead it by 90 degrees 90 DEG.

A menu item allows you to synchronize the output sample frequency to the frequency present at the sync. The connector is a female BNC with the shell grounded to chassis. To permit daisy-chaining sync signals, the input impedance is greater than 1 KQ. This will prevent perform- ance-degrading reflections in the cable.

This is required for both wordclock and AES1 1 id operation. Grounding Very often, grounding is approached in a "hit or miss" manner. However, with care it is possible to wire an audio studio so that it provides maximum protection from power faults and is free from ground loops which induce hum and can cause oscilla- tion. In a modern system with low output impedances and high input impedances, a balanced input will pro- vide common-mode rejection and prevent ground loops — regardless of whether it is driven from a balanced or unbalanced source.

Its subcarrier inputs are unbalanced, but fre- quency response is rolled off at low frequencies to reject hum. Power Ground A Ground the chassis through the third wire in the power cord. I could hardly wait to get one of them on the mic stand to work it out. I concur with Don Cicchetti's post of a couple of days ago Not overly hot in 5k region. Nice round and level response at 12" to 18", but closer proximity effect has a usable warmness for certain situations, never got overly boistrous.

There was no strident, slice your head off, "cheap condensor" upper mids either. Siblance problems were minimal to non-existent with myself and my daughter who is famous for being able to "set the S's off". These initial tests performed on a neutral Mackie 32x8 pre, no Eq. Lettering is engraved into the mic as opposed to just silk screened on. I truly feel these mics are the steal of the century. IMO, anyone taking the pure "those mics are crap" point of view, either haven't actually used them or have nip'd a little too deeply from the snake oil bottle. Put your fears and prejudices aside Michael Broyles Stafire Media.

I got my C1 last week and setup it up over the weekend and boy it sounds so good. One of the greatest things I love about the mic is the proximity. On my inexpensive Audio Technica Condensers ATM31 and MBC I had to get right up on the mics to get decent sound but with the C1 I can put up the wind screen and get 12" away and pretty much stay at unity gain on my preamp and have plenty of signal.

Ya know what I mean? This is a great mic not to mention the unbelievable price. Thanks for taking us Independent Recording Artist to the next level Alan. You've put us little guys in the big leagues. I just completed a CD project for a singer named Beth Ivy. She has a wonderful voice that reminds me, at times, of an early Joni Mitchell with a soulful twist. I set up the AT , and her first response was that it made her sound "nasal. The next day, the Studio Projects C1 that I had ordered came in.

I set it up for the very first time in my vocal session with her. After the first verse, she stopped singing, came out of the vocal booth and said "what kind of mic is this? It makes my voice sound gorgeous! There was no way that I was going to tell her what I had paid for it! I am so pleased that I had bought this mic. I had only read comments and reviews about it on the internet.

I bought one since the price was so good that, even if it sucked - I figured that I could always throw it on some instrument that it would sound decent on. Now the C1 is my main vocal condenser. It has changed my concept that you have to spend thousands to get a great studio mic. I enjoy this mic so much that I have recommended it to two friends who, on my recommendation alone, purchased their own C1s. Thank you. I own just a small home studio, but this mic has revolutionized my sound. Thanks for designing this mic. I still dont know how to use that damn enhancer on the VC1Q, but oh well, I am getting good results with the compressor and the EQ.

I would like to try the Tube mic sometime. Rich Bischoff. A couple of weeks ago I recieved my first C1. I say first because after the first week I had to order another, Which I recieved last week. They are essentially identical sonicaly and physically. I don't know how you do it but I'm glad you did. I've been thirsting after microphones since I was 15 now in my 50's This is an amazing mic and I love the absence of off axis coloration.

Now I've got my eye on the T3. Thanks again. Tom Hagen Tom Hagen Studios. I want to thank you for the C1 mic. I read several reviews on the C1 over the net, and I was curious. I have a moderate studio budget, so I can't afford to buy junk. Every purchase dollar counts. So, after reading several great reviews from educated opinions that I trust, I bought the C1.

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This inexpensive mic has singlehandedly raised the level of vocals that I do at home to a level comparable with any studio vocals that I have recorded elsewhere. I usually rent or borrow high-end mics to do critical vocals at home, or I simply go to a commercial studio. I have always longed for the ability to do vocals that I actually love at home. I have used many of the usual suspects, with moderate success. Finally, I am competitive with the vocal sound that I usually get in commercial studios Thank you for making such a wonderful and useful mic available to "the common man".

From all of us who are "in" on this secret Thank You! Esmond Lewis A friend of mine who is a recording engineer and a very picky guy over the gear he uses tried out my C-1 sample over the weekend. He will not give it back. He said it sounds spectacular. He has recorded with every mic imaginable. Ken is one of the top engineers in Chicago who records world renown rap and house music artist.

He said that most of his clients will get a chance to hear this mic perform. List of artists Ken works with: Bad Boy Bill Chu-Dark Side Productions Faders were at 0. The first run was done with gains set to maximise the signal from each mic, then 2 more recordings with all gains set the same to compare the signal strength of each mic.

We decided to record acoustic guitar, newly strung in preference to voice, and the outcome Yes I know it is supposedly baised to vocals Second lowest signal strength, very "muddy" lacking top-end definition. Imagine having cotton wool stuck in your ears. The guy that owns it was shattered. Third highest or lowest OK it was in the middle signal strength, clear defined sound across the tonal range, somewhat transparent or glassy quality. Definitely not unpleasant. Studio Projects C Signal strength was almost identical to the U, clear defined sound across the tonal range, definitely true to the original acoustic sound.

Neumann U Signal strength, "as above". Again a clear defined sound, but with more bottom end or "body" not heard naturally from the guitar. Am I surprised at the results? Maybe in other circumstances with certain instruments or vocalists it may, but tonight, with an acoustic guitar, this U was eclipsed by the C1. And the guy that owns the AKGs and the U will be buying 2 C1s as soon as he can get the money together. Enough said! For now the skeptic in me has been quieted. Furthermore, me and Drew Daniels www. We also modified the mic Drew is a genious, years as audio engineer at tascam, jbl by inserting a cone shape foam donut around it so as to avoid reflections close to the diagphram.

I told him that i'd be interested to test the T3 against his flat laboratory mic, bruel kjaer pardon the spelling and his C We recorded the sounds on wav and i plan to do a review on it soon and post it on the web. It's SO identical! I bought the C1 a while back and simply could not be happier with the decision. I am not a high powered, world renown engineer or high roller. I use the C1 as main vocal and lead instrument micing. I cant believe what I hear in my headphones as the live shows are aired.

I'm used to hearing the Shure 58s, but when I turned on the phantom power to my C1 the world of Live music lit up and all became crystal clear again for our listeners. I have just place another order for more C1s for my use. By the way the mics and gear belong to me, so the choice of mics used on my StarLight BlueGrass show is mine.

The show has taken on a whole different sound quality since I have upgraded the equipment. The listeners have also noticed the quality, too. Thanks for reading my note and for the great sound gear. I got my C1 yesterday. Good God man, how can you sell these for so little? I had recently set up several pre's and mics to do a shoot out for my own voice. All of them sounded like I'd been singing through a solid cardboard pop filter compared to the C1.

I'm absolutely floored, I'm flabbergasted! I finally know where that beautiful high end sheen you hear on loads of voices on the radio comes from, it's the mic! A good friend had brought over his Blueberry and let me try it and it really suited my voice. These takes were much closer, quality wise, but my partner; Selena listened to them and just said all matter of factly; "The C1 sounds better than the Blueberry. I usually hate the sound of bass and low mids in my voice, but on the C1, my voice sounds great to my own ear.

I have to give you a couple of suggestions. I wish it had a low cut filter at say Hz. I also think it needs a mic stand screw in adapter next to the XLR jack because if the suspension mount were ever missplaced or damaged, you'd have a hand held only mic. Other than that, the C1 is perfect! I now believe every word of all those reviews I've read on the net and know it's not hype. Great case and suspension mount too! You guys are going to definitely change the microphone market.

Thanks again, your product is superb. David Keplinger. I emailed you once as a response in one of the many threads about the C1 and you were kind enough to respond. I finally got my hands on a C1 about a week ago and did some real serious test over the weekend to see if it lived up to its hype. I don't expect you will remember but I am the person who has purchased several 5 mics from audio. By the way - I already felt that my 2 CR mics sounded better than the U What I did for a test is very unscientific. I just set up the C1, U87, U47 and a TLM in a vocal booth and had several singers I work with on commercials come in and do generic readings then actual singing on some basic cover tunes.

The results were as follows. I used on one test for each individual Neve preamps. I then switched to the Sytek preamp, then went stright into the board Mackie D8B. Their opinions never changed. Very unscientific I know but the only truth is in the ears and that is what counts. The only reason I will keep the other mics is because customers ask if you have these mics.

Heck I have slowly been converting folks that use the CR's and CR mics, but I have a session this week with a new band in town so I'm going to give their vocalist the same test since he specifically asked for a U Should be interesting to "hear" the results. Bottom line - if anyone where to compare the 2 C1 and U47 before purchase I think Neuman will have to change their pricing structure. Keep up the good work.

Thanks for a great product. David Artis. While using a C1 as an overhead to record a group acoustic practice, one of the members knocked over the mic stand. It fell hard on a ceramic tile floor with the C1 taking the full impact.