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The finest Solution-D microphone, pure Neumann sound in the digital domain, for use with AES/EBU or AES 42. The signal is converted directly at the capsule

The Georg Neumann GmbH company has repeatedly played a leading role in the field of analog microphone technology. The first condenser microphone, switchable polar patterns, the first stereo microphone and the 48-V phantom power supply were all developed by Neumann and opened up new possibilities in the realm of microphone technology and recording techniques.

With the Solution-D system, Neumann has now succeeded in developing a digital microphone with the dynamic range and signal fidelity of the best analog studio microphones, thus forging the last link in the chain of digital audio production.

By means of a new, high-quality A/D conversion process, a specially developed synchronization method, and remote control of standard microphone parameters and various integrated signal processing functions, the Solution-D system is able to fulfill the highest demands of professional audio production.

The comprehensive features of Solution-D make it possible to achieve excellent quality even in small productions, using only a microphone and a recording device. Neumann is thus once again playing a pioneering role in the field of microphone technology and recording techniques.

The digital Solution-D system consists of three components: the D-01 digital microphone, the DMI-2 digital microphone interface, and the RCS remote control software which operates and remotely controls the microphone. The signal and data transmission of the microphone conform to the new AES 42-2001 standard that governs the transmission of output signals, the power supply and the remote control of microphones with digital outputs. Neumann was instrumental in drafting this new standard, which provides the necessary preconditions for the development of digital microphone technology.

Any sound engineer who is familiar with digital recording technology can begin production with the Solution-D immediately, without extensive training. The same principles of acoustics and recording technology apply as in the case of analog microphones. However the additional features of digital microphones open up new possibilities during production.

Remote control of standard microphone parameters
With the Solution-D, standard microphone settings such as the polar pattern, pre-attenuation and low-cut filter can be controlled remotely. Changing the settings of microphone parameters is greatly simplified, which facilitates rapid testing of different settings so as to optimize the sound quality. It is no longer necessary to make written records of all the parameters.

Integrated digital audio signal processing
An A/D converter developed by Neumann, especially optimized for capsule signal conditions, receives the output signal directly from the capsule. Gain adjustments required for following devices are performed digitally, in the microphone itself. Analog components such as preamplifiers and A/D converters are thus no longer required, resulting in significant cost savings.

The front-end conversion of the analog signal provides a marked improvement in dynamic range. This is noticeable throughout the entire signal chain, and makes level adjustments less critical.

Mixing console functions such as mute and phase reverse, which affect the microphone signal, are likewise integrated into the microphone. Even commands such as On Air (red light) are implemented via remote-controlled LEDs on the microphone. A very special feature is the transient limiter. For the first time, this function can be applied at the most effective point, namely at the signal source, in order to reduce damaging transients that are very short but exhibit a large amplitude. Analog microphones require extremely large headroom in the following signal path to accommodate such signals.

Data transmitted by the microphone
Information transmitted by the microphone includes the name of the manufacturer, the model, the serial number, the software version and a list of all remote-control functions supported by the microphone. Microphone status indicators, including certain warning signals and the ready state, are also transmitted.

The operator can control all the microphones centrally from the control room by means of the Remote Control Software, a graphical user interface which is installed on a desktop or laptop computer.

During production, the sound engineer can continuously monitor all the microphones, since all the important parameters, including signal levels, are displayed on the screen. In addition, a text input field is provided, allowing a channel name to be assigned, e.g. indicating the sound source. All of the microphone settings can be stored in configuration files and retrieved as required.

Why digital signal processing
The digitizing of audio data started more than 15 years ago, beginning at the end of the signal processing chain, with the development of the first digital recording devices. By now, digital versions of almost all audio signal processing components are available.

It is well-known that digital signals provide the basis for precise mathematical calculations and processing, permitting signals to be changed, copied, transmitted and stored as desired, with no loss of quality.

In contrast, analog signal processing is characterized by limited precision, error accumulation and a lack of redundant signal information and error correction procedures. Each analog signal processing step is associated with a degradation of the original signal quality. This results in a gradual decrease in the dynamic range, due to the addition of noise voltages and nonlinear distortion.

In addition, digital processing enables functions to be performed which are difficult or impossible to implement with analog signal processing. This applies particularly to functions which require intermediate storage of data.

A/D Conversion
Despite continuing improvement, integrated circuits available on the market today still constitute a limiting factor with regard to the A/D conversion of audio data.

For example, the best delta-sigma A/D converters currently available as integrated circuits provide a dynamic range of 115-120 dB (A-weighted) for a theoretical word length of 24 bits.

In contrast, a high-quality analog condenser microphone has a dynamic range of up to 130 dB. A high-performance A/D converter is therefore required, in order to avoid adding noise to the audio signal. In addition, the conversion process must be optimally adapted to the signal levels and source impedance in the microphone.

In the case of A/D conversion in a mixing console or other device, as a rule, deterioration in signal quality is to be expected, since the conversion occurs only after level matching has been carried out. As a result, the dynamics are influenced by the headroom and by the characteristics of the microphone preamplifier and A/D converter.

Therefore, the goal was to develop a method for performing high-quality digitization of the capsule signal in the microphone itself. This permits level adjustments and other processing steps to be carried out digitally, ensuring that the signal quality generated by the microphone is not impaired by subsequent processing.

Synchronization
The digital audio data stream transmitted by the microphone must be synchronized to allow processing by the mixing console. This necessitated the development of a reliable synchronization method which could function independently of the length of the microphone cable.

The Neumann company has made a decisive contribution in this area, by developing a process that has become part of the AES 42-2001 standard:

At the microphone signal receiver side, a frequency-phase comparison is performed by a master clock, and a rather slow feedback signal is generated to control a voltage-controlled crystal oscillator (VCXO) in the microphone. This sets up a closed feedback loop, similar to a phase-locked loop (PLL). Following equalization and A/D conversion of the feedback signal, it is transmitted to the microphone as part of the remote signal data stream, conforming to the AES 42-2001 standard. This procedure is not only reliable, but also results in very small jitter amplitudes.

D-01 Digital Microphone
At first glance, the D-01 appears similar to analog microphones. However, immediately following the capsule is an A/D converter developed by Neumann, which is especially optimized for the signal conditions of the microphone capsule. The signal generated by the capsule is immediately converted into a digital signal, resulting in an internal 28-bit signal with a dynamic range of more than 130 dB (A-weighted, including the microphone capsule).

By means of a field programmable gate array (FPGA), the digital signal is then processed further in the microphone itself. This means that parameters such as the polar pattern, pre-attenuation, low-cut filter, gain and various switching functions can be set digitally and controlled remotely. External components such as analog preamplifiers and A/D converters are thus no longer required.

For identification purposes, the microphone transmits information such as the name of the manufacturer, the model, the serial number and the software version to the receiver.

The microphone is equipped with a 3-pin XLR connector. A bi-directional signal conforming to the AES 42-2001 standard is transmitted, which contains the balanced digital microphone output signal, the phantom power supply and a remote control data stream. This data stream also includes a signal for synchronizing the microphone with a master clock.

DMI-2 Digital Microphone Interface
Equipment that supports the new AES 42-2001 standard can process the output signal of the Solution-D microphone directly. Otherwise, the Neumann DMI-2 Digital Microphone Interface must be used. This is a separate, two-channel device which converts the AES 42-2001 data format of the microphone into an AES/EBU signal.

The DMI-2 is operated by means of the Neumann Remote Control Software, which runs on a desktop or laptop computer. The computer and the DMI-2 are connected via a USB port and an interface converter (USB to RS 485). If a large number of microphones is used, several DMI-2s can be cascaded, and each interface can be addressed individually.

In addition to a word clock input and output, the DMI-2 also has an internal master word clock generator. If no master word clock signal (e.g. from a mixing console) exists at the input, the DMI-2 internal master is automatically used to synchronize the two microphone channels and is connected to the word clock output. External commands such as red light for On Air can be controlled via a 9-pin user port.

RCS Remote Control Software
The digital microphone is controlled via the Remote Control Software, a component of the Solution-D system, which runs as an independent program on desktop and laptop computers.

All important parameters are displayed on the screen and can be changed at any time. During production, the sound engineer has complete control over the status and parameters of all the microphones and can change settings quickly and conveniently if necessary.

The parameters displayed include the polar pattern, pre-attenuation, low-cut filter, gain, various microphone status indicators, command indicators, and mute and phase reverse functions. It is also possible to monitor signal levels on the screen.

Information transmitted by the microphone, such as the name of the manufacturer, the model and the serial number, is displayed and serves to identify the microphone. In addition, it is possible to input additional information, such as the name of the sound source. The parameters for the complete microphone setup can of course be stored and retrieved as desired.

Features

  • Extremely wide dynamic range: more than 130 dB
  • Neumann A/D conversion process
  • Pre-amps and external A/D converter are no longer necessary
  • Wide range of new functions through integrated digital signal processing
  • Control and monitoring of all microphone parameters and functions through Remote Control Software

  • Digital Microphone
  • A/D Converter, optimized for direct conversion of the capsule signal
  • New capsule design
  • Additional DSP functions
  • Limiter (incl. de-esser)
  • All parameters and functions remote controllable

  • Digital Microphone Interfaces
  • AES 42-2001 to AES/EBU conversion
  • Reliable synchronization through new PLL process
  • 2 or 8 channels
  • Phantom powering
  • Word clock in and out
  • Integrated word clock generator

  • Remote Control Software
  • Control and monitoring of all parameters
  • DSP function control
  • Peak level meter
  • Gain reduction bargraph for limiter
  • Save, copy, and recall for complete setups
  • Specifications

    Acoustic transducer:
    K 07 large double-diaphragm capsule, diameter 30 mm with protected internal electrodes

    Directional characteristic:
    15 remote controllable polar patterns, from omni to cardioid to figure-8

    Frequency response:
    20 Hz to 20 kHz

    Free-field sensitivity1) 2):
    -44 dBFS

    Equivalent noise level, CCIR3):
    19 dB

    Equivalent noise level, A-weighted3):
    8 dB-A

    Signal-to-noise ratio2), CCIR3):
    75 dB

    Signal-to-noise ratio2), A-weighted3):
    86 dB

    Maximum SPL at 0 dBFS:
    138 dBSPL

    Dynamic range, A-weighted3):
    130 dB

    Latency:
    44,1/48 kHz: 52 samples
    88,2/96 kHz: 61 samples
    176,4/192 kHz: 121 samples (AES 3)

    Supply voltage range:
    +6 V to +10.5 V

    Current consumption:
    max. 220 mA

    Weight:
    approx. 700 g,

    Diameter:
    63.5 mm, Length: 185 mm

    Interface:
    AES42

    Remote controlled functions:
    Polar pattern4)
    Low-cut: flat, 40, 80, 160 Hz
    Pre-attenuation: 0 , -6, -12, -18 dB
    Gain: 0...63 dB in 1 dB steps, clickless
    Testsignal: 1 kHz, pink noise, white noise
    Sampling rates: 44.1, 48, 88.2, 96, 176.4, 192 kHz
    Parametric compressor/limiter (incl. de-esser function)
    Independent peak limiter avoiding any clipping
    Switch functions: soft muting, phase reverse, signal lights
    Signal lights: red4) and blue LEDs
    System functions, firmware download

    A/D conversion:
    Neumann process (patented), 28-bit internal word length

    Digital signal processing:
    Fixed-point, variable internal word length 28 bits to 60 bits

    Synchronization:
    Asynchronous operation (free-running, AES42 - Mode 1), basic frequency accuracy: ± 25 ppm
    Synchronous operation (AES42 - Mode 2), pulling range: Min. ± 100 ppm
    Power supply (phantom power complying with AES42)

    Output:
    XLR3M, 24 bits as per AES/EBU (AES3)



    All data with respekt to 0 dB pre-attenuation and 0 dB gain.
    1) at 1 kHz
    2) re 94 dBSPL
    3) according to IEC 60268-1;
    CCIR-weighting according to CCIR 468-3, quasi peak;
    A-weighting according to IEC 61672-1, RMS
    4) D-01 only
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