With the development of the first digital recording equipment, the digitization of audio data began many years ago, at the end of the signal processing chain. By now, almost all audio signal processing components are available in digital form.

It is well-known that digital signals provide the necessary conditions for mathematically precise calculation and processing, allowing signals to be modified, copied, transmitted and stored as desired, with no loss of quality.

In contrast, analog signal processing is characterized by limited precision, error accumulation, a lack of redundant signal information, and no possibility to include error correction procedures. In the analog signal transmission chain, every processing step is thus associated with a deterioration of signal quality. This results in a progressive decrease in dynamic range, due to the introduction of noise voltages and nonlinear distortion.

Moreover, digital processing permits the performance of functions that are difficult or impossible to implement by means of analog signal processing. This is particularly the case with functions that require intermediate data storage.

With the Solution-D digital microphone system, Neumann has succeeded in bringing the dynamic range and signal fidelity of the best analog studio microphones into the digital domain, thus making possible an entirely digital signal chain for audio production.

Thanks to optimized A/D conversion, especially developed synchronization technology, and the capability of controlling standard microphone parameters and various integrated signal processing functions remotely, Solution-D meets the most demanding requirements of professional audio production. The fundamental principle of the technology permits recordings to be made with no „bottlenecks“ in the signal chain.

An extremely fast peak limiter integrated into the microphone provides constant protection from overloading. The Neumann A/D converter, which is likewise located in the microphone, eliminates the necessity of tedious experimentation with external converters and preamps. The Neumann sound, with optimal quality, is therefore captured directly on the hard drive. Users can rely on this, and thus have more time for the essentials.

The power supply, remote control, synchronization, and signal and data transmission of the digital microphones conform to the AES42 standard. Neumann made a decisive contribution to the development of this standard, which supplies the necessary preconditions for the implementation of digital microphone technology.

A Solution-D digital microphone system consists of the following components: One or more digital microphones, a Digital Microphone Interface (DMI) and the Remote Control Software (RCS), which facilitates the operation and permits the remote control of the microphones. A PC or Mac can serve as the control computer, which of course can also be used simultaneously as Didital Audio Workstation for recording. A DMI permits connection to all subsequent devices that have an AES/EBU interface.

As an alternative to a DMI, a „Connection Kit“ can be used to connect Solution-D microphones to the AES/EBU or S/PDIF interface of a recording system. However, if a Connection Kit is used, it is not possible to control the microphone functions remotely. Thus a control computer is not required. In this case, if it is necessary to synchronize several microphones, a sample rate converter must be used.

All of the possible Solution-D system configurations are illustrated on the following pages.

Starter Sets, containing a microphone and a Connection Kit, provide an economically attractive entry to the world of digital microphones. If users later wish to take advantage of additional adjustable parameters and remotely controlled functions, the system can be expanded at any time through the acquisition of a separate DMI.

The DMI digital microphone interfaces permits familiar microphone settings such as the directional characteristic, pre-attenuation and low-cut filter to be controlled remotely and saved. Changing the settings of microphone parameters is greatly simplified, which makes it possible for settings to be tested rapidly from the monitoring position, in order to optimize the sound quality. All of the settings can be saved together with any desired additional information, thus eliminating the necessity of keeping a hand-written log of the recording procedure.

An A/D converter, developed and patented by Neumann, receives the signal directly from the capsule, and is optimized for the specific signal and impedance parameters of the capsule. Level matching that may be desired for subsequent equipment takes place in the digital domain, in the microphone. Analog components such as preamplifiers and A/D converters are thus no longer required, resulting in considerable cost savings.

The special A/D converter technology makes it possible to have the complete dynamic range of the microphone capsule available in the digital domain, with no restrictions. Setting gain levels is therefore no longer critical.

A particularly significant feature is the peak limiter function. Located for the first time in the most effective position, at the signal source, it reduces transient peaks as the level reaches 0 dBFS, when distortion would normally be inevitable. Analog microphones require extensive headroom in the subsequent signal path to handle such signal peaks, which are short but have a large amplitude. Independently of the peak limiter, a compressor/limiter can also be activated, permitting detailed adjustment via the corresponding parameters.

In addition, functions such as mute and phase reverse are also integrated into the microphone. Visual commands such as „On air“ are implemented by means of remotely controlled LEDs in the D-01 microphone. Acoustic commands in the form of various test signals can be used to facilitate line checks.

The firmware of all the digital microphones can be updated via uploading at any time.

Information transmitted by the microphone includes the name of the manufacturer, the model and serial number, the software version installed in the microphone, and the remotely controllable functions that are supported by the microphone. Status indicators such as „ready for operation“ and specific warning functions are also transmitted.

Despite continuing improvements, integrated circuits available on the market still constitute a limiting factor in the conversion of audio data from analog to digital form. The best currently available delta-sigma A/D converters typically achieve a dynamic range of 115 dB to 120 dB, A-weighted, with a word length of 24 bits.

In comparison, a high-quality analog condenser microphone has a dynamic range of up to 130 dB. A/D conversion with a considerably better performance is therefore required, in order to prevent the addition of noise to the signal. At the same time, this process must be optimally adapted to the signal levels and source impedance found in the microphone.

If the A/D conversion is carried out only after the signal reaches the mixing console or other equipment, this is usually associated with loss of signal quality, since the conversion takes place only after the gain levels have been set. Headroom aspects and noise contributed by the microphone preamplifier and A/D converter thus affect the dynamic range.

Consequently, the development goal was to achieve high-quality digitization of the capsule signal directly in the microphone, so that level matching and other processing steps could be carried out in the digital realm. This is the only way of maintaining the full quality of the microphone signal.

The AES42 standard defines the following two methods of synchronizing the microphone with the receiver (e.g. a mixing console or a DMI digital microphone interface).

Mode 1: The microphone operates asynchronously, using the sampling rate of its internal quartz oscillator. In this case, a sample rate converter is required at the receiver. This mode should be used only if mode 2 synchronization is not possible, since conventional sample rate converters often impair the dynamic range, and increase the latency time.

Mode 2: The microphone is synchronized with a master word clock. This can be either an external word clock or the internal word clock of the DMI. In this case a frequency/phase comparison with the master word clock is carried out in the AES42 receiver (DMI). A control signal is generated that is transmitted to the microphone via the remote control data stream, controlling the frequency of the quartz oscillator in the microphone.

Via the BNC output of the DMI, the internal word clock generator can be used to synchronize additional DMIs and subsequent processing equipment, such as a mixing console.

The signal generated by the capsule is converted directly into a digital signal. The result is a digital output signal with 24 bits and, for example, a dynamic range of 130 dB (A-weighted) in the case of the D-01.

If required, the digital signal processing (DSP) functions integrated into the microphone can be configured and controlled remotely via the DMI digital microphone interface and the RCS remote control software. These functions include gain setting, changing the directional characteristic in the case of the D-01, pre-attenuation, a low-cut filter, a compressor/limiter with an additional de-esser function, and a peak limiter. Here in particular the digital approach can provide a significant advantage. The peak limiter, which receives the output signal almost directly from the capsule, functions as a completely automatic „safety valve“, permitting the safe utilization of the entire available dynamic range even in stressful recording situations.

External components that were previously required, such as analog preamplifiers and A/D converters, are no longer necessary.

To permit clear identification, the microphones send information such as the name of the manufacturer, model, serial number and currently installed software version to the receiver.

The microphones are equipped with three-pin XLR connectors. A bidirectional signal conforming to the AES42 standard is transmitted, containing the balanced digital microphone output signal, the phantom power supply, and a remote control data stream, which includes a signal for synchronizing the microphones with a master clock.

For many years the analog version of the TLM 103 D has played an important role for ambitious home recording and project studios. This microphone has made the Neumann sound available to a broad spectrum of demanding audio engineers and musicians. The TLM 103 D provides all of the sound features of its analog counterpart, in addition to the advantages of digital circuit technology described above.

In addition to the DMI digital microphone interface, Neumann also provides „Connection Kits“ at an attractive price, to permit the simple connection of individual microphones to the widely used S/PDIF and AES/EBU interfaces. This allows numerous users to enjoy easy access to „Neumann sound direct to disk“, without the extensive functionality of a comprehensive DMI. Power is provided by an included plug-in power supply unit.

Of course it is possible to upgrade to a DMI at any time, in order to take advantage of the additional configuration capabilities and DSP functions.

Starter Sets provide a simple and economically very attractive introduction to digital microphone technology. These include a KM 184 D or TLM 103 D microphone and a Connection Kit for an S/PDIF or AES/EBU connection. For entry-level, budget-conscious users who are planning for future recording equipment, an interesting aspect is that an A/D converter is already integrated into the digital Neumann system and, in addition, a preamplifier is no longer required. Moreover, the difficulty of selecting compatible equipment is avoided, since all of the components are provided by a single source. It is thus ensured that the Neumann microphone sound can be captured directly on a digital recording medium with no coloration.

The 2-channel DMI-2 digital microphone interface provides powering and remote control of Solution-D microphones, and outputs their audio data in AES/EBU format.

Microphones of other manufacturers that are compatible with the AES42 standard (e.g. the Sennheiser MKH 8000 with MZD 8000) can of course also be connected to the digital microphone interface, if they can be synchronized.

The Interface is operated via the Neumann RCS remote control software, which is installed on a desktop or laptop computer. The computer is connected to the DMI via a USB port and a USB to RS 485 interface converter. If a large number of microphones is used, several DMIs can be cascaded. In this case, each digital microphone interface can be addressed individually.

In addition to a word clock input and output, the DMIs also have an internal word clock generator. If no master word clock signal (e.g. from a mixing console) is present at the input, the DMI internal word clock is used automatically to synchronize the microphone channels, and the signal is switched to the word clock output.

External commands such as „On Air“ (red light) can be controlled via a 9-pin user port.

Overview of the most important functions:

- AES42 to EBU conversion
- Powering, remote control and synchronization of the microphones without a sample rate converter (AES42, mode 2*)
- Automatic word clock or AES11 synchronization
- All sample rates, from 44.1 kHz to 192 kHz
- Complete control and storage of the settings via Mac or PC

* Explanation Mode 2:
The DMI performs a frequency/phase comparison between the word clock signal recovered from the mic signal, and the word clock of the DMI. From the phase deviation thus determined, the DMI calculates a control signal which is transmitted via the remote control data stream to the mic, where it controls the frequency of the internal quartz oscillator.

The DMI-8, an eight-channel version of the digital microphone interface is considerably simplifying the setup for multichannel applications.

In addition to the proven functions of the two-channel DMI-2 the DMI-8 also offers the following features:

- The capability of cascading up to 128 channels
- Level meter and gain setting on the front panel, operable even without a computer
- D-sub 25 outputs with Tascam® and Yamaha® pin assignments
- Optical ADAT® interface
- Open architecture for later connection to other multi-channel interfaces and audio networks, such as Ether Sound® and MADI

All parameters are displayed on the screen, and can be changed at any time. During production, the audio engineer can monitor the operating status and parameters of all of the connected microphones and, if necessary, can change the settings quickly and easily.

The parameters displayed include the directional characteristic, pre-attenuation, low-cut filter, gain, various microphone status indicators, command indicators, and mute and phase reverse functions. Signal levels and the operation of the compressor or limiter can also be monitored on the screen.

Information transmitted by the microphone, such as the name of the manufacturer, model and serial number, is also displayed for clear identification of the connected microphones. Moreover, it is possible to input additional information such as the name of the sound source. Settings for the complete microphone setup can of course be stored and retrieved as required.

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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