The merits of different coding schemes are discussed with special focus on OCF (optimum coding in the frequency domain). The main requirements for a codec suitable for ISDN channels are high coding efficiency, sufficient tolerance against transmission errors, and superior synchronization capability. The results obtained by applying these codecs to real-world channels like ISDN and satellite links are discussed. The OCF codec for 64 kb/s has been tested in conjunction with ISDN. First experiments were carried out successfully using a local loopback configuration. The use of ISDN in a local switching network is reported. The transmission of stereo compact disc signals by ISDN between Erlangen and Mannheim proved to be very reliable and achieved excellent audio quality.>

A description is given of the noise-to-mask ratio (NMR) meter, a device used to measure the audibility of error signals according to the laws of psychoacoustics. The authors review the working principles of the NMR and describe its real-time implementation aspects. The application of the NMR to several source coding schemes is shown, and future developments in this area are discussed. It is concluded that NMR measurements currently cannot replace listening tests completely. What they can do is deliver objective, reproducible results that help to point out critical pieces of music as well as weak parts of algorithms. It is noted that even subtle changes in the audio channel under test will usually result in measurable differences even when listening tests are not applicable.>

A real-time implementation of a source coding method allowing bit rate reduction from 16 bits to 3 bits per sample at sampling rates up to 48 kHz is described. Using an adaptive transform coding technique optimized for low computing complexity and a system based on the AT&T DSP 32 single-chip signal processor, a very good compromise between sound quality and hardware complexity has been achieved. Distortions were audible only by trained listeners for critical pieces of music. One signal-processor module is needed for each of the two coders and decoders. Each one fits on a single-height Eurocard.>

The predicted wordlength assignment system (PWA) is a digital speech interpolation method which avoids speech clipping and "freeze-out" distortion. Inactive sources are excluded by a speech detector. The active speech signals are coded with variable wordlengths (3-8 bits) at a sampling rate of 8 kHz. In an overload case, all active sources are still served, but at reduced wordlength. The required wordlength is calculated using only the signal history, which is also available at the receiver. Therefore, no auxiliary information about the individual wordlength is transmitted. A system with up to 128 telephone conversation speech sources has been studied using computer simulation. The signal-to-noise ratio (SNR) is employed to describe speech quality. With an input of 128 sources (40 percent activity) and a transmission rate per source of 21 kbits/s, an SNR of 34 dB can be achieved. Above a bit rate of 16 kbits/s, distortions are not audible. As a first step towards implementation, a specially designed fast microprocessor has been used to simulate the most important PWA system functions, such as speech detection, linear prediction, and coding algorithm.