The 108th Audio Engineering Society Convention.(Critical Essay)

Palais des Congres, Paris, France, 19-22 February 2000

The 108th Audio Engineering Society (AES) Convention, a biannual event alternatively held in Europe and North America, was held this past February in Paris. The convention comprised a large exhibition of audio and broadcasting equipment, paper sessions, technical and standard committee meetings, student activities, and workshops. As I mainly participated in the paper sessions, my review is concentrated on these, in particular Audio Coding, Music Instrument Acoustics and Electronic Music Technology, and Signal Processing.

In the audio industry, the AES is best known as an authority of standards, such as the AES/EBU (European Broadcasting Union) digital connector and communication protocols. Coinciding with the Technical Committee meeting for Audio Coding and the Workshop for "MPEG-4 Version 2," the Audio Coding paper sessions also included a few papers on MPEG-4 applications. Riitta Vaananen from Helsinki demonstrated his article on "Synthetic Audio Tools in MPEG-4 Standard." These tools include specifications for sound coding based on structured, parametric descriptions of algorithmic synthesis methods along with wavetable synthesis and speech synthesis from text data. A German group from the Fraunhofer Institution for Integrated Circuits and DSP Solutions presented "Real-Time Implementation of the MPEG-4 Low Delay Advanced Coding Algorithm (AAC-LD) on Motorola DSP56300." For some Computer Music Journal readers, the Fraunhofer Institution will be known for its MP3 compression algorithm used in sound tools such as Digidesign ProTools or Syntrillium Cool Edit. The paper discusses the selection of a fixed-point DSP platform and describes the implementation and effects of a functional real-time AAC-LD codec on the DSP56300. The authors argue that for non-speech signals the algorithm delivers a better performance at lower bit-rates than specialized speech codecs. Markus Erne and George Moschytz from the Eidegeossische Technische Hochschule in Zurich presented their paper on "A Bit-Allocation Scheme for an Embedded and Signal-Adaptive Audio Coder." Their bit-allocation scheme for a wavelet-based audio coder is based on embedded zero-tree coding, EZW, that has mainly been used in video compression. They claim that the scheme additionally enables increased audio quality due to a non-integer quantization estimation.

The AES seemed to be opening its doors to computer/electroacoustic music researchers by programming a special paper session, "Music Instrument Acoustics and Electronic Music Technology." The contents of the session, however, were rather a mixed bag compared to sessions of the International Computer Music Conference (ICMC). One of the most remarkable papers of this session was presented by Jason Flaks of Gibson Guitar Corp, "Global Musical Instrument Communication Standard (GMICS): An Integrated Digital Audio and Control Communication Specification for Instruments." Since the implementation of MIDI in the early 1980s, there have been several attempts to replace the 8-bit signal-based standard, such as ZIPI (see Computer Music Journal 18:4). GMICS is based on the 100 Mbit Ethernet physical layer with RJ-45 connectors and provides low latency digital audio and control signals for instruments in live performance. The audio signal layer is capable of up to 16 channels in 32-bit format at 96 kHz sampling rate. One of the notable features is the allocation of phantom power (24 V DC, delivering 500 mA to 1 A) to unused connections of the RJ-45 units under the IEEE 802.3 standard. After the presentation, there was an exchange of opinions on the phantom power feature, ...