Session G: RECORDING, RESTORATION AND PROCESSING
Saturday, May 11, 14:00 18:00 h The aesthetics and techniques used in the creation of early analog tape-based electronic music were influenced by music compositional practices from the 1880s through the first half of the 20th century. Examining these earlier acoustic pieces and a their notable compositional techniques has helped provide an aesthetic context for students studying early tape-based electronic music and their analog recording projects at the University of Colorado at Denver. The paper is a systematic approach to the various factors that go into a sound recording and its reproduction and links them in a novel way with the expressed views of professionals as regards their philosophies for performance, recording, dissemination, etc. Applying the systematic approach to well-known and frequently touted publications as well as more obscure sources, it becomes possible to structure the desires of a producer and performer and perform a comparison with the obtained result. Producers and Tonmeisters in the classical field (such as Culshaw, Burkowitz, Grubb) discussed their philosophies about the same time that the performance practice movement gathered momentum, and producers in popular music exploited the new possibilities (such as George Martin). Recently perception psychology has been introduced while the technical possibilities including surround have expanded and (hopefully) standardized. Most analogue recordings have deliberate departures from a level frequency response and these were made with the assumption that they would be equalized on playback, either through published data specifying the appropriate curve or through the provision of specialized hardware. This paper examines the possible choices and their implications vis-à-vis performing these equalizations prior to or post the analogue / digital conversion. We consider two main categories: 1) Stationary curves such as RIAA for disc, IEC and NAB for tape and 2) Non stationary curves such as noise reduction systems by Dolby, EMT, dbx. In addition to the purely technical trade-offs consideration is also given to the demands on the knowledge of the transfer engineer and the ability of the chosen method(s) to compensate either instantaneously or subsequently for erroneous decisions made at the time of transfer. This paper addresses the restoration of audio signals proceeding from old recordings, and focuses on long-pulse removal. We propose a new two-stage method to estimate the waveform of each long pulse from the observed noisy signal. First, an initial estimate for the pulse shape is obtained via a non-linear filtering scheme called two-pass splitting-window (TPSW) filtering. Then, this estimate is further smoothed through a piecewise polynomial fitting. The degree of smoothness of the estimate can be controlled by adjusting either the TPSW parameters or the length of the segments to be fitted. The proposed method has low computational complexity, it is not constrained by the assumption of shape similarity among pulse waveforms, and can be successfully applied for removing overlapping pulses. In this paper we present several novel techniques for incorporating fault tolerance in content-based audio search. Our algorithms extend a recently proposed framework for fast index-based search in score-like audio material. Considering queries given as a sequence of notes and the task of matching those queries to a data base of musical tunes or melodies, we investigate possible deviations such a wrong notes, missing notes, or differences in the underlying tempo curves. It turns out that our fast index-based search methods may be quite naturally adapted to tolerate the former kinds of deviations, while the case of tempo changes requires a more careful treatment. Here, we propose a new technique for incorporating a tempo tracking mechanism into our fast search algorithms. Our methods have been successfully implemented and tested within a query-by-whistling application presented at the 2001 Internationale Funkausstellung (IFA) in Berlin, Germany. We describe this application and give an overview on our extensive tests. Due to wiring questions, computer integration, or simply as a way to increase the features of audio systems, the need for an audio networking solution increases every day. However, it is possible to use the know-how of decades of computer networking, and use winning technologies such as Ethernet. With a few adjustments, we can have an audio and midi networking solution, which will achieve our goals and scale for the future, without hurting our pockets. The problem of partially extreme loudness differences in radio and television programs is well known for a long time. With respect to the introduction of new digital techniques combined with parallel transmission of digital and analogue signals the problem of loudness differences again is especially significant. Based upon relevant leveling recommendations and a newly developed loudness algorithm solutions avoiding loudness differences in radio and television are presented. We present an investigation into signal processing models appropriate for audio, and especially high quality musical signals, by means of Bayesian atomic decompositions. At present, many models rely on short-term stationarity of the audio, or highly limiting forms of non-stationarity. Moreover, they are well-suited only to low-level inference tasks. We seek to formulate a new generation of audio models that will address the main limitations of the existing ones and permit high-level inference. As we show, such models result from the marriage of an over complete dictionary of time-frequency atoms with structured hierarchical prior probability distributions developed specifically for audio signals, in order to model coefficient correlation in time and frequency. |
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