Archives of Acoustics

The 14th International Symposium on Sound Engineering and Tonmeistering will be held on May 19-21, 2011, in Wrocław. The Symposium is organized by the Chair of Acoustics, Institute of Telecommunications, Teleinformatics and Acoustics, Wrocław University of Technology, under auspicious of the Polish Section of the Audio Engineering Society. The organizers cordially invite sound engineers, music producers, acousticians, and specialists in sound reinforcement, scientists who deal with sound engineering, sound recording and related areas, students, and employees of the audio industry to participate in the Symposium. The Symposium programme will include lecture sessions and workshop presentations.

In June 2011, Polish Section of the Audio Engineering Society will celebrate its 20th anniversary. On this occasion, the society officers, ending their second, two-year term, present a short summary of the Section's activity during the past four years. The Section's structure comprises the main unit - that is the Polish AES section, and three student sections, associated with the Department of Multimedia Systems at the Gdansk University of Technology, the Chair of Acoustics at the Wroc?aw University of Technology, and the Department of Mechanics and Vibroacoustics at the AGH University of Science and Technology in Cracow. The latter section began its activity only last year. In addition to formal student sections, in the activities of the Polish AES Section also participate students of the Department of Sound Engineering, at the The Fryderyk Chopin University of Music in Warsaw, and students of the Department of Elecroacoustics, at the Warsaw University of Technology.

This paper has two distinct parts. Section 1 includes general discussion of the phenomenon of "absolute pitch" (AP), and presentation of various concepts concerning definitions of "full", "partial" and "pseudo" AP. Sections 2-4 include presentation of the experiment concerning frequency range in which absolute pitch appears, and discussion of the experimental results. The experiment was performed with participation of 9 AP experts selected from the population of 250 music students as best scoring in the pitch-naming piano-tone screening tests. Each subject had to recognize chromas of 108 pure tones representing the chromatic musical scale of nine octaves from E0 to D#9. The series of 108 tones was presented to each subject 60 times in random order, diotically, with loudness level about 65 phon. Percentage of correct recognitions (PC) for each tone was computed. The frequency range for the existence of absolute pitch in pure tones, perceived by sensitive AP possessors stretches usually over 5 octaves from about 130.6 Hz (C3) to about 3.951 Hz (B7). However, it was noted that in a single case, the upper boundary of AP was 9.397 Hz (D9). The split-halves method was applied to estimate the reliability of the obtained results.

Active acoustics offers potential benefits in music halls having acoustical short-comings and is a relatively inexpensive alternative to physical modifications of the enclosures. One critical benefit of active architecture is the controlled variability of acoustics. Although many improvements have been made over the last 60 years in the quality and usability of active acoustics, some problems still persist and the acceptance of this technology is advancing cautiously. McGill's Virtual Acoustic Technology (VAT) offers new solutions in the key areas of performance by focusing on the electroacoustic coupling between the existing room acoustics and the simulation acoustics. All control parameters of the active acoustics are implemented in the Space Builder engine by employing multichannel parallel mixing, routing, and processing. The virtual acoustic response is created using low-latency convolution and a three-way temporal segmentation of the measured impulse responses. This method facilitates a sooner release of the virtual room response and its radiation into the surrounding space. Field tests are currently underway at McGill University involving performing musicians and the audience in order to fully assess and quantify the benefits of this new approach in active acoustics.

The development of digital microphones and loudspeakers adds new and interesting possibilities of their applications in different fields, extended from industrial, medical to consumer audio markets. One of the rapidly growing field of applications is mobile multimedia, such as mobile phones, digital cameras, laptop and desktop PCs, etc. The advances have also been made in digital audio, particularly in direct digital transduction, so it is now possible to create the all-digital audio recording and reproduction chains potentially having several advantages over existing analog systems.

We are exploring the relationship between accents and expression in piano performance. Accents are local events that attract a listener's attention and are either evident from the score (immanent) or added by the performer (performed). Immanent accents are associated with grouping (phrasing), metre, melody and harmony. In piano music, performed accents involve changes in timing, dynamics, articulation, and pedalling; they vary in amplitude, form, and duration. We analyzed the first eight bars of Chopin Prelude op. 28 n. 6. In a separate study, music theorists had marked grouping, melodic and harmonic accents on the score and estimated the importance (salience) of each. Here, we mathematically modeled timing and dynamics in the prelude in two ways using Director Musices (DM) - a software package for automatic rendering of expressive performance. The first rendering focused on phrasing following existing and tested procedures in DM. The second focused on accents - timing and dynamics in the vicinity of the accents identified by the theorists. In an informal listening test, 10 out of 12 participants (5 of 6 musicians and 5 of 6 non-musicians) preferred the accent-based formulation, and several stated that it had more variation of timing and dynamics from one phrase to the next.

In the paper, various approaches to automatic music audio summarization are discussed. The project described in detail, is the realization of a method for extracting a music thumbnail - a fragment of continuous music of a given duration time that is most similar to the entire music piece. The results of subjective assessment of the thumbnail choice are presented, where four parameters have been taken into account: clarity (representation of the essence of the piece of music), conciseness (the motifs are not repeated in the summary), coherence of music structure, and overall quality of summary usefulness.

Whenever the recording engineer uses stereo microphone techniques, he/she has to consider a recording angle resulting from the positioning of microphones relative to sound sources, besides other acoustic factors. The recording angle, the width of a captured acoustic scene and the properties of a particular microphone technique are closely related. We propose a decision supporting method, based on the mapping of the actual position of a sound source to its position in the reproduced acoustic scene. This research resulted in a set of localisation curves characterising four most popular stereo microphone techniques. The curves were obtained by two methods: calculation, based on appropriate engineering formulae, and experiment consisting in the recording of sources and estimation of the perceived position in listening tests. The analysis of curves brings several conclusions important in the recording practice.

The work presents a comparison of some sound attributes perceived at a multichannel and stereo playback of musical recordings. The width of the virtual source, coherence impression, total size of sound scene, general quality and balance were the subjects of interest after the format reduction in accordance with the ITU recommendation. The results showed that evaluation of these attributes depends on the way the original audiosphere has been created in the surround system, for example, for a narrow virtual source the mix-down process causes only a small change in its size but for a broad source the observed degradation is significant. In addition, different ways of conversion from the multichannel to stereo format have been tested for compatibility.

The implemented online urban noise pollution monitoring system is presented with regard to its conceptual assumptions and technical realization. A concept of the noise source parameters dynamic assessment is introduced. The idea of noise modeling, based on noise emission characteristics and emission simulations, was developed and practically utilized in the system. Furthermore, the working system architecture and the data acquisition scheme are described. The method for increasing the speed of noise map calculation employing a supercomputer is explained. The practical implementation of noise maps generation and visualization system is presented, together with introduced improvements in the domain of continuous noise monitoring and acoustic maps creation. Some results of tests performed using the system prototype are shown. The main focus is put on assessing the efficiency of the acoustic maps created with the discussed system, in comparison to results obtained with traditional methods.

Independent Component Analysis (ICA) can be used for single channel audio separation, if a mixed signal is transformed into time-frequency domain and the resulting matrix of magnitude coefficients is processed by ICA. Previous works used only frequency (spectral) vectors and Kullback-Leibler distance measure for this task. New decomposition bases are proposed: time vectors and time-frequency components. The applicability of several different measures of distance of components are analysed. An algorithm for clustering of components is presented. It was tested on mixes of two and three sounds. The perceptual quality of separation obtained with the measures of distance proposed was evaluated by listening tests, indicating "beta" and "correlation" measures as the most appropriate. The "Euclidean" distance is shown to be appropriate for sounds with varying amplitudes. The perceptual effect of the amount of variance used was also evaluated.

The present study was carried out to determine whether recorded musical tones played at various pitches on a clarinet, a flute, an oboe, and a trumpet are perceived as being equal in loudness when presented to listeners at the same A-weighted level. This psychophysical investigation showed systematic effects of both instrument type and pitch that could be related to spectral properties of the sounds under consideration. Level adjustments that were needed to equalize loudness well exceeded typical values of JNDs for signal level, thus confirming the insufficiency of A-weighting as a loudness predictor for musical sounds. Consequently, the use of elaborate computational prediction is stressed, in view of the necessity for thorough investigation of factors affecting the perception of loudness of musical sounds.

The task of electroacoustic devices is a transmission of audio signals. The transmitted signal should be distorted as little as possible. Nonlinear distortions are the distortions depending on signal level. The types of nonlinear distortions as well as their measures are presented in the paper. The weakest device in an electroacoustic chain is a loud-speaker. It causes the greatest degradation of the signal. It is usually the most nonlinear part of the electroacoustic system. The nonlinearities in loudspeakers are described in details. Other types of nonlinear distortions as transient intermodulation in power amplifiers and distortions caused by the A/C sampling are also presented.

This paper reviews parametric audio coders and discusses novel technologies introduced in a low-complexity, low-power consumption audio decoder and music synthesizer platform developed by the authors. The decoder uses parametric coding scheme based on the MPEG-4 Parametric Audio standard. In order to keep the complexity low, most of the processing is performed in the parametric domain. This parametric processing includes pitch and tempo shifting, volume adjustment, selection of psychoacoustically relevant components for synthesis and stereo image creation. The decoder allows for good quality 44.1 kHz stereo audio streaming at 24 kbps. The synthesizer matches the audio quality of industry-standard sample-based synthesizers while using a twenty times smaller memory footprint soundbank. The presented decoder/synthesizer is designed for low-power mobile platforms and supports music streaming, ringtone synthesis, gaming and remixing applications.

In Western music culture instruments have been developed according to unique instrument acoustical features based on types of excitation, resonance, and radiation. These include the woodwind, brass, bowed and plucked string, and percussion families of instruments. On the other hand, instrument performance depends on musical training, and music listening depends on perception of instrument output. Since musical signals are easier to understand in the frequency domain than the time domain, much effort has been made to perform spectral analysis and extract salient parameters, such as spectral centroids, in order to create simplified synthesis models for musical instrument sound synthesis. Moreover, perceptual tests have been made to determine the relative importance of various parameters, such as spectral centroid variation, spectral incoherence, and spectral irregularity. It turns out that the importance of particular parameters depends on both their strengths within musical sounds as well as the robustness of their effect on perception. Methods that the author and his colleagues have used to explore timbre perception are: 1) discrimination of parameter reduction or elimination; 2) dissimilarity judgments together with multidimensional scaling; 3) informal listening to sound morphing examples. This paper discusses ramifications of this work for sound synthesis and timbre transposition.

In this article some key events concerning founding Polish Section of the Audio Engineering Society were presented. In addition, the history covering International Symposia on Sound Engineering and Mastering was outlined. Also, papers contained in this issue were shortly reviewed.

Organologic and campanologic acoustical problems due to applications to sacral objects are characterized on ground of numerous reviewed publications and engineering reports. Participation of several involved research centres, mostly Polish, at solving these problems is evaluated. Some desirable future developments are indicated. Appendices bring examples of documentation on selected investigated objects.

The aim of this paper is to describe the process of choosing the best surround microphone technique for recording of choir with an instrumental ensemble. First, examples of multichannel microphone techniques including those used in the recording are described. Then, the assumptions and details of music recording in Radio Gdansk Studio are provided as well as the process of mixing of the multichannel recording. The extensive subjective tests were performed employing a group of sound engineers and students in order to find the most preferable recording techniques. Because the final recording is based on the mix of "direct/ambient" and "direct-sound all-around" approaches, a subjective quality evaluation was conducted and on this basis the best rated multichannel techniques were chosen. The results show that listeners might consider different factors when choosing the best rated multichannel techniques in separate tasks, as different systems were chosen in the two tests.

Reed woodwind instruments differ in both their geometry (mainly cylindrical or mainly conical) and their excitation mechanism (single or double reed). How much of the resulting sound is due to the single/double reed, and how much to the geometry of the instrument? Measurements done by Almeida et al. (J. Acoust. Soc. Am., 121, 1, 536-546, 2007) show that the flow vs pressure characteristic curve of an oboe reed is not that different from that of a clarinet reed, the only difference probably being due to pressure recovery inside the conical staple. Is it possible to make a single reed mouthpiece for an oboe, while keeping the conical staple, that would still give the oboe its characteristic sound? To find it out, a mouthpiece with the following characteristics was made: A standard clarinet Bb reed can be attached to it, its volume is approximately that of the missing part of the instrument cone, and a standard French oboe staple can be inserted to it, so that it can be inserted in the usual way in any french oboe. In this paper, the first prototype of the mouthpiece is shown. Also, a sound comparison of the oboe sounds played with this mouthpiece and a standard double reed by a professional player is presented.