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1

Psychoacoustic metrics in the psychological diagnosis of noise annoyance

Radosz Jan

Vibrations in Physical Systems

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2023

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Vol. 34, nr 1

art. no. 2023101

EN

Aim of the study was to asses noise annoyance in relation to psychoacoustic metrics of sound in an office environment. The Vienna Test System was used for this purpose. Virtual office acoustic environments were developed with sources of different psychoacoustic parameters (loudness, sharpness, fluctuation strength, roughness) but with a constant A-weighted sound pressure level of 55 dB - sound environment with conversations, sound environment with office equipment (computers, printers, telephones) and sound environment with all office noise sources together. The reference environment was a quiet office room with no additional noise sources. Recorded real noise sources were transferred to a virtual 3D sound environment and converted into binaural sound, which was then played back on headphones. During the exposure to each of the acoustic environments, the subjects performed the ALS test (work performance series) and COG test (measurement of attention and concentration) and then assessed the given environment using a questionnaire. The paper presents the results of the statistical analysis - despite different psychoacoustic metrics of office noise sources in the examined acoustic environments, no statistically significant differences were observed in the results of psychological tests.

2

Nuisance assessment of different annoying sounds based on psychoacoustic metrics and electroencephalography

Pamuła H., Kasprzak C., Kłaczyński M.

Diagnostyka

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2016

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Vol. 17, No. 3

67--74

EN

The aim of the study was to assess whether four psychoacoustic parameters (sharpness, roughness, fluctuation strength and tonality) are useful in describing the perceived annoyance of selected noise sources with respect to an objective assessment based on the acoustics standards. Second goal was to verify if the perceived annoyance of such noises correlates with dominant frequency in electroencephalography (EEG) frequency bands. Twenty sound sources, varying in the degree of nuisance, have been assessed by 178 respondents in an Internet-based psychoacoustic test. Obtained annoyance grades were correlated with calculated psychoacoustic and normative parameters and the positive correlation between perceived annoyance and three psychoacoustic parameters (sharpness, roughness and fluctuation strength) was found. In the second part EEG study during listening of recorded sounds was performed on 18 healthy volunteers. Spearman’s rank correlation confirmed that dominant frequencies in alpha (7-14 Hz) and beta2 band (20-30 Hz) were rising with the increasing annoyance of the sounds. Results obtained may be useful in specifying and clarifying permissible noise levels for annoying sounds.

PL

Celem prac było zbadanie czy psychoakustyczne parametry takie jak ostrość, szorstkość, siła fluktuacji, tonalność mogą być użyteczne do opisu uciążliwości wybranych źródeł hałasu, w odniesieniu do obiektywnej oceny opartej o akustyczne standardy. Drugim celem pracy było sprawdzenie, czy postrzegana uciążliwość tych dźwięków koreluje z częstotliwościami dominującymi w pasmach częstotliwości stosowanych w elektroencefalografii (EEG). W pierwszej części badań 178 osób wypełniło ankietę, oceniając uciążliwość prezentowanych dźwięków. Otrzymane oceny zostały skorelowane z wyliczonymi psychoakustycznymi i normatywnymi parametrami. Potwierdzono, że wraz ze wzrostem wartości parametrów takich jak ostrość, szorstkość i siła fluktuacji, wzrasta także postrzegana przez słuchaczy uciążliwość dźwięków. W drugiej części eksperymentu wykonano badania EEG 18 osób podczas odsłuchu powyższych 20 nagrań. Test korelacji rang Spearmana potwierdził, że dominujące częstotliwości w paśmie alfa (7-14Hz) i beta2 (20-30Hz) wzrastały wraz ze wzrostem średniej oceny uciążliwości. Wyniki przeprowadzonych badań mogą być przydatne do doprecyzowania parametrów oceny hałasu i ich dopuszczalnych wartości.

3

Is a Multi-Slider Interface Layout Responsible for a Stimulus Spacing Bias in the MUSHRA Test?

Zieliński S.

Archives of Acoustics

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2015

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Vol. 40, No. 4

585--594

EN

The multi-stimulus test with hidden reference and anchors (MUSHRA) is commonly used for subjective quality assessment of audio systems. Despite its wide acceptance in scientific and industrial sectors, the method is not free from bias. One possible source of bias in the MUSHRA method may be attributed to a graphical design of its user interface. This paper examines the hypothesis that replacement of the standard multi-slider layout with a single-slider version could reduce a stimulus spacing bias observed in the MUSHRA test. Contrary to the expectation, the aforementioned modification did not reduce the bias. This outcome formally supports the validity of using multiple sliders in the MUSHRA graphical interface.

4

A robust generalized sidelobe canceller employing speech leakage masking

Borowicz A.

Advances in Computer Science Research

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2014

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Nr 11

17--29

EN

A novel speech enhancement method based on generalized sidelobe canceller (GSC) structure is presented. We show that it is possible to reduce audible speech distortions and preserve residual noise level under acoustic model uncertainties. It can be done by constraining a speech leakage power according to masking phenomena and conditional minimizing the residual noise power. We implemented the proposed approach using a simple delay-and-sum beamformer model. Finally a comparative evaluation of the selected methods is performed using objective speech quality measures. The results show that the novel method outperforms conventional one providing lower speech distortions.

PL

Prezentowana jest nowa metoda uzdatniania mowy w oparciu o strukturę uogólnionego tłumika listków bocznych. Wykazujemy, ze możliwe jest zmniejszenie słyszalnych zniekształceń mowy przy zachowaniu stałego poziomu szumu rezydualnego, dla modeli przybliżonych środowiska akustycznego. Może to być dokonane poprzez uwarunkowanie poziomu mocy przecieku mowy zgodnie ze zjawiskiem maskowania oraz minimalizację warunkową mocy szumu rezydualnego. Proponowane podejście zaimplementowano w oparciu o prosty model beamformera opóźniająco-sumującego. Ostatecznie przeprowadzono ocenę porównawczą wybranych metod z wykorzystaniem obiektywnych miar jakości mowy. Wyniki pokazują, że nowa metoda przewyższa konwencjonalną zapewniając mniejsze zniekształcenia mowy.

5

A Perception-Based Method for the Noise Control of Construction Machines

Carletti E.

Archives of Acoustics

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2013

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Vol. 38, No. 2

253--258

EN

During operation, construction machines generate high noise levels which can adversely affect the health and the job performance of operators. The noise control techniques currently applied to reduce the noise transmitted into the operator cab are all based on the decrease of the sound pressure level. Merely reducing this noise parameter may be suitable for the compliance with the legislative requirements but, unfortunately, it is not sufficient to improve the subjective human response to noise. The absolute necessity to guarantee comfortable and safe conditions for workers, requires a change of perspective and the identification of different noise control criteria able to combine the reduction of noise levels with that of psychophysical descriptors representing those noise attributes related to the subjective acoustical discomfort. This paper presents the results of a study concerning the “customization” of a methodology based on Sound Quality for the noise control of construction machines. The purpose is to define new hearing-related criteria for the noise control able to guarantee not only reduced noise levels at the operator position but also a reduced annoyance perception.

6

Concert Hall Sound Clarity: A Comparison of Auditory Judgments and Objective Measures

Miśkiewicz A., Rogala T., Rościszewska T., Rudzki T., Fidecki T.

Archives of Acoustics

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2012

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Vol. 37, No. 1

41-46

EN

The purpose of the study was to compare auditory judgments of sound clarity of music examples recorded in a concert hall with predictions of clarity made from the impulse response signal recorded in the same hall. Auditory judgments were made with the use of two methods: by rating sound clarity on a numerical scale with two endpoints, and by absolute magnitude estimation. Results obtained by both methods were then compared against the values of clarity indices, C80 and C50, determined from the impulse response of the concert hall, measured in places in which the microphone was located during recording of music examples. Results show that auditory judgments of sound clarity and predictions made from the C80 index yield a similar rank order of data, but the relation between the C80 scale and perceived sound clarity is nonlinear. The data also show that the values of C80 and C50 indices are in very close agreement.

7

A Perceptionist's View on Psychoacoustics

Blauert J.

Archives of Acoustics

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2012

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Vol. 37, No. 3

365-371

EN

Psychoacoustics is traditionally based on a world model that assumes a physical world existing inde- pendently of any observer – the so-called objective world. Being exposed to this world, an observer is impinged upon by a variety of stimuli reaching his/her sensory organs. These stimuli, if physiologically adequate, may cause biological transduction and signal processing in the sensory organs and its afferent pathways in such a way that finally a specific excitation of the cortex takes place, which results in sen- sations to appear in the observer’s perceptual world. The sensations are understood as being subjective, since they require an observer to exist. This world model – also known as (objectivistic) realism – reaches its limits when it comes to explaining more complex phenomena of perception. Thereupon, in this paper, an alternative world model is emphasized and applied to psychoacoustics, namely the perceptionist’s model. Like realism, perceptionism has a long tradition in epistemology. It appears to be suitable to improve our understanding of perceptual organization.

8

Parametric Audio Based Decoder and Music Synthesizer for Mobile Applications

Szczerba M., Oomen W., Therssen D.

Archives of Acoustics

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2011

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Vol. 36, No. 2

461-478

EN

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 samplebased 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.

9

Towards spike-based speech processing: a biologically plausible approach to simple acoustic classification

Uysal I., Sathyendra H., Harris J. G.

International Journal of Applied Mathematics and Computer Science

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2008

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Vol. 18, no 2

129-137

EN

Shortcomings of automatic speech recognition (ASR) applications are becoming more evident as they are more widely used in real life. The inherent non-stationarity associated with the timing of speech signals as well as the dynamical changes in the environment make the ensuing analysis and recognition extremely difficult. Researchers often turn to biology seeking clues to make better engineered systems, and ASR is no exception with the usage of feature sets such as Mel frequency cepstral coefficients, which employ filter banks similar to cochlear filter banks in frequency distribution and bandwidth. In this paper, we delve deeper into the mechanics of the human auditory system to take this biological inspiration to the next level. The main goal of this research is to investigate the computation potential of spike trains produced at the early stages of the auditory system for a simple acoustic classification task. First, various spike coding schemes from temporal to rate coding are explored, together with various spike-based encoders with various simplicity levels such as rank order coding and liquid state machine. Based on these findings, a biologically plausible system architecture is proposed for the recognition of phonetically simple acoustic signals which makes exclusive use of spikes for computation. The performance tests show superior performance on a noisy vowel data set when compared with a conventional ASR system.

10

Perceptually constrained signal subspace method for speech enhancement : approximate solutions

Borowicz A., Petrovsky A.

Elektronika : konstrukcje, technologie, zastosowania

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2008

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Vol. 49, nr 4

38-44

EN

This paper is concerned with recently proposed perceptually constrained signal subspace (PCSS) method for speech enhancement. Two simplifications of the PCSS method are presented. The first approach is based on approximate diagonalization of the covariance matrix of noise energies in the transformed domain. The approximate solution is presented in a new form which provides perceptually optimal resi-dual noise shaping and does not require a whitening transformation. The second approach is a realization of the PCSS method in the frequency-domain. This is done using an assumption that the covariance matrices are circulant. The resulting estimator is almost identical to the well known IND (Inaudible Noise Distortion) rule. An evaluation of selected methods is performed using objective speech quality mea-sures and informal listening tests. The results show that the sub-optimal methods offer comparable speech quality as the exact solution in common situations.

PL

Artykuł dotyczy zaproponowanej ostatnio metody podprzestrzeni sygnału z ograniczeniami percepcyjnymi (PCSS). Prezentowane są dwa uproszczenia metody PCSS. Pierwsze podejście opiera się na przybliżonej diagonalizacji macierzy kowariancji energii szumu w dziedzinie transformaty. Rozwiązanie przybliżone umożliwia optymalne w sensie percepcyjnym kształtowanie widma szumu resztkowego i nie wymaga transformacji wybielających. Drugie podejście stanowi realizację metody PCSS w dziedzicznie częstotliwości. Osiąga się to wykorzystując założenie, że macierze kowariancji są macierzami okresowymi. Uzyskany estymator okazuje się niemal identyczny z dobrze znaną regułą IND. Przeprowadzana jest ocena wybranych metod przy użyciu obiektywnych miar jakościowych oraz nieformalnych testów odsłuchowych. Wyniki wskazują, że metody przybliżone oferują porównywalną jakość mowy do metody dokładnej w typowych warunkach.

11

Badanie zdolności percepcji przestrzennej sceny akustycznej u ludzi

Wołoszyn P., Jabłoński M., Malicki Ł.

Automatyka / Akademia Górniczo-Hutnicza im. Stanisława Staszica w Krakowie

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2008

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T. 12, z. 3

733-740

PL

Przedmiotem prac prowadzonych przez autorów jest próba oceny zdolności człowieka do postrzegania i interpretowania przestrzennej sceny akustycznej, którą stanowi zespół dźwięków pochodzących ze zbioru źródeł rozmieszczonych w przestrzeni wokół słuchacza. Opisywany w niniejszym artykule etap badań skupia się przede wszystkim na zbudowaniu odpowiedniego zaplecza sprzętowego pozwalającego na wykonywanie eksperymentów akustycznych oraz poszukiwanie odpowiedzi na pytanie, czy człowiek jest zdolny rozróżniać lokalizacje wielu źródeł dźwięku równocześnie i jakie warunki muszą zachodzić, by taka przestrzenna percepcja była możliwa.

EN

Human ability to discern sounds coming from different locations is based on interaural differences and head-related filtration of acoustic signals. It allows to precisely localize a single sound source with high acuity. Our research focuses on more difficult task of perceiving and recognizing complex spatial sound patterns consisting of several acoustic signals emitted simultaneously from multiple sources. In this paper we describe our hardware platform used to perform such psychoacoustic experiments. We also discuss the results of some experiments performed to check what requirements must be met in order to make perception of spatial sound scene possible.

12

Roughness of two simultaneous harmonic complex tones in various pitch registers

Szczepańska-Antosik J.

Archives of Acoustics

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2008

|

Vol. 33, No. 1

73-78

EN

The purpose of the study was to determine the dependence of perceived roughness on the frequency ratio of two simultaneous harmonic complex tones. A set of 36 dyads forming musical intervals of various tuning systems was presented in three pitch registers. Twelve sound engineering students judged each dyad for roughness by the method of absolute magnitude estimation. Results show that roughness considerably varies with the frequency ratio of the two complex tones, what is a well-known phenomenon. A new finding, being in contrast to published theories of roughness, is that some of the equally-tempered intervals are perceived less rough than their counterparts based on integer frequency ratios. This effect is attributed to slow beats that arise between the harmonics of two complex tones when the frequency ratio of the tones slightly departs from the integer ratio.

13

Warped discrete fourier transform in perceptual speech and audio processing

Petrovsky A., Borowicz A., Parfieniuk M., Petrovsky A.

Prace Naukowe Instytutu Telekomunikacji i Akustyki Politechniki Wrocławskiej. Konferencje

|

2004

|

Vol. 85, nr 29

143-152

EN

This paper presents an unconventional approach to perceptual sound processing, utilizing the Warped Discrete Fourier Transform. Unlike ordinary Discrete Fourier Transform, its novel mutation allows nonuniform sampling of the z-transform over the unit circle. Moreover, the warping can be adjusted to approximate nonlinear frequency resolution of human ear. Thus some aspects of the psy-choacoustic analysis and processing can be improved, what was verified in three practical applications. Firstly, the advanced speech enhancement system operating in the perceptually warped spectrum domain was configured. And recently the same idea was employed in speech and audio compression.

PL

Artykuł prezentuje niekonwencjonalne podejście do perceptualnego przetwarzania dźwięku oparte na Spaczonej Dyskretnej Transformacie Fouriera. W odróżnieniu od zwykłej Dyskretnej Transformaty Fouriera, jej nowa mutacja pozwala na nierównomierne próbkowanie transformaty z na okręgu jednostkowym. Co więcej, spaczenie może być dopasowane tak, by aproksymowało ono nieliniową rozdzielczość częstotliwościową ucha ludzkiego. Dzięki temu pewne aspekty analizy psychoakustycznej i przetwarzania mogą zostać poprawione, co zostało zweryfikowane w trzech praktycznych zastosowaniach. Najpierw zbudowano zaawansowany system uzdatniania mowy operujący w dziedzinie spaczonego widma. Ostatnio ideę wykorzystano także w kompresji mowy i audio.

14

The perceptual high quality audio coding based on dynamic wavelet packet transform

Petrovsky A., Petrovsky A.

Prace Naukowe Instytutu Telekomunikacji i Akustyki Politechniki Wrocławskiej. Konferencje

|

2004

|

Vol. 85, nr 29

153-159

EN

The new perceptual high quality audio coding based on a dynamic wavelet packet transform is presented in the given paper. Tme-frequency signal decomposition is adapted to the source using wavelet packet (WP) or time-shift wavelet packet (TSWP) transforms. Due to TSWP it is possible to increase the flexibility of the analysis filter bank in a time scale and copression level. The masking thresholds are estimated in a wavelet domain and dynamic reconfigurable WP tree structure is formed basing on perceptual entropy. The advantages of this approach are better viewed by considering the waveled packet growing as a spilting process., i.e. the temporal construction WP tree created for each signal frame presents an ideal decision for real time processing implemented in a reconfigurable handware.

15

A computer-based system for objective studies of human psychomotorics

Ossowski A., Smyrnova J.

Journal of Medical Informatics & Technologies

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2002

|

Vol. 3

MI71--79

EN

Firstly the idea of objective psychological tests and their characteristics related to various features of human psychophysiology are introduced. Examples of objective tests are given. Next application of data mining algorithms to analyse data obtained from different tests are outlined. The general concept of a computer system for objective psychological studies of psychomotoric processes in humans is then described. Finally the possibility of implementation of the system in medical and psychoacoustical studies is pointed out.

16

Source enhanced linear prediction of speech incorporating simultaneously masked spectral weighting

Lukasiak J., Burnett I.S.

Journal of Telecommunications and Information Technology

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2001

|

nr 3

15-23

EN

Linear prediction is the cornerstone of most modern speech compression algorithms. This paper proposes modifying the calculation of the linear predictor coefficients to incorporate a weighting function based on the simultaneous masking property of the ear. The resultant prediction filter better models the perceptual characteristics of the source and results in the removal of more perceptually important information from the input speech signal than a standard LP filter. When employed in a low rate speech codec the net effect is an improvement in subjective quality, with no increase in transmission rate and only a modest increase in computational complexity.

17

Computer investigations of sound perception and noise annoyance

Ossowski A., Smyrnova Y.

Journal of Medical Informatics & Technologies

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2001

|

Vol. 2, Part. 2

SN27--SN32

EN

The problem of influence of adverse sound environments on human psychophysical state and health has been outlined. Basic sound characteristics have been defined and the fundamental question of the relation between subjective and objective characteristics has been formulated. The general conception of a computer system for studying human perception of environmental sound and noise has been described. The usefulness of application of data mining algorithms in order to find practical rules that enable to determine subjective sound and noise parameters from the knowledge of physical ones has been pointed out.