Ing. František Rund, Ph.D.

This paper evaluates the ability of several algorithms to detect impulse distortions (clicks) in audio signals. The systems are evaluated against data from a listening test conducted using real audio signals provided by a vinyl manufacturer. Some of the signals contained clicks due to damage during the manufacturing process. An evaluation of click detection algorithms against listening test results focuses on the ability of the click-detection algorithms to detect perceptible clicks. The results presented in this paper show that an algorithm that employs a hearing model detected audible clicks with a lower false detection rate than the other algorithms in the test and that the wavelet transform-based algorithm with a dynamic threshold outperformed the other algorithms.

This paper presents a rate-code model of binaural interaction inspired by recent neurophysiological findings. The model consists of a peripheral part and a binaural part. The binaural part is composed of models of the medial superior olive (MSO) and the lateral superior olive (LSO), which are parts of the auditory brainstem. The MSO and LSO model outputs are preprocessed in the interaural time difference (ITD) and interaural level difference (ILD) central stages, respectively, which give absolute values of the predicted lateralization at their outputs, allowing a direct comparison with psychophysical data. The predictions obtained with the MSO and LSO models are compared with subjective data on the lateralization of pure tones and narrowband noises, discrimination of the ITD and ILD, and discrimination of the phase warp. The lateralization and discrimination experiments show good agreement with the subjective data. In the case of the phase-warp experiment, the models agree qualitatively with the subjective data. The results demonstrate that rate-code models of MSO and LSO can be used to explain psychophysical data considering lateralization and discrimination based on binaural cues.

Monitoring of the audio content quality is important for Digital Audio Broadcasting. Although the subjective methods asses quality more reliably, the objective methods have to be considered when needed big amount of the repeated tests. The highest differences between results of objective and subjective assessment were proved for modern high efficient codecs in the case of the lower bitrates. In this paper, two objective methods, PEAQ and PEMO-Q, are compared to be able to choose the more reliable method for audio quality monitoring. It is shown that PEMO-Q gives better results in the case of lower bitrates for the both High Efficiency profiles of Advanced Audio Coding codec. But the improvement is not essential considering the differences between results of the objective and the subjective assessment methods. Moreover, for music samples, PEMO-Q method brings worse results in the case of lower bitrates for Low Complexity profile of Advanced Audio Coding codec.

Binaural hearing is an important ability of humans. While utilizing a binaural hearing model with Artificial Neural Network it is possible to simulate localization of sound source position by humans. In this paper localization of drum kit sounds as well as localization prediction dependence on head related functions used to synthesize these drum kit sounds are concerned. Three separate tests were performed using two classification algorithms: for horizontal plane only, for both horizontal and vertical planes (Koshkina E., Bouse J., Proc. of Poster 2017). In the first test localization of auralized drum kit sounds through different HRTF sets at fixed positions was performed to observe elevation and azimuth prediction differences. In the other two tests localization at unfixed positions for drum kit sounds with and without room reflections was performed.

For a given type of headphones the sound pressure levels of pure tones at which an adequate number of young listeners without hearing loss just perceived the tones are called Reference Equivalent Threshold Sound Pressure Levels (RETSPLs). RETSPLs for the Sennheiser HD 650 circumaural headphones were measured in this paper for twenty five young listeners whose age was in the range of 19 to 28 years. The data were measured at frequencies between 125 Hz and 16 kHz. Since RETSPLs for this type of headphones are not currently available this paper may be useful for researchers who employ this type of headphones in experiments. The paper also compares RETSPLs for various types of supra-aural and circumaural headphones, which are available in standards.

It has been suggested that the stimulus frequency otoacoustic emissions (SFOAEs) are generated due to coherent reflections of forward-Traveling waves. In this paper, the emissions were simulated by a two-dimensional hydrodynamic cochlea model with parameters based on real dimensions of the human cochlea. Simulations were done using a low-level pure tone at frequencies between 0.5 and 1.3 kHz, and 3 and 6 kHz. In agreement with animal experimental data, the ratio between the simulated SFOAE group delays and group delays estimated from the basilar membrane transfer functions were between 1 and 2. In agreement with the coherent reflection filtering theory, the SFOAE delay was affected by the frequency selectivity of the cochlear filters. However, the simulated results gave different tuning ratios for two models differing only in tuning of their cochlear filters.

This study is focused on the perceived roughness of two simultaneous harmonic complex tones with ratios between their fundamental frequencies set to create intervals on just-tempered (JT) and equal-tempered (ET) scales. According to roughness theories, ET intervals should produce more roughness. However, previous studies have shown the opposite for intervals in which the lower fundamental frequency of the complex was equal to 261.6 Hz. The aim of this study is to verify and explain these results by using intervals composed of complexes whose spectral components were generated with either a sine starting phase or with a random starting phase. Results of the current study showed the same phenomenon as previous studies. To examine whether the explanation of the phenomenon lies in the function of the peripheral ear, three roughness models based upon this function were used: the Daniel and Weber (1997) model, the synchronization index (SI) model, and the model based on a hydrodynamic cochlear model. For most of the corresponding JT and ET intervals, only the Daniel and Weber (1997) model predicted less roughness in the ET intervals. In addition to this, the intervals were analyzed by a model simulating the auditory periphery. The results showed that a possible cause for the roughness differences may be in the frequencies of fluctuations of the signal in the peripheral ear. For JT intervals the fluctuations in the adjacent places on the simulated basilar membrane had either the same frequency or integer multiples of that frequency and were synchronized. Since a previous study showed that synchronized fluctuations in adjacent auditory filters lead to higher roughness than out of phase fluctuations (Terhardt, 1974), this may cause more roughness across JT and ET intervals.

Tento článek popisuje návrh a implementaci vlastního řešení pro úlohu identifikace nahrávky, se zaměřením na identifikaci nahrávek z gramofonové desky. Algoritmus je implementován v programovém prostředí MATLAB. Pro klasifikaci nahrávky a vyhledávání v databázi nahrávek je zvolena metoda strojového učení K-nearest neighbors (k-nejbližších sousedů, KNN). Pro parametrizaci nahrávky je použito již připravených parametrů z Music Information Retrieval (MIR) Toolboxu, nahrávka je klasifikována pomocí parametrů tempo a mode. Pro testování algoritmu byla jako vedlejší funkcionalita programu vytvořena možnost přidat typické zvukové atributy gramofonových desek do existujících nahrávek. Po natrénování na množině všech 269 dostupných masterů nahrávek byl program schopen rozpoznat 92,8 % nahrávek uměle znehodnocených atributy gramofonové desky (posunutí začátku, snížení amplitudy, přidání šumu a ztráta vysokých frekvencí) a zároveň všech 5 dostupných nahrávek pořízených z gramofonové desky. Pro ovládání programu bylo také vytvořené jednoduché grafické rozhraní

This article presents specific methods for reducing artifacts, which may occur within virtual sound source positioning method employing Differential Head-Related Transfer Function (DHRTF), and their subjective and objective assessment. The artifacts result from a process of obtaining the DHRTF, where spike-like spectral peak(s) may occur in the module of the DHRTF. These spectral features result in an undesirable whistling-like sound component that distorts both timbre and spatial perception of the virtual sound. Therefore, the DHRTF requires to be pre-processed by an appropriate method that is able to remove or at least sufficiently reduce the artifacts while preserving both the natural character of the timbre and the spatial perception of the positioned sound. Three selected methods based on limitation and/or smoothening of the DHRTF magnitude by a low-pass filter are introduced here and compared by paired comparison listening tests and by objective assessment method. The tests showed significant decrease of the artifact occurrence with the best results for a method based on limiting the DHRTF magnitude and its smoothening by a moving average convolution kernel.

This study presents a system which detects clicks in sound (audible degradations). The system is based on a computational model of the peripheral ear. In order to train and verify the system, a listening test was conducted using 89 short samples of analog (vinyl) records. The samples contained singing voice, music (rock’n’roll), or both. We randomly chose 30 samples from the set and used it to train the system; then we tested the system using the 59 remaining samples. The system performance expressed as a percentage of correct detections (78.1%) and false alarms (3,9%) is promising.

Maximum-length sequences (MLS) are widely used for measurement of impulse responses of linear time-invariant systems in acoustic and audio systems. It is usually believed that one of the drawbacks of the MLS technique is a requirement on a synchronization of the devices used for the generation and acquisition of the MLS signals. The study presented in this paper shows that the MLS technique can easily be improved and applied to devices that are not synchronous, or even operating at different sampling frequencies. To show the efficiency of the proposed modification to MLS technique we provide several experiments such as a measurement with devices working at 44.1 kHz on the generation side and 96 kHz on the acquisition side, or a measurement of the frequency response function of a cheap mobile phone in which the synchronous clocking would be impossible to realize.

This paper analyzes the performance of Differential Head-Related Transfer Function (DHRTF), an alternative transfer function for headphone-based virtual sound source positioning within a horizontal plane. This experimental one-channel function is used to reduce processing and avoid timbre affection while preserving signal features important for sound localization. The use of positioning algorithm employing the DHRTF is compared to two other common positioning methods: amplitude panning and HRTF processing. Results of theoretical comparison and quality assessment of the methods by subjective listening tests are presented. The tests focus on distinctive aspects of the positioning methods: spatial impression, timbre affection and loudness fluctuations. The results show that the DHRTF positioning method is applicable with very promising performance; it avoids perceptible channel coloration that occurs within the HRTF method, and it delivers spatial impression more successfully than the simple amplitude panning method.

It has been shown that masked thresholds for complex tone maskers may depend on the relative phase between the spectral components of said maskers. Since these masker phase effects are less pronounced in hearing-impaired listeners, it indicates a possible role of peripheral compression. In order to study this phenomenon, we used a previously published physical model of the cochlea. We implemented the model into a version of the temporal-window model and used it to predict masked thresholds in harmonic complex tone maskers. The predicted thresholds were qualitatively similar with behavioral data (reproduced from the literature) of normal-hearing and hearing-impaired listeners: differences between the maximal and minimal masked thresholds decreased with increasing bandwidth of auditory filters (i.e., with increasing loss of peripheral compression). This effect was independent on the duration of the temporal window. In the predicted data, correlations between the maximum masking difference and the bandwidth of the cochlear filters were significant. However in the behavioral data, the correlations were significant but less pronounced. A loss of peripheral compression affects the isointensity responses of the simulated cochlear filters – broadens the response magnitudes and changes the curvature of the response phases. In the cochlear model, the broadening of the cochlear filters did not compensate for the decrease of the maximum masking difference caused by the changed phase curvature. If the relation between the filter bandwidth and compression loss is the same in the cochlear model and in the real cochlea, the less pronounced correlations observed in the behavioral data may indicate changed temporal resolution or other impairment in the auditory system.

This article presents a system for estimation of the direction of multiple speakers and zooming the sound of one of them. The proposed system is combined of two levels; namely, sound source direction estimation and acoustical zooming. The sound source direction estimation uses so-called the energetic analysis method for estimation of the direction of multiple speakers, whereas the acoustical zooming is based on modifying the parameters of the directional audio coding (DirAC) in order to zoom the sound of a selected speaker among the others. Listening tests are performed in order to evaluate this system using different time-frequency transforms.

HRTF (Head Related Transfer Function) je často používaným prostředkem k vytvoření virtuální reality. Tento článek se zabývá implementací automatizovaného systému pro rychlé měření HRTF v prostředí MATLAB. Velká pozornost byla věnována návrhu GUI, které bude sloužit k ovládání celého systému. To zahrnuje ovládání otočné židle v horizontální rovině, generování a příjem měřicích signálů přes externí zvukovou kartu a dále zpracování získaných dat a jejich export.

Virtual Acoustic Space (VAS) is created with Head Related Impulse Response (HRIR) and its Fourier transform – Head Related Transfer Function (HRTF). Both these functions are very dependent on each person’s individual features. This article presents Matlab implementation in basic HRTF personalization, i.e. composing from HRTFs of the subjects with similar anthropometric parameters. The aim is to compose a HRTF that will resemble the measured one to a certain degree of accuracy. The result is the set of functions that compares people’s anthropometric features, creates a personalized HRTF and saves it in the database.

Augmented Reality Audio (ARA) is mostly employed in terms of adding virtual sound source into an existing auditory scene. Common goal of such system is to preserve the highest level of fidelity and natural character of both real and virtual components. This paper introduces another approach to the ARA systems based on intentional modification of the native parameters of both the real and the virtual auditory segments. The system employs binaural microphone-equipped earphones used as microphone-hearthrough device, which are able to capture and immediately reproduce the auditory scene. A processing unit is included in the microphone-earphone signal path. The unit enables to apply various audio effects to both the captured and the virtual sound allowing warped perception of the auditory reality. The real-time processing algorithms were implemented in Pure Data environment. The proposed system provides immersive audio performance and has potential to be used in specific virtual reality applications, such as simulation of reality perception by people with mental disorders.

Sound quality is a major issue of concern for all modern telecom and audiovisual systems. The evaluation of sound quality includes both objective and subjective methods. The subjective methods reflect the listeners perception of the sound and the objective methods try to evaluate the sound parameters affecting its quality in correlation with the results of the subjective methods. The subjective tests are time consuming, which is inconvenient during the development of new algorithms of audio signal processing and tuning its parameters. This article investigates the possibility of applying objective methods for audio quality assessment on an acoustic zooming system and it compares their results and the results of the subjective intelligibility test.

Tento příspěvek popisuje návrh komplexního systému pro měření HRTF (Head Related Transfer Function) s využitím prostředí MATLAB. Automatizovaný systém je navržen tak, aby byl univerzální a kompaktní, poskytoval rychlé a přesné výsledky a umožňoval snadnou práci s daty, která mohou být poté zpracovávána. Otáčení subjektu je řízeno z prostředí Matlab ve kterém je také implementována celá měřicí metoda. Celý systém byl experimentálně testován při měření směrových charakteristik mikrofonů.

Virtual Auditory Space (VAS) can be used in a various fields, such as entertainment, medicine or assistive technologies. Virtual sound sources are usually positioned by filtering of the input signals with Head Related Transfer Function (HRTF). A new method for verification of created VAS was presented in our previous papers. The paper is devoted to the applied improvements of that method, which are functions for image edge detection and automatic graph plotting for a set of subjects.

The virtual sound sources can be used for numerous purposes such as computer games, entertainment (film industry etc.) and more importantly assistive devices. The most straightforward method for the virtual sound source positioning is filtering of the input sound signal with Head Related Transfer Function (HRTF). The HRTF can be measured on the chosen subject or synthesized. When the measurement is done the HRTF must be verified whether is suitable for the purposes of the VAS. The aim of this paper is to simplify such measurements and verifications by using GUI made in the Matlab environment.

Vytvoření virtuálního akustického prostoru má svůj význam pro celou řadu aplikací. K nejpoužívanějším způsobům pro vytvoření iluze zvuku přicházejícího z určitého směru patří využití konvoluce vstupního zvuku s HRIR (Head Related Impulse Response - tj. impulzní odezva vztažená k hlavě). Tento článek prezentuje implementaci zmíněného systému v reálném čase, využívající DSP systém Soundart Chameleon a prostředí Matlab. Prostředí Matlab je využíváno nejen pro editaci naměřených impulzních odezev a vytvoření HRIR sady v příslušném formátu, ale také pro automatické generování souborů potřebných pro překlad assembleru pro DSP. Pro změnu banky filtrů proto není vyžadována znalost programování DSP, stačí jen přeložit kód připravený Matlabem a načíst jej do DSP.

This article deals with issue of virtual sound source positioning for the purposes of acoustic navigation in unknown spaces and extends our research published last year. Navigation experiment was based on using PERSEUS assistive device. No appropriate standards in the field of performing subjective localization a navigation tests can be found. Our methods of design and evaluation of the listening and navigation tests are proposed and discussed. The following text is considered as an overview with references to our previous articles, which contain more detailed info.

This paper deals with the options of creating Virtual Acoustic Space for the purposes of navigation of visually impaired people in unknown environment. The idea of acoustic navigation is based on using virtually positioned sounds presented through the headphones, which are performed to the client by camera operator. This proposal requires a closer aim on head motion effect and ability of the camera operator to lead the client by the sounds stimuli avoiding present obstacles. The presented article follows our previous research, extends first results and suggestions, and discovers new aspects necessary to be examined.

This paper compares methods to acquire the HRTF (Head Related Transfer Function) applicable to navigation purposes, especially for the visually impaired. The most frequently used method for the HRTF acquisition (in general case) is the direct measurement of the HRTF. However, this method is very time-demanding. The alternative method have to be found, more suitable for navigation purposes. Possible methods (less time-demanding) are: the HRTF synthesis and the HRTF personalization. The scope of the paper is to compare the methods. In the paper the HRTFs of the one subject for 15 directions were measured. For the same subject the HRTFs were synthesized and personalized, using said methods described in the paper. The resulting HRTFs were compared. The results were verified by means of a simple listening test.

This paper focuses on a new approach to virtual sound source positioning. The classical method, when two Head Related Transfer Functions are used for processing the stereo signal, is substituted by the approach of affecting only one ear channel signal. The proposed method claims only the difference in spectral features in both channels is essential for ability of sound source perception. This fact allows to reduce the usual required positioning data and also computing operations to only a half. In this article, the process of proposed method is described and compared to the usual method on an example of harmonic signals.

Vytvoření virtuálního akustického prostoru má svůj význam pro celou řadu aplikací. K nejpoužívanějším způsobům pro vytvoření iluze zvuku přicházejícího z určitého směru patří využití HRTF (Head Related Transfer Function - tj. přenosová funkce vztažená k hlavě). Vzhledem náročnosti získání HRTF měřením je výhodná možnost syntézy HRTF na základě antropometrie uživatele. Tento článek prezentuje sadu uživatelských rozhraní umožňujících syntézu HRTF a testování lokalizace virtuálních zdrojů zvuku vytvořených touto metodou. V článku je také popsána použitá metoda syntézy HRTF a prezentovány výsledky dosažené s použitím popsaných uživatelských rozhraní.

Common problem in virtual auditory space (VAS) is so-called front-back confusion. As there is no binaural difference between signals coming from the position ahead of and behind of the listener, the listener slightly moves (turns) his head in order to determine the azimuth of the real sound source. However, in case of artificially created (virtual) sound source, presented via headphones, head rotation gives no effect. A head-tracking device can be used for enhancing orientation in the VAS. The paper is devoted to simple implementation of the above mentioned system in Matlab. The implementation details and the results of the experiment are presented in the paper. Due to experimental results the impact of head-tracking on virtual sound source localization was proved.

Estimation of pure tone hearing threshold is a part of hearing tests conducted in order to know the sensitivity of the human auditory system. This paper presents manual audiometer implemented in Matlab which can be used to conduct such measurements. Audiometer was calibrated by means of an artificial ear and pistonphone. Modified Hughson-Westlake method usually used in manual audiometry is described.

This paper focuses on examination of features of Differential Head Related Transfer Function (DHRTF) and it's issues of its processing. DHRTF is being developed and is considered as a new method in virtual sound source positioning with advantages of reduction of data storage and data processing requirements. Processing by DHRTF is based on affecting only one of the pair of the common stereo channels. Anyway, this approach carries several limitations and issues to be solved. The appropriate solutions requires a good comprehension of DHRTF behavior and exploring and visualization of important spatial information which it carries. Matlab offers a sufficient tool for exploring this area. According to the preliminary results a method for increasing DHRTF processing efficiency is introduced.

A virtual sound source can be positioned in a virtual auditory space with various methods, as shown in our other articles. In our last year paper, we shown an application of virtual sound source positioning for assistive technology: for helping blind people to orientate in real space. In the case, an operator can watch video simulating vision of sighted people and create a virtual sound source which navigates direction of user's walk. An appropriate positioning device (e.g. joystick) has to be used for determining the direction. The paper is devoted to improved version of last year simple implementation of the above mentioned system in Matlab. The Matlab-code from last year version was rewritten and optimized. This implementation is used for design of the system and testing the algorithms for virtual sound source positioning.

In this work sound records during phonation of vowel /a:/ were acquisited and were properly associated with magnetic resonance (MR) image data as a base of computational model of vocal tract.

Using own measured HRTF for each user is considered as the best way to obtain a high-quality perception of sound source virtual position. Without proper equipment the measuring process is very time consuming, so we decided to use the virtual auditory space with small resolution only at first. Then a series of virtually positioned sound were created. Necessity of equalization the room, loudspeaker and headphones for later listening is also discussed. Finally, we made a test sequence fixed to verify orientation in virtual acoustic space. All 6 subjects whose HRTF was obtained were with no sight disability. Last subject was completely sightless, he did not attend the measuring procedure and HRTF sequence of another subject had to be used instead.

For acoustic navigation of visually impaired persons the artificially created sound signals are used. One way to create the virtual audio space is by using binaural signals obtained by an artificial head. To design and create these signals it is necessary to know electro-acoustic properties of the system used. For that reason, this contribution deals with measurement and detailed analysis of transfer functions, non-linear distortion and crosstalk of a complete electro-acoustic test system with an artificial head. Analysis includes electro-acoustic transducers, such as headphones and microphones as well as the influence of outer ear and eventually may include a correction for the particular sound perception of individual persons.

Virtual sound positioning is an efficient approach in case of developing devices for assistive technologies, particularly for visually impaired. Our goal was to compare suitability of different methods, especially method using HRTF or HRIR. A set of HRTF (Head Related Transfer Function) was measured and used for creating virtual acoustic space. After required signal processing an orientation in these spaces was investigated on subjects with no visual handicap and also on visually impaired subject, where multiple stimuli were used. Positioning algorithms were modeled, implemented and tested in MATLAB and as VST plugin. HRTF behavior was also analyzed by means of an auditory model.

This paper deals with the options in developing Virtual Auditory Spaces (VAS) for the purposes of visually impaired in assistive technology. Our approach of complex tests consisting of testing the subject´s hearing, choosing members of the test groups, and the design of the main space orientation tests is presented. Results of the first basic tests aimed on Just noticeable Difference (JND) between visually impaired and subjects with no sight defect are also presented. The results of the following tests are the most important factors for future design of the VAS for the assistive technology purposes.

A virtual sound source can be positioned in virtual auditory space with various methods, as shown in our other articles. There are many applications of virtual sound source positioning, e.g. as assistive technology for helping blind people to orientate in real space. In the case, an operator can watch video simulating vision of sighted people and create the virtual sound source which navigates direction of user's walk. An appropriate positioning device (e.g. joystick) has to be used for determining the direction. The paper is devoted to simple implementation of the system in Matlab. The picture from user's camera is shown, an operator have to point the joystick in direction, where virtual sound source have to be generated in users headphones. This implementation is used for design of the system and testing the algorithms for virtual sound source positioning.

Using Head Related Transfer Function (HRTF) is a key for creating an effective three dimensional sound. Nowadays, two main methods are used to get a set of HRTF in effective quality - modeling according to basic anthropometrical parameters or direct measurement on particular subject or acoustic manikin. This interface was created for comparing perception of sounds adjusted by modeled or measured HRTF.

Pinna, head and body causes distortion of the perceived sound and it, in turn, help to sound source localization in medial plane, where the interaural time and level difffferences between the signals coming to both ears are the same. Auditory model is used in this paper to analyse which spectral parts of transient sound signal are mostly affffected by the pinna, head and body in dependence on the sound source position in space. The presented results show that mainly infuenced bands are between 5 and 12 kHz, as was already shown in literature, and additionally to these data, some other bands located in lower frequencies (between 1 and 4 kHz). Subjective measurements need to be done in order to evaluate these results that lower frequencies are also important for the sound source localization in the medial plane.

Using own measured HRTF for each user is considered as the best way to obtain a high-quality perception of sound source virtual position. Without proper equipment the measuring process is very time consuming, so we decided to use the virtual auditory space with small resolution only at first. In this work, we got the HRTF in 15 positions for 7 different subjects with using MLS signal. Then a series of virtually positioned sound was created using MATLAB algorithm. Necessity of equalization the room, loudspeaker and headphones for later listening is also discussed. Finally, we made a test sequence fixed to verify orientation in virtual acoustic space.

Příspěvek podává přehled o aktuálních trendech ve velkoplošném zobrazování a prostorovém zvuku.

Tento článek představuje uživatelská rozhraní vytvořená v prostředí Matlab, která umožňují ilustraci základních principů binaurální lokalizace. První z popisovaných rozhraní umožňuje subjektivně sledovat změnu lokalizace zvukového podnětu při intenzitních a/nebo časových rozdílech mezi signály reprodukovanými do jednotlivých uší posluchače. Druhé rozhraní ilustruje komplexnější přístup při simulaci filtrace zvuku směrově závislou přenosovou funkcí vztaženou k hlavě (HRTF - Head Related Transfer Function).

Studium zvukového pole vytvářeného soustavou jednotlivých zvukových zdrojů má význam pro mnoho aplikací v oblasti zvukové techniky. Tento článek popisuje realizaci jednoduché simulace 2D pole zdrojů zvuku v prostředí Matlab. Popisovaná simulace umožňuje simulovat realizaci virtuálního zdroje (myšleného zdroje zvuku umístěného v konkrétní pozici) pomocí různých uspořádání 2D pole zdrojů zvuku. Navržená simulace je zamýšlena k využití pro ilustrační a demonstrační účely, pro rychlý odhad rozložení zvukového pole pro konkrétní rozložení reproduktorů, zejména při studiu systémů pro vícekanálový zvuk.

This article results from the last year article at the conference, which designed structure of an experimental audio encoder. The designed structure was implemented by Matlab functions, each function represents one system block. Consequently was made testing implementation of a simple perceptual audio coder which respects structure that was designed in the article. Presented simple encoder substitutes psychoacoustic model by a static one which takes into account frequency dependency of the hearing threshold only. Masking effects are not considered by this static psychoacoustic model, which is not usual but demonstrative enough. Coding is performed in the frequency domain that is obtained by the Fast Fourier Transform. Frequency components are then re-quantized with a different bit-length according to the designed psychoacoustic model. Appropriate decoder was also implemented to prove correct function of the encoder.

Digitální zpracování audio a video signálů je v dnešní době již téměř samozřejmostí a analogové zpracování je již na ústupu. Příspěvek je rozdělen na dvě části, kde první se zabývá zpracováním audiosignálu a druhá část zpracováním obrazového signálu a videosignálu. Úvodní odstavce v obou částech přibližují čtenáři problematiku zpracování audiovisuální informace obecně. Dále je vyjmenováno a popsáno několik zajímavých metod zpracování zvuku a obrazu.

Všechny dnes široce rozšířené systémy pro kompresi zvuku vycházejí ze stejných psychoakustických principů a využívají velmi podobnou vnitřní blokovou strukturu. To nám umožňuje vytvoření jediného modulárního systému, který bude mít definován pouze formát dat vstupujících, respektive vystupujících z jednotlivých systémových bloků. Vnitřní struktura daného bloku není přímo definována a závisí pouze na realizovaném kompresním algoritmu.

Oblast multimediální techniky je široká, spadá do ní vše, co se dotýká lidských smyslů, především zraku a sluchu. V užším významu se pod touto oblastí rozumí oblast audiovizuální techniky, rozhlasu a televize, která přináší rozsáhlé a rychlé změny technologie. Pro výuku předmětů ze zmiňovaných oblastí byla na našem pracovišti vybudována nová multimediální počítačová laboratoř, jejíž vybavení umožňuje vytvoření nových a moderních úloh. Vybavení laboratoře zahrnuje technické prostředky pro snímání, zdrojové kódování, záznam, zpracování a reprodukci zvukové a obrazové informace. Pro výuku ve vyjmenovaných oblastech lze s výhodou využít programové prostředí Matlab, jak ukazuje tento článek.

Tento článek se zabývá problematikou nelineárních systémů, s nimiž se setkáváme denodenně, a z nichž mnohé považujeme pro zjednodušení za lineární. Cílem článku je upozornit na chyby vzniklé při nelineárním zpracování signálu, které nejsou na první pohled zřetelné a jež jsou velmi často zanedbávány, či v horším případě úplně opomíjeny. V článku jsou ukázány některé důsledky zanedbání nelinearit ať už z pohledu zpracování signálu a tvorby efektů v digitální doméně - v prostředí Matlab, nebo při měření zařízení jakými mohou být zesilovač, reproduktor apod. Článek je zaměřen na tři části: chyby při měření soustav s nelinearitami, jež považujeme za lineární; chyby při měření nelineárního zkreslení a chyby při samotném zpracování či tvorbě efektů. Ve všech případech mohou být buď naměřeny chybné výsledky, nebo může dojít k nesprávnému zpracování signálu. Všechny tyto negativní důsledky pak mohou z hlediska vnímání signálu působit rušivě.

In relation to lossy audio compression algorithms, the need for assessment of perceived sound quality of the compressed signal arises. Objective sound quality evaluation methods require appropriate models of human hearing system. In the paper a binaural psychoacoustic model is presented. The model comprises a monaural and a binaural processing stage, the former being based on an active nonlinear cochlear model including the effect of IHC, the latter consisting of a two-dimensional comparison of the right-ear and left-ear signal. Approximation of the influence of outer and middle ear employs the transfer properties of acoustic waveguides.

Psychoakustické modely jsou hojně využívány především v systémech pro redukci datového toku a v algoritmech objektivního hodnocení kvality zvuku. Přínos modelování lidského sluchového ústrojí však může být širší, neboť poznání nejpodstatnějších principů přenosu informace sluchovou drahou může vést např. ke zlepšení funkce kochleárních implantátů. Předkládaný model sluchové dráhy čerpá z nejnovějších biologických poznatků a snaží se přiblížit se fysiologii při zachování únosné výpočetní náročnosti.

Chceme-li hodnotit kvalitativní stránku zpracování zvukové informace, je nejprve nutné stanovit objektivní parametry systémů pro toto zpracování použitých, jako např. přenosové vlastnosti. Pro měření těchto parametrů se často s výhodou používá tzv. posloupností maximální délky (MLS). Při praktickém použití metod využívajících MLS je možné setkat se s určitými problémy, zvláště je-li nutné generovat měřicí signál jedním zařízením a zaznamenávat a vyhodnocovat ho na jiném zařízení. Tento článek se zabývá vlivem špatné synchronizace vzorkovacích kmitočtů při použití MLS signálů pro měření přenosových vlastností. Problém je zkoumán nejdříve pomocí harmonického signálu a poté pomocí modelování tohoto jevu v prostředí Matlab. Model byl porovnán s výsledky reálného měření.

Popisovaná metoda rozšířuje použití metody elektro-akustických analogií i pro systémy, které obsahují rozprostřené prvky. Náhradní obvody pro vlnovody různých typů již byly publikovány, ale popisovaná metoda je založená na obecnějším přístupu a je vhodnější pro použití v počítačovém modelování.

Pro studium kvalitativních aspektů zpracování zvukové informace v procesu slyšení je důležitá detailní znalost zpracování této informace, ke které dochází ve sluchové dráze. Majoritní podíl na tomto zpracování má vnitřní ucho a navazující sluchová centra v mozku, ovšem při skutečně detailním rozboru tohoto zpracování nemůžeme opomenout vliv středního a vnějšího ucha. Tento článek se zabývá sestavením modelu středního a vnějšího ucha v Matlabu se zvláštním zřetelem na modelování vnějšího zvukovodu lidského ucha.

Acoustic waveguides are widely used in many acoustic applications, for example in musical acoustics (music instruments), electro-acoustics (loudspeakers) etc. The underlying theory is also applicable to other special disciplines, e.g. for modeling of human ear canal or human vocal tract. The achieved results can be used not only in medicine but also in psychoacoustics, as a part of the overall model of human auditory system. In the paper a Matlab script with a GUI is presented, which can be used for modeling of transfer properties of acoustic waveguides.

Pochopení způsobu jakým funguje lidské slyšení je základním předpokladem pro návrh nových metod zpracování zvuku. Tento článek se zabývá studiem vlivu přenosové charakteristiky vnějšího zvukovodu na zvukový signál vnímaný ve vnitřním uchu. Přenosová funkce vnějšího ucha je také často dávána do souvislosti s chronickým akustickým traumatem. Ale stejně často se také uvádí, že vliv vnějšího ucha (zvukovodu) je kompenzován při průchodu zvuku středním uchem. Tento článek se snaží přispět ke studiu tohoto problému pomocí modelování vnějšího a středního ucha.

V příspěvku je popsána tvorba fyzického modelu vokálního traktu člověka na kterém byla testována shodnost měřených a vypočtených akustických charakteristik.

The models of human ear canal are useful for many applications, for example in electroacoustics, physiological and psychological acoustic and for medical purposes. This paper deals with a human ear canal modeling which uses the electro-acoustic analogy. Human ear canal is divided into several parts, and each of them is considered as a non-uniform waveguide with a suitable shape. The description for each of parts in terms of chain matrix is derived. The impact of the boundary conditions (pinna, eardrum, middle and inner ear, etc.) and losses in ear canal is also taken in the account.

Příspěvek se zabývá popisem akustických vlnovodů a uvádí výhody a nevýhody použití metody elektro-akustických analogií pro tento popis. Použití zmíněné metody je ilustrováno na modelování vnějšího zvukovodu lidského ucha.

Akustické vlnovody proměnného průřezu jsou použitelné pro celou řadu aplikací, od elektroakustických až po medicínské. Dvě hlavní skupiny těchto vlnovodů - Salmonovy a Besselovy - je možné popsat pomocí kaskádních matic. V tomto příspěvku je na základě studování parametrů kaskádní matice vlnovodu sledována souvislost mezi parametry tvarové funkce vlnovodu a přenosovou funkcí pro akustický tlak. Dále se studuje kaskádní matice spojení dvou vlnovodů pro ověření vlivu parametrů tohoto spojení (spojitost, hladkost) na přenosovou funkci výsledného vlnovodu.

The paper describes a measurement on two cadaverous temporal bones. The Human Ear Canal Transfer function was determined and results are discussed.