bluuebunny/crossref_metadata_2025 huggingface dataset:
Reconstructing individual monophonic instruments from musical mixtures using scene completion.
Authors: Han Jinyu , Pardo Bryan
Published: October 2010

Monaural sound source separation is the process of separating sound sources from a single channel mixture. In mixtures of pitched musical instruments, the problem of overlapping harmonics poses a significant challenge to source separation and reconstruction. One standard method to resolve overlapped harmonics is based on the assumption that harmonics of the same source have correlated amplitude envelopes: common amplitude modulation (CAM). Based on CAM, overlapped harmonics are approximated using the amplitude envelope from the non-overlapped harmonics of the same note. CAM assumes non-overlapped harmonics from the same noteare available and have similar amplitude envelopes to the overlapped harmonics. This is not always the case. A technique is proposed for harmonic temporal envelope estimation based on the idea of scene completion. The system learns the harmonic envelope for each instrument’s notes from the non-overlapped harmonics of other notes played by that instrument, wherever they occur in the recording. This model is used to reconstruct the overlapped harmonic envelopes for obstructed harmonics. This allows reconstruction of completely overlapped notes, yet does not require predetermined instrument models. Experiments show the proposed algorithm performs better than an existing system based on CAM when the harmonics of pitched instrument are strongly overlapped.
Including parametric models in spectrogram decomposition.
Authors: Bertrand David , Romain Hennequin , Roland Badeau , Jean-Louis Durrieu
Published: October 2010

Regarding the decomposition of an audio stream in elementary sound objects, the non-negative matrix factorization (NMF) is now one of the prominent tools that can be employed. For instance, in the context of an automatic music transcription application, the interest of the technique relies on its ability to use redundancies over a whole piece to hopefully be able to identify separated sound objects as musical notes. In this ideal case, the spectrogram factorizes into a matrix of inferred spectral atoms and a matrix of temporal activations associated to these atoms. However, the unconstrained version of the NMF does not guarantee that the resulting atoms are actually musical notes: the spectrum of a single note can be split into several atoms or, conversely, a single atom could represent more than one note. The work that will be presented here deals with parametric models designed to describe the spectral modifications that occur along the progress of a single audio event (e.g., a musical note) in order to obtain a more meaningful decomposition from a musical point of view.
Urban noise and transport as a base of environmental quality strategy.
Authors: Orozco Medina Martha , Arturo Figueroa Montano
Published: October 2010

In recent years, urban noise has become an essential element in the environmental analysis since the study allows for a support tool for the analysis of environmental quality in the cities. Additionally, residents of large cities are demanding more options for transportation improvements from the standpoint of energy efficiency, low emissions of pollutants, and strategic routes, among others. This proposal presents the results of a diagnosis of urban noise on an avenue with high traffic flow in the city of Guadalajara, and analyzed during the construction of a rapid rail transit preferential and once it is running. The results of the noise levels actually indicate that the entry of a fast and environmental friendly bus significantly reduces noise emissions and contributes to the acoustic quality of the avenue with the benefits it represents for the people.
Noise level over 40 dB.
Authors: Barroeta Carlos , Ortiz Floriberto R. , Novoa Colin Juan Francisco , Salinas Salinas Marisol
Published: October 2010

Noise in cities is one of the most important problems for the population of the world. Traffic and industry noise is possibly producing new illness; 40–90 dB is the range of tolerance value for the human ear; depending on the application, some noise was analyzed in the real world. These parameters should be observed in design tactics for getting a better environment policy. Corporations and citizens must work for less pollution and more opportunities for development.
Internal mapping of the normal modes of a liquid filled spherical resonator.
Authors: Mobley Joel , Raymond Jason L. , Prather Wayne E. , Gaitan D. Felipe
Published: October 2010

We report on the measurements of the pressure fields set up inside a water-filled steel shell over a spectral range (10–50 kHz) that includes 75 modes of the composite system. The 9.5-in.-diameter shell has a wall thickness to radius ratio of h/R = 0.158, which is beyond the thin shell limit. Hydrophone scans of the acoustic pressure fields in the fluid were performed, from which 46 modes of the composite resonator were identified. The modal frequencies are in good agreement with those predicted by Mehl for a fluid-filled shell of arbitrary wall thickness. The radial pressure profiles of the modes are shown to fit the expected spherical Bessel profiles for solutions of the Helmholtz equation under spherical symmetry. The results are interpreted in terms of the idealized rigid shell and empty shell limits.
Bubble size distribution and wave attenuation in an ultrasonic cavitation field.
Authors: Labouret Stéphane , Frohly Jacques
Published: October 2010

The bubble size distributions of a cavitation bubble field in a standing wave have been determined with an electromagnetic method, for bubble radii from 1.2 μm to about 10 μm. Results will be displayed for 349 kHz and 1.1 MHz ultrasound frequencies in continuous mode. A stabilization of the size distribution by the wave attenuation in the bubble field is proposed for a cavitation regime of low activity at 349 kHz. Results in chopped irradiation mode are presented at both frequencies. Measurements in continuous irradiation in the presence of several concentration of sodium dodecyle sulfate will also be displayed.
Coherent backscattering enhancement in cavities: Simple shape cavity revisited.
Authors: Stefan Catheline , Thomas Gallot , Philippe Roux , Guillemette Ribay , Julien de Rosny
Published: October 2010

Coherent backscattering effect (CBE) is classically introduced in disordered, random, or chaotic media. In this work, the attention is focused on simple parallelepipedic cavities since, contrary to a widespread idea, CBE can also be observed for a pure-tone source in a 1-D cavity. This approach is of two-fold interest. First, analytical computations predict a dimensional dependence of the coherent backscattering enhancement according to a (3/2)d law, d being the dimensionality of the cavity, that have not yet been compared to experiments. Second, it opens new ballistic interpretations for which each multiply reverberated path is associated with more (rectangle and parallelepipedic cavities) or less (1-D cavity) than one single reciprocal counterpart. This paper is the first of two; the second paper dealing with some impacts of symmetry on CBE.
Coherent backscattering enhancement in cavities: Highlight of the role of symmetry.
Authors: Gallot Thomas , Stefan Catheline , Roux Philippe
Published: October 2010

Through experiments and simulations, some consequences of symmetry on coherent backscattering enhancement (CBE) in cavities are reported here. First, CBE outside the source is observed (a) on a single symmetric point in a 1-D cavity, in a 2-D circular membrane, and in a 2-D symmetric chaotic cavity; (b) on three points in a rectangle; and (c) seven in a parallelepiped. Second, the existence of enhanced intensity lines and plans in 2-D or 3-D cavities is demonstrated. Third, we show how antisymmetry is responsible for the existence of a coherent backscattering decrement with a dimensional dependence of (12)d law, d being the dimensionality of the cavity.
Underwater sound radiation by a layered infinite plate with discontinuity introduced by a signal conditioning plate.
Authors: Pan Jie , Zhang Yanni
Published: October 2010

Adding a layer of elastic foam to a baseplate reduces underwater sound radiation from the vibrating baseplate and sound reflection of incoming sound. To increase the local signal to noise ratio of a detecting hydrophone placed in front of the foam, a signal conditioning plate of finite size is introduced between the foam and hydrophone. This paper considers the effect of the discontinuity by the signal conditioning plate on the general properteis of the sound radiation of a layered plate. Both far field sound directionality and near field sound distribution are calculated to illustrate the properties. The scattering of structural waves in the elastic foam by the signal conditioning plate is used to explain the observed properties.
Interior holography.
Authors: Williams Earl G.
Published: October 2010

Interior spaces with vibrating walls embrace a broad set of structural acoustic problems (planes, trains and automobiles). In the hunt for efficient measurement schemes that provide details of both the acoustic and structural fields, inverse holographic methods are very promising. These methods use a surface of measured data and mathematically back-track the field in time and space filling in a 3-D volume around the array in the process and are particularly attractive since they can image the time averaged acoustic intensity vector, localizing and quantifying surface sources. There is a score of holographic back-propagation methods in use today. In almost all cases single measurement surfaces are not accurate due to reverberation and dual conformal surfaces are needed to separate sound fields coming from forward and backward directions. The spherical array (open or rigid) is a noticeable exception, an accurate mathematical approach is obtained with only one surface of data. This paper presents an overview of the inverse problem in interior spaces paying particular attention to the rigid spherical array, discussing improvements that extend the frequency range and volume of study, and simple regularization methods to stabilize the ill-posed nature of the volumetric reconstructions. [Work supported by ONR.]
The role of speech readability in the intelligibility of visual speech signals produced by cued speech transliterators.
Authors: Krause Jean C. , Bennett Abby N.
Published: October 2010

Although it is common practice for deaf individuals to use interpreters as a means of accessing spoken information, very little is known about the intelligibility of visual signals produced by interpreters [Kluwin and Stewart, Odyssey 2, 15–17 (2001)]. One of the first studies to evaluate interpreter intelligibility [Pelley, MS thesis, USF (2008)] recently examined 12 interpreters who use a speech-based visual communication mode known as cued speech. The interpreters, or transliterators, ranged in intelligibility from 52% to 90% (as evaluated by deaf individuals proficient in cued speech), and the accuracy of their cues (hand signals) accounted for only 26% of the variance in intelligibility. Because mouth movements are also part of the cued speech signal, this experiment examined the contribution of speechreadability to the intelligibility of the messages. Eight individuals who were unfamiliar with cued speech were presented with visual stimuli excised from the transliterated messages and asked to transcribe the stimuli. Speech readability of the visual speech signals was measured as the percentage of words correctly received. Results show speech readability is independent of accuracy and accounts for roughly 13% (on average) or more (when transliterator experience is controlled) of the variance in intelligibility. [Work supported by NIH/NIDCD Grant 5-R03-DC007355.]
The judgment of voices does not conform to the evaluative/potency/activity framework.
Authors: Hinz Verlin , Berglund Luke , Maw Natasha
Published: October 2010

The human voice is indicative of a variety of characteristics. It may be particularly influential in judgments such as of charisma. But what features of the voice are indicators of these characteristics? Osgood et al. (1957) suggested that a vast array of stimuli are judged by humans along the three dimensions of evaluation, potency, and activity. We tried to determine whether voices are also judged according to these three dimensions. Male and female judges rated 36 short (2 s) segments of voice segments from a variety of languages on Osgood et al.’s dimensions, using 9 different scales. Listeners were able to make their judgments quickly, even though the stimuli were short. Confirmatory factor analysis failed to produce the three factor structure from the listener’s judgments of the voices. Gender and language of the voices did not help clarify the structure of the judgments. Hence, this study demonstrates that human voices are unique stimuli that may be judged using a more complex framework.
Spectral modulation transfer function and vowel identification.
Authors: Phillips Jillian , Wiese Sarah , Liu Chang , Eddins David
Published: October 2010

Psychophysical and physiological studies have demonstrated selectivity for spectral modulation frequency in the auditory system, suggesting that auditory perception of complex spectra might be based on spectral modulation channels. However, an early study [Liu and Eddins (2008)] reported that vowel identification was significantly reduced by filtering in spectral modulation frequency domain, although there was no significant correlation between spectral modulation detection thresholds and vowel identification. The latter result may have been due to the small number of listeners (n = 5) in the previous study. Using a larger number of listeners, the present study measured spectral modulation transfer functions and vowel identification performance with and without spectral modulation filtering in 11 normal-hearing listeners. Vowel identification performance was significantly correlated with spectral modulation detection thresholds at 0.5 cyc/oct (i.e., higher spectral modulation detection thresholds associated with poorer vowel identification scores). In addition, average spectral modulation detection thresholds were significantly correlated with the change in vowel identification associated with filtering in the spectral modulation domain. Both of these results highlight the relation between the perception of sinusoidal spectral modulation and vowel identification.
Effects of training and exposure on the perception of spectrally degraded speech and environmental sounds.
Authors: Shafiro Valeriy , Sheft Stanley , Ho Kim , Gygi Brian
Published: October 2010

Loebach and Pisoni [J. Acoust. Soc. Am. 123, 1126–1139 (2008)] demonstrated perceptual improvements following a brief training with spectrally degraded environmental sounds generalized to speech. The present study investigated whether benefit could also result from mere exposure over a longer time period. Participants received two pretests, one week apart, prior to beginning a week-long environmental sound training regimen, which was followed by two post-test sessions, separated by another week without training. Spectrally degraded stimuli, processed with a four-channel vocoder, consisted of a 160-item environmental sound test, word and sentence tests (CNC, SPIN), and a battery of psychoacoustic and cognitive tests. Results indicated significant improvements in all speech and environmental sound scores between the initial pretest and the last post-test, with performance increments following both exposure and training. For environmental sounds (the stimulus class that was trained), the magnitude of positive change that accompanied training was much greater than that due to exposure alone, with improvement for untrained categories roughly comparable to the speech benefit from exposure. Speech and environmental sound performance were differentially correlated with tests of spectral and temporal-fine-structure processing abilities, while working memory and executive function were correlated with speech, but not environmental sound perception. [Work supported by NIH.]
Intelligibility of temporally interrupted dual-band speech.
Authors: Sheft Stanley , Shafiro Valeriy , Kuvadia Sejal , Lorenzi Christian
Published: October 2010

To evaluate potential factors underlying the speech benefit obtained when combining acoustic and electrical stimulation, the perception of temporally interrupted dual-band speech (lowpass at 800 Hz and highpass at 1600 Hz) was investigated with HINT sentences quasi square-wave gated at rates ranging from 0.8 to 12.5 Hz. For the highpass band alone, intelligibility improved with gating rate, while for the lowpass band, performance was consistently poor at all rates. Concurrent presentation of both bands interrupted at the same rate led to significantly better performance that exceeded predictions based on probability summation. Similar performance benefits were obtained when the two bands were gated at different rates. Specific aspects of dual-band performance could be accounted for through detection theory with appropriate selection of set size for 1-of-n detection and cross-band correlation. However, no single parameter set led to a complete description of results. An alternative approach is to view the consequence of combining speech bands as an enhancement of definition of stimulus structure to allow for greater information transmission. A similar effect may contribute to the benefit obtained when combining acoustic and electric speech stimulation with acoustic elements providing structure through enhancement of signal coherence and segmentation. [Work supported by NIH.]
Improving speech intelligibility for hearing aid users using a speech segregation algorithm.
Authors: Vishnubhotla Srikanth , Espy-Wilson Carol
Published: October 2010

Users of hearing aid devices encounter great difficulty in speech recognition in noisy environments, and show inferior performance in situations with interfering background noise or competing speakers as compared to the case of a quiet background. In such devices, a pre-processing stage, which first removes the background interference, should greatly boost the speech recognition performance in noisy environments. In this work, the performance of a previously reported speech extraction algorithm is investigated for the purpose of improving the speech intelligibility of hearing aid users. Recently, we performed segregation of speech mixtures using our speech extraction algorithm and showed improvement in the speech recognition performance of normal hearing listeners. In this work, we investigate the performance of our algorithm for users of hearing aids on the same perceptual task of recognizing speech in the presence of either interfering speech or noise. We focus on tuning the segregation algorithm for hearing aid users, and emphasize on the preservation of the appropriate envelope and fine structure cues which will prove most helpful in the recognition of the target speech signal.
A simple hearing screener up to 17 kHz for use in speech perception research.
Authors: Composeo Elizabeth , Luque Jenna , Maye Jessica , Siegel Jonathan
Published: October 2010

Speech scientists are well aware that hearing ability affects the perception of speech sounds. However, hearing is rarely directly tested in studies of normal speech perception. Furthermore, standard audiological testing does not test above 8 kHz, although speech spans frequencies as high as 17 kHz. Given that amplification of frequencies above 8 kHz aids speech perception in hearing impaired subjects, higher frequencies appear to play a role in speech perception. Thus, speech perception research would be improved by incorporating direct tests of speech frequency hearing in all participants. We developed a simple, computer-based hearing screener for use in speech perception research. Using easily accessible equipment and simple calibration techniques, the screener tests bilateral hearing sensitivity between 250 Hz–17 kHz in about 5 min and is easily implemented by non-audiologists. A total of 111 ears from 56 participants were tested using both the simple screener and a full audiological workup using modified Bekesy tracking up to 17 kHz. The hearing screener identified hearing loss across a wide range of frequencies with a high level of both sensitivity (0.85395) and specificity (0.83118). This study demonstrates that a quick and simple hearing screener can provide an accurate assessment of hearing ability across the speech frequency range.
Quantification of context effects in phoneme and word recognition by non-native listeners.
Authors: Shi Lu , Koenig Laura
Published: October 2010

Non-native listeners have difficulty perceiving context cues in speech. The current study intended to quantify the gain in non-native listeners’ perception of English speech due to context by adopting the Boothroyd–Nittrouer model [J. Acoust. Soc. Am. 84, 101–114 (1988)], in which context effects are expressed via factors j and k. These factors are mathematically equal to the total number of components in a speech stimulus with no context, but are reduced when context is present. Ten normal-hearing, non-native and 10 native listeners participated in this study. To investigate the effect of acoustic-phonetic cues, 12 lists of CVC English words and 12 lists of nonsense words were employed. To investigate the effect of semantic-syntactic cues, three sets of 20 four-word sentences were employed. These sentences differed in the amount of semantic and syntactic cues contained in the stimuli. All stimuli were presented binaurally at 45 dB HL. A concomitant speech-weighted steady-state noise was added at 39 and 45 dB HL for words and sentences, respectively. The non-native listeners’ verbal responses were recorded digitally. It was expected that j and k are larger in value in non-native than native listeners, and that the magnitude of the difference is associated with the non-native listeners’ English learning history.
The interrelation between stimulus range and the number of response categories in vowel categorization.
Authors: Benders Titia , Escudero Paola
Published: October 2010

The categorization of speech sounds is influenced by the preceding sound context. Specifically, listeners shift their boundary between two categories toward the values of the preceding sounds. Such boundary shifts are larger if listeners can respond with fewer categories than are actually present in the stimuli. This study investigates the influence of the stimulus range on boundary shifts between vowels and whether these shifts are smaller if listeners can choose from more categories than are present in the stimuli. The F1 continuum between Spanish /i/ and /e/ was presented to Peruvian listeners in three ranges. Half of the listeners responded with /i/ and /e/, the other half chose from the five Spanish vowels. Boundary shifts between /i/ and /e/ as a function of the stimulus range were observed, which were smaller when listeners could choose from five categories. In subsequent analyzes, the influence of the preceding sound context appeared smaller for listeners that could choose from five categories. Also, these listeners shifted their boundaries after more stimuli than listeners who responded with only two response categories. It is proposed that the effect of the number of response categories on boundary shifts is due to an automatic process in speech perception.
The high redundancy of speech sounds in sentential context.
Authors: Stilp Christian E.
Published: October 2010

Readers are exceptionally good at guessing which letter comes next in printed text, even with little contextual information. This was quantified by Shannon [Bell Syst. Tech. J., (1948); Bell Syst. Tech. J., (1951)], who demonstrated that printed English text is approximately 50% redundant (predictable). Here, redundancy of American English speech was examined in a modified “guessing game” paradigm, as listeners guessed which speech sound came next throughout an entire sentence. Four TIMIT sentences were presented to 19 listeners, and guesses were made by clicking buttons on a computer screen corresponding to each consonant and vowel sound in American English. Listeners received feedback after each guess and heard each sentence (up to their most recent correct guess) as often as they wished. Numbers of guesses were at chance levels at sentence and word onsets, and were much better than chance with the addition of phonemic and sentential context. Equiprobable speech sounds predict 0% redundancy (chance performance) and entropy of 5.29 bits/sound. Listener performance demonstrates a mean sentence redundancy of 40% (3.18 bits/sound), within a narrow range of 36–43. Behavioral results will be evaluated with reference to simple phoneme frequency and transitional probabilities between phonemes. [Work supported by NIDCD.]
Frequency maDDers, but phonology and formality maTTer, too.
Authors: Gafter Roey , Kurumada Chigusa , Pineda Marisa , Sumner Meghan
Published: October 2010

In cases of allophonic variation, the frequency of a variant has been claimed to influence phonological representations, affecting the perception and recognition of words. For example, Connine (2004) found stronger Ganong effects in words with frequent variants (e.g., intervocalic tap in the word pretty) than in words with infrequent variants (e.g., [t] in the word pretty). Specifically, in a phoneme-monitoring task, listeners reported more PRETTY responses when listening to a pre[D]y-bre[D]y continuum than a pre[t]y-bre[t]y continuum. If this bias is due to variant frequency, we might expect this effect to generalize across variants, independent of other factors (context, within-paradigm alternations). However, a diverse set of variants and alternate explanatory factors contributing to the effect have not yet been investigated. For example, the roles of paradigm variants (e.g., baiting-bait versus pretty-*pret) and filler type (e.g., testing for intervocalic [t] in the context of carefully versus casually articulated fillers) might also explain these effects. This study builds on Connine (2004) and examines both paradigmatic relations and filler type across two allophonic alternations (tapping, final [t] glottalization). Ultimately, the results show that variant frequency alone is insufficient to account for this effect. An integrated theory of speech perception is proposed.
Phonetic features guide phonotactic generalizations in perception.
Authors: Cristià Alejandrina , Mielke Jeff , Peperkamp Sharon
Published: October 2010

Although features are useful theoretical constructs (they allow parsimonious descriptions of phonological inventories, patterns, and alternations), their etology remains debated. Three hypotheses have been advanced: (1) Phonological patterns involving phonetically natural classes arise from historical changes affecting similar sounds, but features have no psychological reality; (2) features emerge in the listeners’ phonology on the basis of phonetic and phonological experience (including exposure to patterns resulting from historical changes); (3) features are abstract symbols provided by universal grammar (UG), and they are independent of phonetic implementation. To test these alternatives, French and English listeners were exposed to a constraint on obstruent voicing, and tested implicitly on their generalization of the constraint to untrained obstruents. If the historical hypothesis is true, no generalization would occur. Since voiced stops and fricatives form a phonetic natural class in French but not in English, emergent features would favor generalization only in this language. In contrast, voiced stops and fricatives are natural classes in terms of abstract phonological features in both languages; therefore, the UG hypothesis predicts both groups will generalize. Current results support the emergent hypothesis [French generalized: t(11) = 2.5, p<0.03; English did not: t(8) = 1.69, p>0.1], but additional data are being collected.
Automatic speech recognition technical for Náhuatl language isolated words.
Authors: Suarez Guerra Sergio , Oropeza Rodriguez Jose Luis , Flores Paulin Juan Carlos
Published: October 2010

Automatic speech recognition (ASR) represents a great expectative in communications systems where interaction between computers and humans is present. In the last three decades it has been grounded. The use of computer techniques helps people who speak in a native language or with physical limitation to make transactions, leave messages, obtain information, or control some device using voice expressions. This work shows results obtained when using different techniques of ASR into the Náhuatl language. Náhuatl language is a native language spoken in Mexico and is the most important because there are many people speaking it. Náhuatl language has around 49 variants. Preserving language is important because language is the most efficient means of transmitting a culture and it is the owners of that culture who lose the most when the language dies. Mel cepstrum coefficients, vector quantization, and hidden Markov models of continuous density were used. The experiments reported 99% accuracy in ASR on Náhuatl numerals corpus employed. This work represents the first step for implementing an automatic speech recognition translator into Spanish from the Náhuatl language, with closed sentences.
Voice analysis for a cochlear implant.
Authors: Becerra Garcia Jorge , Flores Martinez Joel , Ramirez Rangel Patricia Lorena
Published: October 2010

In the laboratory of acoustics in the ESIME, IPN, Mexico, a project for the design of a cochlear implant is under development. The most important part of this project is the processing of acoustical signals for the recognition of voice. The main objetive is an individual analysis of perception to every participating person in an appropriate manner when such signals reach the auditory system.
Time/frequency resolution of acoustic features for automatic speech recognition.
Authors: Zahorian Stephen A. , Hu Hongbing , Wu Jiang
Published: October 2010

The underlying assumption for spectral/temporal features for use in automatic speech recognition is that the frequency resolution should be emphasized in relation to temporal resolution. Accordingly, Mel frequency cepstral coefficients are typically computed using an approximately 25-ms frame length with a 10-ms frame spacing, and using 3–5 frames to represent temporal derivative information. In phone recognition experiments based on the TIMIT database using discrete cosine transform coefficients for spectral information and discrete cosine series coefficients for their temporal evolution, substantially higher phone accuracies were obtained with much shorter frame lengths (8 ms), much shorter frame spacings (2 ms), and much longer time intervals for capturing spectral tracks (on the order of 500 ms). Experimental results with various conditions are given for phone recognition using the TIMIT database. The implications of these results are that spectral/temporal evolution features, emphasizing the temporal aspects, are of great importance for automatic speech recognition.
Objective and subjective speech quality evaluation of wideband noise reduction algorithms.
Authors: Parsa Vijay , Pourmand Nazanin , Cowley Angela , Gupta Malay , Forrester Chris
Published: October 2010

Subjective and/or objective measurements of speech quality are important in benchmarking speech enhancement algorithms. Subjective measures include ratings of speech quality by listeners, whereas objective measures compute a metric based on the clean and enhanced speech samples. While subjective quality ratings are the “gold-standard,” they are also time- and resource-consuming. An objective metric that correlates highly with subjective data is attractive, as it can act as a substitute for benchmarking and fine-tuning the noise reduction algorithms. In this paper, the performance of several noise reduction algorithms for wideband (50–7000 Hz) telephony application was evaluated both subjectively and objectively. A custom wideband noise reduction database was created that contained speech samples corrupted by different background noises at different signal to noise ratios and processed by seven different noise reduction algorithms. Speech samples in the database were subsequently rated by a group of 32 listeners with normal hearing capabilities. Several objective metrics including log-likelihood ratio, weighted spectral slope, PESQ, and the loudness pattern distortion (LPD) measure based on the Moore–Glasberg auditory model were used to predict the subjective ratings. Results showed that the subjective ratings were highly reliable and the LPD metric correlated the best with subjective ratings of enhanced wideband speech.
Speech command recognition using a system based on the digital signal processor TMS320F2812 of Texas Instruments.
Authors: Herrera Abel , Hernandez Carlos
Published: October 2010

This paper describes the hardware design of a multi-purpose electronic system for speech command recognition based on the digital signal processor TMS320F2812 of Texas Instruments. In the paper, we utilize this system for speech command recognition. Specifically, we design and build an electronic system capable of recognizing speech commands. The main blocks are a power source, oscillator, adc, amplifier, jtag, reset boot mode, and monitor LED. The technique used for recognition is a multi section vector quantization, which is a well known technique in the field. This was selected over techniques such as discrete hidden Markov models (HMMs), Gaussian mixture models, or dynamic time warping. It is faster and equally precise compared to these techniques for the purpose of a very limited vocabulary, although we are aware that several types of HMMs are the techniques more widely used until now for continuous speech recognition systems. Preliminary tests for five words give us a rate of 96% in recognition accuracy for 10 repetitions for training and 10 for testing (one subject, under a 20 dB of signal-to-noise ratio). In these months, the system will be tested for more words and subjects. Also, it will be tested in real situations.
Head-related transfer function compression using Legendre polynomials and autoregressive model.
Authors: Shekarchi Sayedali , Hallam John , Christensen-Dalsgaard Jakob
Published: October 2010

Head-related transfer functions (HRTFs) are generally large datasets and thus a constraint for real-time applications. We propose a method to reduce redundancy and compress the datasets. In this method, HRTFs are transformed into time domain head-related impulse responses (HRIRs) and compressed by conversion into autoregressive (AR) filters. The AR coefficients are calculated using Prony’s method and the order is determined using the minimum eigenvalue of the HRIR covariance matrix. Such filters are specified by a few coefficients which can generate the full HRIRs. Next, Legendre polynomials (LPs) are used to compress the AR filter coefficients. LPs are derived on the sphere and form an orthonormal basis set for spherical functions. (Higher-order LPs capture increasingly fine spatial detail; the number of LPs needed to represent an HRTF, therefore, is indicative of its spatial complexity.) The results indicate that compression ratios can exceed 95% while maintaining an error of less than 10% in the recovered HRTFs. [Work funded by the EU framework program 7 under Contract No. IST 215370.]
Double-talk detector based on audio watermarking.
Authors: Czyzewski Andrzej , Ciarkowski Andrzej
Published: October 2010

The double-talk detection (DTD) algorithm is a vital part of an acoustic echo cancellation system commonly used in the telecommunication applications. The role of the DTD algorithm is to stop the adaptation process of the adaptive filter used for the estimation of echo path response in the presence of double-talk condition. The typically used DTD solutions are often a compromise between high efficiency and affordable computational cost. The algorithm introduced by the authors is based on a novel principle related to the so-called semi-fragile watermarking, making it possible to achieve satisfactory performance with limited processing burden. A hidden signature is embedded into the arriving far-end speaker signal just before it is replayed to the near-end speaker. Consequently, the presence of that signature is detected in the signal recorded by the near-end speaker microphone. The appropriate choice of the employed watermarking technique causes the signature to become undetectable during the near-end speaker’s talkspurts, precisely identifying the double-talk periods. The use of echo-hiding watermarking method allows fulfilling the algorithm requirements. The results of performed experiments prove the good performance of the proposed DTD solution. [Research funded by the Polish Ministry of Science and Higher Education within Grant No. PBZ-MNiSW-02-II-2007.]
A model for under-ice-target acoustic scattering.
Authors: Bishop Garner C.
Published: October 2010

A model is described that has been developed to calculate the scatter of a high-frequency acoustic pulse that originates from a stationary source and is incident on and is scattered from the under-ice surface characteristic of the Arctic and from a stationary target. The under-ice surface is modeled by an ensemble of ice keels, and ice keels are modeled by an ensemble of ice blocks with rectangular facets. The Helmholtz–Kirchhoff integral in the Kirchhoff approximation is used to calculate the scatter from an ice facet. The scatter from an ice keel is given by a coherent sum of the scatter from all the ice facets. A T-matrix is used to calculate the scatter from the stationary target. The first order terms in the multiple scattering process between the target and the ice surface are also calculated, i.e., the scattering from the target of the free field scatter from ice facets and the scatter from ice facets of the free field scatter of the target. Numerical results show the extent to which the scatter from the target in the under-ice environment differs from its free field scatter.
Convolving the impulse response of ocean reverberation.
Authors: Weinberg Henry , Keenan Ruth E
Published: October 2010

The comprehensive acoustic system simulation, a range-dependent standard model for predicting ocean reverberation, has the ability to simulate complex pulses. However, the method is inefficient if the acoustic system is aimed at numerous directions for various pulse shapes. A recent capability uses the standard approach to determine a numerical impulse response as a function of time and azimuth. The impulse response is then convolved with a complex pulse in order to obtain the actual response. Standard and convolved responses agree for the simple test cases that were tried. The standard approach is faster for one look direction and pulse shape. Convolution is an order of magnitude faster for realistic applications. At first glance, the impulse response appeared to be a significant advantage to the standard approach, but theoretical issues have not been resolved to the authors’ satisfaction. Some methods, including the one described here, convolve acoustic pressure squared. Others convolve acoustic pressure. References tend to quote basic convolution theory or require a significant background in signal processing. The former is probably an oversimplification. The latter is significantly more complicated than standard convolution.
An airborne monochromatic acoustic method to measure the hydraulic characteristics of shallow water flows.
Authors: Nichols Andrew , Horoshenkov Kirill V. , Tait Simon J. , Maximov German
Published: October 2010

Discussed is a new acoustic method to measure the dynamic surface characteristics of open channel flows and hence infer various flow parameters. Hydraulic laboratory tests suggest a relationship between the surface shape and the flow’s internal structure and hydraulic boundary conditions. This work investigates acoustic scattering of a monochromatic ultrasonic wave by a dynamically rough air-water interface and analyzes the effects of the water surface pattern on the recorded directivity of the scattered acoustic intensity field and its relation to the spatial correlation function for the dynamically rough interface. Temporal analysis of the acoustic intensity scattered by a dynamically rough interface is also used to relate the first and second statistical moments to the fluctuations in the amplitude and phase of the reflected acoustic field. This acoustic measurement method would ultimately allow for remote monitoring of real channel flows and evaluation of flow parameters such as flow rate, depth, turbulence scale, and channel bed topography.
An airborne broadband acoustic method to measure the characteristics of a dynamically rough, air-water interface.
Authors: Nichols Andrew , Attenborough Keith , Taherzadeh Shahram , Shepherd Simon J.
Published: October 2010

An acoustic method to measure the characteristics of a dynamically rough interface is presented. The theoretical foundation for this method has been adapted from outdoor sound propagation research whereby the boundary roughness is represented by an effective admittance. This model is used to calculate the excess attenuation of a broadband pulse emitted at near grazing incidence by a point source over a dynamically rough air-water interface. The excess attenuation is compared to that measured with an array of microphones. The effects of the water surface pattern on the recorded excess attenuation spectrum and its relation to the spatial correlation function for the dynamically rough interface are then analyzed. A Newton–Raphson root finding method is used to determine the acoustic admittance from the measured excess attenuation data in the frequency range between 0 and 20 kHz. The relationship between the mean roughness height, its standard deviation, and the real and imaginary parts of the admittance are obtained.
Simultaneous Bayesian localization of multiple sources in an uncertain ocean environment.
Authors: Dosso Stan E. , Wilmut Michael J.
Published: October 2010

This paper considers simultaneous localization of multiple acoustic sources when properties of the ocean environment (water column and seabed) are poorly known. A Bayesian formulation is developed in which the environmental parameters, noise variance, and locations and complex strengths (amplitudes and phases) of multiple sources are considered unknown random variables constrained by acoustic data and prior information. Two approaches are considered for estimating source locations and strengths. The first approach, referred to as focalization, maximizes the posterior probability density (PPD) over all parameters using an adaptive hybrid optimization. The second approach, referred to as marginalization, integrates the PPD to produce marginal probability distributions for source positions and strengths, which quantify localization uncertainties. In this approach, 2-D Gibbs sampling is applied to source ranges and depths, and Metropolis–Hastings sampling is applied in principal-component space for environmental parameters. In both approaches, closed-form maximum-likelihood expressions for source strengths and noise variance allow these parameters to be sampled implicitly rather than explicitly, reducing the dimensionality of the inversion. Examples are presented of both approaches applied to single- and multi-frequency localization of multiple sources in an uncertain shallow-water environment.
Automated estimation of marine engine rpm using underwater acoustic sensors.
Authors: Wilson Gary R. , Barlett Martin L.
Published: October 2010

The effect of the underwater noise made by boats and ships on marine mammals is a current topic of research. The source of the noise is primarily the engine, and the intensity and spectrum of the noise are directly related to the rpm of the engine. For systems that monitor and record boat noise, it is useful to be able to estimate the rpm of the engine in order to correlate it with the noise characteristics. In this paper a processing method will be presented that automatically estimates engine rpm from the narrowband characteristics of the boat noise. This method was applied to recordings of a number of different boats in different environments. The accuracy of the results will be demonstrated by comparing the estimates with independent measurements of the engine rpm using instrumented test boats. General trends in broadband spectral intensity versus rpm will be shown for a wide variety of boats.
Developing a matched-filter kernel for localization of complex sources in a dispersive ocean waveguide with array invariance.
Authors: Chauhan Hari , Gong Zheng , Tran Duong D. , Makris Nicholas , Ratilal Purnima
Published: October 2010

The array invariant method was previously developed by Lee and Makris [J. Acoust. Soc. Am. (2006)] for instantaneously localizing both transient and continuous broadband acoustic sources after conventional plane-wave beamforming and matched filtering array measurements in an ocean waveguide. The approach has been demonstrated with experimental data for a source transmitting linear frequency modulated pulses and theoretically for multiple uncorrelated broadband noise sources. Spectral analyzes of field-recorded data acquired during the ocean acoustic waveguide remote sensing Gulf of Maine 2006 Experiment show that acoustic signatures from many random sources, such as marine mammal vocalizations, are complex and exhibit irregular patterns that vary over time. Furthermore, the signals embedded in noise may also contain interfering sources correlated with the signal over some time duration. Here, we extend the array invariant method for range and bearing estimation of such complex sources. A robust matched filter kernel based on the acoustic properties of field-recorded whale vocalizations is developed to maximize the signal-to-noise ratio. Examples that simulate the beamformed and matched filtered array measurements of complex whale vocalizations embedded in noise in a random ocean waveguide are provided to demonstrate that range and bearing estimation using array invariance shows good agreement with true whale location.
Pyramids and basements.
Authors: Beristain Sergio
Published: October 2010

Among pre-Columbian buildings there are a series of sound effects which resemble some particular sounds that can be related to religion or political affairs. Ancient pyramids were employed by priests and politicians as a high basement in order to make his voice heard in the distance, addressing their people, during the religious ceremonies and sometimes by the government or for social events or to promote a war or a defensive action; this was done when public address systems were not available, so it was a good medium to communicate with large audiences. Sometimes directly, and sometimes taking advantage of the stone walls behind the speaker to reinforce his speech level, and of course always without grassing incidence to avoid sound level reduction.
The Northern Group Plaza in Palenque: An archaeoacoustic study.
Authors: Zalaquett Francisca
Published: October 2010

This investigation presents and proposes a methodology for archaeoacoustic analysis of public rituals of plazas in the Mayan area, which includes detailed studies of frequencies on musical instruments, white and pink sound emanations and their reverberations in the plazas, and laboratory studies on the stucco that covered the temples. As a result of the Northern Group study, we got rooms that could be used as scenarios where a musician or a speaker stood and this place allowed the frequency’s increase, causing an effect of “horn” in the sound and quality, so they were able to deliver messages at a greater distance, and therefore more people. On the plaza, we establish areas of musical activity and areas where they could locate the spectators with great acoustics and visibility of representations expressed. In addition we found that in this case, the design and construction of the structures by the Maya of Palenque involved a great knowledge of the sounds and behavior.