Independent Sound Sources Research Articles

ISEE: A Wearable Image-Sound Translation System for Partially Sighted People

As a growing number of blinds, many blind aid systems which use the way of converting image to sound or vibration have been designed. However, phonetic prompt and vibration are very limited in information expression. Therefore, a system called Image-Sound Eye-substitution Equipment (ISEE) is designed in this paper. This system consists of binocular camera, Raspberry Pi, and stereo headsets. It can convert image to sound by mapping each pixel into an independent sound source based on sensory substitution technology and enable blinds to browse pictures, appreciate scenery, and avoid obstacles for its high-density information expression. Its core of data processing design is the Sensory Substitution Technology-based Image-to-Sound Transform (SST-IST) algorithm. The SST-IST mainly consists of 3 steps, i.e., the image reshape, image-to-sound conversion matrix generation, and image-to-sound transmission. First, a single-frequency pitch of pixel is generated, followed by wave interception and localization to obtain independent sound clips to express pixel information. Second, the sound chips based on Head Related Transfer Functions (HRTF) are stereo processed. Third, a series of single-frequency sound chips are accumulated with weights to generate a complete sound. Extensive experiments show that our SST-IST can restore image information well and the sound restore rate can achieve 73.0%. Also, it is universally adaptive and robust to the changes in ambient light and noises within a certain range. The auditory discrimination experiments show that, with 4-5 hours training, the users can recognize basic graphics by using ISEE system.

An improved functional beamforming algorithm for far-field multi-sound source localization based on Hilbert curve

In view of the mismatch problem between the sound sources and searching grid points (non-ideal case), an improved functional beamforming (FBF) algorithm based on the Hilbert curve, so-called HFBF, for far-field localization of multi-sound sources is presented in this paper. In contrast with the traditional FBF, the HFBF may generate the searching grids on a sound source plane and optimize them by reducing the searching plane size, adjusting the function order and the focusing vector direction, so as to improve the efficiency of grid searching and the estimation accuracy of sound source parameters. A computational procedure of the HFBF is proposed and applied to identify two independent sound sources. In terms of the locating error and the estimated error of sound pressure level (SPL), the performances of four algorithms are compared by simulations and experiments. The results suggest that the accuracies of the location coordinates and the estimated SPL values are significantly improved by the Hilbert-based grid searching. The conclusions can be drawn that, for multi-source localization, the HFBF can somewhat break through the Rayleigh limit. If the exponent value is selected properly, the estimated SPL error of the HFBF can be less than 0.1 dB. The HFBF has a larger range of available exponent values, which is negatively correlated with the frequency of a sound source. Generally speaking, in the non-ideal cases, the proposed HFBF is more accurate and effective than the FBF. It can be applied to a large-scale space, such as aircraft, train, traffic environment, etc., for far-field localization of multi-sound sources in engineering.

Single-channel speech separation method based on attention mechanism

In order to improve the efficiency of speech interaction so that both humans and machines can hear it clearly, to solve the problem of single channel speech separation, a singlechannel speech separation method combined with attention mechanism is proposed. Based on the time-domain speech separation network, the method uses Encoder-separator-decoder framework and improves the Separator structure of the framework. First, the encoder transforms the input mixed speech signal into the latent space and extracts the feature of the speech signal to obtain the deep representation of the speech. Then, the deep attention features of speech were extracted by the splitter combined with the attention mechanism, and the mask was estimated for each independent sound source. Finally, the decoder inversely transforms the separated source signals back to the time domain. Experiments on WSJ0 speech separation data set show that the proposed method can effectively improve the SI-SNR of mixed speech separation, which is 0.31dB higher than the original time-domain network, and 0.94dB higher for the mixed case of boys and girls.

Acoustic Hologram Enhanced Phased Arrays for Ultrasonic Particle Manipulation

By combining phased-array technology with recent developments in acoustic holograms, the authors enhance the capabilities of ultrasonic particle manipulation while reducing the cost. A 3D-printed hologram enables them to create high-fidelity force fields and manipulate small objects along complex paths. Attached to this hologram is an array of independent sound sources; controlling the output delay (phase) of these sources allows movement of the force field in space, yielding noncontact ``tweezers'' that can trap particles of almost any shape. Applications include 3D-bioprinting of cells to create designer tissues, as well as holding microorganisms in place for careful biological study.

Effect of Dual-Carrier Processing on the Intelligibility of Concurrent Vocoded Sentences.

The goal of this study was to examine the role of carrier cues in sound source segregation and the possibility to enhance the intelligibility of 2 sentences presented simultaneously. Dual-carrier (DC) processing (Apoux, Youngdahl, Yoho, & Healy, 2015) was used to introduce synthetic carrier cues in vocoded speech. Listeners with normal hearing heard sentences processed either with a DC or with a traditional single-carrier (SC) vocoder. One group was asked to repeat both sentences in a sentence pair (Experiment 1). The other group was asked to repeat only 1 sentence of the pair and was provided additional segregation cues involving onset asynchrony (Experiment 2). Both experiments showed that not only is the "target" sentence more intelligible in DC compared with SC, but the "background" sentence intelligibility is equally enhanced. The participants did not benefit from the additional segregation cues. The data showed a clear benefit of using a distinct carrier to convey each sentence (i.e., DC processing). Accordingly, the poor speech intelligibility in noise typically observed with SC-vocoded speech may be partly attributed to the envelope of independent sound sources sharing the same carrier. Moreover, this work suggests that noise reduction may not be the only viable option to improve speech intelligibility in noise for users of cochlear implants. Alternative approaches aimed at enhancing sound source segregation such as DC processing may help to improve speech intelligibility while preserving and enhancing the background.

Efficient spatialization techniques for immersive real-time soundscapes

A simulated soundscape requires many independent sound sources to create an immersive experience. However, the challenges of rendering many simultaneous sources can force a choice between an interactive real-time simulation with limited sources and a static, pre-rendered soundscape with more sources. We will discuss production techniques for optimizing soundscape auralizations to maximize the number of concurrent sources which can be rendered in real time to achieve an immersive level of detail with the potential for a high level of interactivity. Topics covered with include source prioritization, benefits and limitations of various spatialization techniques, including ambisonics, VBAP, and MIAP, and hybrid techniques for independent rendering of direct and reflected sound.

Performance of active feedforward control systems in non-ideal, synthesized diffuse sound fields.

The acoustic performance of passive or active panel structures is usually tested in sound transmission loss facilities. A reverberant sending room, equipped with one or a number of independent sound sources, is used to generate a diffuse sound field excitation which acts as a disturbance source on the structure under investigation. The spatial correlation and coherence of such a synthesized non-ideal diffuse-sound-field excitation, however, might deviate significantly from the ideal case. This has consequences for the operation of an active feedforward control system which heavily relies on the acquisition of coherent disturbance source information. This work, therefore, evaluates the spatial correlation and coherence of ideal and non-ideal diffuse sound fields and considers the implications on the performance of a feedforward control system. The system under consideration is an aircraft-typical double panel system, equipped with an active sidewall panel (lining), which is realized in a transmission loss facility. Experimental results for different numbers of sound sources in the reverberation room are compared to simulation results of a comparable generic double panel system excited by an ideal diffuse sound field. It is shown that the number of statistically independent noise sources acting on the primary structure of the double panel system depends not only on the type of diffuse sound field but also on the sample lengths of the processed signals. The experimental results show that the number of reference sensors required for a defined control performance exhibits an inverse relationship to control filter length.

On the statistical distributions of active intensity directions

Pressure and pressure differences measured using closely-spaced microphone pairs enable the calculation of active intensity directions. These directions are exploited in various applications such as acoustical analysis and modelling, multichannel audio systems, source localization and real-time blind source separation. Despite their widespread usage, the distribution of the active intensity directions in the presence of interference has not been investigated before, except for the diffuse sound field case for which the distribution becomes uniform. In this paper, the interference problem has been formulated using the pressure and pressure differences. Algebra of random variables, fractional linear transformations and Brownian motion have been used for the exposition of the model distributions. It is shown that the distribution is wrapped Cauchy for a single sound source in the presence of independent interference in an anechoic chamber. In a direct plus diffuse sound field, the distribution is offset Cauchy for a nonstationary target source and von Mises for the special case of a stationary target source, related to the Brownian motion. In the presence of multiple independent sound sources, the distribution can be modeled as mixtures of circular Cauchy distributions and a uniform distribution.

Extending Space in Varèse's Interpolations: Extended Stereo Space

EXTENDING SPACE IN VARÈSE’S INTERPOLATIONS: EXTENDED STEREO SPACE JAIME E. OLIVER LA ROSA 1. INTRODUCTION HE ELECTRONIC INTERPOLATIONS for Déserts, were developed in two stages: first in 1954, at Schaeffer’s Studio D’Essai in Paris, and later in 1961, in the Columbia-Princeton Electronic Music Studio (CPEMC) in New York. Varèse arrived in Paris with sounds he had recorded in factories in New York and with the first interpolation completed and the second one incomplete. He first worked with Ann McMillan in New York, then with Pierre Henry in the Studio D’Essai, and Bülent Arel in the CPEMC. According to Risset, Varèse wanted to make a new version, but due to several complications, he only re-worked the Paris versions in 1961. It is unclear what was the nature of this re-working of the original T Extending Space in Varèse’s Interpolations 263 tapes. After Varèse’s death, Ussachevsky proposed to make a new version of the interpolations, but “Louise Varèse was not sure that Varèse would have approved of it” and he did not make this version.1 2. ETHICAL AND AESTHETIC CONSIDERATIONS There is an ethical issue to consider when approaching the spatialization of a historical work recorded on magnetic tape: Do we have the right to spatialize the work of a composer who is no longer present? Sound diffusion is regarded both as a compositional and performance practice. As a performance practice it is generally understood as the projection of sounds in a space.2 This practice is generally conceived as a “live” practice, where sound projection of a mono or stereo source is achieved manually with a mixer and a set of speakers. Some composers3 have suggested the use of a computer to diffuse sound, as computers provide us with the possibility of controlling sound diffusion with more precision than manual control. We know from his writings that Varèse had thought of sound in space beyond the technological possibilities of his time. Varèse was constantly demanding from technology, what technology couldn’t yet provide. It is clear from Varèse’s writing that sound projection and spatial concerns were a central metaphor in his work: . . . a sense of sound-projection in space by means of the emission of sound in any part or in many parts of the hall, as may be required by the score.4 I shall add a fourth [dimension of music], sound projection—that feeling that sound is leaving us with no hope of being reflected back, a feeling akin to that aroused by beams of light sent forth by a powerful searchlight—for the ear as for the eye, that sense of projection, of a journey into space.5 Furthermore, Varèse later said, “for the first time I heard my music projected into space,”6 referring to the premiere of Poéme électronique at the Phillips Pavilion, in which “three layers of sound material [were projected] through 350 loudspeakers.”7 Considering on one hand that sound diffusion as a performance practice allows a person to distribute a stereo or mono piece into a larger number of speakers, and, on the other, that Varèse clearly had the desire of projecting his music into space, multichannel versions of his work are valid and desirable explorations. 264 Perspectives of New Music 3. PROCEDURE TO EXTEND THE STEREO SPACE Like a lot of early electronic music, the interpolations were realized in two channels or stereo. The technique was so unusual at the time, that in order to achieve the stereo radio transmission, two radio stations were used. In the interpolations, sounds do not move in space by dynamically changing the relative amplitudes of the mono channels. Sounds are generally placed in one of the channels and not in the other one. It is, in any case, extremely clear from the tape that Varèse produced two differentiated sound streams intended to be perceived from independent sound sources. In order to respect this original realization, my approach looked to find a way of extending the original stereo version, therefore obtaining a technique that could also be extended to other...

Distribution pattern of finless porpoises at the junction of the Yangtze River and Poyang Lake observed by towed passive acoustic device

The Yangtze finless porpoise (Neophocaena asiaeorientalis asiaeorientalis) is an endangered freshwater porpoise subspecies unique to the Yangtze River basin. Without immediate conservation measures, it could soon become extinct, just as the Yangtze River dolphin (baiji, Lipotes vexillifer). We report seasonal change in the local distribution of the porpoises living in the conjunction area of the middle reaches of the Yangtze River, side streams, and appended Poyang Lake. A towed stereo acoustic data-logger, A-tag, was used to detect echolocation signals and sound source bearing angles. The independent sound source directions provided the number of animals present, not just the number of sounds. Passive acoustic surveys were performed regularly from May 2007 to August 2010. The water level was highest in August (summer) and lowest in February (winter) and at mid-level in May (spring) and November (autumn). The average number of porpoises detected in 11 surveys conducted in different seasons varied from 0.53 to 1.26 individuals per km. No significant trend of reducing number of porpoises was detected during 3-y monitoring. The distribution of the porpoises was seasonally site-specific. In May and August, the animals were detected more often at river junctions than in the lake, but vice versa from November to February.

Sexual dimorphism and bilateral asymmetry of syrinx and vocal tract in the European starling (Sturnus vulgaris)

Sexually dimorphic vocal behavior in zebra finches (Taeniopygia guttata) is associated with a 100% larger syrinx in males and other morphological adaptations of the sound source. The songbird syrinx consists of two independent sound sources, whose specialization for different spectral ranges may be reflected in morphological properties, but the morphology of labia and syringeal skeleton have not been investigated for lateralized specializations. Similarly, little is known whether the morphology of the songbird vocal tract reflects differences in vocal behavior. Here, we tested the hypothesis that different vocal behavior and specialization is reflected in the morphology. We investigated syringeal and upper vocal tract morphology of male and female European starlings (Sturnus vulgaris). Female starlings exhibit smaller vocal repertoires and sing at lower rates than males. In males, the left syrinx produces mostly low frequencies, while the right one is used for higher notes. Macroscopic and histological techniques were used to record nineteen measurements from the syrinx and the vocal tract which were tested for sexual differences in syrinx and vocal tract and for lateral asymmetry within the syrinx. Sexually dimorphic vocal behavior is reflected in the morphology of the starling syrinx. Males have a larger syrinx with the size difference attributable to increased muscle mass and three enlarged elements of the syringeal skeleton. The upper vocal tract, however, does not differ between males and females. Distinct lateralization was found in two elements of the syringeal skeleton of females, and the labia in the left syrinx are larger than those on the right in both sexes. The sexual dimorphism of the syringeal size is smaller in starlings (35%) than in zebra finches (100%), which is consistent with the different vocal behavior of females in both species. The morphological differences between the two sound sources are discussed in relation to their vocal performance.

Air-borne sound source characterization by patch impedance coupling approach

An approach of independent sound source characterization is discussed. The source is defined by its blocked sound pressure and surface impedance via a suitable enveloping surface. Both the blocked pressure and the impedance are made discrete using the patch averaging concept. The approach is adapted to numerical as well as experimental implementation. The characterization of a source by patches allows for acoustical sub-structuring, which in turn enables the prediction of the sound field created by a source coupled to an arbitrary environment. Numerical simulations are presented which demonstrate the validity of the approach.

Distinguishing between the interior pressures induced from two independent sources within a room using the probabilistic approach

In this paper, a probabilistic approach is introduced and used to distinguish between the interior pressures that are induced from two independent sound sources within a rectangular room. One source is a vibrating wall of the room and the other is an interior point source. The model is set up using the modal analysis method. In the probabilistic method, the probability densities of the uncertain acoustic model parameters are computed. The values of highest probability density are identified and assigned to the uncertain parameters in the acoustic model. The contributions of the sound sources to the total pressure are then obtained from the acoustic model input with the identified parameters. The effects of the modal truncations, the modeling errors, and the measurement locations on the accuracies of the identification process are studied.

Birdsong: The role of the vocal tract

The avian vocal organ, the syrinx, is unique among vertebrates in its morphology and location at or near the base of the trachea. In oscine songbirds, two independent sound sources share the same vocal tract. Despite a growing knowledge of the neural and syringeal mechanisms underlying song, the acoustic role of the suprasyringeal vocal tract in modulating birdsong is still poorly understood. During song, the beak opening, or gape, is typically positively correlated with the dominant frequency of the vocalization, but the mechanism by which beak gape influences vocal tract acoustics is controversial. X-ray cinematography of singing birds reveals song-related movements of the hyoid apparatus and larynx that cause the volume of the upper vocal tract to vary inversely with the songs fundamental frequency. At low fundamental frequencies, the oropharynx and cranial end of the esophagus expand to form a single large chamber, the volume of which is reduced at higher frequencies by collapse of the esophagus and reduction of the oropharyngeal cavity. A computational acoustic model indicates that birds actively tune the volume of their oropharyngeal-esophageal cavity so that its major resonance tracks the fundamental frequency of the song. [Work supported by NIH.]

Estimation of speech embedded in a reverberant and noisy environment by independent component analysis and wavelets

In this paper, we develop a system for enhancement of the speech signal with highest energy from a linear convolutive mixture of n statistically independent sound sources recorded by m microphones, where m<n. In this system we use the concept of independent component analysis (ICA) along with adaptive auditory filter banks and pitch tracking. Computer simulations and real-world experiments carried out in an actual room and measured through objective and subjective measures confirm the validity of the proposed algorithm.

Comparative study: Flow-induced noise on underwater acoustic pressure, particle velocity, and intensity sensors

A neutrally buoyant underwater acoustic p-u intensity probe contains discrete sensors for direct measurement of acoustic pressure and particle velocity [J. A. McConnell and G. C. Lauchle, J. Acoust. Soc. Am. 103, 2755(A) (1998)]. Intensity is calculated from the cross spectrum between these two quantities. Thus a single probe can be categorized as a compound sensor capable of independently measuring acoustic pressure, particle velocity, and intensity. Using this probe, a comparative study concerning the measurement of all three quantities in the presence of mean flow and an independent sound source was performed. Bias errors, which result from the flow-induced noise generated by towing the sensor (e.g., a cylindrical body) in cross flow at low speeds through a channel of water, are presented. Other parameters of interest include analysis of experimental pressure–pressure and pressure–velocity correlation functions. The effect of body geometry (e.g., variation in aspect ratio) on the measured bias errors and correlation functions is also presented. [Work supported by the Office of Naval Research, the Naval Submarine League, and the Acoustics Technology Branch of the Naval Air Warfare Center—Aircraft Division, via the DoD Small Business Innovation Research Program.]

5-2-5 matrix system

A matrix system encodes five discrete audio signals down to a two-channel stereo recording and decodes the recorded stereo signal into at least five stand alone, independent channels to allow placement of specific sounds at any one of 5 or more predetermined locations as individual, independent sound sources, thus producing a 5-2-5 matrix system. One embodiment of the system provides signals to left front, right front, center, left rear, and right rear speaker locations. The matrix system is compatible with all existing stereo materials and material encoded for use with other existing surround systems. Material specifically encoded for this system can be played back through any other existing decoding systems without producing undesirable results.

Adaptive separation of mixed broadband sound sources with delays by a beamforming Herault-Jutten network

The Herault-Jutten network has been used to separate independent sound sources that have been linearly mixed. The problem of separating a mixture of several independent signals in free-field conditions or a signal and echoes in confined spaces is compounded by propagation time delays between the source(s) and the microphones because the conventional Herault-Jutten network cannot tolerate time delays. In this paper, we combine a symmetrically balanced beamforming array with the conventional Herault-Jutten network. The resulting system can adaptively separate signals that include delays introduced by the propagation medium. The proposed algorithm has been simulated in digital communication multipath channels where intersymbol interference exists. The simulation results show two clear advantages of the proposed method over the conventional adaptive equalization: (1) there is no penalty for very long impulse responses caused by long delays, and (2) no training signals are needed for equalization. The design of a multibeamformer to handle the source separation of multiple broad-band signals is also presented. >

On the Ontogeny of Display Behaviour in the Black-Headed Gull: I. the Gradual Emergence of the Adult Forms

On the Ontogeny of Display Behaviour in the Black-Headed Gull: I. the Gradual Emergence of the Adult Forms

Bilateral interactions within the vocal control pathway of birds: two evolutionary alternatives.

AbstractBilateral interactions between the right and left sides of the vocal control pathway of three avian species, the budgerigar (Melopsittacus undulatus), the zebra finch (Poephila guttata), and the canary (Serinus can‐arius), were compared by using microstimulation. Stimulation of the neo‐striatal nucleus, hyperstriatum ventrale pars caudale (HVc), caused a constant‐latency increase in neural activity of the tracheosyringeal (ts) nerve. This nerve supplies the avian vocal organ, the syrinx.Unilateral stimulation of HVc in budgerigars caused equal responses in both right and left ts nerves, whereas in canaries and zebra finches, much larger responses with lower thresholds and shorter latencies were evoked in the ipsilateral ts nerve than in the contralateral ts nerve. Furthermore, contralateral responses in the oscines were seen only at high rates of stimulation, whereas ipsilateral responses could be evoked by a single stimulus pulse.Simultaneous bilateral stimulation of HVc in budgerigars evoked a greater than linear summation of responses on each ts nerve, evidence for convergence of ipsi‐ and contralateral inputs onto the same set of motor neurons. No such summation was detected in zebra finches or canaries.The bilateral distribution of excitation within the song control pathway of budgerigars should lead to coactivation of bilaterally paired muscles on each side of the syrinx. This type of interaction should ensure that the two sides of the parrot syrinx operate as parts of a unitary, or single‐voiced, syrinx. In contrast, the two sides of the song control pathway in canaries and zebra finches are almost completely isolated from each other, and are thus free to act independently. This type of organization lends itself to the separate control of simultaneously produced, harmonically unrelated sound elements by the two independent sound sources present in the songbird's two‐voiced syrinx.