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Auditory scene analysis (ASA) is the process through which the auditory system makes sense of complex acoustic environments by organising sound mixtures into meaningful events and streams. Although music psychology has acknowledged the fundamental role of ASA in shaping music perception, no efficient test to quantify listeners' ASA abilities in realistic musical scenarios has yet been published. This study presents a new tool for testing ASA abilities in the context of music, suitable for both normal-hearing (NH) and hearing-impaired (HI) individuals: the adaptive Musical Scene Analysis (MSA) test. The test uses a simple 'yes–no' task paradigm to determine whether the sound from a single target instrument is heard in a mixture of popular music. During the online calibration phase, 525 NH and 131 HI listeners were recruited. The level ratio between the target instrument and the mixture, choice of target instrument, and number of instruments in the mixture were found to be important factors affecting item difficulty, whereas the influence of the stereo width (induced by inter-aural level differences) only had a minor effect. Based on a Bayesian logistic mixed-effects model, an adaptive version of the MSA test was developed. In a subsequent validation experiment with 74 listeners (20 HI), MSA scores showed acceptable test–retest reliability and moderate correlations with other music-related tests, pure-tone-average audiograms, age, musical sophistication, and working memory capacities. The MSA test is a user-friendly and efficient open-source tool for evaluating musical ASA abilities and is suitable for profiling the effects of hearing impairment on music perception. [ABSTRACT FROM AUTHOR]

Introduction: Auditory scene analysis refers to the system through which the auditory system distinguishes distinct auditory events and sources to create meaningful auditory information. The exact number of directly perceived auditory stimuli is unknown, studies suggest it may range from 3 to 5. This number differs among individuals, and potentially may indirectly indicate the ability to store and process the complex information, related to the memory load, which is combined with human cognitive processes. Aim: This study aims to further identify and quantify the number of sounds that can be perceived simultaneously in a complex auditory environment. Material and methods: Participants were presented with structured acoustic recordings and were asked to identify the exact number of targeted stimuli heard throughout the test. The experiment was designed to assess the auditory load and determine the maximum number of auditory stimuli that a healthy human can perceive at once. Results: Our study showed that on average, participants could identify up to three sounds at once with accuracy of responses declining progressively for four sounds or more. Conclusions: This study aimed to investigate the human capacity to detect and identify multiple sound signals simultaneously in a noisy environment. By understanding this ability, we sought to assess cognitive reserve in individuals. Our objective was to determine if auditory load could serve as a diagnostic tool for cognitive evaluation. We believe that further research will establish the validity of this approach, and we anticipate that it is only a matter of time before it becomes a viable method for assessing cognitive function. [ABSTRACT FROM AUTHOR]

The cocktail party problem refers to people’s auditory scene analysis in the complex auditory scene of everyday life in acoustics, which has also been addressed in modern literature. In this article, I define the literary cocktail party as a literary soundscape characterised by noise, with the dynamics of attention motivating its sound events and overall material relationality. Taking Virginia Woolf’s Between the Acts as an example, this article analyses how the interaction of words and sounds works with the dynamics of passive and active attention – two types of attention working in the bottom-up and top-down neural processes of the brain respectively. In different stages of the literary cocktail party, the development of sound events incorporates sonic materiality into the understanding of literary form, promotes the literary effect of defamiliarisation and creates relations in the soundscape. In the end, the article proposes ‘dip listening’ as a productive listening method for modern literary soundscape that is capable of not only generating an embodied sense of epiphany in daily experience but also creates the betweenness among various agencies in a shared tiredness. [ABSTRACT FROM AUTHOR]

Purpose: One of the most obvious functional effects of aging on the cognitive and processing processes of spatial hearing is the localization problem and the disorder of speech perception in noise. The purpose of the present study is to investigate the performance of dynamic spatial auditory processing in the elderly. Methods: This descriptive and analytical study was conducted on 60 young participants aged from18 to 25 years old and 60 elderly participants aged from 60 to 75 old years, using speech, spatial, and qualities of hearing scale (SSQ) questionnaire, binaural masking level difference (BMLD) and dynamic quick speech in noise (DS-QSIN) tests. Results: Comparing the average scores of the tests and the questionnaire using the independent t test showed a significant difference between the two groups (p < 0.001). It was also found that gender had no effect on the results (p > 0.05). Conclusions: Aging is accompanied by different structural and functional changes in the auditory central nervous system, which leads to a decrease in speech perception in challenging listening environments, as well as a decrease in sound localization abilities, due to the reduction of temporal and spectral information. This problem affects the determination of the source of sound and the spatial cognition of the elderly and leads to a disturbance in the awareness of the auditory environment. Therefore, auditory rehabilitation programs can cause the improvement of spatial auditory processing performance and improve speech perception in noise in the elderly. [ABSTRACT FROM AUTHOR]

In this paper, we propose an online adaptation method for Fourier series-based acoustic transfer function (FS-ATF) models for robot audition systems using microphone array signal processing. The ATF represents the characteristics of signal propagation from a sound source to a microphone, which is essential for sound source localization (SSL) and sound source separation (SSS). For practical applications in dynamically changing real-world environments, ATF models must satisfy two criteria: (1) they must be adaptable to changes in the acoustic environment; and (2) they must be lightweight to be suitable for resource-constrained systems, such as robots with limited memory and computational capacity. The proposed method addresses these challenges using Fourier series expansions for interpolation, which reduces the memory footprint of the ATF model and facilitates online adaptation to acoustic environmental changes. The experimental results demonstrate that the proposed online adaptation method both improves the SSL and SSS performance while reducing the size of the ATF model, which represents a significant improvement over existing online ATF adaptation methods. [ABSTRACT FROM AUTHOR]

In recent years, the relation between Sound Event Detection (SED) and Source Separation (SSep) has received a growing interest, in particular, with the aim to enhance the performance of SED by leveraging the synergies between both tasks. In this paper, we present a detailed description of JSS (Joint Source Separation and Sound Event Detection), our joint-training scheme for SSep and SED, and we measure its performance in the DCASE Challenge for SED in domestic environments. Our experiments demonstrate that JSS can improve SED performance, in terms of Polyphonic Sound Detection Score (PSDS), even without additional training data. Additionally, we conduct a thorough analysis of JSS's effectiveness across different event classes and in scenarios with severe event overlap, where it is expected to yield further improvements. Furthermore, we introduce an objective measure to assess the diversity of event predictions across the estimated sources, shedding light on how different training strategies impact the separation of sound events. Finally, we provide graphical examples of the Source Separation and Sound Event Detection steps, aiming to facilitate the interpretation of the JSS methods. [ABSTRACT FROM AUTHOR]

In recent years, embedded system technologies and products for sensor networks and wearable devices used for monitoring people's activities and health have become the focus of the global IT industry. In order to enhance the speech recognition capabilities of wearable devices, this article discusses the implementation of audio positioning and enhancement in embedded systems using embedded algorithms for direction detection and mixed source separation. The two algorithms are implemented using different embedded systems: direction detection developed using TI TMS320C6713 DSK and mixed source separation developed using Raspberry Pi 2. For mixed source separation, in the first experiment, the average signal-to-interference ratio (SIR) at 1 m and 2 m distances was 16.72 and 15.76, respectively. In the second experiment, when evaluated using speech recognition, the algorithm improved speech recognition accuracy to 95%. [ABSTRACT FROM AUTHOR]

Sound source separation (SSS) is a fundamental problem in audio signal processing, aiming to recover individual audio sources from a given mixture. A promising approach is multichannel non-negative matrix factorization (MNMF), which employs a Gaussian probabilistic model encoding both magnitude correlations and phase differences between channels through spatial covariance matrices (SCM). In this work, we present a dedicated hardware architecture implemented on field programmable gate arrays (FPGAs) for efficient SSS using MNMF-based techniques. A novel decorrelation constraint is presented to facilitate the factorization of the SCM signal model, tailored to the challenges of multichannel source separation. The performance of this FPGA-based approach is comprehensively evaluated, taking advantage of the flexibility and computational capabilities of FPGAs to create an efficient real-time source separation framework. Our experimental results demonstrate consistent, high-quality results in terms of sound separation. [ABSTRACT FROM AUTHOR]

Selective attention is a crucial ability of the auditory system. Computationally, following an auditory object can be illustrated as tracking its acoustic properties, e.g., pitch, timbre, or location in space. The difficulty is related to the fact that in a complex auditory scene, the information about the tracked object is not available in a clean form. The more cluttered the sound mixture, the more time and frequency regions where the object of interest is masked by other sound sources. How does the auditory system recognize and follow acoustic objects based on this fragmentary information? Numerous studies highlight the crucial role of top-down processing in this task. Having in mind both auditory modeling and signal processing applications, we investigated how computational methods with and without top-down processing deal with increasing sparsity of the auditory features in the task of estimating instantaneous voice states, defined as a combination of three parameters: fundamental frequency F0 and formant frequencies F1 and F2. We found that the benefit from top-down processing grows with increasing sparseness of the auditory data. [ABSTRACT FROM AUTHOR]

This document provides technical details about a dataset of binaural room impulse responses (BRIRs) that were measured in a listening room. The dataset captures the spatial dependency of BRIRs on listener positions and orientations, making it useful for various applications such as BRIR modeling, interpolation, crosstalk cancellation, and personal sound zones. The data collection process involved playing back signals from loudspeakers and recording the signals received with binaural microphones. The processed BRIRs were saved in a specific file format and visualized to examine various parameters. The authors acknowledge contributions and declare their funding source. [Extracted from the article]

The task of identification of sound events in a particular surrounding is known as Sound Event Detection (SED) or Acoustic Event Detection (AED). The occurrence of sound events is unstructured and also displays wide variations in both temporal structure and frequency content. Sound events may be non-overlapped (monophonic) or overlapped (polyphonic) in nature. In real-time scenarios, polyphonic SED is most commonly seen as compared to monophonic SED. In this paper, a Mel-Pseudo Constant Q-Transform (MP-CQT) technique is introduced to perform polyphonic SED to effectively learn both monophonic and polyphonic sound events. A pseudo CQT technique is adapted to extract features from the audio files and their Mel spectrograms. The Mel-scale is believed to broadly simulate human perception system. The classifier used is a Convolutional Recurrent Neural Network (CRNN). Comparison of the performance of the proposed MP-CQT technique along with CRNN is presented and a considerable performance improvement is observed. The proposed method achieved an average error rate of 0.684 and average F1 score of 52.3%. The proposed approach is also analyzed for the robustness by adding an additional noise at different Signal to Noise Ratios (SNRs) to the audio files. The proposed method for SED task has displayed improved performance as compared to state-of-the-art SED systems. The introduction of new feature extraction technique has shown promising improvement in the performance of the polyphonic SED system. [ABSTRACT FROM AUTHOR]

The construction of major marine infrastructure projects and the exploration and development of deep-sea mineral resources require fine imaging of seabed strata and structures. The high-resolution marine seismic exploration based on a high broadband sparker source is an important approach to reveal seabed stratum and reservoir structure, and identify geohazard. To optimize the performance of sparker seismic source, we investigated the electro-acoustic characteristics of spark discharge under conditions of different charging voltages and electrode numbers. Results show that the sound source level increased with the increase of the charging voltage, whereas the main frequency decreased when the charging voltage increases. In addition, it was found that the charging capacitance had more obvious influence on the main frequency than the sound source level did. Although the load energy decreased with increasing electrode number, the sound source level still increased but the main frequency decreased. Meanwhile, the primary to bubble (P/B) ratio increased with the increase of the electrode number. To gain a deeper insight into the electro-acoustic characteristics, we investigate the relationship between sound source level and power peak, from which a good correlation was observed. A more practical statistical analysis on the rise rate of current was processed, and a perfect logarithmic function was derived. Furthermore, we found that the main frequency was most possibly subjected to the electrical energy, especially the charging energy per electrode. The results indicate that the charging energy per electrode less than 10 J could increase the main frequency to above 300 Hz. At last, the main frequency could be reduced to 20 Hz when the charging energy of a single-electrode discharge was enhanced to over 4 kJ. This study shall be helpful in developing a sparker seismic source and improving the performance for marine engineering exploration and geohazard assessment. [ABSTRACT FROM AUTHOR]

This document is an editorial published in Frontiers in Human Neuroscience titled "Early development of sound processing in the service of speech and music perception." The editorial discusses the importance of understanding how infants process sound, particularly speech and music, as it provides insights into the neural and processing prerequisites of speech and music perception. The document summarizes empirical studies and reviews in four main research areas: higher-order auditory functions supporting speech and music processing, rhythm, speech processing in infants, and predicting language developmental outcomes. The authors highlight the progress being made in the field of studying the roots and early development of speech and music perception. [Extracted from the article]

Human listeners possess an innate capacity to discern patterns within rapidly unfolding sensory input. Core questions, guiding ongoing research, focus on the mechanisms through which these representations are acquired and whether the brain prioritizes or suppresses predictable sensory signals. Previous work, using fast auditory sequences (tone-pips presented at a rate of 20 Hz), revealed sustained response effects that appear to track the dynamic predictability of the sequence. Here, we extend the investigation to slower sequences (4 Hz), permitting the isolation of responses to individual tones. Stimuli were 50 ms tone-pips, ordered into random (RND) and regular (REG; a repeating pattern of 10 frequencies) sequences; Two timing profiles were created: in "fast" sequences, tone-pips were presented in direct succession (20 Hz); in "slow" sequences, tone-pips were separated by a 200 ms silent gap (4 Hz). Naive participants (N = 22; both sexes) passively listened to these sequences, while brain responses were recorded using magnetoencephalography (MEG). Results unveiled a heightened magnitude of sustained brain responses in REG when compared to RND patterns. This manifested from three tones after the onset of the pattern repetition, even in the context of slower sequences characterized by extended pattern durations (2,500 ms). This observation underscores the remarkable implicit sensitivity of the auditory brain to acoustic regularities. Importantly, brain responses evoked by single tones exhibited the opposite pattern--stronger responses to tones in RND than REG sequences. The demonstration of simultaneous but opposing sustained and evoked response effects reveals concurrent processes that shape the representation of unfolding auditory patterns. [ABSTRACT FROM AUTHOR]

Most auditory environments contain multiple sound waves that are mixed before reaching the ears. In such situations, listeners must disentangle individual sounds from the mixture, performing the auditory scene analysis. Analyzing complex auditory scenes relies on listeners ability to segregate acoustic events into different streams, and to selectively attend to the stream of interest. Both segregation and selective attention are known to be challenging for adults with normal hearing, and seem to be even more difficult for children. Here, we review the recent literature on the development of auditory scene analysis, presenting behavioral and neurophysiological results. In short, cognitive and neural mechanisms supporting stream segregation are functional from birth but keep developing until adolescence. Similarly, from 6 months of age, infants can orient their attention toward a target in the presence of distractors. However, selective auditory attention in the presence of interfering streams only reaches maturity in late childhood at the earliest. Methodological limitations are discussed, and a new paradigm is proposed to clarify the relationship between auditory scene analysis and speech perception in noise throughout development. [ABSTRACT FROM AUTHOR]

This article provides an overview of the challenges and limitations faced in the past when conducting sign language research. The author emphasizes the physical labor involved in searching for resources, note-taking, and videotaping. They also share their personal experiences working with deaf individuals and highlight the importance of sign language. The article acknowledges the lack of understanding and appreciation for sign language at the time and emphasizes the need for further research in this field. The author reflects on their own journey in sign language research, from encountering challenges and knowledge gaps during their clinical fellowship year to pursuing a PhD at McGill University. They conducted experiments to investigate the comprehension of ASL and LSQ, explored the role of facial expression in ASL comprehension, and examined the question of a critical period for sign language development and processing. The author's research has contributed to a deeper understanding of sign language and its significance in the study of language. [Extracted from the article]

Objectives: Cochlear implant (CI) users exhibit large variability in understanding speech in noise. Past work in CI users found that spectral and temporal resolution correlates with speech-in-noise ability, but a large portion of variance remains unexplained. Recent work on normal-hearing listeners showed that the ability to group temporally and spectrally coherent tones in a complex auditory scene predicts speech-in-noise ability independently of the audiogram, highlighting a central mechanism for auditory scene analysis that contributes to speech-in-noise. The current study examined whether the auditory grouping ability also contributes to speech-in-noise understanding in CI users. Design: Forty-seven post-lingually deafened CI users were tested with psychophysical measures of spectral and temporal resolution, a stochastic figure-ground task that depends on the detection of a figure by grouping multiple fixed frequency elements against a random background, and a sentence-in-noise measure. Multiple linear regression was used to predict sentence-in-noise performance from the other tasks. Results: No co-linearity was found between any predictor variables. All three predictors (spectral and temporal resolution plus the figure-ground task) exhibited significant contribution in the multiple linear regression model, indicating that the auditory grouping ability in a complex auditory scene explains a further proportion of variance in CI users' speech-in-noise performance that was not explained by spectral and temporal resolution. Conclusion: Measures of cross-frequency grouping reflect an auditory cognitive mechanism that determines speech-in-noise understanding independently of cochlear function. Such measures are easily implemented clinically as predictors of CI success and suggest potential strategies for rehabilitation based on training with non-speech stimuli. [ABSTRACT FROM AUTHOR]

Listeners often operate in complex acoustic environments, consisting of many concurrent sounds. Accurately encoding and maintaining such auditory objects in short‐term memory is crucial for communication and scene analysis. Yet, the neural underpinnings of successful auditory short‐term memory (ASTM) performance are currently not well understood. To elucidate this issue, we presented a novel, challenging auditory delayed match‐to‐sample task while recording MEG. Human participants listened to 'scenes' comprising three concurrent tone pip streams. The task was to indicate, after a delay, whether a probe stream was present in the just‐heard scene. We present three key findings: First, behavioural performance revealed faster responses in correct versus incorrect trials as well as in 'probe present' versus 'probe absent' trials, consistent with ASTM search. Second, successful compared with unsuccessful ASTM performance was associated with a significant enhancement of event‐related fields and oscillatory activity in the theta, alpha and beta frequency ranges. This extends previous findings of an overall increase of persistent activity during short‐term memory performance. Third, using distributed source modelling, we found these effects to be confined mostly to sensory areas during encoding, presumably related to ASTM contents per se. Parietal and frontal sources then became relevant during the maintenance stage, indicating that effective STM operation also relies on ongoing inhibitory processes suppressing task‐irrelevant information. In summary, our results deliver a detailed account of the neural patterns that differentiate successful from unsuccessful ASTM performance in the context of a complex, multi‐object auditory scene. [ABSTRACT FROM AUTHOR]

The Cocktail party problem, which is tracing and identifying a specific speaker's speech while numerous speakers communicate concurrently is one of the crucial problems still to be addressed for automated speech recognition (ASR) and speaker recognition. In this study, we attempt to thoroughly explore traditional methods for speech separation in a cocktail party environment and further analyze traditional single-channel methods for example source-driven methods like Computational Auditory Scene Analysis (CASA), data-driven methods like non-negative matrix factorization (NMF), model-driven methods, customary multi-channel methods such as beamforming, blind source separation for multi-channel and the newly developed deep learning approaches such as meta-learning based methods, self-supervised learning. This paper further accentuates numerous datasets and evaluation metrics in the domain of speech processing & brings out the comparison between traditional methods and methods based on deep learning for speech separation. This study provides a basic understanding and comprehensive knowledge of state-of-the-art researches in the area of speech separation and serves as a brief overview to new researchers. [ABSTRACT FROM AUTHOR]

In perceptual psychology, audition and introspection have not yet received as much attention as other topics (e.g., vision) and methods (third-person paradigms). Practical examples and theoretical considerations show that it nevertheless seems promising to treat both topics in conjunction to gain insights into basic structures of attention regulation and respective agentive awareness. To this end, an empirical study on voluntary auditory change was conducted with a non-reactive first-person design. Data were analyzed with a mixed methods approach and compared with an analogous study on visual reversal. Qualitative hierarchical coding and explorative statistics yield a cross-modal replication of frequency patterns of mental activity as well as significant differences between the modalities. On this basis, the role of mental agency in perception is refined in terms of different levels of intention and discussed in the context of the philosophical mental action debate as well as of the Global Workspace/Working Memory account. As a main result, this work suggests the existence and structure of a gradual and developable agentive attention awareness on which voluntary attention regulation can build, and which justifies speaking, in a certain sense, of attentional self-perception. [ABSTRACT FROM AUTHOR]

The article studies the acoustic characteristics of Russian speech under conditions of multitalker babble noise and manifestation of the Lombard effect in the context of auditory analysis of a complex scene. We have compared the spectral characteristics of stressed vowels [a], [u], [i] in words uttered by six women in completely quiet conditions and against a background of diotically presented multitalker babble noise at a level of 60 dB, imitating polyphony. In noise conditions, versus quiet, we obtained an increase in voice fundamental frequency (F0) and the first formant (F1) for all identified vowels. No overall pattern in changes in the second formant (F2) was found. When the vowel [i] was spoken in noise conditions, F2 decreased in all speakers; when vowels [u] and [a] were pronounced, it could have both decreased and increased. Thus, in general, the nature of the revealed changes in the spectral characteristics of the vowels of Russian speech in noise conditions corresponded to the features of Lombard speech for a number of European and Asian languages. For the first time, an inversely proportional relationship between F0 of a speaker in quiet conditions and its changes in noise conditions: the higher F0 in quiet conditions, the less its increase on a noise background. The revealed spectral changes reflect the processes of adaptive articulation correction aimed at highlighting the speaker's voice and increasing the intelligibility of his speech against the background of multitalker babble noise. [ABSTRACT FROM AUTHOR]

This article provides a comprehensive overview of how the ears and brain work together to process sound. It explains that when sound enters our ears, it is converted into electrical signals that the brain can interpret. The brain uses nerve cells to separate and recognize different sounds, determining their loudness and direction. The article discusses the different parts of the ear involved in this process and highlights the role of the brain in decoding and understanding the electrical signals. It also emphasizes that our understanding of sound is influenced by our prior knowledge and what we see. The article further explores how the brain processes and separates different sounds, using mechanisms like the medial superior olives to determine location and separate sounds. It concludes by emphasizing the importance of understanding how we hear for effective treatment of hearing problems and the development of technologies like mp3 files and artificial speech recognition. The authors, who include experts in auditory neuroscience and a children's author, provide their perspectives and contact information for further inquiries. [Extracted from the article]

The human auditory system employs a number of principles to facilitate the selection of perceptually separated streams from a complex sound mixture. The brain leverages multi-scale redundant representations of the input and uses memory (or priors) to guide the selection of a target sound from the input mixture. Moreover, feedback mechanisms refine the memory constructs resulting in further improvement of selectivity of a particular sound object amidst dynamic backgrounds. The present study proposes a unified end-to-end computational framework that mimics these principles for sound source separation applied to both speech and music mixtures. While the problems of speech enhancement and music separation have often been tackled separately due to constraints and specificities of each signal domain, the current work posits that common principles for sound source separation are domain-agnostic. In the proposed scheme, parallel and hierarchical convolutional paths map input mixtures onto redundant but distributed higher-dimensional subspaces and utilize the concept of temporal coherence to gate the selection of embeddings belonging to a target stream abstracted in memory. These explicit memories are further refined through self-feedback from incoming observations in order to improve the system's selectivity when faced with unknown backgrounds. The model yields stable outcomes of source separation for both speech and music mixtures and demonstrates benefits of explicit memory as a powerful representation of priors that guide information selection from complex inputs. [ABSTRACT FROM AUTHOR]

This study proposes a novel off-screen sound separation method based on audio-visual pre-training. In the field of audio-visual analysis, researchers have leveraged visual information for audio manipulation tasks, such as sound source separation. Although such audio manipulation tasks are based on correspondences between audio and video, these correspondences are not always established. Specifically, sounds coming from outside a screen have no audio-visual correspondences and thus interfere with conventional audio-visual learning. The proposed method separates such off-screen sounds based on their arrival directions using binaural audio, which provides us with three-dimensional sensation. Furthermore, we propose a new pre-training method that can consider the off-screen space and use the obtained representation to improve off-screen sound separation. Consequently, the proposed method can separate off-screen sounds irrespective of the direction from which they arrive. We conducted our evaluation using generated video data to circumvent the problem of difficulty in collecting ground truth for off-screen sounds. We confirmed the effectiveness of our methods through off-screen sound detection and separation tasks. [ABSTRACT FROM AUTHOR]

Consider that the sound dipole source in the flow field is composed of multiple micro-spherical oscillating sources. An aerodynamic sound source identification method is established by the relationship among the oscillating source, the radiated sound pressure, and the pressure gradient of flow in the near-wall flow field, and the formula for calculating the sound power of the sound dipole source in unsteady flow is derived. It shows that the power of sound dipole sources is proportional to the square of the oscillating force or pressure gradient. The combination of the formula and CFD method is further applied to the flow around the cylinder, which clearly presents the sound power and location characteristics of sound dipole sources. Further, the relationship between the sound source and the flow separation, or flow vortex shedding, is analyzed. The corresponding correlation analysis is also carried out, which indicates that the sound dipole source exists in a finite area of the attached wall. The front end of the area is at the separation point along the circumferential direction of the wall, and the end is at the location where the separation vortex completely falls off and a trailing vortex begins to form. In addition, the thickness of the area exists along the radial direction and gradually increases backward. [ABSTRACT FROM AUTHOR]

Adaptation is an essential feature of auditory neurons, which reduces their responses to unchanging and recurring sounds and allows their response properties to be matched to the constantly changing statistics of sounds that reach the ears. As a consequence, processing in the auditory system highlights novel or unpredictable sounds and produces an efficient representation of the vast range of sounds that animals can perceive by continually adjusting the sensitivity and, to a lesser extent, the tuning properties of neurons to the most commonly encountered stimulus values. Together with attentional modulation, adaptation to sound statistics also helps to generate neural representations of sound that are tolerant to background noise and therefore plays a vital role in auditory scene analysis. In this review, we consider the diverse forms of adaptation that are found in the auditory system in terms of the processing levels at which they arise, the underlying neural mechanisms, and their impact on neural coding and perception. We also ask what the dynamics of adaptation, which can occur over multiple timescales, reveal about the statistical properties of the environment. Finally, we examine how adaptation to sound statistics is influenced by learning and experience and changes as a result of aging and hearing loss. [ABSTRACT FROM AUTHOR]

TWO EXPERIMENTS INVESTIGATED PERCEPTUAL AND emotional consequences of note articulation in music by examining the degree to which participants perceived notes to be separated from each other in a musical phrase. Seven-note piano melodies were synthesized with staccato notes (short decay) or legato notes (gradual/sustained decay). Experiment 1 (n = 64) addressed the impact of articulation on perceived melodic cohesion and perceived emotion expressed through melodies. Participants rated melodic cohesion and perceived emotions conveyed by 32 legato and 32 staccato melodies. Legato melodies were rated more cohesive than staccato melodies and perceived as emotionally calmer and sadder than staccato melodies. Staccato melodies were perceived as having greater tension and energy. Experiment 2 (n = 60) addressed whether articulation is associated with humor and fear in music, and whether the impact of articulation depends on major vs. minor mode. For both modes, legato melodies were scarier than staccato melodies, whereas staccato melodies were more amusing and surprising. The effect of articulation on perceived happiness and sadness was dependent on mode: staccato enhanced perceived happiness for minor melodies; legato enhanced perceived sadness for minor melodies. Findings are discussed in relation to theories of music processing, with implications for music composition, performance, and pedagogy. [ABSTRACT FROM AUTHOR]

Most auditory evoked potential (AEP) studies in echolocating toothed whales measure neural responses to outgoing clicks and returning echoes using short-latency auditory brainstem responses (ABRs) arising a few ms after acoustic stimuli. However, little is known about longer-latency cortical AEPs despite their relevance for understanding echo processing and auditory stream segregation. Here, we used a non-invasive AEP setup with low click repetition rates on a trained harbor porpoise to test the long-standing hypothesis that echo information from distant targets is completely processed before the next click is emitted. We reject this hypothesis by finding reliable click-related AEP peaks with latencies of 90 and 160 ms, which are longer than 99% of click intervals used by echolocating porpoises, demonstrating that some higher-order echo processing continues well after the next click emission even during slow clicking. We propose that some of the echo information, such as range to evasive prey, is used to guide vocal-motor responses within 50–100 ms, but that information used for discrimination and auditory scene analysis is processed more slowly, integrating information over many click-echo pairs. We conclude by showing theoretically that the identified long-latency AEPs may enable hearing sensitivity measurements at frequencies ten times lower than current ABR methods. [ABSTRACT FROM AUTHOR]

Auditory stream segregation and informational masking were investigated in brain-lesioned individuals, age-matched controls with no neurological disease, and young college-age students. A psychophysical paradigm known as rhythmic masking release (RMR) was used to examine the ability of participants to identify a change in the rhythmic sequence of 20-ms Gaussian noise bursts presented through headphones and filtered through generalized head-related transfer functions to produce the percept of an externalized auditory image (i.e., a 3D virtual reality sound). The target rhythm was temporally interleaved with a masker sequence comprising similar noise bursts in a manner that resulted in a uniform sequence with no information remaining about the target rhythm when the target and masker were presented from the same location (an impossible task). Spatially separating the target and masker sequences allowed participants to determine if there was a change in the target rhythm midway during its presentation. RMR thresholds were defined as the minimum spatial separation between target and masker sequences that resulted in 70.7% correct-performance level in a single-interval 2-alternative forced-choice adaptive tracking procedure. The main findings were (1) significantly higher RMR thresholds for individuals with brain lesions (especially those with damage to parietal areas) and (2) a left–right spatial asymmetry in performance for lesion (but not control) participants. These findings contribute to a better understanding of spatiotemporal relations in informational masking and the neural bases of auditory scene analysis. [ABSTRACT FROM AUTHOR]

Listeners use the spatial location or change in spatial location of coherent acoustic cues to aid in auditory object formation. From stimulus-evoked onset responses in normal-hearing listeners using electroencephalography (EEG), we have previously shown measurable tuning to stimuli changing location in quiet, revealing a potential window into the cortical representations of auditory scene analysis. These earlier studies used non-fluctuating, spectrally narrow stimuli, so it was still unknown whether previous observations would translate to speech stimuli, and whether responses would be preserved for stimuli in the presence of background maskers. To examine the effects that selective auditory attention and interferers have on object formation, we measured cortical responses to speech changing location in the free field with and without background babble (+6 dB SNR) during both passive and active conditions. Active conditions required listeners to respond to the onset of the speech stream when it occurred at a new location, explicitly indicating 'yes' or 'no' to whether the stimulus occurred at a block-specific location either 30 degrees to the left or right of midline. In the aggregate, results show similar evoked responses to speech stimuli changing location in quiet compared to babble background. However, the effect of the two background environments diverges somewhat when considering the magnitude and direction of the location change and where the subject was attending. In quiet, attention to the right hemifield appeared to evoke a stronger response than attention to the left hemifield when speech shifted in the rightward direction. No such difference was found in babble conditions. Therefore, consistent with challenges associated with cocktail party listening, directed spatial attention could be compromised in the presence of stimulus noise and likely leads to poorer use of spatial cues in auditory streaming. [ABSTRACT FROM AUTHOR]

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Albert Feng was a pioneer in the field of auditory neuroethology who used frogs to investigate the neural basis of spectral and temporal processing and directional hearing. Among his many contributions was connecting neural mechanisms for sound pattern recognition and localization to the problems of auditory masking that frogs encounter when communicating in noisy, real-world environments. Feng's neurophysiological studies of auditory processing foreshadowed and inspired subsequent behavioral investigations of auditory masking in frogs. For frogs, vocal communication frequently occurs in breeding choruses, where males form dense aggregations and produce loud species-specific advertisement calls to attract potential mates and repel competitive rivals. In this review, we aim to highlight how Feng's research advanced our understanding of how frogs cope with noise. We structure our narrative around three themes woven throughout Feng's research—spectral, temporal, and directional processing—to illustrate how frogs can mitigate problems of auditory masking by exploiting frequency separation between signals and noise, temporal fluctuations in noise amplitude, and spatial separation between signals and noise. We conclude by proposing future research that would build on Feng's considerable legacy to advance our understanding of hearing and sound communication in frogs and other vertebrates. [ABSTRACT FROM AUTHOR]

Current study is focused on human event-related potentials (ERPs) elicited in the auditory spatial masking paradigm. ERPs of the test subjects (n = 18) were recorded in the passive conditions using uncorrelated low-frequency noise signals. Spatial positions of the stimuli were determined by interaural level differences (ILDs). Test signal and masker were colocated or separated by 90 or 180 deg of azimuth and were presented simultaneously. The onset of a test signal was delayed from masker onset in order to separate the onset responses. We estimated the ERPs dependence on angular distance between signal and masker. When the test signal was presented against the background of a masker, the N1 and P2 components decreased in magnitude and increased in latency, as compared to presentation in silence. Spatial release from masking resulted in ERP magnitude increase and latency decrease as the angular distance between test signal and masker changed from 0 to 90 deg. A further spatial separation of the lateralized signal and masker did not result in further release from masking. ERPs magnitude was a more reliable indicator of the angular separation of signals than its latency. These findings suggest that binaural unmasking mechanisms operate at the preconscious stage of complex auditory scene analysis. [ABSTRACT FROM AUTHOR]

In this article, we deal with the problem of increasing the accuracy of the automatic determination of the coordinates of the sound source location. We propose a new algorithm for the identification of the sound source's position coordinates based on a system of three equations of the second order describing the dynamics of acoustic wavefront propagation. The implementation of the algorithm is carried out by a distributed automated system, which includes autonomous sensor-receivers located in the field and connected to the server of this system via wireless communication channels. Sensor-receivers are placed at the vertices of a flat, symmetrical figure with 4 axes of symmetry of the second order (square). The proposed algorithm takes into account the change in the phase speed of the sound wave when the temperature, air humidity, wind direction and speed change and allows for the determination of the coordinates of the position of the sound source with an error of no more than 1%. The experiment with real input data was carried out in a simulated environment, which was created on the Node.js platform. [ABSTRACT FROM AUTHOR]

Sequential auditory scene analysis (ASA) is often studied using sequences of two alternating tones, such as ABAB or ABA_, with "_" denoting a silent gap, and "A" and "B" sine tones differing in frequency (nominally low and high). Many studies implicitly assume that the specific arrangement (ABAB vs ABA_, as well as low-high-low vs high-low-high within ABA_) plays a negligible role, such that decisions about the tone pattern can be governed by other considerations. To explicitly test this assumption, a systematic comparison of different tone patterns for two-tone sequences was performed in three different experiments. Participants were asked to report whether they perceived the sequences as originating from a single sound source (integrated) or from two interleaved sources (segregated). Results indicate that core findings of sequential ASA, such as an effect of frequency separation on the proportion of integrated and segregated percepts, are similar across the different patterns during prolonged listening. However, at sequence onset, the integrated percept was more likely to be reported by the participants in ABA_low-high-low than in ABA_high-low-high sequences. This asymmetry is important for models of sequential ASA, since the formation of percepts at onset is an integral part of understanding how auditory interpretations build up. [ABSTRACT FROM AUTHOR]