Auditory Distances Research Articles

Capturing the dynamics of peripersonal space by integrating expectancy effects and sound propagation properties

BackgroundHumans perceive near space and far space differently. Peripersonal space (PPS), i.e. the space directly surrounding the body, is often studied using paradigms based on audiotactile integration. In these paradigms, reaction time (RT) to a tactile stimulus is measured in the presence of a concurrent auditory looming stimulus. New MethodWe propose here to refine the experimental procedure by disentangling behavioral contributions of the targeted audiotactile integration mechanisms from expectancy effects. To this aim, we added to the protocol a baseline with a fixed sound distance. Furthermore, in order to improve the relevance of the audiotactile integration measures, we took into account sound propagation properties and assessed RTs for logarithmically spaced auditory distances. ResultsExpectation contributed significantly to overall behavioral responses. Subtracting it isolated the audiotactile effect due to the stimulus proximity. This revealed that audiotactile integration effects have to be tested on a logarithmic scale of distances, and that they follow a linear variation on this scale. Comparison with Existing Method(s)The current method allows cleaner and more pertinent sampling measures for evaluating audiotactile integration phenomena linked to PPS. Furthermore, most of the existing methods propose a sigmoid fitting, which rests on the intuitive framework that PPS is an in-or-out zone. Our results suggest that behavioral effects follow a logarithmic decrease, thus a response graduated in space. ConclusionsThe proposed protocol design and method of analysis contribute to sharpen the experimental investigation of the factors influencing and modifying multisensory integration phenomena in the space surrounding the body.

Perception of Auditory Distance in Normal-Hearing and Moderate-to-Profound Hearing-Impaired Listeners.

The auditory system allows the estimation of the distance to sound-emitting objects using multiple spatial cues. In virtual acoustics over headphones, a prerequisite to render auditory distance impression is sound externalization, which denotes the perception of synthesized stimuli outside of the head. Prior studies have found that listeners with mild-to-moderate hearing loss are able to perceive auditory distance and are sensitive to externalization. However, this ability may be degraded by certain factors, such as non-linear amplification in hearing aids or the use of a remote wireless microphone. In this study, 10 normal-hearing and 20 moderate-to-profound hearing-impaired listeners were instructed to estimate the distance of stimuli processed with different methods yielding various perceived auditory distances in the vicinity of the listeners. Two different configurations of non-linear amplification were implemented, and a novel feature aiming to restore a sense of distance in wireless microphone systems was tested. The results showed that the hearing-impaired listeners, even those with a profound hearing loss, were able to discriminate nearby and far sounds that were equalized in level. Their perception of auditory distance was however more contracted than in normal-hearing listeners. Non-linear amplification was found to distort the original spatial cues, but no adverse effect on the ratings of auditory distance was evident. Finally, it was shown that the novel feature was successful in allowing the hearing-impaired participants to perceive externalized sounds with wireless microphone systems.

Auditory Sketches: Very Sparse Representations of Sounds Are Still Recognizable.

Sounds in our environment like voices, animal calls or musical instruments are easily recognized by human listeners. Understanding the key features underlying this robust sound recognition is an important question in auditory science. Here, we studied the recognition by human listeners of new classes of sounds: acoustic and auditory sketches, sounds that are severely impoverished but still recognizable. Starting from a time-frequency representation, a sketch is obtained by keeping only sparse elements of the original signal, here, by means of a simple peak-picking algorithm. Two time-frequency representations were compared: a biologically grounded one, the auditory spectrogram, which simulates peripheral auditory filtering, and a simple acoustic spectrogram, based on a Fourier transform. Three degrees of sparsity were also investigated. Listeners were asked to recognize the category to which a sketch sound belongs: singing voices, bird calls, musical instruments, and vehicle engine noises. Results showed that, with the exception of voice sounds, very sparse representations of sounds (10 features, or energy peaks, per second) could be recognized above chance. No clear differences could be observed between the acoustic and the auditory sketches. For the voice sounds, however, a completely different pattern of results emerged, with at-chance or even below-chance recognition performances, suggesting that the important features of the voice, whatever they are, were removed by the sketch process. Overall, these perceptual results were well correlated with a model of auditory distances, based on spectro-temporal excitation patterns (STEPs). This study confirms the potential of these new classes of sounds, acoustic and auditory sketches, to study sound recognition.

Investigation of auditory distance perception and preferences in concert halls by using virtual acoustics.

Virtual acoustics with multichannel sound reproduction was used to study auditory distance perception in four concert halls with multiple sound sources on stage. Eight subjects reported apparent auditory distances in five seating positions from 10 to 26 m to the middle of the sources on stage. The distance estimates were collected by absolute distance estimation procedure as well as a free modulus estimation procedure including both within and between halls evaluations. In addition, pairwise preferences were collected for two positions within each hall and for one position between halls. Results reveal that the perception of distance is dependent on the hall acoustics and show how the strength factor G and direct-to-reverberant energy ratio covary in relation to perceptual distances in these halls. The results also indicate that in such large spaces the overestimation of short distances may continue up to and further than 10 m from the sound sources. Preference results show that closer seats were liked more than further ones and that the strength of this preference is associated with the difference in perceptual distances.

Acoustic and auditory sketches: Recognition of severely simplified natural sounds by human listeners

Sounds in our environment like voices, animal calls, or musical instruments can be recognized on the basis of timbre alone. The objective of this study was to unravel the features underlying this robust sound recognition. We investigated how severely sounds can be simplified while still being recognizable. A large number of natural sounds were simplified into “auditory or acoustic sketches:” energy peaks were selected on the auditory or acoustic spectrogram, at three levels of simplification (high, medium, and low). The remaining non-sparse information was removed. Listeners were asked to classify the simplified sounds into their four original categories: instruments, birds, vehicles, or voices. We adapted a recent signal detection model to dissociate the perceptual sensitivity (d' scores) from the bias, for each category (4-AFC task). Overall, severely simplified sounds could still be recognized above chance by the listeners. Auditory distances, based on spectro-temporal excitation patterns (STEPs), were then computed. Participants' performances on the 4-AFC task were well correlated with the auditory distances between the four categories. Altogether, our results suggest that sound recognition is a very robust perceptual process, and that basic spectral and spectro-temporal differences between sounds, captured by the auditory distances, can account for this robust recognition.

New single-ended objective measure for non-intrusive speech quality evaluation

This article proposes a new output-based method for non-intrusive assessment of speech quality of voice communication systems and evaluates its performance. The method requires access to the processed (degraded) speech only, and is based on measuring perception-motivated objective auditory distances between the voiced parts of the output speech to appropriately matching references extracted from a pre-formulated codebook. The codebook is formed by optimally clustering a large number of parametric speech vectors extracted from a database of clean speech records. The auditory distances are then mapped into objective Mean Opinion listening quality scores. An efficient data-mining tool known as the self-organizing map (SOM) achieves the required clustering and mapping/reference matching processes. In order to obtain a perception-based, speaker-independent parametric representation of the speech, three domain transformation techniques have been investigated. The first technique is based on a perceptual linear prediction (PLP) model, the second utilises a bark spectrum (BS) analysis and the third utilises mel-frequency cepstrum coefficients (MFCC). Reported evaluation results show that the proposed method provides high correlation with subjective listening quality scores, yielding accuracy similar to that of the ITU-T P.563 while maintaining a relatively low computational complexity. Results also demonstrate that the method outperforms the PESQ in a number of distortion conditions, such as those of speech degraded by channel impairments.

Perceptual non‐intrusive speech quality assessment using a self‐organizing map

PurposeThis paper seeks to propose a new non‐intrusive method for the assessment of speech quality of voice communication systems and evaluate its performance.Design/methodology/approachThe method is based on measuring perception‐based objective auditory distances between the voiced parts of the output speech to appropriately matching references extracted from a pre‐formulated codebook. The codebook is formed by optimally clustering a large number of parametric speech vectors extracted from a database of clean speech records. The auditory distances are then mapped into equivalent subjective mean opinion scores (MOSs). The required clustering and matching processes are achieved by an efficient data‐mining tool known as the self‐organizing map (SOM). The proposed method was examined using a wide range of distortion including speech compression, wireless channel impairments, VoIP channel impairments, and modifications to the signal from features such as AGC.FindingsThe experimental results reported indicate that the proposed method provides a high level of accuracy in predicting the actual subjective quality of the speech. Specifically, the second version of the method, which is based on the use of bark spectrum (BS) analysis, is more accurate in predicting the MOS scores compared with its first and third versions (which are based on BS analysis and mel frequency cepstrum coefficients (MFCC), respectively), and outperforms the ITU‐T PESQ in a large number of test cases, particularly those related to distortion caused by channel impairments and signal level modifications.Research limitations/implicationsIt is believed that the prototype developed of the proposed objective speech quality measure is sufficiently accurate and robust against speaker, utterance and distortion type variations. Nevertheless, there are still possible directions for further improvements and enhancement. In general there are three areas that could be pursued for further improvements: widening the coverage of speaker variations of the system's codebook; formulating and using a perceptual speech model that provides true speaker‐independent representation of speech; and implementing the proposed measure as a stand‐alone system, preferably for real‐time applications.Practical implicationsBeing an output‐based method, the proposed method can be employed for monitoring and assessing telecommunications networks under both live traffic conditions and off‐line evaluation.Originality/valueThe main contribution of this paper is the introduction of a new output‐based, non‐intrusive method for the assessment of speech quality that is sufficiently accurate and robust. To the best of the author's knowledge, no reliable output‐based objective speech quality assessment method has to date been reported or formally recognised.

Song variation and environmental auditory masking in the grasshopper sparrow

Some grassland bird species, in particular grasshopper sparrows (Ammodramus savannarum), sing songs with especially high mean frequencies (7.0–8.0 kHz). Acoustic interference is one potential explanation for the evolution of high frequency vocalizations, particularly in open habitats. We tested predictions from a model of effective auditory communication distances to understand the potential effects of vocal production and environmental auditory masking on vocal behavior and territoriality. Variation in the spectral structure of songs and the size and shape of territories was measured for grasshopper sparrows in typical grassland habitats. Median territory areas were 1629 m2 at a site in the center of the species range in Nebraska, and 1466 m2 at our study site in Maryland, with average territory diameters measuring 20.2 m. Species densities and sound pressure levels also were determined for stridulating insects and other noise sources in the habitat. Based on current models of effective communication distances, known noise levels, and information on hearing abilities, our results suggest that auditory sensitivity and environmental noise could be factors influencing the mean frequency and spatial dynamics of territorial behavior in grassland birds. [Work supported by NIH and the CRFRC.]

The nature of vowel structures.

The Dispersion-Focalization Theory (DFT) attempts to predict vowel systems by using the competition between two perceptual costs: (i) dispersion based on inter-vowel distances, (ii) local focalization based on intra-vowel spectral salience related to formants proximity. The first cost is related to the global structure of the system and the second to the internal structure of each vowel element. The competition takes place in an auditory (formant-based) space, and it is controlled by two parameters, namely λ which sets the respective weight of F1 and higher formants in auditory distances, and α which sets the respective weights of the dispersion and focalization costs. We describe a new methodology for testing the DFT predictions: for a given number of vowels, the so-called “phase spaces” allow us to determine the DFT winner in the (λ, α) space. We present a refined analysis of the UPSID database inventory of vowel systems. From the comparison between experimental phase spaces and UPSID data we not only derive a (λ, α) region for which DFT predictions fit quite well with the phonological inventories and are compatible with preferred 3-to-7 vowels systems, but also what the possible variants in the systems are and in which order they can appear.

Weight of phonetic substance in the structure of sound inventories

In the research field initiated by Lindblom & Liljencrants in 1972, we illustrate the possibility of giving substance to phonology, predicting the structure of phonological systems with nonphonological principles, be they listener-oriented (perceptual contrast and stability) or speaker-oriented (articulatory contrast and economy). We proposed for vowel systems the Dispersion-Focalisation Theory (Schwartz et al., 1997b). With the DFT, we can predict vowel systems using two competing perceptual constraints weighted with two parameters, respectively λ and α. The first one aims at increasing auditory distances between vowel spectra (dispersion), the second one aims at increasing the perceptual salience of each spectrum through formant proximities (focalisation). We also introduced new variants based on research in physics - namely, phase space (λ,α) and polymorphism of a given phase, or superstructures in phonological organisations (Vallée et al., 1999) which allow us to generate 85.6% of 342 UPSID systems from 3- to 7-vowel qualities. No similar theory for consonants seems to exist yet. Therefore we present in detail a typology of consonants, and then suggest ways to explain plosive vs. fricative and voiceless vs. voiced consonants predominances by i) comparing them with language acquisition data at the babbling stage and looking at the capacity to acquire relatively different linguistic systems in relation with the main degrees of freedom of the articulators; ii) showing that the places “preferred” for each manner are at least partly conditioned by the morphological constraints that facilitate or complicate, make possible or impossible the needed articulatory gestures, e.g. the complexity of the articulatory control for voicing and the aerodynamics of fricatives. A rather strict coordination between the glottis and the oral constriction is needed to produce acceptable voiced fricatives (Mawass et al., 2000). We determine that the region where the combinations of Ag (glottal area) and Ac (constriction area) values results in a balance between the voice and noise components is indeed very narrow. We thus demonstrate that some of the main tendencies in the phonological vowel and consonant structures of the world’s languages can be explained partly by sensorimotor constraints, and argue that actually phonology can take part in a theory of Perception-for-Action-Control.

The Dispersion-Focalization Theory of vowel systems

The Dispersion-Focalization Theory (DFT) attempts to predict vowel systems based on the minimization of an energy function summing two perceptual components: global dispersion, which is based on inter-vowel distances; and local focalization, which is based on intra-vowel spectral salience related to the proximity of formants. The computation takes place in an auditory (formant-based) space, and is controlled by two parameters, λ,which sets the respective weight of F 1and higher formants in auditory distances, and α,which sets the respective weights of the dispersion and focalization components. In this study, we first sample the acoustic space with 33 “prototypes” associated with the major primary articulations for vowels. Then, for a given number of vowels, we define two different tools, “phase spaces” for dealing with optimal systems and “stability ” for characterizing sub-optimal ones. The corresponding predictions are compared with the main trends of vowel systems analyzed in a companion paper (Schwartz, Boë, Vallée & Abry, this issue). We then derive a region in the ( λ, α) space for which theory predictions fit quite well with phonological inventories, with a special concern for two problems, namely peripheral vowels and front rounded vowels. Finally we stress the difficulty of predicting some trends of vowel systems linked to the organization of vowels in series, which could constitute a limit to substance-based predictions of vowel inventories.