Masking Curves Research Articles

Loudness of double impulsive sounds and its relation to temporal masking curves

The loudness of double impulsive sounds is examined from the viewpoint of temporal masking in that why the loudness of double impulsive sound becomes louder with the shorter duration compared with longer duration. The stimuli are set of various rise/fall times and various durations with the energy inside the stimuli remained unchanged. And to compare the loudness, it is least with the equal rise/fall times that with the short rise and long fall time and long rise time and short fall time. This is also true for single impulsive sounds. And together with this, the loudness is louder with the duration of the stimuli in that the louder the loudness the shorter the stimuli. The results of temporal masking curves are shown for explaining the cause of this loudness variations why it becomes louder with the shorter durations.

Analysis of the application of college popular music education relying on the elite teaching optimization algorithm

ABSTRACT This paper presents a novel approach to enhancing college popular music education by incorporating an elite teaching optimization algorithm into a customized model. A key feature of the proposed model is the utilization of a high-fidelity wideband audio coding algorithm, which employs a time-frequency analysis module based on the MDCT fast algorithm to reduce storage requirements. The psychoacoustic analysis module takes advantage of the high frequency domain resolution of FFT and accounts for differences between MDCT and DFT by calculating masking curves separately. The proposed model is evaluated using the elite teaching optimization algorithm, and results demonstrate that it effectively improves the effectiveness of college popular music education.

The effect of stimulus level on excitation patterns of individual electrode contacts in cochlear implants

ObjectiveSpread of excitation (SOE) in cochlear implants (CI) is a measure linked to the specificity of the electrode-neuron interface. The SOE can be estimated objectively by electrically evoked compound action potential (eCAP) measurements, recorded with the forward-masking paradigm in CI recipients. The eCAP amplitude can be plotted as a function of the roving masker, resulting in a spatial forward masking (SFM) curve. The eCAP amplitudes presented in the SFM curves, however, reflect an interaction between a masker and probe stimulus, making the SFM curves less reliable for examining SOE effects at the level of individual electrode contacts. To counter this, our previously published deconvolution method estimates the SOE at the electrode level by deconvolving the SFM curves (Biesheuvel et al., 2016). The aim of this study was to investigate the effect of stimulus level on the SOE of individual electrode contacts by using SFM curves analyzed with our deconvolution method. DesignFollowing the deconvolution method, theoretical SFM curves were calculated by the convolution of parameterized excitation density profiles (EDP) attributable to masker and probe stimuli. These SFM curves were subsequently fitted to SFM curves from CI recipients by iteratively adjusting the EDPs. We first improved the EDP parameterization to account for stimulus-level effects and validated this updated parameterization by comparing the EDPs to simulated excitation density profiles (sEDP) from our computational model of the human cochlea. Secondly, we analyzed SFM curves recorded with varying probe stimulus level in 24 patients, all implanted with a HiFocus Mid-Scala electrode array. With the deconvolution method extended to account for stimulus level effects, the SFM curves measured with varying probe stimulus levels were converted into EDPs to elucidate the effects of stimulus level on the SOE. ResultsThe updated EDP parameterization was in good agreement with the sEDPs from the computational model. Using the extended deconvolution method, we found that higher stimulus levels caused significant widening of EDPs (p < 0.001). The stimulus level also affected the EDP amplitude (p < 0.001) and the center of excitation (p < 0.05). Concerning the raw SFM curves, an increase in current level led to higher SFM curve amplitudes (p < 0.001), while the width of the SFM curves did not change significantly (p = 0.62). ConclusionThe extended deconvolution method enabled us to study the effect of stimulus level on excitation areas in an objective way, as the EDP parameterization was in good agreement with sEDPs from our computational model. The analysis of SFM curves provided new insights into the effect of the stimulus level on SOE. We found that the EDPs, and therefore the SOE, mainly became wider when the stimulus level increased. Lastly, the comparison of the EDP parameterization with simulations in our computation model provided new insights about the validity of the deconvolution method.

A study on launch vehicle on board acoustic data compression

The field of data compression has been evolved over the last decades. In this way, several techniques to reduce the amount of acquired data from the sensor required to be transmitted have been developed. Those techniques are usually classified by lossless or lossy, where, for the lossless techniques, all acquired data is recovered, while the lossy techniques introduce errors to these data. Each of these techniques presents advantages and drawbacks, being the analyst responsible for choosing the appropriate technique for a specific application. This work presents a comparison study using Vorbis lossy audio format with a low pass filter and a custom psychoacoustic masking curve to reduce the error in compressing a vibro-acoustic signal from a launch vehicle.

Comparing nonredundant masking and filled-aperture kernel phase for exoplanet detection and characterization

The limitations of adaptive optics and coronagraph performance make exoplanet detection close to {\lambda}/D extremely difficult with conventional imaging methods. The technique of non-redundant masking (NRM), which turns a filled aperture into an interferometric array, has pushed the planet detection parameter space to within {\lambda}/D. For high Strehl, the related filled-aperture kernel phase technique can achieve resolution comparable to NRM, without the associated dramatic decrease in throughput. We present non-redundant masking and kernel phase contrast curves generated for ground- and space-based instruments. We use both real and simulated observations to assess the performance of each technique, and discuss their capabilities for different exoplanet science goals such as broadband detection and spectral characterization.

Investigating time-efficiency of forward masking paradigms for estimating basilar membrane input-output characteristics.

It is well known that pure-tone audiometry does not sufficiently describe individual hearing loss (HL) and that additional measures beyond pure-tone sensitivity might improve the diagnostics of hearing deficits. Specifically, forward masking experiments to estimate basilar-membrane (BM) input-output (I/O) function have been proposed. However, such measures are very time consuming. The present study investigated possible modifications of the temporal masking curve (TMC) paradigm to improve time and measurement efficiency. In experiment 1, estimates of knee point (KP) and compression ratio (CR) of individual BM I/Os were derived without considering the corresponding individual “off-frequency” TMC. While accurate estimation of KPs was possible, it is difficult to ensure that the tested dynamic range is sufficient. Therefore, in experiment 2, a TMC-based paradigm, referred to as the “gap method”, was tested. In contrast to the standard TMC paradigm, the maker level was kept fixed and the “gap threshold” was obtained, such that the masker just masks a low-level (12 dB sensation level) signal. It is argued that this modification allows for better control of the tested stimulus level range, which appears to be the main drawback of the conventional TMC method. The results from the present study were consistent with the literature when estimating KP levels, but showed some limitations regarding the estimation of the CR values. Perspectives and limitations of both approaches are discussed.

Characterizing impairments in compression and filter shape to establish their role in hidden hearing loss

In the study of hidden hearing loss, it is important to characterize an individual’s peripheral losses accurately while minimizing measurement time for clinical applications. In this talk, we show how peripheral losses involved in compression and auditory filtering can be estimated simultaneously and accurately using the compressive GammaChirp (cGC) filter and notched noise (NN) maskers. Sets of thresholds were gathered with asymmetric maskers and fitted with the cGC filter model using several constraints associated with cochlear filtering. The slopes of the resultant input-output functions were within the range established with temporal masking curves above 1000 Hz. Studies of the fitting process also revealed how to reduce the number of threshold measurements without unduly increasing estimation error. These improvements led to the extension of the cGC filter model to explain hearing impairment of cochlear origin. The model also led to a hearing impairment simulator which can analyze everyday sounds inc...

Real-time simulation of hearing impairment: Application to speech-in noise intelligibility

Accurately simulating cochlear hearing loss in real time is of paramount importance for various applications, including easier testing of the impact of cochlear damage on auditory perception in real-life situations, evaluating sound qualities as perceived by hearing-impaired listeners, and pedagogical demonstrations. Following an idea originally proposed by Irino and colleagues, a real-time hearing-loss simulator was developed. Using an inverse compressive gammarchirp filters analysis, the simulator aims to replicate the increase of hearing thresholds, the level-dependent widening of auditory filters, and modification of the loudness curve (recruitment). The characteristics of the simulator, and basic psychoacoustical measurements performed with and without the simulator (absolute thresholds, auditory filters estimated with notch-noise data, cochlear input/output functions estimated with temporal masking curves) will be presented. Finally, an experiment comparing the intelligibility of speech in noise for real and simulated hearing-impaired listeners with a French version of the Four Alternative Auditory Feature Test (FAAF) will be presented. On average, the results of normal-hearing listeners using the simulator were found to be similar to the results obtained in individuals with hearing loss of presumed cochlear origin, suggesting that the simulator is valid.

Penyandi Persepsi Isyarat Audio Berdasar pada Model Modulasi Frekuensi (FM)

Data compression or source coding has been playing an important role in many areas in daily life. There are several audio encoders available with various methods. Each audio encoder works at a certain range bitrate depending on the signal model used and the method implemented. Based on literature reviews it is to know that there are only a view of audio encoder working at bit rate below 16 kbit/s, especially for music signals. The togetherness in having harmonically components between the audio signals and frequency modulated signals is the basic idea of proposing the perceptual encoder based on FM signal model. In estimating and extracting FM parameters, the psychoacoustic model is used to exploit the perceptual nature of humans ears. The errors of FM modeling is distributed below the masking curve of psychoacoustic model. The developed encoder allows sending music signals on the GSM channel. The developed encoder has four cascaded main processes, namely sinusoid components extraction, harmonic components separation, FM parameters estimation, and FM parameter encoding. The simulation results show the proposed method is successful to encode some audio signal types.

Exploration of a physiologically-inspired hearing-aid algorithm using a computer model mimicking impaired hearing

Objective: To use a computer model of impaired hearing to explore the effects of a physiologically-inspired hearing-aid algorithm on a range of psychoacoustic measures. Design: A computer model of a hypothetical impaired listener’s hearing was constructed by adjusting parameters of a computer model of normal hearing. Absolute thresholds, estimates of compression, and frequency selectivity (summarized to a hearing profile) were assessed using this model with and without pre-processing the stimuli by a hearing-aid algorithm. The influence of different settings of the algorithm on the impaired profile was investigated. To validate the model predictions, the effect of the algorithm on hearing profiles of human impaired listeners was measured. Study sample: A computer model simulating impaired hearing (total absence of basilar membrane compression) was used, and three hearing-impaired listeners participated. Results: The hearing profiles of the model and the listeners showed substantial changes when the test stimuli were pre-processed by the hearing-aid algorithm. These changes consisted of lower absolute thresholds, steeper temporal masking curves, and sharper psychophysical tuning curves. Conclusion: The hearing-aid algorithm affected the impaired hearing profile of the model to approximate a normal hearing profile. Qualitatively similar results were found with the impaired listeners’ hearing profiles.

Frequency Tuning of the Efferent Effect on Cochlear Gain in Humans.

Cochlear gain reduction via efferent feedback from the medial olivocochlear bundle is frequency specific (Guinan, Curr Opin Otolaryngol Head Neck Surg 18:447-453, 2010). The present study with humans used the Fixed Duration Masking Curve psychoacoustical method (Yasin et al., J Acoust Soc Am 133:4145-4155, 2013a; Yasin et al., Basic aspects of hearing: physiology and perception, pp 39-46, 2013b; Yasin et al., J Neurosci 34:15319-15326, 2014) to estimate the frequency specificity of the efferent effect at the cochlear level. The combined duration of the masker-plus-signal stimulus was 25 ms, within the efferent onset delay of about 31-43 ms (James et al., Clin Otolaryngol 27:106-112, 2002). Masker level (4.0 or 1.8 kHz) at threshold was obtained for a 4-kHz signal in the absence or presence of an ipsilateral 60 dB SPL, 160-ms precursor (200-Hz bandwidth) centred at frequencies between 2.5 and 5.5 kHz. Efferent-mediated cochlear gain reduction was greatest for precursors with frequencies the same as, or close to that of, the signal (gain was reduced by about 20 dB), and least for precursors with frequencies well removed from that of the signal (gain remained at around 40 dB). The tuning of the efferent effect filter (tuning extending 0.5-0.7 octaves above and below the signal frequency) is within the range obtained in humans using otoacoustic emissions (Lilaonitkul and Guinan, J Assoc Res Otolaryngol 10:459-470, 2009; Zhao and Dhar, J Neurophysiol 108:25-30, 2012). The 10 dB bandwidth of the efferent-effect filter at 4000 Hz was about 1300 Hz (Q(10) of 3.1). The FDMC method can be used to provide an unbiased measure of the bandwidth of the efferent effect filter using ipsilateral efferent stimulation.

Effect of cueing on stability of behavioral measurements of basilar membrane responses with a precursor

Stimulation of the olivocochlear efferent system can reduce the gain of the basilar membrane (BM) response to a sound. This gain reduction may be accompanied by a reduction in BM compression, although behavioral data are inconclusive. In this study, the effect of efferent stimulation on compression is assessed using fixed-duration masking curves (FDMCs) with a precursor. In a pilot experiment, presenting a broadband noise precursor before the masker resulted in large within-listener variability for the shortest target durations, although a three-interval forced-choice task was used. To reduce possible confusion effects and increase the dynamic range of BM response measurement, a contra-lateral broadband cue signal, gated simultaneously with the precursor and masker, was added. FDMCs with a 150-ms precursor were measured with and without cue for target durations of 3, 9, and 15 ms. The total masker-target duration was fixed at 25 ms. Listeners were trained at least 2 h, then each condition was measured three times. In contrast to the literature, the presence of the cue neither reduced within-listener variability nor increased sensitivity. Alternative methods are thus required to improve the stability and dynamic range of behavioral measurements of BM responses with a precursor. [Work supported by the FWF project I-1362-N30.]

Perceptual and Bitrate-Scalable Coding of Haptic Surface Texture Signals

Applications involving indirect interpersonal communication, such as collaborative design/assembly/exploration of physical objects, can benefit strongly from the transmission of contact-based haptic media, in addition to the more traditional audiovisual media. Inclusion of haptic media has been shown to improve immersiveness, task performance, and the overall experience of task execution. While several decades of research have been dedicated to the acquisition, processing, coding, and display of audio and video streams, similar aspects for haptic streams have been addressed only recently. Simultaneous masking is a perceptual phenomenon widely exploited in the compression of audio data. In the first part of this paper, to the best of our knowledge, we present first-time empirical evidence for masking in the perception of wideband vibrotactile signals. Our results show that this phenomenon for haptics is very similar to its auditory analog. Signals closer in frequency to a powerful masker ( 25 dB above detection threshold) are masked more strongly (peak threshold-shifts of up to 28 dB) than those away from the masker (threshold-shifts of 15-20 dB). The masking curves approximately follow the masker's spectral profile. In the second part of this paper, we present a bitrate scalable haptic texture codec, which incorporates the masking model and describe its subjective and objective performance evaluation. Experiments show that we can drive down the codec output bitrate to a very low value of 2.3 kbps, without the subjects being able to reliable discriminate between the codec input and distorted output texture signals.

Time-efficient multidimensional threshold tracking method

Traditionally, adaptive methods have been used to reduce the time it takes to estimate psychoacoustic thresholds. However, even with adaptive methods, there are many cases where the testing time is too long to be clinically feasible, particularly when estimating thresholds as a function of another parameter, such as in temporal masking curves or when characterizing auditory filters. Here we present a new method, the “grid” method, which adaptively varies multiple parameters during each experimental run. By changing the way the parameter-response space is sampled, the method increases the proportion of experimental time spent in the vicinity of the sought-after threshold curve. The resulting increase in time-efficiency is substantial and can make some measurements clinically feasible. Thresholds from temporal masking curves obtained with the grid method are compared with those from one of the most time-efficient standard methods (single-interval up-down adaptive method of Lecluyse, 2013). Overall, individuals’ results from both methods are very highly correlated, but the grid method was an order of magnitude faster in estimating thresholds. The application of the grid method to other measurements, such as characterizing auditory filters, will also be discussed.

Forward-masking recovery and the assumptions of the temporal masking curve method of inferring cochlear compression.

The temporal masking curve (TMC) method is a behavioral technique for inferring human cochlear compression. The method relies on the assumptions that in the absence of compression, forward-masking recovery is independent of masker level and probe frequency. The present study aimed at testing the validity of these assumptions. Masking recovery was investigated for eight listeners with sensorineural hearing loss carefully selected to have absent or nearly absent distortion product otoacoustic emissions. It is assumed that for these listeners basilar membrane responses are linear, hence that masking recovery is independent of basilar membrane compression. TMCs for probe frequencies of 0.5, 1, 2, 4, and 6 kHz were available for these listeners from a previous study. The dataset included TMCs for masker frequencies equal to the probe frequencies plus reference TMCs measured using a high-frequency probe and a low, off-frequency masker. All of the TMCs were fitted using linear regression, and the resulting slope and intercept values were taken as indicative of masking recovery and masker level, respectively. Results for on-frequency TMCs suggest that forward-masking recovery is generally independent of probe frequency and of masker level and hence that it would be reasonable to use a reference TMC for a high-frequency probe to infer cochlear compression at lower frequencies. Results further show, however, that reference TMCs were sometimes shallower than corresponding on-frequency TMCs for identical probe frequencies, hence that compression could be overestimated in these cases. We discuss possible reasons for this result and the conditions when it might occur.

Exploring the impacts of consistency in sound masking

Electronic sound masking systems control the noise side of the signal-to-noise ratio in interior environments. Their effectiveness relates directly to how consistently the specified masking curve is achieved. Current system specifications generally allow a relatively wide range in performance, in large part reflecting expectations set by legacy technologies. This session presents a case study of sound masking measurements and speech intelligibility calculations conducted in office spaces. These are used as a foundation to discuss the impacts of local inconsistencies in the masking sound and to begin a discussion of appropriate performance requirements for masking systems.

Computational modeling of individual differences in behavioral estimates of cochlear nonlinearities.

Temporal masking curves (TMCs) are often used to estimate cochlear compression in individuals with normal and impaired hearing. These estimates may yield a wide range of individual differences, even among subjects with similar quiet thresholds. This study used an auditory model to assess potential sources of variance in TMCs from 51 listeners in Poling et al. [J Assoc Res Otolaryngol, 13:91-108 (2012)]. These sources included threshold elevation, the contribution of outer and inner hair cell dysfunction to threshold elevation, compression of the off-frequency linear reference, and detection efficiency. Simulations suggest that detection efficiency is a primary factor contributing to individual differences in TMCs measured in normal-hearing subjects, while threshold elevation and the contribution of outer and inner hair cell dysfunction are primary factors in hearing-impaired subjects. Approximating the most compressive growth rate of the cochlear response from TMCs was achieved only in subjects with the highest detection efficiency. Simulations included off-frequency nonlinearity in basilar membrane and inner hair cell processing; however, this nonlinearity did not improve predictions, suggesting that other sources, such as the decay of masking and the strength of the medial olivocochlear reflex, may mimic off-frequency nonlinearity. Findings from this study suggest that sources of individual differences can play a strong role in behavioral estimates of compression, and these sources should be considered when using forward masking to study cochlear function in individual listeners or across groups of listeners.

Across-frequency behavioral estimates of the contribution of inner and outer hair cell dysfunction to individualized audiometric loss.

Identifying the multiple contributors to the audiometric loss of a hearing impaired (HI) listener at a particular frequency is becoming gradually more useful as new treatments are developed. Here, we infer the contribution of inner (IHC) and outer hair cell (OHC) dysfunction to the total audiometric loss in a sample of 68 hearing aid candidates with mild-to-severe sensorineural hearing loss, and for test frequencies of 0.5, 1, 2, 4, and 6 kHz. It was assumed that the audiometric loss (HLTOTAL) at each test frequency was due to a combination of cochlear gain loss, or OHC dysfunction (HLOHC), and inefficient IHC processes (HLIHC), all of them in decibels. HLOHC and HLIHC were estimated from cochlear I/O curves inferred psychoacoustically using the temporal masking curve (TMC) method. 325 I/O curves were measured and 59% of them showed a compression threshold (CT). The analysis of these I/O curves suggests that (1) HLOHC and HLIHC account on average for 60–70 and 30–40% of HLTOTAL, respectively; (2) these percentages are roughly constant across frequencies; (3) across-listener variability is large; (4) residual cochlear gain is negatively correlated with hearing loss while residual compression is not correlated with hearing loss. Altogether, the present results support the conclusions from earlier studies and extend them to a wider range of test frequencies and hearing-loss ranges. Twenty-four percent of I/O curves were linear and suggested total cochlear gain loss. The number of linear I/O curves increased gradually with increasing frequency. The remaining 17% I/O curves suggested audiometric losses due mostly to IHC dysfunction and were more frequent at low (≤1 kHz) than at high frequencies. It is argued that in a majority of listeners, hearing loss is due to a common mechanism that concomitantly alters IHC and OHC function and that IHC processes may be more labile in the apex than in the base.

Behavioral measures of cochlear compression and temporal resolution as predictors of speech masking release in hearing-impaired listeners

Hearing-impaired (HI) listeners often show less masking release (MR) than normal-hearing listeners when temporal fluctuations are imposed on a steady-state masker, even when accounting for overall audibility differences. This difference may be related to a loss of cochlear compression in HI listeners. Behavioral estimates of compression, using temporal masking curves (TMCs), were compared with MR for band-limited (500-4000 Hz) speech and pure tones in HI listeners and age-matched, noise-masked normal-hearing (NMNH) listeners. Compression and pure-tone MR estimates were made at 500, 1500, and 4000 Hz. The amount of MR was defined as the difference in performance between steady-state and 10-Hz square-wave-gated speech-shaped noise. In addition, temporal resolution was estimated from the slope of the off-frequency TMC. No significant relationship was found between estimated cochlear compression and MR for either speech or pure tones. NMNH listeners had significantly steeper off-frequency temporal masking recovery slopes than did HI listeners, and a small but significant correlation was observed between poorer temporal resolution and reduced MR for speech. The results suggest either that the effects of hearing impairment on MR are not determined primarily by changes in peripheral compression, or that the TMC does not provide a sufficiently reliable measure of cochlear compression.

Novel frequency masking curves for noise-robust automatic speech recognition

We investigate the incorporation of frequency masking curves in the feature extraction module of automatic speech recognition systems to improve noise robustness. Frequency-masking curves are mathematically derived based on an auditory model, in which a basilar membrane is modeled as a cascade system of damped simple harmonic oscillators. Based on the analysis of the motion under speech signals, we derive the relationship between the amplitudes of neighboring oscillators and convert it into frequency-masking curves, which are used in the computation of spectral-masking thresholds to modify the speech spectrum. Evaluated on the Aurora 2.0 noisy-digit speech database, the proposed methodology achieves a significant improvement in noise-robustness.