Multi-response deconvolution of auditory evoked potentials in a reduced representation space.

The estimation of auditory evoked potentials requires deconvolution when the duration of the responses to be recovered exceeds the inter-stimulus interval. Based on least squares deconvolution, in this article we extend the procedure to the case of a multi-response convolutional model, that is, a model in which different categories of stimulus are expected to evoke different responses. The computational cost of the multi-response deconvolution significantly increases with the number of responses to be deconvolved, which restricts its applicability in practical situations. In order to alleviate this restriction, we propose to perform the multi-response deconvolution in a reduced representation space associated with a latency-dependent filtering of auditory responses, which provides a significant dimensionality reduction. We demonstrate the practical viability of the multi-response deconvolution with auditory responses evoked by clicks presented at different levels and categorized according to their stimulation level. The multi-response deconvolution applied in a reduced representation space provides the least squares estimation of the responses with a reasonable computational load. matlab/Octave code implementing the proposed procedure is included as supplementary material.

Auditory brainstem responses obtained with randomised stimulation level.

OBJECTIVE: To present randomised stimulation level (RSL) - a stimulation paradigm in which the level of the stimuli is randomised, rather than presented sequentially as in the conventional paradigm. DESIGN: The value of RSL was evaluated by (i) comparing the morphology of auditory brainstem responses (ABRs) elicited by the conventional and RSL paradigms, and by (ii) an online survey investigating the hearing comfort of the stimulus sequence. STUDY SAMPLE: ABRs were obtained from 11 normal-hearing adults (8 females, 25-29 years). The online survey was administered to 238 adults from the general community. RESULTS: Results showed that (i) both stimulation paradigms elicit ABR signals of similar morphology, (ii) RSL provides a faster comprehensive representation of the ABR session, and that (iii) the general population found RSL stimuli to be more comfortable. CONCLUSIONS: The simultaneous evaluation of all ABR traces of the session provided by RSL has potential to improve the identification of ABR components by enabling clinicians to make use of the response tracking strategy from the start of the test, which is critical in situations where ABRs present an abnormal morphology. New research opportunities and the clinical potential of RSL are discussed.

The hunt for hidden hearing loss in humans: From preclinical studies to effective interventions.

Many individuals experience hearing problems that are hidden under a normal audiogram. This not only impacts on individual sufferers, but also on clinicians who can offer little in the way of support. Animal studies using invasive methodologies have developed solid evidence for a range of pathologies underlying this hidden hearing loss (HHL), including cochlear synaptopathy, auditory nerve demyelination, elevated central gain, and neural mal-adaptation. Despite progress in pre-clinical models, evidence supporting the existence of HHL in humans remains inconclusive, and clinicians lack any non-invasive biomarkers sensitive to HHL, as well as a standardized protocol to manage hearing problems in the absence of elevated hearing thresholds. Here, we review animal models of HHL as well as the ongoing research for tools with which to diagnose and manage hearing difficulties associated with HHL. We also discuss new research opportunities facilitated by recent methodological tools that may overcome a series of barriers that have hampered meaningful progress in diagnosing and treating of HHL.

Subspace-constrained deconvolution of auditory evoked potentials.

Auditory evoked potentials can be estimated by synchronous averaging when the responses to the individual stimuli are not overlapped. However, when the response duration exceeds the inter-stimulus interval, a deconvolution procedure is necessary to obtain the transient response. The iterative randomized stimulation and averaging and the equivalent randomized stimulation with least squares deconvolution have been proven to be flexible and efficient methods for deconvolving the evoked potentials, with minimum restrictions in the design of stimulation sequences. Recently, a latency-dependent filtering and down-sampling (LDFDS) methodology was proposed for optimal filtering and dimensionality reduction, which is particularly useful when the evoked potentials involve the complete auditory pathway response (i.e., from the cochlea to the auditory cortex). In this case, the number of samples required to accurately represent the evoked potentials can be reduced from several thousand (with conventional sampling) to around 120. In this article, we propose to perform the deconvolution in the reduced representation space defined by LDFDS and present the mathematical foundation of the subspace-constrained deconvolution. Under the assumption that the evoked response is appropriately represented in the reduced representation space, the proposed deconvolution provides an optimal least squares estimation of the evoked response. Additionally, the dimensionality reduction provides a substantial reduction of the computational cost associated with the deconvolution. matlab/Octave code implementing the proposed procedures is included as supplementary material.

Latency-dependent filtering and compact representation of the complete auditory pathway response.

Auditory evoked potentials (AEPs) include the auditory brainstem response (ABR), middle latency response (MLR), and cortical auditory evoked potentials (CAEPs), each one covering a specific latency range and frequency band. For this reason, ABR, MLR, and CAEP are usually recorded separately using different protocols. This article proposes a procedure providing a latency-dependent filtering and down-sampling of the AEP responses. This way, each AEP component is appropriately filtered, according to its latency, and the complete auditory pathway response is conveniently represented (with the minimum number of samples, i.e., without unnecessary redundancies). The compact representation of the complete response facilitates a comprehensive analysis of the evoked potentials (keeping the natural continuity related to the neural activity transmission along the auditory pathway), which provides a new perspective in the design and analysis of AEP experiments. Additionally, the proposed compact representation reduces the storage or transmission requirements when large databases are manipulated for clinical or research purposes. The analysis of the AEP responses shows that a compact representation with 40 samples/decade (around 120 samples) is enough for accurately representing the response of the complete auditory pathway and provides appropriate latency-dependent filtering. MatLab/Octave code implementing the proposed procedure is included in the supplementary materials.

Self-Referenced Multifrequency Phase-Resolved Luminescence Spectroscopy.

Phase-resolved luminescence chemical sensors provide the analyte determination based on the estimation of the luminescence lifetime. The lifetime is estimated from an analysis of the amplitudes and/or phases of the excitation and emission signals at one or several modulation frequencies. This requires recording both the excitation signal (used to modulate the light source) and the emission signal (obtained from an optical transducer illuminated by the luminescent sensing phase). The excitation signal is conventionally used as reference, in order to obtain the modulation factor (the ratio between the emission and the excitation amplitudes) and/or the phase shift (the difference between the emission and the excitation phases) at each modulation frequency, which are used to estimate the luminescence lifetime. In this manuscript, we propose a new method providing the luminescence lifetimes (based either on amplitudes or phases) using only the emission signal (i.e., omitting the excitation signal in the procedure). We demonstrate that the luminescence lifetime can be derived from the emission signal when it contains at least two harmonics, because in this case the amplitude and phase of one of the harmonics can be used as reference. We present the theoretical formulation as well as an example of application to an oxygen measuring system. The proposed self-referenced lifetime estimation provides two practical advantages for luminescence chemical sensors. On one hand, it simplifies the instrument architecture, since only one analog-to-digital converter (for the emission signal) is necessary. On the other hand, the self-referenced estimation of the lifetime improves the robustness against degradation of the sensing phase or variations in the optical coupling, which reduces the recalibration requirements when the lifetimes are based on amplitudes.

Matrix-based formulation of the iterative randomized stimulation and averaging method for recording evoked potentials.

The iterative randomized stimulation and averaging (IRSA) method was proposed for recording evoked potentials when the individual responses are overlapped. The main inconvenience of IRSA is its computational cost, associated with a large number of iterations required for recovering the evoked potentials and the computation required for each iteration [involving the whole electroencephalogram (EEG)]. This article proposes a matrix-based formulation of IRSA, which is mathematically equivalent and saves computational load (because each iteration involves just a segment with the length of the response, instead of the whole EEG). Additionally, it presents an analysis of convergence that demonstrates that IRSA converges to the least-squares (LS) deconvolution. Based on the convergence analysis, some optimizations for the IRSA algorithm are proposed. Experimental results (configured for obtaining the full-range auditory evoked potentials) show the mathematical equivalence of the different IRSA implementations and the LS-deconvolution and compare the respective computational costs of these implementations under different conditions. The proposed optimizations allow the practical use of IRSA for many clinical and research applications and provide a reduction of the computational cost, very important with respect to the conventional IRSA, and moderate with respect to the LS-deconvolution. matlab/Octave implementations of the different methods are provided as supplementary material.

An automatic algorithm for blink-artifact suppression based on iterative template matching: application to single channel recording of cortical auditory evoked potentials.

OBJECTIVE: Artifact reduction in electroencephalogram (EEG) signals is usually necessary to carry out data analysis appropriately. Despite the large amount of denoising techniques available with a multichannel setup, there is a lack of efficient algorithms that remove (not only detect) blink-artifacts from a single channel EEG, which is of interest in many clinical and research applications. This paper describes and evaluates the iterative template matching and suppression (ITMS), a new method proposed for detecting and suppressing the artifact associated with the blink activity from a single channel EEG. APPROACH: The approach of ITMS consists of (a) an iterative process in which blink-events are detected and the blink-artifact waveform of the analyzed subject is estimated, (b) generation of a signal modeling the blink-artifact, and (c) suppression of this signal from the raw EEG. The performance of ITMS is compared with the multi-window summation of derivatives within a window (MSDW) technique using both synthesized and real EEG data. MAIN RESULTS: Results suggest that ITMS presents an adequate performance in detecting and suppressing blink-artifacts from a single channel EEG. When applied to the analysis of cortical auditory evoked potentials (CAEPs), ITMS provides a significant quality improvement in the resulting responses, i.e. in a cohort of 30 adults, the mean correlation coefficient improved from 0.37 to 0.65 when the blink-artifacts were detected and suppressed by ITMS. SIGNIFICANCE: ITMS is an efficient solution to the problem of denoising blink-artifacts in single-channel EEG applications, both in clinical and research fields. The proposed ITMS algorithm is stable; automatic, since it does not require human intervention; low-invasive, because the EEG segments not contaminated by blink-artifacts remain unaltered; and easy to implement, as can be observed in the Matlab script implemeting the algorithm provided as supporting material.

Selective processing of auditory evoked responses with iterative-randomized stimulation and averaging: A strategy for evaluating the time-invariant assumption.

The recording of auditory evoked potentials (AEPs) at fast rates allows the study of neural adaptation, improves accuracy in estimating hearing threshold and may help diagnosing certain pathologies. Stimulation sequences used to record AEPs at fast rates require to be designed with a certain jitter, i.e., not periodical. Some authors believe that stimuli from wide-jittered sequences may evoke auditory responses of different morphology, and therefore, the time-invariant assumption would not be accomplished. This paper describes a methodology that can be used to analyze the time-invariant assumption in jittered stimulation sequences. The proposed method [Split-IRSA] is based on an extended version of the iterative randomized stimulation and averaging (IRSA) technique, including selective processing of sweeps according to a predefined criterion. The fundamentals, the mathematical basis and relevant implementation guidelines of this technique are presented in this paper. The results of this study show that Split-IRSA presents an adequate performance and that both fast and slow mechanisms of adaptation influence the evoked-response morphology, thus both mechanisms should be considered when time-invariance is assumed. The significance of these findings is discussed.

Auditory brainstem and middle latency responses recorded at fast rates with randomized stimulation.

Randomized stimulation and averaging (RSA) allows auditory evoked potentials (AEPs) to be recorded at high stimulation rates. This method does not perform deconvolution and must therefore deal with interference derived from overlapping transient evoked responses. This paper analyzes the effects of this interference on auditory brainstem responses (ABRs) and middle latency responses (MLRs) recorded at rates of up to 300 and 125 Hz, respectively, with randomized stimulation sequences of a jitter both greater and shorter than the dominant period of the ABR/MLR components. Additionally, this paper presents an advanced approach for RSA [iterative-randomized stimulation and averaging (I-RSA)], which includes the removal of the interference associated with overlapping responses through an iterative process in the time domain. Experimental results show that (a) RSA can be efficiently used in the recording of AEPs when the jitter of the stimulation sequence is greater than the dominant period of the AEP components, and (b) I-RSA maintains all the advantages of RSA and is not constrained by the restriction of a minimum jitter. The significance of the results of this study is discussed.

A flexible and inexpensive high-performance auditory evoked response recording system appropriate for research purposes.

Recording auditory evoked responses (AER) is done not only in hospitals and clinics worldwide to detect hearing impairments and estimate hearing thresholds, but also in research centers to understand and model the mechanisms involved in the process of hearing. This paper describes a high-performance, flexible, and inexpensive AER recording system. A full description of the hardware and software modules that compose the AER recording system is provided. The performance of this system was evaluated by conducting five experiments with both real and artificially synthesized auditory brainstem response and middle latency response signals at different intensity levels and stimulation rates. The results indicate that the flexibility of the described system is appropriate to record AER signals under several recording conditions. The AER recording system described in this article is a flexible and inexpensive high-performance AER recording system. This recording system also incorporates a platform through which users are allowed to implement advanced signal processing methods. Moreover, its manufacturing cost is significantly lower than that of other commercially available alternatives. These advantages may prove useful in many research applications in audiology.

Automatic quality assessment and peak identification of auditory brainstem responses with fitted parametric peaks.

The recording of the auditory brainstem response (ABR) is used worldwide for hearing screening purposes. In this process, a precise estimation of the most relevant components is essential for an accurate interpretation of these signals. This evaluation is usually carried out subjectively by an audiologist. However, the use of automatic methods for this purpose is being encouraged nowadays in order to reduce human evaluation biases and ensure uniformity among test conditions, patients, and screening personnel. This article describes a new method that performs automatic quality assessment and identification of the peaks, the fitted parametric peaks (FPP). This method is based on the use of synthesized peaks that are adjusted to the ABR response. The FPP is validated, on one hand, by an analysis of amplitudes and latencies measured manually by an audiologist and automatically by the FPP method in ABR signals recorded at different stimulation rates; and on the other hand, contrasting the performance of the FPP method with the automatic evaluation techniques based on the correlation coefficient, FSP, and cross correlation with a predefined template waveform by comparing the automatic evaluations of the quality of these methods with subjective evaluations provided by five experienced evaluators on a set of ABR signals of different quality. The results of this study suggest (a) that the FPP method can be used to provide an accurate parameterization of the peaks in terms of amplitude, latency, and width, and (b) that the FPP remains as the method that best approaches the averaged subjective quality evaluation, as well as provides the best results in terms of sensitivity and specificity in ABR signals validation. The significance of these findings and the clinical value of the FPP method are highlighted on this paper.

A study of adaptation mechanisms based on ABR recorded at high stimulation rate.

OBJECTIVE: This paper analyzes the fast and slow mechanisms of adaptation through a study of latencies and amplitudes on ABR recorded at high stimulation rates using the randomized stimulation and averaging (RSA) technique. METHODS: The RSA technique allows a separate processing of auditory responses, and is used, in this study, to categorize responses according to the interstimulus interval (ISI) of their preceding stimulus. The fast and slow mechanisms of adaptation are analyzed by the separated responses methodology, whose underlying principles and mathematical basis are described in detail. RESULTS: The morphology of the ABR is influenced by both fast and slow mechanisms of adaptation. These results are consistent with previous animal studies based on spike rate. CONCLUSIONS: Both fast and slow mechanisms of adaptation are present in all subjects. In addition, the distribution of the jitter and the sequencing of the stimuli may be critical parameters when obtaining reliable ABRs. SIGNIFICANCE: The separated responses methodology enables for the first time the analysis of the fast and slow mechanisms of adaptation in ABR obtained at stimulation rates greater than 100 Hz. The non-invasive nature of this methodology is appropriate for its use in humans.

Ultrasound indoor positioning system based on a low-power wireless sensor network providing sub-centimeter accuracy.

This paper describes the TELIAMADE system, a new indoor positioning system based on time-of-flight (TOF) of ultrasonic signal to estimate the distance between a receiver node and a transmitter node. TELIAMADE system consists of a set of wireless nodes equipped with a radio module for communication and a module for the transmission and reception of ultrasound. The access to the ultrasonic channel is managed by applying a synchronization algorithm based on a time-division multiplexing (TDMA) scheme. The ultrasonic signal is transmitted using a carrier frequency of 40 kHz and the TOF measurement is estimated by applying a quadrature detector to the signal obtained at the A/D converter output. Low sampling frequencies of 17.78 kHz or even 12.31 kHz are possible using quadrature sampling in order to optimize memory requirements and to reduce the computational cost in signal processing. The distance is calculated from the TOF taking into account the speed of sound. An excellent accuracy in the estimation of the TOF is achieved using parabolic interpolation to detect of maximum of the signal envelope at the matched filter output. The signal phase information is also used for enhancing the TOF measurement accuracy. Experimental results show a root mean square error (rmse) less than 2 mm and a standard deviation less than 0.3 mm for pseudorange measurements in the range of distances between 2 and 6 m. The system location accuracy is also evaluated by applying multilateration. A sub-centimeter location accuracy is achieved with an average rmse of 9.6 mm.

Analysis of electrical thresholds and maximum comfortable levels in cochlear implant patients.

OBJECTIVE: It is well known that a proper fitting of the cochlear implant processor is relevant to provide good quality in speech perception. The aim of this study is to extract statistical information to be applied for fitting the processor. METHODS: This study is based on the programming maps of 121 patients, aged from 18 months to 68 years at the moment of implantation. All subjects were implanted with the COMBI 40+ cochlear implant at San Cecilio University Hospital, Granada (Spain). The patients were classified into groups based on their age at implantation: younger than 5 years, between 5 and 16 years, and older than 16 years. The patients in each age-based group were divided into two subgroups, considering whether they had recent hearing experience or not. A special group including patients affected by severe damages in the cochlea was also defined. RESULTS: Relationships between the programming parameters and factors like the age at implantation, the hearing experience and the presence of severe cochlear damage were found. The THR levels for patients younger than 5 years were significantly lower than those for patients implanted between 5 and 16 years, and this group presented significantly lower THR levels than adults. The MCL levels were not significantly influenced by the age at implantation. A significant increment was observed for both, MCL and THR levels, when patients were affected by severe cochlear damage. A significant increment in the THR levels were observed for patients with no recent hearing experience, while no significant differences were found for MCL levels. This study also analyzes the distribution along the cochlea of the stimulation levels. In the case of patients not affected by severe cochlear damage, the most basal electrodes presented a significant increment in the stimulation levels with respect to the rest of electrodes. CONCLUSION: This work provides information of great value for programming the speech processors, particularly when the subjective responses of the patients are not sufficient. The application in our ENT Service has reduced substantially the average time needed to obtain an acceptable fitting of the processor, especially in children. Our study also shows that electrical thresholds are a good indicator of the functionality of the auditory nerve. The analysis of this parameter highlights the importance of an early intervention as well as a deep insertion of the electrode carrier in order to obtain the maximum functionality from the cochlear implant.

Identification of active compounds in vegetal extracts based on correlation between activity and HPLC-MS data.

We propose a method identifying candidates for active compounds in vegetal extracts. From a collection of samples, the method requires, for each sample, a HPLC-MS analysis and a measurement of the activity. By applying a correlation analysis between the activity and the chromatographic area for each interval of elution time and m/z ratio, the peaks corresponding to candidates for active compounds can be identified. Additionally, when peaks are identified, a model can be estimated to predict the activity in new samples. Both methods are evaluated in one experiment involving the phenolic extract (PE) from 22 samples of extra virgin olive oil (EVOO) where the activity is a cytotoxicity index against JIMT-1 breast cancer cells. In this experiment, the samples were separated into two disjunct partitions: one was used for training (identification of candidates and estimation of prediction model), while the other was used for validation (by comparing the predicted and the measured activities). Three compounds were identified as candidates to be responsible for the cytotoxicity of the EVOO-PE against JIMT-1 cells. The prediction model provided an accurate estimation of the activity.

Long-term evolution of the electrical stimulation levels for cochlear implant patients.

OBJECTIVES: The stimulation levels programmed in cochlear implant systems are affected by an evolution since the first switch-on of the processor. This study was designed to evaluate the changes in stimulation levels over time and the relationship between post-implantation physiological changes and with the hearing experience provided by the continuous use of the cochlear implant. METHODS: Sixty-two patients, ranging in age from 4 to 68 years at the moment of implantation participated in this study. All subjects were implanted with the 12 channels COMBI 40+ cochlear implant at San Cecilio University Hospital, Granada, Spain. Hearing loss etiology and progression characteristics varied across subjects. RESULTS: The analyzed programming maps show that the stimulation levels suffer a fast evolution during the first weeks after the first switch-on of the processor. Then, the evolution becomes slower and the programming parameters tend to be stable at about 6 months after the first switch-on. The evolution of the stimulation levels implies an increment of the electrical dynamic range, which is increased from 15.4 to 20.7 dB and improves the intensity resolution. A significant increment of the sensitivity to acoustic stimuli is also observed. For some patients, we have also observed transitory changes in the electrode impedances associated to secretory otitis media, which cause important changes in the programming maps. CONCLUSION: We have studied the long-term evolution of the stimulation levels in cochlear implant patients. Our results show the importance of systematic measurements of the electrode impedances before the revision of the programming map. This report also highlights that the evolution of the programming maps is an important factor to be considered in order to determine an adequate calendar fitting of the cochlear implant processor.

Recording of auditory brainstem response at high stimulation rates using randomized stimulation and averaging.

The recording of auditory brainstem response (ABR) at high stimulation rates is of great interest in audiology. It allows a more accurate diagnosis of certain pathologies at an early stage and the study of different mechanisms of adaptation. This paper proposes a methodology, which we will refer to as randomized stimulation and averaging (RSA) that allows the recording of ABR at high stimulation rates using jittered stimuli. The proposed method has been compared with quasi-periodic sequence deconvolution (QSD) and conventional (CONV) stimulation methodologies. Experimental results show that RSA provides a quality in ABR recordings similar to that of QSD and CONV. Compared with CONV, RSA presents the advantage of being able to record ABR at rates higher than 100 Hz. Compared with QSD, the formulation of RSA is simpler and allows more flexibility on the design of the pseudorandom sequence. The feasibility of the RSA methodology is validated by an analysis of the morphology, amplitudes, and latencies of the most important waves in ABR recorded at high stimulation rates from eight normal hearing subjects.

Using evoked compound action potentials to assess activation of electrodes and predict C-levels in the Tempo+ cochlear implant speech processor.

OBJECTIVE: In this study, we analyze how electrically evoked compound action potential (ECAP) responses can be used to assess whether electrodes should be activated in the map and to estimate C levels in the Med-El Tempo+ Cochlear Implant Speech Processor. DESIGN: ECAP thresholds were measured using the ECAP Recording System of the Pulsar CI implant. Twenty-one postlingually and 28 prelingually deafened patients participated in this study. The relationship between ECAP responses and the activation of electrodes was analyzed. Because an error in the estimation of T levels (behavioral thresholds) has less effect on hearing quality than an error in the estimation of C levels in the Tempo+ cochlear implant speech processor (maximum comfort levels), correlation and regression analyses were performed between ECAP thresholds and C levels. RESULTS: The observation of an evoked potential generally implied that the electrode was activated because only 3.5% of electrodes that yielded measurable evoked responses were deactivated, because of collateral stimulations or an unpleasant hearing sensation. In contrast, the absence of an evoked potential did not imply that an electrode should be deactivated, because 20% of these electrodes provided a useful auditory sensation. ECAP responses did not predict the absolute behavioral comfort levels because of the excessive error between behavioral C levels and those derived from ECAP thresholds (the mean relative error is 43.78%). However, by applying a normalization procedure, ECAP measurements allowed the C-level profile to be predicted with a mean relative error of 6%; that is, they provided useful data to determine the C level of each electrode relative to the average C level of the patient. CONCLUSIONS: ECAP is a reliable and an useful objective measurement that can assist in the fitting of the Tempo+ cochlear implant speech processor. From results presented in this work, a protocol is proposed for fitting this cochlear implant system. This protocol facilitates appropriate cochlear implant fitting, particularly for children or uncooperative patients.

Reducing blanking artifact in electrically evoked compound action potentials.

The main source of distortion in the recording of the electrically evoked compound action potentials is the stimulus artifact. The popular hardware blanking technique tends to reduce this artifact, but generates a blanking artifact as a consequence of the transient state in the amplifier. In this paper we propose two techniques to deal with the blanking artifact. The proposed techniques are combined with conventional and generalized alternating stimulation in order to reduce both stimulus and blanking artifacts in the recording of the evoked potentials. A comparison over 126 evoked potential recordings reveals that the proposed blanking artifact reduction methods improve the quality of electrically evoked compound action potential recordings.