The Effect of Rate on Tone Burst Extratympanic Electrocochleography in Adults with Normal Hearing

The recording of electrocochleography (ECochG) to tone burst stimuli with a high rate is hypothesized to provide advantages over standard click ECochG with slow rate. Specifically, the use of tone burst stimuli presented at a high rate may enhance the summating potential (SP) while also reducing neural contributions in the response. To allow for the analysis of the complex ECochG responses to high rates, the Continuous Loop Averaging Deconvolution (CLAD) technique was employed to deconvolve the responses. This study examined the effect of high rate and stimulus duration on the SP amplitude of tone burst extratympanic ECochG in 20 adult females with normal hearing. ECochG was recorded using 500 Hz and 2000 Hz stimuli with varied durations (12 ms, 6 ms, 3 ms) and five rates ranging from 7.1/s to 234.38/s. Within-subject repeated measures (rate x duration) analysis of variance were conducted. For both 500 Hz and 2000 Hz, the mean deconvolved SP amplitudes were larger at faster rates compared to slower rates, and larger at shorter duration than longer duration. With the long durations, the increase in SP amplitude with high rate is thought to be due to changes in cochlear mechanics and nonlinearity of the inner hair cells. However, the large amplitude measured in the short duration conditions is mostly due to the contribution of the action potential in the recording. Our study supports the use of the CLAD technique with tone burst ECochG and provides normative data upon which further research can build

Determination and evaluation of clinically efficient stopping criteria for the multiple auditory steady-state response technique

Background: Although the auditory steady-state response (ASSR) technique utilizes objective statistical detection algorithms to estimate behavioural hearing thresholds, the audiologist still has to decide when to terminate ASSR recordings introducing once more a certain degree of subjectivity. Aims: The present study aimed at establishing clinically efficient stopping criteria for a multiple 80-Hz ASSR system. Methods: In Experiment 1, data of 31 normal hearing subjects were analyzed off-line to propose stopping rules. Consequently, ASSR recordings will be stopped when (1) all 8 responses reach significance and significance can be maintained for 8 consecutive sweeps; (2) the mean noise levels were ≤ 4 nV (if at this “≤ 4-nV” criterion, p-values were between 0.05 and 0.1, measurements were extended only once by 8 sweeps); and (3) a maximum amount of 48 sweeps was attained. In Experiment 2, these stopping criteria were applied on 10 normal hearing and 10 hearing-impaired adults to asses the efficiency. Results: The application of these stopping rules resulted in ASSR threshold values that were comparable to other multiple-ASSR research with normal hearing and hearing-impaired adults. Furthermore, in 80% of the cases, ASSR thresholds could be obtained within a time-frame of 1 hour. Investigating the significant response-amplitudes of the hearing-impaired adults through cumulative curves indicated that probably a higher noise-stop criterion than “≤ 4 nV” can be used. Conclusions: The proposed stopping rules can be used in adults to determine accurate ASSR thresholds within an acceptable time-frame of about 1 hour. However, additional research with infants and adults with varying degrees and configurations of hearing loss is needed to optimize these criteria

Neural Correlates of Binaural Interaction Using Aggregate-System Stimulation in Cochlear Implantees

The importance of binaural cues in auditory stream formation and sound source differentiation is widely accepted. When treating one ear with a cochlear implant (CI) the peripheral auditory system gets partially replaced and processing delays get added potentially, thus important interaural time encoding gets altered. This is a crucial problem because factors like the interaural time delay between the receiving ears are known to be responsible for facilitating such cues, e.g., sound source localization and separation. However, these effects are not fully understood, leaving a lack of systematic binaural fitting strategies with respect to an optimal binaural fusion. To gain new insights into such alterations, we suggest a novel method of free-field evoked auditory brainstem response (ABR) analysis in CI users. As a result, this method does not bypass the technically induced intrinsic delays of the hearing device while leaving the complete electrode array active, thus the most natural way of stimulation is provided and the comparable testing of real world stimuli gets facilitated. Unfortunately, ABRs acquired in CI users are additionally affected by the prominent artifact caused by their electrical stimulation, which severely distorts the desired neural response, thus challenging their analysis. To circumvent this problem, we further introduce a novel narrowband filtering CI artifact removal technique capable of obtaining neural correlates of ABRs in CI users. Consequently, we were able to compare brainstem-level responses collected of 12 CI users and 12 normal hearing listeners using two different stimuli (i.e., chirp and click) at four different intensities each, what comprises an adaption of the prominent brainstem evoked response audiometry serving as an additional evaluation criterion. We analyzed the responses using the average of 2,000 trials in combination with synchronized regularizations across them and found consistent results in their deflections and latencies, as well as in single trial relationships between both groups. This method provides a novel and unique perspective into the natural CI users’ brainstem-level responses and can be practical in future research regarding binaural interaction and fusion. Furthermore, the binaural interaction component (BIC), i.e., the arithmetical difference between the sum of both monaurally evoked ABRs and the binaurally evoked ABR, has been previously shown to be an objective indicator for binaural interaction. This component is unfortunately known to be rather fragile and as a result, a reliable, objective measure of binaural interaction in CI users does not exist to the present date. It is most likely that implantees would benefit from a reliable analysis of brainstem-level and subsequent higher-level binaural interaction, since this could objectively support fitting strategies with respect to a maximization of interaural integration. Therefore, we introduce a novel method capable of obtaining neural correlates of binaural interaction in bimodal CI users by combining recent advances in the field of fast, deconvolution-based ABR acquisitions with the introduced narrowband filtering technique. The proposed method shows a significant improvement in the magnitude of resulting BICs in 10 bimodal CI users and a control-group of 10 normal hearing subjects when compensating the interaural latency difference caused by the technical devices. In total, both proposed studies objectively demonstrate technical-driven interaural latency mismatches. Thus, they strongly emphasize potential benefits when balancing these interaural delays to improve binaural processing by significant increases in associated neural correlates of successful binaural interaction. These results and also the estimated latency differences should be investigated in larger group sizes to further consolidate the results, but confirm the demand for rather binaural solutions than treating hearing losses in an isolated monaural manner.Zusammenfassung Die Notwendigkeit binauraler Verarbeitungsprozesse in der auditorischen Wahrnehmung ist weitestgehend akzeptiert. Bei der Therapie eines Ohres mit einem Cochlea-Implantat (engl. cochlear implant (CI)) wird das periphere auditorische System teilweise ersetzt und verändert, sodass natürliche, interaurale Zeitauflösungen beeinflusst werden. Dieses Problem ist entscheidend, denn Faktoren wie interaurale Laufzeitunterschiede zwischen den aufnehmenden Ohren sind verantwortlich für die Umsetzung der erwähnten binauralen Verarbeitungsprozesse, z.B. Schallquellenlokalisation und -separation. Allerdings sind diese Effekte nicht ausreichend verstanden, weshalb bis heute binaurale Anpassstrategien mit Rücksicht auf eine optimale Fusionierung fehlen. Um neue Einsichten in solche zeitlichen Verzerrungen zu erhalten, schlagen wir ein neues Verfahren der Freifeld evozierten auditorischen Hirnstammpotentiale (engl. auditory brainstem response (ABR)) in CI-Nutzern vor. Diese Methode beinhaltet explizit technisch-induzierte Laufzeiten verwendeter Hörhilfen, sodass eine natürliche Stimulation unter Verwendung von realitätsnahen Stimuli ermöglicht wird. Unglücklicherweise sind ABRs von CI-Nutzern zusätzlich mit Stimulationsartefakten belastet, wodurch benötigte neurale Antworten weiter verzerrt werden und eine entsprechende Analyse der Signale deutlich erschwert wird. Um dieses Problem zu umgehen, schlagen wir eine neue Artefakt- Reduktionstechnik vor, welche auf spektraler Schmalbandfilterung basiert und somit den Erhalt entsprechender, neuraler ABR Korrelate ermöglicht. Diese Methoden erlaubten die Interpretation neuraler Korrelate auf Hirnstammebene unter Verwendung von zwei verschiedenen Stimuli (Chirps und Klicks) unter vier verschiedenen Lautstärken in 12 CI-Nutzern und 12 normalhörenden Probanden. Die beschriebene Prozedur adaptiert somit die weitläufig bekannte Hirnstammaudiometrie (engl. brainstem evoked response audiometry (BERA)), deren Ergebnisse zur zusätzlichen Evaluation des vorgestellten Verfahrens dienten. Die Untersuchung der aus 2000 Einzelantworten erhaltenen Mittelwerte in Kombination mit der Analyse synchronisierter Regularitäten über den Verlauf der Einzelantworten ergab dabei konsistente Beobachtungen in gefundenen Amplituden, Latenzen sowie in Abhängigkeiten zwischen Einzelantworten in beiden Gruppen. Das vorgestellte Verfahren erlaubt somit auf einzigartige Weise neue und ungesehene Einsichten in natürliche, neurale Antworten auf Hirnstammebene von CI-Nutzern, welche in zukünftigen Studien verwendet werden können, um binaurale Interaktionen und Fusionen weiter untersuchen zu können. Interessanterweise hat sich, die auf ABRs basierende, binaurale Interaktionskomponente (engl. binaural interaction component (BIC)) als objektiver Indikator binauraler Integration etabliert. Diese Komponente (d.h. die arithmetische Differenz zwischen der Summe der monauralen Antworten und der binauralen Antwort) ist leider sehr fragil, wodurch ein sicherer und objektiver Nachweis in CI-Nutzern bis heute nicht existiert. Dabei ist es sehr wahrscheinlich, dass gerade Implantatsträger von einer entsprechenden Analyse auf Hirnstammebene und höherrangigen Ebenen deutlich profitieren würden, da dies objektiv Anpassstrategien mit Rücksicht auf eine bestmögliche binaurale Integration ermöglichen könnte. Deshalb stellen wir ein weiteres, neuartiges Verfahren zum Erhalt von neuralen Korrelaten binauraler Interaktion in bimodal versorgten CI-Trägern vor, welches jüngste Erfolge im Bereich der schnellen, entfalltungsbasierten ABR Akquisition und der bereits vorgestellten Schmalband- filterung zur Reduktion von Stimulationsartefakten kombiniert. Basierend auf diesem Verfahren konnten signifikante Verbesserungen in der BIC-Amplitude in 10 bimodal versorgten Patienten sowie 10 normalhörenden Probanden, basierend auf umgesetzte, interaurale Laufzeitkompensationen technischer Hörhilfen, aufgezeigt werden. Insgesamt demonstrieren beide vorgestellten Studien technisch-induzierte, interaurale Laufzeitunterschiede und betonen demnach sehr deutlich potenzielle Vorteile in assoziierten neuralen Korrelaten binauraler Interaktionen, wenn solche Missverhältnisse zeitlich ausgeglichen werden. Die aufgezeigten Ergebnisse sowie die getätigte Abschätzungen technischer Laufzeiten sollte in größeren Gruppen weiter untersucht werden, um die Aussagekraft weiter zu steigern. Dennoch unterstreichen diese Einsichten das Verlangen nach binauralen Lösungsansätzen in der zukünftigen Hörrehabilitation, statt bisheriger isolierter und monauraler Therapien

Efficient Acquisition and Denoising of Full-Range Event-Related Potentials Following Transient Stimulation of the Auditory Pathway

This body of work relates to recent advances in the field of human auditory event-related potentials (ERP), specifically the fast, deconvolution-based ERP acquisition as well as single-response based preprocessing, denoising and subsequent analysis methods. Its goal is the contribution of a cohesive set of methods facilitating the fast, reliable acquisition of the whole electrophysiological response generated by the auditory pathway from the brainstem to the cortex following transient acoustical stimulation. The present manuscript is divided into three sequential areas of investigation : First, the general feasibility of simultaneously acquiring auditory brainstem, middle-latency and late ERP single responses is demonstrated using recordings from 15 normal hearing subjects. Favourable acquisition parameters (i.e., sampling rate, bandpass filter settings and interstimulus intervals) are established, followed by signal analysis of the resulting ERP in terms of their dominant intrinsic scales to determine the properties of an optimal signal representation with maximally reduced sample count by means of nonlinear resampling on a logarithmic timebase. This way, a compression ratio of 16.59 is achieved. Time-scale analysis of the linear-time and logarithmic-time ERP single responses is employed to demonstrate that no important information is lost during compressive resampling, which is additionally supported by a comparative evaluation of the resulting average waveforms - here, all prominent waves remain visible, with their characteristic latencies and amplitudes remaining essentially unaffected by the resampling process. The linear-time and resampled logarithmic-time signal representations are comparatively investigated regarding their susceptibility to the types of physiological and technical noise frequently contaminating ERP recordings. While in principle there already exists a plethora of well-investigated approaches towards the denoising of ERP single-response representations to improve signal quality and/or reduce necessary aquisition times, the substantially altered noise characteristics of the obtained, resampled logarithmic-time single response representations as opposed to their linear-time equivalent necessitates a reevaluation of the available methods on this type of data. Additionally, two novel, efficient denoising algorithms based on transform coefficient manipulation in the sinogram domain and on an analytic, discrete wavelet filterbank are proposed and subjected to a comparative performance evaluation together with two established denoising methods. To facilitate a thorough comparison, the real-world ERP dataset obtained in the first part of this work is employed alongside synthetic data generated using a phenomenological ERP model evaluated at different signal-to-noise ratios (SNR), with individual gains in multiple outcome metrics being used to objectively assess algorithm performances. Results suggest the proposed denoising algorithms to substantially outperform the state-of-the-art methods in terms of the employed outcome metrics as well as their respective processing times. Furthermore, an efficient stimulus sequence optimization method for use with deconvolution-based ERP acquisition methods is introduced, which achieves consistent noise attenuation within a broad designated frequency range. A novel stimulus presentation paradigm for the fast, interleaved acquisition of auditory brainstem, middle-latency and late responses featuring alternating periods of optimized, high-rate deconvolution sequences and subsequent low-rate stimulation is proposed and investigated in 20 normal hearing subjects. Deconvolved sequence responses containing early and middle-latency ERP components are fused with subsequent late responses using a time-frequency resolved weighted averaging method based on cross-trial regularity, yielding a uniform SNR of the full-range auditory ERP across investigated timescales. Obtained average ERP waveforms exhibit morphologies consistent with both literature values and the reference recordings obtained in the first part of this manuscript, with all prominent waves being visible in the grand average waveforms. The novel stimulation approach cuts acquisition time by a factor of 3.4 while at the same time yielding a substantial gain in the SNR of obtained ERP data. Results suggest the proposed interleaved stimulus presentation and associated postprocessing methodology to be suitable for the fast, reliable extraction of full-range neural correlates of auditory processing in future studies.Diese Arbeit steht im Zusammenhang mit aktuellen Entwicklungen auf dem Gebiet der ereigniskorrelierten Potentiale (EKP) des humanen auditorischen Systems, insbesondere der schnellen, entfaltungsbasierten EKP-Aufzeichnung sowie einzelantwortbasierten Vorverarbeitungs-, Entrauschungs- und nachgelagerten Analysemethoden. Ziel ist die Bereitstellung eines vollständigen Methodensatzes, der eine schnelle, zuverlässige Erfassung der gesamten elektrophysiologischen Aktivität entlang der Hörbahn vom Hirnstamm bis zum Cortex ermöglicht, die als Folge transienter akustischer Stimulation auftritt. Das vorliegende Manuskript gliedert sich in drei aufeinander aufbauende Untersuchungsbereiche : Zunächst wird die generelle Machbarkeit der gleichzeitigen Aufzeichnung von Einzelantworten der auditorischen Hirnstammpotentiale zusammen mit mittelspäten und späten EKP anhand von Referenzmessungen an 15 normalhörenden Probanden demonstriert. Es werden hierzu geeignete Erfassungsparameter (Abtastrate, Bandpassfiltereinstellungen und Interstimulusintervalle) ermittelt, gefolgt von einer Signalanalyse der resultierenden EKP im Hinblick auf deren dominante intrinsische Skalen, um auf dieser Grundlage die Eigenschaften einer optimalen Signaldarstellung mit maximal reduzierter Anzahl an Abtastpunkten zu bestimmen, die durch nichtlineare Neuabtastung auf eine logarithmische Zeitbasis realisiert wird. Hierbei wird ein Kompressionsverhältnis von 16.59 erzielt. Zeit-Skalen-Analysen der uniform und logarithmisch abgetasteten EKP-Einzelantworten zeigen, dass bei der kompressiven Neuabtastung keine relevante Information verloren geht, was durch eine vergleichende Auswertung der resultierenden, gemittelten Wellenformen zusätzlich gestützt wird - alle prominenten Wellen bleiben sichtbar und sind hinsichtlich ihrer charakteristischen Latenzen und Amplituden von der Neuabtastung weitgehend unbeeinflusst. Die uniforme und logarithmische Signalrepräsentation werden hinsichtlich ihrer Anfälligkeit für die üblicherweise bei der EKP-Aufzeichnung auftretenden physiologischen und technischen Störquellen vergleichend untersucht. Obwohl bereits eine Fülle von gut etablierten Ansätzen für die Entrauschung von EKP-Einzelantwortdarstellungen zur Verbesserung der Signalqualität und/oder zur Reduktion der benötigten Erfassungszeiten existiert, erfordern die wesentlich veränderten Störeigenschaften der vorliegenden, logarithmisch abgetasteten Einzelantwortdarstellungen im Gegensatz zu ihrem uniformen Äquivalent eine Neubewertung der verfügbaren Methoden für diese Art von Daten. Darüber hinaus werden zwei neuartige, effiziente Entrauschungsalgorithmen geboten, die auf der Koeffizientenmanipulation einer Sinogramm-Repräsentation bzw. einer analytischen, diskreten Wavelet-Zerlegung der Einzelantworten basieren und gemeinsam mit zwei etablierten Entrauschungsmethoden einer vergleichenden Leistungsbewertung unterzogen werden. Um einen umfassenden Vergleich zu ermöglichen, werden der im ersten Teil dieser Arbeit erhaltene EKP-Messdatensatz sowie synthetischen Daten eingesetzt, die mithilfe eines phänomenologischen EKP-Modells bei verschiedenen Signal-Rausch-Abständen (SRA) erzeugt wurden, wobei die individuellen Anstiege in mehreren Zielmetriken zur objektiven Bewertung der Performanz herangezogen werden. Die erhaltenen Ergebnisse deuten darauf hin, dass die vorgeschlagenen Entrauschungsalgorithmen die etablierten Methoden sowohl in den eingesetzten Zielmetriken als auch mit Blick auf die Laufzeiten deutlich übertreffen. Weiterhin wird ein effizientes Reizsequenzoptimierungsverfahren für den Einsatz mit entfaltungsbasierten EKP-Aufzeichnungsmethoden vorgestellt, das eine konsistente Rauschunterdrückung innerhalb eines breiten Frequenzbands erreicht. Ein neuartiges Stimulus-Präsentationsparadigma für die schnelle, verschachtelte Erfassung auditorischer Hirnstammpotentiale, mittlelspäter und später Antworten durch alternierende Darbietung von optimierten, dichter Stimulussequenzen und nachgelagerter, langsamer Einzelstimulation wird eingeführt und in 20 normalhörenden Probanden evaluiert. Entfaltete Sequenzantworten, die frühe und mittlere EKP enthalten, werden mit den nachfolgenden späten Antworten fusioniert, wobei eine Zeit-Frequenz-aufgelöste, gewichtete Mittelung unter Berücksichtigung von Regularität über Einzelantworten hinweg zum Einsatz kommt. Diese erreicht einheitliche SRA der resultierenden EKP-Signale über alle untersuchten Zeitskalen hinweg. Die erhaltenen, gemittelten EKP-Wellenformen weisen Morphologien auf, die sowohl mit einschlägigen Literaturwerten als auch mit den im ersten Teil dieses Manuskripts erhaltenen Referenzaufnahmen konsistent sind, wobei alle markanten Wellen deutlich in den Gesamtmittelwerten sichtbar sind. Das neuartige Stimulationsparadigma verkürzt die Erfassungszeit um den Faktor 3.4 und vergrößert gleichzeitig den erreichten SRA erheblich. Die Ergebnisse deuten darauf hin, dass die vorgeschlagene verschachtelte Stimuluspräsentation und die nachgelagerte EKP-Verarbeitungsmethodik zur schnellen, zuverlässigen Extraktion neuronaler Korrelate der gesamten auditorischen Verarbeitung im Rahmen zukünftiger Studien geeignet sind.Bundesministerium für Bildung und Forschung | Bimodal Fusion - Eine neurotechnologische Optimierungsarchitektur für integrierte bimodale Hörsysteme | 2016-201

Human sensory-evoked responses differ coincident with either "fusion-memory" or "flash-memory", as shown by stimulus repetition-rate effects

BACKGROUND: A new method has been used to obtain human sensory evoked-responses whose time-domain waveforms have been undetectable by previous methods. These newly discovered evoked-responses have durations that exceed the time between the stimuli in a continuous stream, thus causing an overlap which, up to now, has prevented their detection. We have named them "A-waves", and added a prefix to show the sensory system from which the responses were obtained (visA-waves, audA-waves, somA-waves). RESULTS: When A-waves were studied as a function of stimulus repetition-rate, it was found that there were systematic differences in waveshape at repetition-rates above and below the psychophysical region in which the sensation of individual stimuli fuse into a continuity. The fusion phenomena is sometimes measured by a "Critical Fusion Frequency", but for this research we can only identify a frequency-region [which we call the STZ (Sensation-Transition Zone)]. Thus, the A-waves above the STZ differed from those below the STZ, as did the sensations. Study of the psychophysical differences in auditory and visual stimuli, as shown in this paper, suggest that different stimulus features are detected, and remembered, at stimulation rates above and below STZ. CONCLUSION: The results motivate us to speculate that: 1) Stimulus repetition-rates above the STZ generate waveforms which underlie "fusion-memory" whereas rates below the STZ show neuronal processing in which "flash-memory" occurs. 2) These two memories differ in both duration and mechanism, though they may occur in the same cell groups. 3) The differences in neuronal processing may be related to "figure" and "ground" differentiation. We conclude that A-waves provide a novel measure of neural processes that can be detected on the human scalp, and speculate that they may extend clinical applications of evoked response recordings. If A-waves also occur in animals, it is likely that A-waves will provide new methods for comparison of activity of neuronal populations and single cells

A convolutional neural-network model of human cochlear mechanics and filter tuning for real-time applications

Auditory models are commonly used as feature extractors for automatic speech-recognition systems or as front-ends for robotics, machine-hearing and hearing-aid applications. Although auditory models can capture the biophysical and nonlinear properties of human hearing in great detail, these biophysical models are computationally expensive and cannot be used in real-time applications. We present a hybrid approach where convolutional neural networks are combined with computational neuroscience to yield a real-time end-to-end model for human cochlear mechanics, including level-dependent filter tuning (CoNNear). The CoNNear model was trained on acoustic speech material and its performance and applicability were evaluated using (unseen) sound stimuli commonly employed in cochlear mechanics research. The CoNNear model accurately simulates human cochlear frequency selectivity and its dependence on sound intensity, an essential quality for robust speech intelligibility at negative speech-to-background-noise ratios. The CoNNear architecture is based on parallel and differentiable computations and has the power to achieve real-time human performance. These unique CoNNear features will enable the next generation of human-like machine-hearing applications

Comparison of LFP-Based and Spike-Based Spectro-Temporal Receptive Fields and Cross-Correlation in Cat Primary Auditory Cortex

Multi-electrode array recordings of spike and local field potential (LFP) activity were made from primary auditory cortex of 12 normal hearing, ketamine-anesthetized cats. We evaluated 259 spectro-temporal receptive fields (STRFs) and 492 frequency-tuning curves (FTCs) based on LFPs and spikes simultaneously recorded on the same electrode. We compared their characteristic frequency (CF) gradients and their cross-correlation distances. The CF gradient for spike-based FTCs was about twice that for 2–40 Hz-filtered LFP-based FTCs, indicating greatly reduced frequency selectivity for LFPs. We also present comparisons for LFPs band-pass filtered between 4–8 Hz, 8–16 Hz and 16–40 Hz, with spike-based STRFs, on the basis of their marginal frequency distributions. We find on average a significantly larger correlation between the spike based marginal frequency distributions and those based on the 16–40 Hz filtered LFP, compared to those based on the 4–8 Hz, 8–16 Hz and 2–40 Hz filtered LFP. This suggests greater frequency specificity for the 16–40 Hz LFPs compared to those of lower frequency content. For spontaneous LFP and spike activity we evaluated 1373 pair correlations for pairs with >200 spikes in 900 s per electrode. Peak correlation-coefficient space constants were similar for the 2–40 Hz filtered LFP (5.5 mm) and the 16–40 Hz LFP (7.4 mm), whereas for spike-pair correlations it was about half that, at 3.2 mm. Comparing spike-pairs with 2–40 Hz (and 16–40 Hz) LFP-pair correlations showed that about 16% (9%) of the variance in the spike-pair correlations could be explained from LFP-pair correlations recorded on the same electrodes within the same electrode array. This larger correlation distance combined with the reduced CF gradient and much broader frequency selectivity suggests that LFPs are not a substitute for spike activity in primary auditory cortex

Distinct Features of Auditory Steady-State Responses as Compared to Transient Event-Related Potentials

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Noise Attenuation Estimation for Maximum Length Sequences in Deconvolution Process of Auditory Evoked Potentials

The use of maximum length sequence (m-sequence) has been found beneficial for recovering both linear and nonlinear components at rapid stimulation. Since m-sequence is fully characterized by a primitive polynomial of different orders, the selection of polynomial order can be problematic in practice. Usually, the m-sequence is repetitively delivered in a looped fashion. Ensemble averaging is carried out as the first step and followed by the cross-correlation analysis to deconvolve linear/nonlinear responses. According to the classical noise reduction property based on additive noise model, theoretical equations have been derived in measuring noise attenuation ratios (NARs) after the averaging and correlation processes in the present study. A computer simulation experiment was conducted to test the derived equations, and a nonlinear deconvolution experiment was also conducted using order 7 and 9 m-sequences to address this issue with real data. Both theoretical and experimental results show that the NAR is essentially independent of the m-sequence order and is decided by the total length of valid data, as well as stimulation rate. The present study offers a guideline for m-sequence selections, which can be used to estimate required recording time and signal-to-noise ratio in designing m-sequence experiments