Showing papers on "Digital hearing aid published in 2017"
TL;DR: An adaptive feedback canceller, which is trained using a set of sparse adaptive algorithms is designed in this paper to take advantage of the sparseness of the acoustic feedback path in a hearing aid.
28 citations
TL;DR: This implementation aims at having all the subbands collectively updating the fullband adaptive filter, without the need for a subband to fullband weight conversion and offers improved feedback cancellation at reduced computational load in comparison with a delayless subband implementation of a PEM-based feedback canceller.
Abstract: Acoustic feedback cancellation is one of the challenging tasks in the design of a behind the ear digital hearing aid. This feedback cancellation is usually achieved by using an adaptive filter. The finite correlation between the desired microphone input signal and the input signal to the loudspeaker results in a biased estimation of the adaptive filter, which may produce disturbances in the hearing aid. Prediction error method (PEM) has been used in literature to reduce the bias effects. The convergence of a PEM-based feedback canceller can be improved by implementing the adaptive filter in the subband domain. However, a direct subband implementation results in aliasing issues, band-edge problems, and introduces a delay due to analysis and synthesis filters. In order to reduce the aliasing and delay issues, a delayless subband implementation of a PEM-based feedback canceller is designed in this paper. A delayless multiband-structured subband implementation of the feedback canceller is also attempted to further reduce the aliasing and band-edge effects. This implementation aims at having all the subbands collectively updating the fullband adaptive filter, without the need for a subband to fullband weight conversion and offers improved feedback cancellation at reduced computational load in comparison with a delayless subband implementation of a PEM-based feedback canceller. In addition, an attempt has been made to further improve the convergence behavior by using an improved proportionate learning scheme. The improved convergence offered by the proposed scheme is evident from the simulation study. The improvement has been further quantified using a perceptual evaluation of speech quality and the proposed approach has been shown to provide enhanced speech quality.
22 citations
01 Mar 2017
TL;DR: A novel Bayesian Blind Deconvolution approach with exponentially decaying kernel is proposed and its application in extracting the invariant part of the feedback path measurements of a digital hearing aid is shown.
Abstract: Acoustic Feedback Path in a digital hearing aid is not only affected by the user's head and ear, but also by different acoustic environments. But some of these effects are common for a specific style of hearing aid and individual ear, i.e., this part will be invariant to the different acoustic environments and can be interpreted as the effects associated with that specific hearing aid and ear characteristics. In this article we propose a novel Bayesian Blind Deconvolution approach with exponentially decaying kernel and show its application in extracting the invariant part of the feedback path measurements of a digital hearing aid. Efficacy of our proposed approach in extracting the invariant part has been measured by using the extracted invariant part to model unseen test Feedback Path (FBP) measured from the same hearing aid but in a different acoustic environment, over existing methods.
10 citations
TL;DR: A self-fitting algorithm based on an improved interactive evolutionary computation (IEC) algorithm and expert system, which enables the patients to fit the hearing aid by themselves, is proposed.
Abstract: The traditional hearing aid fitting method, which mainly relies on the audiologist, is timeconsuming and messy. To improve this situation, a self-fitting algorithm based on an improved interactive evolutionary computation (IEC) algorithm and expert system, which enables the patients to fit the hearing aid by themselves, is proposed. The algorithm takes the band gain as the fitting target and uses the patient’s subjective evaluation to iteratively update the algorithm parameters based on the improved IEC algorithm. In addition, a real-time updated expert system is constructed to assist in the optimization of the initial and iterative parameters of the fitting based on the patient’s audiogram and personal information. To verify the performance of the algorithm, a self-fitting software for the hearing aid is designed. Through this software, the test signal is generated for the patient to evaluate the audio quality on a five-level scale. Based on the evaluation results, the algorithm iteratively optimizes the algorithm parameters until the patient is satisfied with the generated audio. Compared with the fitting algorithm based on Gaussian processes algorithm or the interactive evolutionary algorithm, the average subjective speech recognition rate of the proposed algorithm increase at least 11%. The average recognition rate for environmental sound is also improved by at least 2.9%. In addition, the fitting time of the proposed algorithm is shortened by at least 10 min compared to others two algorithms.
10 citations
TL;DR: It is shown in this paper that matching error, delay and hardware complexity in terms of the number of multipliers required for the prototype filter are lower than the existing techniques for all types of hearing losses.
Abstract: This paper proposes the design of a simple digital hearing aid using non-uniform modified discrete Fourier transform filter bank. Here, the non-uniform filter bank is obtained by first designing a uniform filter bank and then merging suitable bands to obtain the desired non-uniform structure. In hearing aids, narrow channels with adjustable gains can be allocated to those frequency regions of the audiogram with sharp transition, whereas wider passband channels can be allocated to frequency regions with slower transitions. The latter can be done by merging the bands in those regions. It is shown in this paper that matching error, delay and hardware complexity in terms of the number of multipliers required for the prototype filter are lower than the existing techniques for all types of hearing losses.
10 citations
TL;DR: It is possible to conclude that measuring the hearing aid benefit with the self-assessment questionnaires will assist the clinicians in making judgments about the areas in which a patient is experiencing more difficulty in everyday listening environment and in revising the possible technologies.
Abstract: Introduction For many reasons, it is important for audiologists and consumers to document improvement and benefit from amplification device at various stages of uses of amplification device. Professional are also interested to see the impact of amplification device on the consumer's auditory performance at different stages i.e. immediately after fitting and over several months of use. Objective The objective of the study was to measure the hearing aid benefit following 6 months – 1-year usage, 1 year – 1.5 years usage, and 1.5 years – 2 years' usage. Methods A total of 45 subjects participated in the study and were divided equally in three groups: hearing aid users from 6 months to 1 year, 1 year to 1.5 year, and 1.5 year to two years. All subjects responded to the Hearing Aid Benefit Questionnaire (63 questions), which assesses six domains of listening skills. Result Results showed the mean scores obtained were higher for all domains in the aided condition, as compared with unaided condition for all groups. Results also showed a significant improvement in the overall score between first-time users with hearing aid experience of six months to one year and hearing aid users using hearing aids for a period between 1.5 and 2 years. Conclusion It is possible to conclude that measuring the hearing aid benefit with the self-assessment questionnaires will assist the clinicians in making judgments about the areas in which a patient is experiencing more difficulty in everyday listening environment and in revising the possible technologies.
7 citations
Patent•
24 May 2017
TL;DR: In this article, a BP-artificial-neural-network-based intelligent matching algorithm for a digital hearing aid is presented, in which an initial weight value and a threshold of a BP artificial neural network are optimized on the basis of the genetic algorithm principle.
Abstract: The invention discloses a BP-artificial-neural-network-based intelligent matching algorithm for a digital hearing aid. On the basis of a BP artificial neural network, a network is trained by using lots of training data to obtain a satisfactory mature network; and the network is corrected by using a self-built formula model, thereby obtaining a mature intelligent matching algorithm. On the basis of the genetic algorithm principle, an initial weight value and a threshold of a BP artificial neural network are optimized; the BP neural network is trained by using an existing audiogram and a spectrum gain response as raining data; and the network is corrected based on a matching formula model to obtain a mature BP artificial neural network to replace the existing matching prescription formula, so that parameters like all channel gains, the maximum sound output, the compression rate, and the compression inflection point of the digital hearing aid can be obtained.
6 citations
01 Sep 2017
TL;DR: A new processing scheme is developed that decomposes the signal into perceptually matched sliding bands and implements combined noise reduction and acoustic feedback suppression (AFS) and the proposed AFS algorithm is based on spectral subtraction rule.
Abstract: The paper presents an implementation of an improved smartphone-based hearing aid which originates from our free smartphone application “Petralex” released for iOS and Android devices. In the present contribution we develop a new processing scheme that decomposes the signal into perceptually matched sliding bands and implements combined noise reduction and acoustic feedback suppression (AFS). The proposed AFS algorithm is based on spectral subtraction rule. The algorithm is robust to rapid changes in acoustic feedback path and considerably increases maximum stable gain.
5 citations
01 Sep 2017
TL;DR: The functionality and technical performance of DSP hearing aids are more efficient compared to analog hearing aids, and they are self-adjusting and flexible.
Abstract: Hearing loss, also known as hearing impairment is a partial or total inability to hear. Noise is one of the cause of hearing loss. Hearing aid is an electroacoustic device which is designed to amplify sound. 5.3 percent of earth’s population suffer from hearing loss. The functionality and technical performance of DSP hearing aids are more efficient compared to analog hearing aids. Digital hearing aids are self-adjusting and flexible. In this way, the sound it transmits matches the specific pattern of hearing loss. Adaptive filtering is used which effectively cancels out the additive noise by changing the filter coefficients over time and adapts to the changing signal characteristics according to an optimization algorithm. Gain is selectively added at higher frequency according to the person’s hearing loss profile. This allows for a convenient, detailed control of the frequency response of the hearing aid. Finally the processed signal undergoes power check to assure that the signal power is less than the threshold of pain. A spectrogram which gives a representation of how the frequency content of a signal changes with time is plotted. The MATLAB provides graphics for visualizing data and toolboxes for a wide range of engineering and scientific applications.
5 citations
11 May 2017
TL;DR: A reconfigurable filter bank for digital hearing aid in Canonical Sign Digit Space is proposed based on fractional interpolation which enables the realization of the entire system with two different prototype filters, thus reducing the system complexity.
Abstract: In this paper, a reconfigurable filter bank for digital hearing aid in Canonical Sign Digit Space is proposed to reduce the complexity while implementing hardware. The major requirement of hearing aid is that it should be able to provide different sound decomposition schemes in-order to cope with the needs of different patients. In this paper, the filter bank is based on fractional interpolation which enables the realization of the entire system with two different prototype filters, thus reducing the system complexity. A 4-bit control signal is used to alter the subband schemes without modifying the configuration of the filter bank. The gain of each subband is then optimized using Minimax algorithm. The proposed filter bank can achieve satisfactory performance in audiogram matching which is shown by means of examples.
3 citations
Patent•
13 Jun 2017
TL;DR: In this article, a coarse and fine focusing audiometry method based on an intelligent cell phone is proposed, in which a digital hearing aid is used as an audio source of a pure tone test, and an instruction program for controlling the hearing aid and a human-machine interface comprising an operation prompt and patient response are operated on the intelligent phone.
Abstract: The invention discloses a rapid pure tone audiometry method based on an intelligent cell phone, namely a coarse and fine focusing audiometry method. A digital hearing aid is used as an audio source of a pure tone test; the intelligent cell phone is used as a human-machine interface and an audiometry program control; single-audio-frequency signals with different frequencies and different volumes are generated in auditory meatuses of patient with hearing damages; whether the patients hear sounds or feel uncomfortable or not is responded and analyzed to obtain a hearing threshold value and an uncomfortable threshold value which are used for describing hearing conditions of the patients. In a pure tone testing process, the human-machine interface comprising an operation prompt and patient response, and an instruction program for controlling the hearing aid are operated on the intelligent cell phone; a pure tone signal is generated by the digital hearing aid; the intelligent cell phone is used for controlling the hearing aid in a manner of being in wireless communication with the hearing aid, so that pure tones with different frequencies and different volumes are generated; the sound volumes which are heard by the patients or not and under which the patients feel uncomfortable or not are acquired and analyzed by the intelligent cell phone, so as to finish a hearing test.
Patent•
04 Jan 2017
TL;DR: In this paper, an adaptive fitting method of a digital hearing aid is proposed, in which a summation form of two-dimensional Gauss distribution is used as a fitting model, and interactive evolutionary computation is employed as an optimization method, and the fitting model can be determined by human-computer interaction within 30 generations.
Abstract: The invention discloses an adaptive fitting method of a digital hearing aid. The fitting process of the traditional hearing aid cannot be realized without a hearing expert, the fitting result directly depends on the professional level of the hearing expert, in addition, for developing countries, the medical conditions of a considerable part of regions are not up to the fitting standard, but adaptive fitting of the hearing aid can be effectively realized by adoption of the method, namely the fitting process of the hearing aid is realized without the hearing expert. According to the method disclosed by the invention, a summation form of two-dimensional Gauss distribution is used as a fitting model, interactive evolutionary computation is used as an optimization method, and the fitting model of the digital hearing aid can be determined by human-computer interaction within 30 generations in combination with a human-computer interaction interface designed in the method disclosed by the invention. Experimental results indicate that the adaptive fitting of the digital hearing aid can be effectively realized by the method provided by the invention, thereby getting rid of the limitation of the hearing expert to the fitting process of the digital hearing aid.
Journal Article•
TL;DR: In bimodal group, aural/oral performance was significantly improved in quiet and noise situations in comparison with unilateral group, due to the advantage of binaural processing and low frequency information provided by the hearing aid.
Abstract: Background and Aim: Sound processors in cochlear implant (CI) cannot encode low frequency information and discard much of the temporal fine structure required to perceive fundamental frequency Hearing aids can transmit low frequency information, which is important for pitch perception and provides many advantages for the users This study aimed to compare aural/oral performance of bimodal cochlear implants with unilateral ones in children using parents' evaluation of aural/oral performance of children (PEACH) questionnaire Methods: Twenty children with unilateral cochlear implant and 20 ones with bimodal cochlear implants were selected for this study Of them, 23 had cochlear devices, 10 possessed Med-El ones, and 7 wore advanced bionics ones Bimodal group had at least 7 months of hearing experience with digital hearing aid in non-implanted ear In order to compare the aural/oral performance in these groups, we used the PEACH questionnaire Results: In unilateral and bimodal groups, age of implantation and age of testing and hearing experience before CI use were not significantly different However, there was a significant difference in quiet score, noise score, and total score between unilateral and bimodal groups (p<005) Conclusion: In bimodal group, aural/oral performance was significantly improved in quiet and noise situations in comparison with unilateral group This improvement is due to the advantage of binaural processing and low frequency information provided by the hearing aid
TL;DR: DHA fittings were generally successful, as most patients (1,137/1,597 [71.2%]) did not require any follow-up appointment during the study period, and only 41 of the 1,597 first or new fittings (2.6%) were considered unsatisfactory by patients and necessitated a follow- up reprogramming appointment.
Abstract: We conducted a retrospective study to determine the success rate of initial fittings in digital hearing aid (DHA) users. We also addressed the implications of national health systems’ continuing to...
TL;DR: This research paper expounds microcontroller based binaural digital hearing aids for hearing-impaired people by making use of ATmega328 microcontroller and other circuitries to process the audio signal input by either increasing or reducing the gain level of input audio signal, filter background noise, frequencies compression, save battery power and minimize circuit.
Abstract: This research paper expounds microcontroller based binaural digital hearing aids for hearing-impaired people by making use of ATmega328 microcontroller and other circuitries to process the audio signal input by either increasing or reducing the gain level of input audio signal, filter background noise, frequencies compression, save battery power and minimize circuit by making use of the internal ADC of the microcontroller and two PMW pins of the microcontroller as DAC. Hearing impairment among the youths and adults nowadays are in the increase, due wrong use of phones of which every minute of the day someone’s earphone is on listening to one type of music or the other. In other to solve the problem created so to say this research work was conceived and given birth to. The different stages of digital hearing aid are designed and then simulated first in Proteus software which then was implemented using PCB-board. The main components of this system were the audio input unit which consists of the microphone and its pre-amplifier, the microcontroller (ATmega328) which consists of the ADC, the DAC and the audio signal processing, the filter stage and control codes (frequencies compression codes, power saver codes, acoustic feedback control codes, signal level control and adaptive adjustment codes etc.), the power amplifier and volume control unit and then the earphones (output). The control codes were written in C language while Ardinuo Uno compiler was used to write the codes into ATmega328. The prototype has an overall system gain of 27dB and the power output of 32.5mW. The prototype was tested with a patient that has a hearing impairment and the patient was satisfactory with the device. http://dx.doi.org/10.4314/njt.v36i3.34
TL;DR: It is suggested that the use of bimodal fitting can be considered as an effective management option in order to obtain binaural hearing benefits in children who undergo unilateral cochlear implantation.
Abstract: Introduction: There are clear benefits of having bilateral inputs to the auditory nervous system. Hearing-impaired children are, therefore, generally fitted with two hearing aids so that they can benefit from hearing binaurally. Children who use a cochlear implant in one ear and no amplification in the opposite ear are, however, deprived of these advantages. The current study was undertaken to determine the benefits of bimodal stimulation in pediatric population. Methods and Materials: This study comprised of 20 children between 6-11 years of age with profound bilateral sensorineural hearing loss with cochlear implant in one ear and fitted with digital hearing aid in non-implanted ear. Mean sound localization score was compared in children with cochlear implant only and those with both cochlear implant and hearing aid. Result: A statistically significant difference was found between mean sound localization in both test condition under quiet surrounding. Conclusion: Hence it is suggested that the use of bimodal fitting can be considered as an effective management option in order to obtain binaural hearing benefits in children who undergo unilateral cochlear implantation.
TL;DR: An improved perceptual multiband spectral subtraction (MBSS) noise reduction algorithm (NRA) and a novel robust voice activity detection (VAD) based on the amended sub-band SNR, which can considerably increase the accuracy of discrimination between noise and speech frame.
Abstract: This paper first reviews the state-of-the-art noise reduction methods and points out their vulnerability in noise reduction performance and speech quality, especially under the low signal-noise ratios (SNR) environments. Then this paper presents an improved perceptual multiband spectral subtraction (MBSS) noise reduction algorithm (NRA) and a novel robust voice activity detection (VAD) based on the amended sub-band SNR. The proposed SNR-based VAD can considerably increase the accuracy of discrimination between noise and speech frame. The simulation results show that the proposed NRA has better segmental SNR (segSNR) and perceptual evaluation of speech quality (PESQ) performance than other noise reduction algorithms especially under low SNR environments. In addition, a fully operational digital hearing aid chip is designed and fabricated in the 0.13 μm CMOS process based on the proposed NRA. The final chip implementation shows that the whole chip dissipates 1.3 mA at the 1.2 V operation. The acoustic test result shows that the maximum output sound pressure level (OSPL) is 114.6 dB SPL, the equivalent input noise is 5.9 dB SPL, and the total harmonic distortion is 2.5%. So the proposed digital hearing aid chip is a promising candidate for high performance hearing-aid systems. key words: digital hearing aid, noise reduction, multiband spectral subtraction, voice activity detection
Patent•
21 Nov 2017
TL;DR: In this paper, a noise elimination method based on a generalized sidelobe canceller in a digital hearing aid is proposed, where a wavelet threshold denoising method and a beamforming algorithm are effectively combined, and coherent noise in the signals is eliminated.
Abstract: The invention discloses a noise elimination method based on a generalized sidelobe canceller in a digital hearing aid According to the invention, a wavelet threshold denoising method and a beamforming algorithm are effectively combined, wavelet threshold denoising is performed on each path of signals of a microphone array at first, and coherent noise in the signals is eliminated so as to reduce time delay errors and leakage signals caused by the noise In the wavelet threshold denoising algorithm, a threshold function directly determines the quality of an enhancement effect, so that an improved continuous threshold function with adjustment parameters is proposed The threshold function processes a wavelet coefficient of noisy speech signals in a wavelet domain, and acquires speech signals with the coherent noise being eliminated after reconstruction processing An adaptive beamformer of the generated sidelobe canceller adopts an improved LMS algorithm and improves the update speed of the weight coefficient The result of a simulation test shows that the method effectively improves the signal-to-noise ratio and the intelligibility of speech signals compared with the traditional generalized sidelobe canceller