Showing papers on "Digital hearing aid published in 2016"
TL;DR: General system-identification analyses of chinchilla auditory nerve fiber responses to Gaussian noise are used to uncover pronounced distortions in coding of rapidly varying acoustic temporal fine structure and slower envelope cues following noise trauma.
Abstract: People with cochlear hearing loss have substantial difficulty understanding speech in real-world listening environments (e.g., restaurants), even with amplification from a modern digital hearing aid. Unfortunately, a disconnect remains between human perceptual studies implicating diminished sensitivity to fast acoustic temporal fine structure (TFS) and animal studies showing minimal changes in neural coding of TFS or slower envelope (ENV) structure. Here, we used general system-identification (Wiener kernel) analyses of chinchilla auditory nerve fiber responses to Gaussian noise to reveal pronounced distortions in tonotopic coding of TFS and ENV following permanent, noise-induced hearing loss. In basal fibers with characteristic frequencies (CFs) >1.5 kHz, hearing loss introduced robust nontonotopic coding (i.e., at the wrong cochlear place) of low-frequency TFS, while ENV responses typically remained at CF. As a consequence, the highest dominant frequency of TFS coding in response to Gaussian noise was 2.4 kHz in noise-overexposed fibers compared with 4.5 kHz in control fibers. Coding of ENV also became nontonotopic in more pronounced cases of cochlear damage. In apical fibers, more classical hearing-loss effects were observed, i.e., broadened tuning without a significant shift in best frequency. Because these distortions and dissociations of TFS/ENV disrupt tonotopicity, a fundamental principle of auditory processing necessary for robust signal coding in background noise, these results have important implications for understanding communication difficulties faced by people with hearing loss. Further, hearing aids may benefit from distinct amplification strategies for apical and basal cochlear regions to address fundamentally different coding deficits. SIGNIFICANCE STATEMENT Speech-perception problems associated with noise overexposure are pervasive in today9s society, even with modern digital hearing aids. Unfortunately, the underlying physiological deficits in neural coding remain unclear. Here, we used innovative system-identification analyses of auditory nerve fiber responses to Gaussian noise to uncover pronounced distortions in coding of rapidly varying acoustic temporal fine structure and slower envelope cues following noise trauma. Because these distortions degrade and diminish the tonotopic representation of temporal acoustic features, a fundamental principle of auditory processing, the results represent a critical advancement in our understanding of the physiological bases of communication disorders. The detailed knowledge provided by this work will help guide the design of signal-processing strategies aimed at alleviating everyday communication problems for people with hearing loss.
40 citations
TL;DR: This paper proposes the use of a variable bandwidth filter, using Farrow subfilters, for this purpose, and results show that lower order filters and better audiogram matching with lesser matching errors are obtained using F arrow structure.
34 citations
TL;DR: Here, the strength of hybrid evolutionary algorithms is explored and their various combinations are compared to select a proper coefficient representation for the Farrow based filter, which results in low complexity implementation.
22 citations
Patent•
06 Jul 2016
TL;DR: In this article, a voice enhancement method for fusing phase estimation and human ear hearing characteristics in a digital hearing aid was proposed, comprising steps of obtaining a frequency domain expression mode containing noise through Fourier transformation, adopting a minimum value control recursive average method to obtain a noise power spectrum, obtaining the initial enhancement voice and noise through correcting the phase of the voice and the noise through improving the phase estimation of voice distortion under the low signal-to-noise ratio environment, performing filtering processing on the initial voice and background noise through a gammatone filter assembly which simulates
Abstract: The invention discloses a voice enhancement method for fusing phase estimation and human ear hearing characteristics in a digital hearing aid, comprising steps of obtaining a frequency domain expression mode containing noise through Fourier transformation, adopting a minimum value control recursive average method to obtain a noise power spectrum, obtaining initial enhancement voice and a noise amplitude spectrum, obtaining the initial enhancement voice and noise through correcting the phase of the voice and the noise through improving the phase estimation of the voice distortion under the low signal-to-noise ratio environment, performing filtering processing on the initial enhancement voice and the noise through a gammatone filter assembly which simulates the working mechanism of the artificial cochlea, performing analysis on the time frequency of the gammatone filter assembly to obtain the time frequency expression form consisting of time frequency units, using the hearing characteristics of the human ear to calculate binary mast containing noise in the time frequency domain, and using the mask value to obtain the enhanced voice after synthesis.
13 citations
Patent•
25 May 2016
TL;DR: In this paper, a depth and breadth neural network combined speech enhancement algorithm of a digital hearing aid belonging to the speech signal processing technical field is presented. But the method comprises following steps: first, carrying out speech activity detection to noise contained speech signals; extracting features such as autocorrelation function maximum values and variances of frequency bands of the noise contained audio signals; building a two-value decision device by using a BP neural network; judging speech segments and noise segments; extracting MFCC and first level MFCC features; detecting noise types by a depth neural network, wherein
Abstract: The invention discloses a depth and breadth neural network combined speech enhancement algorithm of a digital hearing aid belonging to the speech signal processing technical field The method comprises following steps: firstly, carrying out speech activity detection to noise contained speech signals; extracting features such as autocorrelation function maximum values and variances of frequency bands of the noise contained speech signals; building a two-value decision device by using a BP neural network; judging speech segments and noise segments; secondly, extracting MFCC and first level MFCC features; detecting noise types by a depth neural network, wherein the depth neural network is formed by cascading a learning vector quantization neural network and the BP neural network; finally, building a breadth neural network formed by connecting various networks in parallel; automatically selecting corresponding neural networks by the breadth neural network according to the noise types; removing the noises, thus obtaining the enhanced speech and improving the speech intelligibility of the hearing aid output speech According to the algorithm, the training processes of the neural networks are finished offline; the complexity of the test algorithm of the trained networks is low; and therefore, the real timeliness is satisfied
12 citations
TL;DR: Low-cut modified amplification improved aided speech perception compared to standard NAL-NL1 prescriptive amplification in individuals with ANSD and the use of low-cut amplification with a low number of channels in the hearing aid for individuals with AnSD is recommended.
Abstract: Objective: The present study was conducted to determine if low-cut modified amplification improved aided speech perception compared to standard NAL-NL1 prescriptive amplification in individuals with ANSD. The study further aimed to check the influence of the number of channels in a hearing aid on speech perception with low-cut modification of amplification. Study design: A total of 22 individuals with ANSD in the age range 15–42 years were recruited for the study. The unaided and aided speech identification scores were obtained with standard amplification and low-cut amplification settings in both a four- and a 16-channel digital hearing aid. Results: The results showed that low-cut amplification was slightly better than standard amplification for aided speech perception. Such an improvement could be attributed to the elimination of upward spread of masking during low-cut modification of amplification. In addition, improvement was greater with a four-channel hearing aid compared to a 16-channel hearing ai...
8 citations
TL;DR: The design and implementation of digital hearing aids requires a detailed knowledge of various digital signal processing techniques used in hearing aids like Wavelet Trans-forms, uniform and non-uniform Filter Banks and Fast Fourier Transform.
Abstract: The design and implementation of digital hearing aids requires a detailed knowledge of various digital signal processing techniques used in hearing aids like Wavelet Trans-forms, uniform and non-uniform Filter Banks and Fast Fourier Transform (FFT). In this paper the design and development of digital part of hearing aid is divided into three different phases. In the first phase review and Matlab simulation of various signal processing techniques used in the digital hearing aids is presented. In the second phase a software implementation was carried out and the firmware was designed for the Xilinx Microblaze softcore processor system. In the third phase everything was moved into hardware using VHDL hardware description language. The implementation was done on Xilinx Field Programmable Gate Array (FPGA) Development Board.
8 citations
Patent•
09 Mar 2016
TL;DR: In this article, a digital hearing aid voice noise elimination method is proposed, which mainly comprises the following steps: S1) carrying out subband division and framing on sampled signals of input voice to obtain each frame of sub-band noise speech signal; S2) calculating a gain function of each frame.
Abstract: The invention relates to the technical field of voice signal processing and especially relates to a digital hearing aid voice noise elimination method. The method mainly comprises the following steps: S1) carrying out sub-band division and framing on sampled signals of input voice to obtain each frame of sub-band noise speech signal; S2) calculating a gain function of each frame of sub-band noise speech signal; S3) damping each frame of sub-band noise speech signal according to the gain function to obtain each frame of sub-band enhancing signal; and S4) enabling the sub-band enhancing signals to pass through synthesis filter banks and be added to obtain noise-reduced voice and then, outputting the noise-reduced voice. According to the method, computation complexity is reduced, and thus time delay and power consumption is reduced; and meanwhile, an improved algorithm enables signal-to-noise ratio of the noise-reduced voice to be improved by more than 5 dB, so that noise in the speech signals is effectively suppressed, intelligibility of the speech is improved and higher practical value is achieved.
7 citations
04 Mar 2016
TL;DR: In this article, a sliding-band compression algorithm is proposed for low-level sounds in a smartphone app for use as a hearing aid, which is realized using FFTbased analysis-synthesis and can be integrated with other such techniques for computational efficiency.
Abstract: Listeners with sensorineural hearing loss have degraded speech perception due to frequency-dependent elevation of hearing thresholds, reduced dynamic range, and increased temporal and spectral masking. Signal processing in hearing aids for such listeners uses frequency-selective amplification and dynamic range compression for restoring normal loudness of low-level sounds without making the high-level sounds uncomfortably loud. Sliding-band compression has been reported earlier for reducing the temporal and spectral distortions generally associated with currently used single and multiband compression techniques. The paper presents implementation of sliding-band compression as a smartphone app for use as a hearing aid. The processing involves a frequency-dependent gain function calculated on the basis of critical bandwidth based short-time power spectrum and the specified hearing thresholds, compression ratios, and attack and release times. It is realized using FFT-based analysis-synthesis and can be integrated with other such techniques for computational efficiency.
6 citations
03 Mar 2016
TL;DR: In this article, the simulation of the simple digital hearing aid was developed using MATLAB programming language and the implementation of this configurable digital hearing aids (DHA) system includes noise reduction filter, frequency-dependent amplification and amplitude compression.
Abstract: A great proportion of human population suffers from hearing loss. Hearing loss is a measure of shift in auditory system compared to that of a normal ear for detection of a pure tone. But with the availability of modern day technologies and the recent developments in signal processing area, sophisticated artificial hearing aid systems can be designed that relax the job of damaged auditory systems to a great extent and make much of the sound available to the hearing impaired. In this project, the simulation of the simple digital hearing aid was developed using MATLAB programming language. The implementation of this configurable digital hearing aid (DHA) system includes noise reduction filter, frequency-dependent amplification and amplitude compression. We tested our filters on a mock patient and successfully reduced white Gaussian noise, increased the gain for frequencies which were difficult to hear, and shaped the amplitude to prevent any of the frequencies from becoming too loud. Finally, future trends and expected innovations in the hearing aid industry are discussed.
5 citations
Patent•
15 Jun 2016
TL;DR: In this article, the authors proposed a digital hearing aid noise reduction method based on improved sub-band signal-to-noise ratio estimation, the method comprises the steps: a decomposition filter decomposes an original signal into a plurality of sub-bands; a cross-correlation function and each mean square value of two adjacent frame signals in each subband is calculated, so that a signal to noise ratio of the sub band is estimated, the gain of each sub Band is calculated according to the estimated signal to noise ratio, and the gain multiplied by a sub Band signal
Abstract: The invention provides a digital hearing aid noise reduction method based on improved sub-band signal-to-noise ratio estimation, the method comprises the steps: a decomposition filter decomposes an original signal into a plurality of sub-bands; a cross-correlation function and each mean square value of two adjacent frame signals in each sub-band is calculated, so that a signal-to-noise ratio of the sub-band is estimated; gain of each sub-band is calculated according to the estimated signal-to-noise ratio, and the gain multiplied by a sub-band signal is a corrected sub-band signal; finally, the corrected sub-band signals are combined to obtain a noise reduction processed voice. According to the invention, the accurater sub-band signal-to-noise ratio estimation method makes the background noise inhibitory effect being better, and auditory fatigue of hearing-aid users is reduced; the method is simple and has high efficiency, inverse Fourier transform is avoided, so that the time delay performance is greatly improved, the method improved for 60.6% from a traditional spectral subtraction, and the method improved for 40.7% from a traditional Wiener filtering method.
Patent•
06 Jan 2016
TL;DR: In this paper, a wide dynamic compression method of a digital hearing aid based on sound pressure level segmentation is proposed, where the sound pressure is thinned in 8 segments to obtain more accurate input/output curves; and the 6-order IIR decomposition synthesis filter bank conforming to human hearing characteristics can be used for obtaining a compensation gain value better conformed to the actual demand of the patient.
Abstract: The invention discloses a wide dynamic compression method of a digital hearing aid based on sound pressure level segmentation, wherein the wide dynamic compression method comprises the following steps: framing a voice signal, and filtering the framed signal through a 16-channelnonuniform spaced 6-order IIR decomposition filter bank; then, calculating the sound pressure levels of the voice signals of the channels, and obtaining a hearing compensation curve of a patient in combination with an audiogram of the patient; carrying out channel-separated hearing compensation on the patient according to the hearing compensation curve, integrating compensated multichannel signals to obtain a compensated useful signal, and providing the useful signal for the patient. The wide dynamic compression method disclosed by the invention has the beneficial effects that, the sound pressure level is thinned in 8 segments to obtain more accurate input/output curves; and the 6-order IIR decomposition synthesis filter bank conforming to human hearing characteristics can be used for obtaining a compensation gain value better conforming to the actual demand of the patient.
12 Dec 2016
TL;DR: A digital hearing aid is implemented as an Android-based smartphone app that uses sliding-band dynamic range compression for restoring normal loudness of low-levelSounds without making the high-level sounds uncomfortably loud and for reducing the perceptible temporal and spectral distortions associated with currently used single and multiband compression techniques.
Abstract: Persons with sensorineural hearing loss suffer from degraded speech perception caused by frequency-dependent elevation of hearing thresholds, reduced dynamic range and abnormal loudness growth, and increased temporal and spectral masking. For improving speech perception by persons with moderate loss of this type, a digital hearing aid is implemented as an Android-based smartphone app. It uses sliding-band dynamic range compression for restoring normal loudness of low-level sounds without making the high-level sounds uncomfortably loud and for reducing the perceptible temporal and spectral distortions associated with currently used single and multiband compression techniques. The processing involves application of a frequency dependent gain function calculated on the basis of critical bandwidth based short-time power spectrum and is realized using FFT-based analysis-synthesis. The implementation has a touch-controlled graphical user interface enabling the app user to fine tune the frequency-dependent parameters in an interactive and real-time mode.
15 Jun 2016
TL;DR: The proposed design adopts an efficient prescription-fitting algorithm to reduce inter-band interference, enabling the proposed quasi-ANSI filter bank to compensate any type of hearing loss using the NAL-NL1 or HSE prescription formulas.
Abstract: The ANSI S1.11 1/3-octave filter bank is suitable for digital hearing aids, but its large group delay and high compu- tational complexity complicate matters considerably. This study presents a 10-ms 18-band quasi-ANSI S1.11 1/3-octave filter bank for processing 24 kHz audio signals. We first discuss a filter order optimization algorithm to define the quasi-ANSI filters. The group delay constraint of filters is limited to 10 ms. The proposed design adopts an efficient prescription-fitting algorithm to reduce inter-band interference, enabling the proposed quasi-ANSI filter bank to compensate any type of hearing loss (HL) using the NAL-NL1 or HSE prescription formulas. Simulation results re- veal that the maximum matching error in the prescriptions of the mild HL, moderate HL, and severe-to-profound HL is less than 1.5 dB. This study also investigates the complexity-effective multirate IFIR quasi-ANSI filter bank. For an 18-band digital hearing aid with a 24 kHz sampling rate, the proposed architecture eliminates approximately 93% of the multiplications and up to 74% of the storage elements, compared with a parallel FIR filters architec- ture. The proposed analysis filter bank (AFB) was designed in UMC 90 nm CMOS high-VT technology, and on the basis of post-layout simulations, it consumes 73 W. By voltage scaling (to 0.6 V), the simulation results show that the power consumption decreases to 27 W, which is approximately 30% of that consumed by the most energy-efficient AFB available in the literature for use in hearing aids.
01 Apr 2016
TL;DR: A technique to generate additive white gaussian noise (AWGN) or other types of filtered noise using coefficients stored in program memory of the DSP is proposed and an implementation of the technique is carried out on a dsPIC from Microchip®.
Abstract: Treatment of tinnitus by means of masking sounds allows to obtain a significant improve of the quality of life of the individual that suffer that condition. In view of that, it is possible to develop noise synthesizers based on random number generators in digital signal processors (DSP), which are used in almost any digital hearing aid devices. DSP architecture have limitations to implement a pseudo random number generator, due to it, the noise statistics can be not as good as expectations. In this paper, a technique to generate additive white gaussian noise (AWGN) or other types of filtered noise using coefficients stored in program memory of the DSP is proposed. Also, an implementation of the technique is carried out on a dsPIC from Microchip®. Objective experiments and experimental measurements are performed to analyze the proposed technique.
TL;DR: A technique to generate adittive white gaussian noise (AWGN) or noise with uniform distribution using coefficients stored in memory of the DSP program is proposed.
Abstract: Treatment of tinnitus with masking sounds has reach a significant developed in recent years. It is mainly because it has been possible to implement noise sinthesizers based on random number generators in digital signal processors (DSP), which form a part of almost any digital hearing aid device. One limitation of these methods is that limitations of the DSP architecture prevent pseudo white noise of being generated conform to a real white noise statistics. In this paper, a technique to generate adittive white gaussian noise (AWGN) or noise with uniform distribution using coefficients stored in memory of the DSP program is proposed. An implementation of the technique is carried out on a dsPIC from Microchip and subjective experiments and experimental measurements are performed to validate the performance of the developed technique.
Patent•
15 Jun 2016
TL;DR: In this article, a digital hearing aid having a wireless receiving function is described, which consists of at least one hearing aid module and a wireless relay capable of being connected to at least 1 external device.
Abstract: In the present invention, disclosed is a digital hearing aid having a wireless receiving function. According to the present invention, the digital hearing aid comprises: at least one hearing aid module; and a wireless relay capable of being connected to at least one external device which can connect wireless communication with each hearing aid module and can connect wireless communication with at least one external device. The wireless relay is wireless communication connected to at least one external device to receive sound data from at least one selected external device. The wireless relay sends the sound data to at least one hearing aid module which is wireless communication connected. At least one hearing aid module receives the sound data and converts the sound data into voice data to provide the converted voice data to a wearer.
01 Nov 2016
TL;DR: This paper proposes an implementation for a bit-serial multiplier for DSP in a hearing aid with high working frequency and low power consumption, and adopts the Booth encoding to reduce the number of partial products in multiplication and reduce the calculation time and power consumption.
Abstract: An increase of half-hearing person caused by progressive aging of society in our country leads to an increase in demand for a digital hearing aid with a DSP. Because of a hard physical limit for battery capacity which stems from its wearing form, the battery life of an existing digital hearing aid comes up to only about few days. In this paper, we proposed an implementation for a bit-serial multiplier for DSP in a hearing aid with high working frequency and low power consumption. To reduce the power consumption associated with clock generation, we use a ring oscillator to dynamically generate clock pulse only in the period of calculation. In addition, we adopt the Booth encoding to reduce the number of partial products in multiplication and reduce the calculation time and power consumption associated with it. We implement the proposed multiplier and show the effectiveness of it through the comparison experiments.
01 Oct 2016
TL;DR: In this article, a review of basics of hearing aid, challenges faced by hearing impaired and various filters or speech compression techniques used to enhance speech is presented. But this review is limited to hearing aid.
Abstract: Understanding of speech in noisy environment is great challenge to hearing impaired even with hearing aid. To enhance speech in noisy background various noise reduction algorithms used in digital hearing aid. With noise reduction power consumption and memory optimization considering size is difficult. Selection of algorithms reduces complexity as well as power and memory requirement. This paper take review of basics of hearing aid, challenges faced by hearing impaired and various filters or speech compression techniques used to enhance speech.
01 Oct 2016
TL;DR: The article will tell echo cancelation principle, introduce several common algorithms, such as RLS algorithm, LMS algorithm and NLMS algorithm, and discuss their advantages and disadvantages, and puts forward a kind of algorithm which is a combination of NLMS algorithms, and BL MS algorithm, named NBLMS_M-K.
Abstract: With the aging of the population increase, the need of hearing aids is increasing, along with the wide application of digital signal processing technology, applied to adaptive echo cancellation algorithms in digital hearing aid has been significantly improved and developed. The article will to tell echo cancelation principle, to introduce several common algorithms, such as RLS algorithm, LMS algorithm and NLMS algorithm, and discuss their advantages and disadvantages, puts forward a kind of algorithm which is a combination of NLMS algorithm, and BLMS algorithm, named NBLMS_M-K. Finally, discuss the limitations of the adaptive echo cancellation algorithms in digital hearing aid and the future development of algorithm.
Patent•
06 Jul 2016
TL;DR: In this paper, the authors proposed a digital hearing-aid self-adaptive sound feedback elimination method, which is based on a zero-attracting-least mean norm algorithm.
Abstract: The invention provides a digital hearing-aid self-adaptive sound feedback elimination method. The processing procedures of the digital hearing-aid self-adaptive sound feedback elimination method are as follows: firstly, obtaining a signal received by a digital hearing-aid; then obtaining an ideal output signal based on a known input signal and calculating an error between the practically received signal and an ideal received signal; and finally carrying out iteration on each discrete time point by applying a zero-attracting-least mean norm algorithm, and carrying out estimation and updating adjustment on an unknown sound feedback path. Compared with the prior art, the digital hearing-aid self-adaptive sound feedback elimination method has the following advantages that influence on a sound feedback elimination effect is reduced; better compromise in the aspects of convergence rate and estimation accuracy is achieved, and computation complexity is reduced greatly; and robustness and adaptability of the digital hearing-aid can be improved efficiently.
04 Oct 2016
TL;DR: An implementation of digital hearing aid based on the mobile computing platform iPhone has a low delay, takes into account characteristics of the computing platform and allows to perform the correction of sensorineural hearing loss.
Abstract: An implementation of digital hearing aid based on the mobile computing platform iPhone is proposed. The developed signal processing scheme has a low delay, takes into account characteristics of the computing platform and allows to perform the correction of sensorineural hearing loss. The cor-rection is carried out by linear frequency-dependent amplification and wideband dynamic range com-pression of the signal. In order to take into consideration magnitude characteristic of loudspeakers, the audiometry is performed using iPhone directly. The experimental results of with proposed hearing aid are given.
Patent•
24 Feb 2016
TL;DR: In this paper, the authors proposed a frequency resolution enhancing method for a digital hearing aid, which mainly comprises the following steps of: S1, obtaining a frequency differential threshold and a frequency value of the frequency resolution decreasing frequency point of a patient; S2, according to the frequency differential thresholds and the frequency value, determining a frequency stretching area and an area in which the frequency compression area needs to be enhanced; and S3, carrying out non-linear frequency stretching on the frequency stretch area of voice signals input into the digital hearing aids.
Abstract: The invention relates to the technical field of voice signal processing and especially relates to a frequency resolution enhancing method of a digital hearing aid The frequency resolution enhancing method mainly comprises the following steps of: S1, obtaining a frequency differential threshold and a frequency value of a frequency resolution decreasing frequency point of a patient; S2, according to the frequency differential threshold and the frequency value, determining a frequency stretching area and a frequency compression area in which the frequency resolution needs to be enhanced; and S3, carrying out non-linear frequency stretching on the frequency stretching area of voice signals input into the digital hearing aid, and carrying out non-linear frequency compression on the frequency compression area According to the invention, signal frequency intervals of the areas in which the frequency resolution decreases are increased, the frequency sensitivity of the patient is improved, the frequency resolution of the voice signals input into the digital hearing aid is improved, and the language identification capability and the language communication capability of the patient are further improved
30 Apr 2016
TL;DR: This proposed paper represents an efficient FIR filter design using adaptive algorithm to match the audiogram with minimum errors with the filter coefficients so that the signal to noise ratio (SNR) is increased and noise in minimized.
Abstract: ISSN: 2231-5381 http://www.ijettjournal.org Page 106 Abstract: Finite impulse response (FIR) filters and filter banks contains specific properties as good stability as well as linear-phase can be easily achieved. hence, they are popular in many applications such as communication systems, audio signal processing, biomedical instruments. Based on these properties we can implement FIR filter bank design in Digital Hearing Aid processing. Most of the currently available hearing aid designs provide the filter bank with fixed bands (uniform or nonuniform). Thus the patients unable to take the full advantage to improve their specific auditive performance by using the hearing aid with limited number of fixed bands. This reduces the potential flexibility in matching of hearing loss with steeply sloping audiograms. One method of improving the same is to use an instrument with higher number of frequency bands for matching the audiogram with minimum matching error. This proposed paper represents an efficient FIR filter design using adaptive algorithm to match the audiogram with minimum errors with the filter coefficients so that the signal to noise ratio(SNR) is increased and noise in minimized.
15 Mar 2016
TL;DR: It is concluded that the socio-economic status of the hearing aid user might be playing an important role in gaining hearing aid knowledge as the analog user mostly belongs to lower socio- economic status with intellectual inefficiency than the digital hearing aid users.
Abstract: Background and Objective: Clinical experience indicates audiologists typically spend at least 15–20 min for a hearing aid orientation (HAO). Audiologists demonstrate the viewpoint that the information disseminated during an HAO is important for patients to learn and remember in order to use hearing aids effectively and independently. The present study was an investigation of new hearing aid users’ ability to remember the information presented in a typical HAO session. The aim of the study was to compare between new analog and digital hearing aid users for recognition of the hearing aid orientation.
Method: In the present study, 57 subjects within the age range of 35 to 65 years were selected for administering the Hearing Aid Knowledge Inventory (HAKI; Reese, 2001) after hearing aid orientation.
Result: Result showed that the digital hearing aid users were better oriented towards the program, showing greater percentage of people following the prescribed procedure for hearing aid maintenance and proper utilization compared to analog users.
Conclusion: From the present findings we can conclude, that the socio-economic status of the hearing aid user might be playing an important role in gaining hearing aid knowledge as the analog user mostly belongs to lower socio-economic status with intellectual inefficiency than the digital hearing aid user.
Key Words: HAKI, HAO, Analog, Digital, Hearing aid users
Patent•
09 Jun 2016
TL;DR: In this paper, a low-complexity adjustable filter bank for a digital hearing aid and an operating method therefor is presented. The filter bank comprises a masking module and a multi-band generation module which are connected in sequence, wherein the masking modules are used for dividing received voice signals into three regions, i.e. a low frequency region, a medium frequency region and a high frequency region in accordance with frequencies, and the multiband generator module is used for giving three different subband decomposition states to each of the three regions.
Abstract: A low-complexity adjustable filter bank for a digital hearing aid and an operating method therefor. The filter bank comprises a masking module and a multi-band generation module which are connected in sequence, wherein the masking module is used for dividing received voice signals into three regions, i.e. a low frequency region, a medium frequency region and a high frequency region in accordance with frequencies, and the multi-band generation module is used for giving three different subband decomposition states to each of the three regions, i.e. the low frequency region, the medium frequency region and the high frequency region, and then outputting subband signals. The three different subband decomposition states comprise: state 1: decomposing each region into subbands of which the bandwidth is π/3; state 2: uniformly decomposing each region into two subbands of which the bandwidth is π/6; and state 3: uniformly decomposing each region into four subbands of which the bandwidth is π/12. The method is low in complexity and small in delay, so that the division states of frequency bands can be changed by controlling parameters in the case where the structure of the filter bank is not changed, thereby achieving the goal of performing voice decomposition according to hearing loss characteristics of a patient.
Patent•
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13 Jan 2016
TL;DR: In this paper, a digital hearing aid has been provided with a transmission wire, the ingenious couples together ear mold device and long and thin structural devices, and new digital audio aid has contained duct formula and ear-level hearing aid advantage between them simultaneously, provides for the user and has used and test the convenience of joining in marriage, and the detachable connected mode has also made things convenient for carrying of user.
Abstract: The utility model discloses a digital hearing aid, include, wear in the ear mold device of duct department and with the long and thin structural devices that the ear mold device is connected, wherein, long and thin structural devices includes hard of hearing partial component and transmission wire, the transmission wire has first plug and second plug, be equipped with first socket and second socket on ear mold device and the hard of hearing partial component respectively, first plug and second plug can coordinate respectively and insert in first socket and the second socket, realize that ear mold device and long and thin structural devices can dismantle the connection, wherein, transmission wire still cladding has toughness and flexible protection tube. The utility model discloses a be provided with a transmission wire, the ingenious couples together ear mold device and long and thin structural devices, and new digital hearing aid has contained duct formula and ear -level hearing aid advantage between them simultaneously, provides for the user and has used and test the convenience of joining in marriage, and the detachable connected mode has also made things convenient for carrying of user.