Showing papers on "Digital hearing aid published in 2004"

Otologics middle ear transducer™ ossicular stimulator: Performance results with varying degrees of sensorineural hearing loss

Abstract: Objectives This study was conducted to demonstrate the safety and efficacy of the Otologics Middle Ear Transducer™ (MET™) Ossicular Stimulator and, in particular, to compare the audiologic benefits of this novel form of electromechanical stimulation with those of conventional acoustical amplification. Material and Methods A total of 282 patients were implanted with the device in Europe and the USA. Pure-tone audiometry, speech recognition and subjective assessment of benefit were tested before the surgery and 2, 3, 6 and 12 months afterwards. The US patients were fitted with a digital hearing aid for a minimum of 4 weeks prior to surgery, and the same benefit measures were performed with the digital hearing aid and their “walk-in” hearing aid. Results Group mean postoperative bone and air conduction thresholds did not change significantly from preoperative levels. Postoperative air conduction thresholds decreased slightly in some individual patients, due to the mass loading effect exerted by the transduce...

The effect on speech intelligibility of varying compression time constants in a digital hearing aid.

Abstract: The identification of nonsense syllables in quiet and in three types of background (babble, cafeteria and single female speaker) was measured using four hearing aid compression algorithms differing in attack and release time constants, and using linear amplification. The speech level was always 65 dB SPL. The compression algorithms, which were implemented in a Phonak Claro ITE hearing aid, were: (1) ‘very fast’—the attack time was 8 ms and the release time was 32 ms, for all 20 channels; (2) ‘slow–fast’—the attack and release times decreased from 500 ms for low frequencies to about 100 ms for high frequencies; (3) ‘fast–slow’—the attack and release times increased from about 50 ms for low frequencies to 500 ms for high frequencies; and (4) ‘slow+fast’—a very slow-acting gain control signal was combined with a fast-acting gain control signal, for each channel in a 10-channel system. Acoustical stimuli were presented monaurally via a circumaural headphone mounted over the hearing aid. The linear condition d...

Relative benefits of linear analogue and advanced digital hearing aids.

Abstract: Speech recognition performance and self-reported benefitfrom linear analogue and advanced (digital) hearing aidswere compared in 100 first-time hearing aid users withmild-to-moderate sensorineural hearing loss fitted monaurally with a behind-the-ear (BTE) hearing aid in a single-blind randomized crossover trial. Subjects usedeach aid for 5 weeks in turn, with aid order balancedacross subjects. Three alternative models of digital hearing aid were assigned to subjects according to a balanceddesign. Aid type was disguised to keep subjects blind within practical limitations. Aided speech recognition performance in noise was measured at speech levels of 65 and 75 dB at a speech-to-noise ratio (SNR) of _2 dB forclosed sets of single words. Self-rated benefit was measured using the Abbreviated Profile of Hearing Aid Benefit (APHAB) and the Glasgow Hearing Aid Benefit Profile (GHABP). Quality of life, hearing aid use and user preferences were also assessed. Speech recognition scores with the digital aids were sig...

Comparison of three procedures for initial fitting of compression hearing aids. II. Experienced users, fitted unilaterally

Abstract: This paper is the second in a series comparing three procedures for the initial fitting of multichannel compression hearing aids. The first paper reported the results for a group of 10 experienced hearing aid users fitted bilaterally. This paper reports the results for a different group of 10 experienced hearing aid users fitted unilaterally. The three procedures were: (1) CAMEQ, which aims to amplify speech so as to give equal loudness per critical band over the frequency range 500-5000 Hz, and to give similar overall loudness to normal over a wide range of speech levels; (2) CAMREST, which aims to amplify speech so as to restore normal specific loudness patterns, over a wide range of speech levels; and (3) DSL [i/o], which aims to map the dynamic range of normal-hearing people into the reduced dynamic range of hearing-impaired people, with full restoration of audibility. Each subject was fitted with one Danalogic 163D digital hearing aid, using each of the three fitting procedures in turn; the order was counter-balanced across subjects. Prescribed insertion gains for 55 and 80 dB SPL input levels were verified using real-ear measurements. Immediately after fitting with a given procedure, and 1 week after fitting. the gains were adjusted, when required, by the minimum amount necessary to achieve acceptable fittings. On average, the adjustments were smallest for the CAMREST procedure, slightly larger for the CAMEQ procedure, and largest of all for DSL [i/o]. For the DSL [i/o] the gain changes were mostly negative, especially for high frequencies and the higher input level. After these gain adjustments, users wore the aids for at least 3 weeks before speech reception thresholds (SRTs) for sentences in quiet and in steady and fluctuating background noise were measured. The APHAB questionnaire was also administered. The hearing aids were then refitted with the next procedure. SRTs and APHAB scores did not differ significantly between the three procedures. We conclude that the CAMEQ and CAMREST procedures provide a more appropriate initial fitting than DSL [i/o] for unilaterally experienced hearing aid wearers. Comparison with our earlier study based on bilateral fittings suggests that the preferred gains are similar for unilateral and bilateral fittings.

Hearing aids system for impaired peoples

Abstract: Traditional analog hearing aids are similar to a simple radio They can be tuned and adjusted for volume, bass and treble But hearing loss is not just a technical loss of volume Rather, hearing deficiency can increase sensitivity and reduce tolerance to certain sounds while diminishing sensitivity to others For instance, digital technology can tell the difference between speech and background noise, allowing one in while filtering out the other Approximately 10% of the world's population suffers from some type of hearing loss, yet only a small percentage of this statistic use a hearing aid The stigma associated with wearing a hearing aid, customer dissatisfaction with hearing aid performance, and the cost associated with a high performance solution are all causes of low market penetration Through the use of digital signal processing, digital hearing aid now offers what the analog hearing aid cannot offer It proposes the possibility of performing signal-to noise enhancement, flexible gain-processing, digital feedback reduction, etc In this paper, the simulation of simple digital hearing aid was developed using MATLAB programming language The implementation of this configurable digital hearing aid (DHA) system includes the noise reduction filter, frequency shaper function, and amplitude compression function This digital hearing aid system is design to adapt for mild and moderate hearing loss patient since different gain can be set to map different levels of hearing loss

Speech Recognition in Hearing Aids

Abstract: A variety of algorithms intended for the new generation of hearing aids is presented in this thesis. The main contribution of this work is the hidden Markov model (HMM) approach to classifying listening environments. This method is efficient and robust and well suited for hearing aid applications. This thesis shows that several advanced classification methods can be implemented in digital hearing aids with reasonable requirements on memory and calculation resources. A method for analyzing complex hearing aid algorithms is presented. Data from each hearing aid and listening environment is displayed in three different forms: (1) Effective temporal characteristics (Gain-Time), (2) Effective compression characteristics (Input-Output), and (3) Effective frequency response (Insertion Gain). The method works as intended. Changes in the behavior of a hearing aid can be seen under realistic listening conditions. It is possible that the proposed method of analyzing hearing instruments generates too much information for the user. An automatic gain controlled (AGC) hearing aid algorithm adapting to two sound sources in the listening environment is presented. The main idea of this algorithm is to: (1) adapt slowly (in approximately 10 seconds) to varying listening environments, e.g. when the user leaves a disciplined conference for a multi-babble coffee-break; (2) switch rapidly(in about 100 ms) between different dominant sound sources within one listening situation, such as the change from the user's own voice to a distant speaker's voice in a quiet conference room; (3) instantly reduce gain for strong transient sounds and then quickly return to the previous gain setting; and (4) not change the gain in silent pauses but instead keep the gain setting of the previous sound source. An acoustic evaluation shows that the algorithm works as intended. A system for listening environment classification in hearing aids is also presented. The task is to automatically classify three different listening environments: 'speech in quiet', 'speech in traffic', and 'speech in babble'. The study shows that the three listening environments can be robustly classified at a variety of signal-to-noise ratios with only a small set of pre-trained source HMMs. The measured classification hit rate was 96.7-99.5% when the classifier was tested with sounds representing one of the three environment categories included in the classifier. False alarm rates were0.2-1.7% in these tests. The study also shows that the system can be implemented with the available resources in today's digital hearing aids. Another implementation of the classifier shows that it is possible to automatically detect when the person wearing the hearing aid uses the telephone. It is demonstrated that future hearing aids may be able to distinguish between the sound of a face-to-face conversation and a telephone conversation, both in noisy and quiet surroundings. However, this classification algorithm alone may not be fast enough to prevent initial feedback problems when the user places the telephone handset at the ear. A method using the classifier result for estimating signal and noise spectra for different listening environments is presented. This evaluation shows that it is possible to robustly estimate signal and noise spectra given that the classifier has good performance. An implementation and an evaluation of a single keyword recognizer for a hearing instrument are presented. The performance for the best parameter setting gives 7e-5 [1/s] in false alarm rate, i.e. one false alarm for every four hours of continuous speech from the user, 100% hit rate for an indoors quiet environment, 71% hit rate for an outdoors/traffic environment and 50% hit rate for a babble noise environment. The memory resource needed for the implemented system is estimated to 1820 words (16-bits). Optimization of the algorithm together with improved technology will inevitably make it possible to implement the system in a digital hearing aid within the next couple of years. A solution to extend the number of keywords and integrate the system with a sound environment classifier is also outlined.

Integrated circuit technology in hearing aids

Abstract: In recent years, digital technology has enabled the provision of many new hearing aid features such as noise reduction, feedback suppression and environment classification. As silicon technology evolves, more and more features will be possible within the existing package size and power budget. This paper provides an overview of integrated circuit technology and its application to hearing aid design. The architecture of a digital hearing aid circuit is described in light of the system constraints. The talk will conclude with a brief discussion of expected technological advances and their impact on hearing aid design.

Real-time feedback cancellation in hearing aids

Abstract: The real-time adaptive feedback canceller in a digital hearing aid during telephone use is studied. Detection of feedback onset involves five criteria such as continous and limited input signal level variation, continous modulation of input signal level, duration of specific input signal, and difference between the current gain to the maximum deliverable gain. It is observed that when feedback is detected, the adaptive feedback canceller modifies the system gain around the knee-point. It is suggested to detect the onset of the feedback build-up pattern to prevent any audible feedback reaching the hearing aid user's ear.