Preferred delay and phase-frequency response of open-canal hearing aids with music at low insertion gain

Zakis, Justin A.; Fulton, Bernadette; Steele, Brenton R.

doi:10.3109/14992027.2012.701020

Cited by 8 publications

(2 citation statements)

References 14 publications

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“…Therefore, in developing open ear hearing assistive devices that utilize beamforming, the designers need to be less worried on latency constraints from digital signal processing algorithms, and more concerned with distractions that may arise from perceiving one's own voice, as well as audio-visual mismatch. A distorted perception of one's own voice can happen anywhere between 2 and 50 ms (see [22] for a review). This provides a strict constraint of processing delay and would likely need creative solutions to work around.…”

Section: Plos Onementioning

confidence: 99%

Negative impacts from latency masked by noise in simulated beamforming

Drew

Brimijoin

2021

PLoS ONE

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Those experiencing hearing loss face severe challenges in perceiving speech in noisy situations such as a busy restaurant or cafe. There are many factors contributing to this deficit including decreased audibility, reduced frequency resolution, and decline in temporal synchrony across the auditory system. Some hearing assistive devices implement beamforming in which multiple microphones are used in combination to attenuate surrounding noise while the target speaker is left unattenuated. In increasingly challenging auditory environments, more complex beamforming algorithms are required, which increases the processing time needed to provide a useful signal-to-noise ratio of the target speech. This study investigated whether the benefits from signal enhancement from beamforming are outweighed by the negative impacts on perception from an increase in latency between the direct acoustic signal and the digitally enhanced signal. The hypothesis for this study is that an increase in latency between the two identical speech signals would decrease intelligibility of the speech signal. Using 3 gain / latency pairs from a beamforming simulation previously completed in lab, perceptual thresholds of SNR from a simulated use case were obtained from normal hearing participants. No significant differences were detected between the 3 conditions. When presented with 2 copies of the same speech signal presented at varying gain / latency pairs in a noisy environment, any negative intelligibility effects from latency are masked by the noise. These results allow for more lenient restrictions for limiting processing delays in hearing assistive devices.

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Section: Plos Onementioning

confidence: 99%

Negative impacts from latency masked by noise in simulated beamforming

Drew

Brimijoin

2021

PLoS ONE

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“…Although studies testing delays in the range of 1 to >10 ms in simulation have demonstrated negative effects on sound quality as judged by normal and impaired hearing listeners (Stone & Moore, 1999, 2002, 2003, 2005Stone, Moore, Meisenbacker, & Derleth, 2008), these negative effects were not evident when testing in real hearing instruments worn by hearing impaired listeners (Groth & Sondergard, 2004). One study by Zakis, Fulton, and Steele (2012) examined the effect of throughput delay on the sound quality of music in real hearing instruments. In this study, an attempt was made to create a worst case scenario by using open-canal hearing instruments with the gain set such that the likelihood of comb filtering was maximized.…”

Section: Throughput Delaymentioning

confidence: 99%

Analog-to-Digital Conversion to Accommodate the Dynamics of Live Music in Hearing Instruments

Hockley¹,

Bahlmann²,

Fulton³

2012

Trends in Amplification

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Hearing instrument design focuses on the amplification of speech to reduce the negative effects of hearing loss. Many amateur and professional musicians, along with music enthusiasts, also require their hearing instruments to perform well when listening to the frequent, high amplitude peaks of live music. One limitation, in most current digital hearing instruments with 16-bit analog-to-digital (A/D) converters, is that the compressor before the A/D conversion is limited to 95 dB (SPL) or less at the input. This is more than adequate for the dynamic range of speech; however, this does not accommodate the amplitude peaks present in live music. The hearing instrument input compression system can be adjusted to accommodate for the amplitudes present in music that would otherwise be compressed before the A/D converter in the hearing instrument. The methodology behind this technological approach will be presented along with measurements to demonstrate its effectiveness.Keywords hearing aids, hearing instruments, live music, musicians and hearing loss A/D conversion peak sound pressure level of music

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