scholarly journals Effect of cochlear implant n-of-m strategy on signal-to-noise ratio below which noise hinders speech recognition

2019 ◽  
Vol 145 (5) ◽  
pp. EL417-EL422
Author(s):  
Lucas Stam ◽  
S. Theo Goverts ◽  
Cas Smits
2020 ◽  
Vol 24 ◽  
pp. 233121652097034
Author(s):  
Florian Langner ◽  
Andreas Büchner ◽  
Waldo Nogueira

Cochlear implant (CI) sound processing typically uses a front-end automatic gain control (AGC), reducing the acoustic dynamic range (DR) to control the output level and protect the signal processing against large amplitude changes. It can also introduce distortions into the signal and does not allow a direct mapping between acoustic input and electric output. For speech in noise, a reduction in DR can result in lower speech intelligibility due to compressed modulations of speech. This study proposes to implement a CI signal processing scheme consisting of a full acoustic DR with adaptive properties to improve the signal-to-noise ratio and overall speech intelligibility. Measurements based on the Short-Time Objective Intelligibility measure and an electrodogram analysis, as well as behavioral tests in up to 10 CI users, were used to compare performance with a single-channel, dual-loop, front-end AGC and with an adaptive back-end multiband dynamic compensation system (Voice Guard [VG]). Speech intelligibility in quiet and at a +10 dB signal-to-noise ratio was assessed with the Hochmair–Schulz–Moser sentence test. A logatome discrimination task with different consonants was performed in quiet. Speech intelligibility was significantly higher in quiet for VG than for AGC, but intelligibility was similar in noise. Participants obtained significantly better scores with VG than AGC in the logatome discrimination task. The objective measurements predicted significantly better performance estimates for VG. Overall, a dynamic compensation system can outperform a single-stage compression (AGC + linear compression) for speech perception in quiet.


2020 ◽  
Author(s):  
chaofeng lan ◽  
yuanyuan Zhang ◽  
hongyun Zhao

Abstract This paper draws on the training method of Recurrent Neural Network (RNN), By increasing the number of hidden layers of RNN and changing the layer activation function from traditional Sigmoid to Leaky ReLU on the input layer, the first group and the last set of data are zero-padded to enhance the effective utilization of data such that the improved reduction model of Denoise Recurrent Neural Network (DRNN) with high calculation speed and good convergence is constructed to solve the problem of low speaker recognition rate in noisy environment. According to this model, the random semantic speech signal with a sampling rate of 16 kHz and a duration of 5 seconds in the speech library is studied. The experimental settings of the signal-to-noise ratios are − 10dB, -5dB, 0dB, 5dB, 10dB, 15dB, 20dB, 25dB. In the noisy environment, the improved model is used to denoise the Mel Frequency Cepstral Coefficients (MFCC) and the Gammatone Frequency Cepstral Coefficents (GFCC), impact of the traditional model and the improved model on the speech recognition rate is analyzed. The research shows that the improved model can effectively eliminate the noise of the feature parameters and improve the speech recognition rate. When the signal-to-noise ratio is low, the speaker recognition rate can be more obvious. Furthermore, when the signal-to-noise ratio is 0dB, the speaker recognition rate of people is increased by 40%, which can be 85% improved compared with the traditional speech model. On the other hand, with the increase in the signal-to-noise ratio, the recognition rate is gradually increased. When the signal-to-noise ratio is 15dB, the recognition rate of speakers is 93%.


2019 ◽  
Vol 28 (1) ◽  
pp. 101-113 ◽  
Author(s):  
Jenna M. Browning ◽  
Emily Buss ◽  
Mary Flaherty ◽  
Tim Vallier ◽  
Lori J. Leibold

Purpose The purpose of this study was to evaluate speech-in-noise and speech-in-speech recognition associated with activation of a fully adaptive directional hearing aid algorithm in children with mild to severe bilateral sensory/neural hearing loss. Method Fourteen children (5–14 years old) who are hard of hearing participated in this study. Participants wore laboratory hearing aids. Open-set word recognition thresholds were measured adaptively for 2 hearing aid settings: (a) omnidirectional (OMNI) and (b) fully adaptive directionality. Each hearing aid setting was evaluated in 3 listening conditions. Fourteen children with normal hearing served as age-matched controls. Results Children who are hard of hearing required a more advantageous signal-to-noise ratio than children with normal hearing to achieve comparable performance in all 3 conditions. For children who are hard of hearing, the average improvement in signal-to-noise ratio when comparing fully adaptive directionality to OMNI was 4.0 dB in noise, regardless of target location. Children performed similarly with fully adaptive directionality and OMNI settings in the presence of the speech maskers. Conclusions Compared to OMNI, fully adaptive directionality improved speech recognition in steady noise for children who are hard of hearing, even when they were not facing the target source. This algorithm did not affect speech recognition when the background noise was speech. Although the use of hearing aids with fully adaptive directionality is not proposed as a substitute for remote microphone systems, it appears to offer several advantages over fixed directionality, because it does not depend on children facing the target talker and provides access to multiple talkers within the environment. Additional experiments are required to further evaluate children's performance under a variety of spatial configurations in the presence of both noise and speech maskers.


2021 ◽  
pp. 019459982110492
Author(s):  
Allan M. Henslee ◽  
Christopher R. Kaufmann ◽  
Matt D. Andrick ◽  
Parker T. Reineke ◽  
Viral D. Tejani ◽  
...  

Objective Electrocochleography (ECochG) is increasingly being used during cochlear implant (CI) surgery to detect and mitigate insertion-related intracochlear trauma, where a drop in ECochG signal has been shown to correlate with a decline in hearing outcomes. In this study, an ECochG-guided robotics-assisted CI insertion system was developed and characterized that provides controlled and consistent electrode array insertions while monitoring and adapting to real-time ECochG signals. Study Design Experimental research. Setting A research laboratory and animal testing facility. Methods A proof-of-concept benchtop study evaluated the ability of the system to detect simulated ECochG signal changes and robotically adapt the insertion. Additionally, the ECochG-guided insertion system was evaluated in a pilot in vivo sheep study to characterize the signal-to-noise ratio and amplitude of ECochG recordings during robotics-assisted insertions. The system comprises an electrode array insertion drive unit, an extracochlear recording electrode module, and a control console that interfaces with both components and the surgeon. Results The system exhibited a microvolt signal resolution and a response time <100 milliseconds after signal change detection, indicating that the system can detect changes and respond faster than a human. Additionally, animal results demonstrated that the system was capable of recording ECochG signals with a high signal-to-noise ratio and sufficient amplitude. Conclusion An ECochG-guided robotics-assisted CI insertion system can detect real-time drops in ECochG signals during electrode array insertions and immediately alter the insertion motion. The system may provide a surgeon the means to monitor and reduce CI insertion–related trauma beyond manual insertion techniques for improved CI hearing outcomes.


2017 ◽  
Vol 28 (05) ◽  
pp. 404-414 ◽  
Author(s):  
Dorothy Neave-DiToro ◽  
Adrienne Rubinstein ◽  
Arlene C. Neuman

Background: Limited attention has been given to the effects of classroom acoustics at the college level. Many studies have reported that nonnative speakers of English are more likely to be affected by poor room acoustics than native speakers. An important question is how classroom acoustics affect speech perception of nonnative college students. Purpose: The combined effect of noise and reverberation on the speech recognition performance of college students who differ in age of English acquisition was evaluated under conditions simulating classrooms with reverberation times (RTs) close to ANSI recommended RTs. Research Design: A mixed design was used in this study. Study Sample: Thirty-six native and nonnative English-speaking college students with normal hearing, ages 18–28 yr, participated. Intervention: Two groups of nine native participants (native monolingual [NM] and native bilingual) and two groups of nine nonnative participants (nonnative early and nonnative late) were evaluated in noise under three reverberant conditions (0.03, 0.06, and 0.08 sec). Data Collection and Analysis: A virtual test paradigm was used, which represented a signal reaching a student at the back of a classroom. Speech recognition in noise was measured using the Bamford–Kowal–Bench Speech-in-Noise (BKB-SIN) test and signal-to-noise ratio required for correct repetition of 50% of the key words in the stimulus sentences (SNR-50) was obtained for each group in each reverberant condition. A mixed-design analysis of variance was used to determine statistical significance as a function of listener group and RT. Results: SNR-50 was significantly higher for nonnative listeners as compared to native listeners, and a more favorable SNR-50 was needed as RT increased. The most dramatic effect on SNR-50 was found in the group with later acquisition of English, whereas the impact of early introduction of a second language was subtler. At the ANSI standard’s maximum recommended RT (0.6 sec), all groups except the NM group exhibited a mild signal-to-noise ratio (SNR) loss. At the 0.8 sec RT, all groups exhibited a mild SNR loss. Conclusion: Acoustics in the classroom are an important consideration for nonnative speakers who are proficient in English and enrolled in college. To address the need for a clearer speech signal by nonnative students (and for all students), universities should follow ANSI recommendations, as well as minimize background noise in occupied classrooms. Behavioral/instructional strategies should be considered to address factors that cannot be compensated for through acoustic design.


2014 ◽  
Vol 25 (10) ◽  
pp. 952-968 ◽  
Author(s):  
Stephen Julstrom ◽  
Linda Kozma-Spytek

Background: In order to better inform the development and revision of the American National Standards Institute C63.19 and American National Standards Institute/Telecommunications Industry Association-1083 hearing aid compatibility standards, a previous study examined the signal strength and signal (speech)-to-noise (interference) ratio needs of hearing aid users when using wireless and cordless phones in the telecoil coupling mode. This study expands that examination to cochlear implant (CI) users, in both telecoil and microphone modes of use. Purpose: The purpose of this study was to evaluate the magnetic and acoustic signal levels needed by CI users for comfortable telephone communication and the users’ tolerance relative to the speech levels of various interfering wireless communication–related noise types. Research Design: Design was a descriptive and correlational study. Simulated telephone speech and eight interfering noise types presented as continuous signals were linearly combined and were presented together either acoustically or magnetically to the participants’ CIs. The participants could adjust the loudness of the telephone speech and the interfering noises based on several assigned criteria. Study Sample: The 21 test participants ranged in age from 23–81 yr. All used wireless phones with their CIs, and 15 also used cordless phones at home. There were 12 participants who normally used the telecoil mode for telephone communication, whereas 9 used the implant’s microphone; all were tested accordingly. Data Collection and Analysis: A guided-intake questionnaire yielded general background information for each participant. A custom-built test control box fed by prepared speech-and-noise files enabled the tester or test participant, as appropriate, to switch between the various test signals and to precisely control the speech-and-noise levels independently. The tester, but not the test participant, could read and record the selected levels. Subsequent analysis revealed the preferred speech levels, speech (signal)-to-noise ratios, and the effect of possible noise-measurement weighting functions. Results: The participants' preferred telephone speech levels subjectively matched or were somewhat lower than the level that they heard from a 65 dB SPL wideband reference. The mean speech (signal)-to-noise ratio requirement for them to consider their telephone experience “acceptable for normal use” was 20 dB, very similar to the results for the hearing aid users of the previous study. Significant differences in the participants’ apparent levels of noise tolerance among the noise types when the noise level was determined using A-weighting were eliminated when a CI-specific noise-measurement weighting was applied. Conclusions: The results for the CI users in terms of both preferred levels for wireless and cordless phone communication and signal-to-noise requirements closely paralleled the corresponding results for hearing aid users from the previous study, and showed no significant differences between the microphone and telecoil modes of use. Signal-to-noise requirements were directly related to the participants’ noise audibility threshold and were independent of noise type when appropriate noise-measurement weighting was applied. Extending the investigation to include noncontinuous interfering noises and forms of radiofrequency interference other than additive audiofrequency noise could be areas of future study.


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