Effects of noise suppression and envelope dynamic range compression on the intelligibility of vocoded sentences for a tonal language

Vocoder simulation studies have suggested that the carrier signal type employed affects the intelligibility of vocoded speech. The present work further assessed how carrier signal type interacts with additional signal processing, namely, single-channel noise suppression and envelope dynamic range compression, in determining the intelligibility of vocoder simulations. In Experiment 1, Mandarin sentences that had been corrupted by speech spectrum-shaped noise (SSN) or two-talker babble (2TB) were processed by one of four single-channel noise-suppression algorithms before undergoing tone-vocoded (TV) or noise-vocoded (NV) processing. In Experiment 2, dynamic ranges of multiband envelope waveforms were compressed by scaling of the mean-removed envelope waveforms with a compression factor before undergoing TV or NV processing. TV Mandarin sentences yielded higher intelligibility scores with normal-hearing (NH) listeners than did noise-vocoded sentences. The intelligibility advantage of noise-suppressed vocoded speech depended on the masker type (SSN vs 2TB). NV speech was more negatively influenced by envelope dynamic range compression than was TV speech. These findings suggest that an interactional effect exists between the carrier signal type employed in the vocoding process and envelope distortion caused by signal processing

Efficacy in noise of the Starkey Surflink Mobile 2 technology in directional versus omnidirectional microphone mode with experienced adult hearing aid users

The Starkey SurfLink Mobile 2 is a remote microphone accessory. Starkey claims that by placing the SurfLink’s internal microphone in the directional microphone setting, the participant will hear better in noise over the omnidirectional setting. This study aims to test the thisthe claim about the devic

Coding Strategies for Cochlear Implants Under Adverse Environments

Cochlear implants are electronic prosthetic devices that restores partial hearing in patients with severe to profound hearing loss. Although most coding strategies have significantly improved the perception of speech in quite listening conditions, there remains limitations on speech perception under adverse environments such as in background noise, reverberation and band-limited channels, and we propose strategies that improve the intelligibility of speech transmitted over the telephone networks, reverberated speech and speech in the presence of background noise. For telephone processed speech, we propose to examine the effects of adding low-frequency and high- frequency information to the band-limited telephone speech. Four listening conditions were designed to simulate the receiving frequency characteristics of telephone handsets. Results indicated improvement in cochlear implant and bimodal listening when telephone speech was augmented with high frequency information and therefore this study provides support for design of algorithms to extend the bandwidth towards higher frequencies. The results also indicated added benefit from hearing aids for bimodal listeners in all four types of listening conditions. Speech understanding in acoustically reverberant environments is always a difficult task for hearing impaired listeners. Reverberated sounds consists of direct sound, early reflections and late reflections. Late reflections are known to be detrimental to speech intelligibility. In this study, we propose a reverberation suppression strategy based on spectral subtraction to suppress the reverberant energies from late reflections. Results from listening tests for two reverberant conditions (RT60 = 0.3s and 1.0s) indicated significant improvement when stimuli was processed with SS strategy. The proposed strategy operates with little to no prior information on the signal and the room characteristics and therefore, can potentially be implemented in real-time CI speech processors. For speech in background noise, we propose a mechanism underlying the contribution of harmonics to the benefit of electroacoustic stimulations in cochlear implants. The proposed strategy is based on harmonic modeling and uses synthesis driven approach to synthesize the harmonics in voiced segments of speech. Based on objective measures, results indicated improvement in speech quality. This study warrants further work into development of algorithms to regenerate harmonics of voiced segments in the presence of noise

Evaluation of a wireless remote microphone in bimodal cochlear implant recipients

Objective: To evaluate the benefit of a wireless remote microphone (MM) for speech recognition in noise in bimodal adult cochlear implant (CI) users both in a test setting and in daily life. Design: This prospective study measured speech reception thresholds in noise in a repeated measures design with factors including bimodal hearing and MM use. The participants also had a 3-week trial period at home with the MM. Study sample: Thirteen post-lingually deafened adult bimodal CI users. Results: A significant improvement in SRT of 5.4 dB was found between the use of the CI with the MM and the use of the CI without the MM. By also pairing the MM to the hearing aid (HA) another improvement in SRT of 2.2 dB was found compared to the situation with the MM paired to the CI alone. In daily life, participants reported better speech perception for various challenging listening situations, when using the MM in the bimodal condition. Conclusion: There is a clear advantage of bimodal listening (CI and HA) compared to CI alone when applying advanced wireless remote microphone techniques to improve speech understanding in adult bimodal CI users

The Effect of Binaural Beamforming Technology on Speech Intelligibility in Bimodal Cochlear Implant Recipients

Although the benefit of bimodal listening in cochlear implant users has been agreed on, speech comprehension remains a challenge in acoustically complex real-life environments due to reverberation and disturbing background noises. One way to additionally improve bimodal auditory performance is the use of directional microphones. The objective of this study was to investigate the effect of a binaural beamformer for bimodal cochlear implant (CI) users. This prospective study measured speech reception thresholds (SRT) in noise in a repeated-measures design that varied in listening modality for static and dynamic listening conditions. A significant improvement in SRT of 4.7 dB was found with the binaural beamformer switched on in the bimodal static listening condition. No significant improvement was found in the dynamic listening condition. We conclude that there is a clear additional advantage of the binaural beamformer in bimodal CI users for predictable/static listening conditions with frontal target speech and spatially separated noise sources

Multi-microphone adaptive noise reduction strategies for coordinated stimulation in bilateral cochlear implant devices

This is the published version, also available here: http://dx.doi.org/10.1121/1.3372727.Bilateral cochlear implant (BI-CI) recipients achieve high word recognition scores in quiet listening conditions. Still, there is a substantial drop in speech recognition performance when there is reverberation and more than one interferers. BI-CI users utilize information from just two directional microphones placed on opposite sides of the head in a so-called independent stimulation mode. To enhance the ability of BI-CI users to communicate in noise, the use of two computationally inexpensive multi-microphone adaptive noise reduction strategies exploiting information simultaneously collected by the microphones associated with two behind-the-ear (BTE) processors (one per ear) is proposed. To this end, as many as four microphones are employed (two omni-directional and two directional) in each of the two BTE processors (one per ear). In the proposed two-microphone binaural strategies, all four microphones (two behind each ear) are being used in a coordinated stimulation mode. The hypothesis is that such strategies combine spatial information from all microphones to form a better representation of the target than that made available with only a single input. Speech intelligibility is assessed in BI-CI listeners using IEEE sentences corrupted by up to three steady speech-shaped noise sources. Results indicate that multi-microphone strategies improve speech understanding in single- and multi-noise source scenarios

Improving the Speech Intelligibility By Cochlear Implant Users

In this thesis, we focus on improving the intelligibility of speech for cochlear implants (CI) users. As an auditory prosthetic device, CI can restore hearing sensations for most patients with profound hearing loss in both ears in a quiet background. However, CI users still have serious problems in understanding speech in noisy and reverberant environments. Also, bandwidth limitation, missing temporal fine structures, and reduced spectral resolution due to a limited number of electrodes are other factors that raise the difficulty of hearing in noisy conditions for CI users, regardless of the type of noise. To mitigate these difficulties for CI listener, we investigate several contributing factors such as the effects of low harmonics on tone identification in natural and vocoded speech, the contribution of matched envelope dynamic range to the binaural benefits and contribution of low-frequency harmonics to tone identification in quiet and six-talker babble background. These results revealed several promising methods for improving speech intelligibility for CI patients. In addition, we investigate the benefits of voice conversion in improving speech intelligibility for CI users, which was motivated by an earlier study showing that familiarity with a talker’s voice can improve understanding of the conversation. Research has shown that when adults are familiar with someone’s voice, they can more accurately – and even more quickly – process and understand what the person is saying. This theory identified as the “familiar talker advantage” was our motivation to examine its effect on CI patients using voice conversion technique. In the present research, we propose a new method based on multi-channel voice conversion to improve the intelligibility of transformed speeches for CI patients

Head orientation benefit to speech intelligibility in noise for cochlear implant users and in realistic listening conditions

Cochlear implant (CI) users suffer from elevated speech-reception thresholds and may rely on lip reading. Traditional measures of spatial release from masking quantify speech-reception-threshold improvement with azimuthal separation of target speaker and interferers and with the listener facing the target speaker. Substantial benefits of orienting the head away from the target speaker were predicted by a model of spatial release from masking. Audio-only and audio-visual speech-reception thresholds in normal-hearing (NH) listeners and bilateral and unilateral CI users confirmed model predictions of this head-orientation benefit. The benefit ranged 2–5 dB for a modest 30� orientation that did not affect the lip-reading benefit. NH listeners’ and CI users’ lip-reading benefit measured 3 and 5 dB, respectively. A head-orientation benefit of �2 dB was also both predicted and observed in NH listeners in realistic simulations of a restaurant listening environment. Exploiting the benefit of head orientation is thus a robust hearing tactic that would benefit both NH listeners and CI users in noisy listening conditions

Speech comprehension across multiple CI processor generations: Scene dependent signal processing

Objectives In clinical practice, characterization of speech comprehension for cochlear implant (CI) patients is typically administered by a set of suprathreshold measurements in quiet and in noise. This study investigates speech comprehension of the three most recent cochlear implant sound processors; CP810, CP910, and CP1000 (Cochlear Limited). To compare sound processor performance across generations and input dynamic range changes, the state-of-the art signal processing technologies available in each sound processor were enabled. Outcomes will be assessed across a range of stimulation intensities, and finally analyzed with respect to normal hearing listeners. Methods In a prospective study, 20 experienced postlingually deafened CI patients who received a Nucleus CI in the ENT department of the University Hospital of SH in Kiel were recruited. Speech comprehension was measured in quiet at 40, 50, and 65 dBSPL with monosyllabic words as well as by speech reception threshold for two-digit numbers. In noise, speech reception thresholds were measured with the adaptive German matrix test with speech and noise in front. Results We found that high levels of open-set speech comprehension are achieved at suprathreshold presentation levels in quiet. However, results at lower test levels have remained mostly unchanged for tested sound processors with default dynamic range. Expanding the lower limit of the acoustic input dynamic range yields better speech comprehension at lower presentation levels. In noise the application of ForwardFocus improves the speech reception. Overall, a continuous improvement for speech perception across three generations of CI sound processors was found. Conclusions Findings motivate further development of signal pre-processing, an additional focus of clinical work on lower stimulation levels, and automation of ForwardFocus. Level of evidence 2