Tobias Herzke

Comparing Binaural Pre-processing Strategies I: Instrumental Evaluation

In a collaborative research project, several monaural and binaural noise reduction algorithms have been comprehensively evaluated. In this article, eight selected noise reduction algorithms were assessed using instrumental measures, with a focus on the instrumental evaluation of speech intelligibility. Four distinct, reverberant scenarios were created to reflect everyday listening situations: a stationary speech-shaped noise, a multitalker babble noise, a single interfering talker, and a realistic cafeteria noise. Three instrumental measures were employed to assess predicted speech intelligibility and predicted sound quality: the intelligibility-weighted signal-to-noise ratio, the short-time objective intelligibility measure, and the perceptual evaluation of speech quality. The results show substantial improvements in predicted speech intelligibility as well as sound quality for the proposed algorithms. The evaluated coherence-based noise reduction algorithm was able to provide improvements in predicted audio signal quality. For the tested single-channel noise reduction algorithm, improvements in intelligibility-weighted signal-to-noise ratio were observed in all but the nonstationary cafeteria ambient noise scenario. Binaural minimum variance distortionless response beamforming algorithms performed particularly well in all noise scenarios.

Effects of instantaneous multiband dynamic compression on speech intelligibility

The recruitment phenomenon, that is, the reduced dynamic range between threshold and uncomfortable level, is attributed to the loss of instantaneous dynamic compression on the basilar membrane. Despite this, hearing aids commonly use slow-acting dynamic compression for its compensation, because this was found to be the most successful strategy in terms of speech quality and intelligibility rehabilitation. Former attempts to use fast-acting compression gave ambiguous results, raising the question as to whether auditory-based recruitment compensation by instantaneous compression is in principle applicable in hearing aids. This study thus investigates instantaneous multiband dynamic compression based on an auditory filterbank. Instantaneous envelope compression is performed in each frequency band of a gammatone filterbank, which provides a combination of time and frequency resolution comparable to the normal healthy cochlea. The gain characteristics used for dynamic compression are deduced from categorical loudness scaling. In speech intelligibility tests, the instantaneous dynamic compression scheme was compared against a linear amplification scheme, which used the same filterbank for frequency analysis, but employed constant gain factors that restored the sound level for medium perceived loudness in each frequency band. In subjective comparisons, five of nine subjects preferred the linear amplification scheme and would not accept the instantaneous dynamic compression in hearing aids. Four of nine subjects did not perceive any quality differences. A sentence intelligibility test in noise (Oldenburg sentence test) showed little to no negative effects of the instantaneous dynamic compression, compared to linear amplification. A word intelligibility test in quiet (one-syllable rhyme test) showed that the subjects benefit from the larger amplification at low levels provided by instantaneous dynamic compression. Further analysis showed that the increase in intelligibility resulting from a gain provided by instantaneous compression is as high as from a gain provided by linear amplification. No negative effects of the distortions introduced by the instantaneous compression scheme in terms of speech recognition are observed.

Indication criteria for cochlear implants and hearing aids: impact of audiological and non-audiological findings

Owing to technological progress and a growing body of clinical experience, indication criteria for cochlear implants (CI) are being extended to less severe hearing impairments. It is, therefore, worth reconsidering these indication criteria by introducing novel testing procedures. The diagnostic evidence collected will be evaluated. The investigation includes postlingually deafened adults seeking a CI. Prior to surgery, speech perception tests [Freiburg Speech Test and Oldenburg sentence (OLSA) test] were performed unaided and aided using the Oldenburg Master Hearing Aid (MHA) system. Linguistic skills were assessed with the visual Text Reception Threshold (TRT) test, and general state of health, socio-economic status (SES) and subjective hearing were evaluated through questionnaires. After surgery, the speech tests were repeated aided with a CI. To date, 97 complete data sets are available for evaluation. Statistical analyses showed significant correlations between postsurgical speech reception threshold (SRT) measured with the adaptive OLSA test and pre-surgical data such as the TRT test (r=−0.29), SES (r=−0.22) and (if available) aided SRT (r=0.53). The results suggest that new measures and setups such as the TRT test, SES and speech perception with the MHA provide valuable extra information regarding indication for CI.

Comparing Binaural Pre-processing Strategies II Speech Intelligibility of Bilateral Cochlear Implant Users

Several binaural audio signal enhancement algorithms were evaluated with respect to their potential to improve speech intelligibility in noise for users of bilateral cochlear implants (CIs). 50% speech reception thresholds (SRT50) were assessed using an adaptive procedure in three distinct, realistic noise scenarios. All scenarios were highly nonstationary, complex, and included a significant amount of reverberation. Other aspects, such as the perfectly frontal target position, were idealized laboratory settings, allowing the algorithms to perform better than in corresponding real-world conditions. Eight bilaterally implanted CI users, wearing devices from three manufacturers, participated in the study. In all noise conditions, a substantial improvement in SRT50 compared to the unprocessed signal was observed for most of the algorithms tested, with the largest improvements generally provided by binaural minimum variance distortionless response (MVDR) beamforming algorithms. The largest overall improvement in speech intelligibility was achieved by an adaptive binaural MVDR in a spatially separated, single competing talker noise scenario. A no-pre-processing condition and adaptive differential microphones without a binaural link served as the two baseline conditions. SRT50 improvements provided by the binaural MVDR beamformers surpassed the performance of the adaptive differential microphones in most cases. Speech intelligibility improvements predicted by instrumental measures were shown to account for some but not all aspects of the perceptually obtained SRT50 improvements measured in bilaterally implanted CI users.

A Binaural CI Research Platform for Oticon Medical SP/XP Implants Enabling ITD/ILD and Variable Rate Processing

We present the first portable, binaural, real-time research platform compatible with Oticon Medical SP and XP generation cochlear implants. The platform consists of (a) a pair of behind-the-ear devices, each containing front and rear calibrated microphones, (b) a four-channel USB analog-to-digital converter, (c) real-time PC-based sound processing software called the Master Hearing Aid, and (d) USB-connected hardware and output coils capable of driving two implants simultaneously. The platform is capable of processing signals from the four microphones simultaneously and producing synchronized binaural cochlear implant outputs that drive two (bilaterally implanted) SP or XP implants. Both audio signal preprocessing algorithms (such as binaural beamforming) and novel binaural stimulation strategies (within the implant limitations) can be programmed by researchers. When the whole research platform is combined with Oticon Medical SP implants, interaural electrode timing can be controlled on individual electrodes to within ±1 µs and interaural electrode energy differences can be controlled to within ±2%. Hence, this new platform is particularly well suited to performing experiments related to interaural time differences in combination with interaural level differences in real-time. The platform also supports instantaneously variable stimulation rates and thereby enables investigations such as the effect of changing the stimulation rate on pitch perception. Because the processing can be changed on the fly, researchers can use this platform to study perceptual changes resulting from different processing strategies acutely.

The ABCIT research platform

We present a portable, binaural, real-time research platform for cochlear implant devices. The platform consists of hardware and software parts. The research hardware is responsible for the 4-channel input audio signal as well as for the electric stimulation of the cochlea. The software includes a highly flexible research environment for the development of sound pre-processing (“speech processing”) and stimulation strategies. The interface between hardware and software is fully bi-directional, supporting recording capabilities of objective measures specific to the connected implant device. Hence, this platform enables the researchers to perform a diversity of experiments related to cochlear implant research.

Open portable platform for hearing aid research

The NIDCD has recently funded a number of projects to develop portable signal processing tools that enable real-time processing of the acoustic environment. The overarching goal is to provide a large group of researchers with the means to efficiently develop and evaluate, in collaborative multi-center environments, novel signal processing schemes, individualized fitting procedures, and technical solutions and services for hearing apparatus such as hearing aids and assistive listening devices. We report on the specific goals and results of two such projects. In one of them (R01DC015429), an open source software platform for real-time runtime environments is developed: The open Master Hearing Aid (openMHA). It provides an extendible set of algorithms for hearing aid signal processing and runs under Linux, Windows, and Mac operating systems on standard PC platforms and on small-scale ARM-based boards. An optimized version of openMHA is provided for the companion SBIR project (R44DC016247), which is a portable, rigid, versatile, and wearable platform featuring an ARM Cortex®-A8 processor. The resulting Portable Hearing Aid Community Platform consists of both hardware elements to provide the advanced desired functionality and software routines to provide for all the features that researchers may need to develop new algorithms.The NIDCD has recently funded a number of projects to develop portable signal processing tools that enable real-time processing of the acoustic environment. The overarching goal is to provide a large group of researchers with the means to efficiently develop and evaluate, in collaborative multi-center environments, novel signal processing schemes, individualized fitting procedures, and technical solutions and services for hearing apparatus such as hearing aids and assistive listening devices. We report on the specific goals and results of two such projects. In one of them (R01DC015429), an open source software platform for real-time runtime environments is developed: The open Master Hearing Aid (openMHA). It provides an extendible set of algorithms for hearing aid signal processing and runs under Linux, Windows, and Mac operating systems on standard PC platforms and on small-scale ARM-based boards. An optimized version of openMHA is provided for the companion SBIR project (R44DC016247), which is a portabl...

Open community platform for hearing aid algorithm research

The project “Open community platform for hearing aid algorithm research” funded by the National Institutes of Health (NIH Grant 1R01DC015429-01) aims at sustainable, focused research toward improvement and new types of assistive hearing systems. To this end, an open-source software platform for real-time audio signal processing will be developed and made available to the research community including a standard set of reference algorithms. Furthermore, novel algorithms for dynamic and frequency compression, auditory-scene-analysis based noise suppression and speech enhancement, and feedback management will be investigated. For a realistic assessment of the benefits of hearing aid algorithms and combinations thereof, instrumental measures of performance in virtual acoustic environments of varying complexity will be included in the algorithm design and optimization. With such a quasi-standard set of benchmarks and the means to develop and integrate own signal-processing methods and measures in the same frame...

A mobile SoC-based platform for evaluating hearing aid algorithms and architectures

Current research on audio signal processing algorithms for digital hearing aid devices is extremely pushing the performance demands. Nowadays, there is a trend of using several microphones in such systems (e.g., binaural systems) to improve the speech perception of a hearing impaired person. However, there is a lack of mobile platforms, capable of processing such algorithms in real-time. This paper presents a new mobile SoC-based evaluation and development platform (including a multi-channel audio extension board), specially thought not only for evaluating new hearing aid signal processing algorithms but also to develop new hardware co-processor architectures, that could be integrated in current hearing aid devices to improve their performance with a minimal extra energy consumption.