To enhance speech recognition, as well as Mandarin tone recognition in noice, we proposed a speech coding strategy called zero-crossing of fine structure in low frequency (LFFS) for cochlear implant based on low frequency non-uniform sampling (LFFS for short). In the range of frequency perceived boundary of human ear, we used zero-crossing time of the fine structure to generate the stimulus pulse sequences based on the frequency selection rule. Acoustic simulation results showed that although on quiet background the performance of LFFS was similar to continuous interleaved sampling (CIS), on the noise background the performance of LFFS in Chinese tones, words and sentences were significantly better than CIS. In addition to this, we also got better Mandarin recognition factors distribution by using the improved index distribution model. LFFS contains more tonal information which was able to effectively improve Mandarin recognition of the cochlear implant.
In order to improve the speech quality and auditory perceptiveness of electronic cochlear implant under strong noise background, a speech enhancement system used for electronic cochlear implant front-end was constructed. Taking digital signal processing (DSP) as the core, the system combines its multi-channel buffered serial port (McBSP) data transmission channel with extended audio interface chip TLV320AIC10, so speech signal acquisition and output with high speed are realized. Meanwhile, due to the traditional speech enhancement method which has the problems as bad adaptability, slow convergence speed and big steady-state error, versiera function and de-correlation principle were used to improve the existing adaptive filtering algorithm, which effectively enhanced the quality of voice communications. Test results verified the stability of the system and the de-noising performance of the algorithm, and it also proved that they could provide clearer speech signals for the deaf or tinnitus patients.
Cochlear implant (CI) in present Chinese environment will lose pitch information and result in low speech recognition. In order to research Chinese feature-based speech processing strategy for cochlear implant contrapuntally and to improve the speech recognition for CI recipients, we improve the CI front-end signal acquisition platform and research the signal features. Our search includes the waveform, spectrogram, energy intensity, pitch and formant parameters for different speech processing strategies of cochlear implant. Features in two kinds of speech processing strategies are analyzed and extracted for the study of parameter characteristics. Therefore, the proposed aim of this paper is to extend the research on Chinese-based CI speech processing strategy.
Cochlear implant (CI) is the only method for efficacious treatment of congenital severe deafness at present. However, for children with congenital severe deafness after CI, the mechanism of the structural and functional changes of their cerebral cortex is not clear. This study was based on the cross-modal reorganization of deaf patients. Event related potential (ERP) and source localization technique were used to visualize the change of cortical activity in children with congenital severe deafness during 1-year period (0, 1, 3, 6, 9 and 12 months after CI). We aimed to investigate the association between hearing restoration and cross-modal reorganization in children with congenital severe deafness after CI. The results showed that the cross-modal reorganization exists in children with congenital severe deafness. During hearing restoration, the function of the cross-modal reorganization reversed to the normal state. The method and conclusions of this study may be of significance in guiding the training and evaluation of hearing rehabilitation after CI in patients.
Speech enhancement methods based on microphone array adopt many microphones to record speech signal simultaneously. As spatial information is increased, these methods can increase speech recognition for cochlear implant in noisy environment. Due to the size limitation, the number of microphones used in the cochlear implant cannot be too large, which limits the design of microphone array beamforming. To balance the size limitation of cochlear implant and the spatial orientation information of the signal acquisition, we propose a speech enhancement and beamforming algorithm based on dual thin uni-directional / omni-directional microphone pairs (TP) in this paper. Each TP microphone contains two sound tubes for signal acquisition, which increase the overall spatial orientation information. In this paper, we discuss the beamforming characteristics with different gain vectors and the influence of the inter-microphone distance on beamforming, which provides valuable theoretical analysis and engineering parameters for the application of dual microphone speech enhancement technology in cochlear implants.
Microphone array based methods are gradually applied in the front-end speech enhancement and speech recognition improvement for cochlear implant in recent years. By placing several microphones in different locations in space, this method can collect multi-channel signals containing a lot of spatial position and orientation information. Microphone array can also yield specific beamforming mode to enhance desired signal and suppress ambient noise, which is particularly suitable to be applied in face-to-face conversation for cochlear implant users. And its application value has attracted more and more attention from researchers. In this paper, we describe the principle of microphone array method, analyze the microphone array based speech enhancement technologies in present literature, and further present the technical difficulties and development trend.