| Hearing is the most important way for people to communicate in social life, but with the development of industrialization and the aging intensifies, the number of people with hearing loss disease continues to increase. Under the existing level of medical conditions, the most convenient and commonly mean to improve patient’s hearing ability is wearing a hearing aid. The loudness compensation algorithm is the core technology of all the key algorithms for digital hearing aids, which amplify the original signal and make the speech signal within the patient’s auditory dynamic range, so that the patient can not only hear the sound that can’t be heard when not wearing a digital hearing aid, but also feel comfortable.This paper focuses on the loudness compensation algorithm that applied to digital hearing aids. Based on the review of previous work and traditional algorithms, combining with the human auditory, the loudness compensation algorithm which is more in line with the actual needs of the patient is studied, and some research results are concluded in this pape:In this paper, the basic working principle of digital hearing aids, the mechanism of hearing and the cause of hearing loss are studied.The commonly used test methods and related strategies of hearing therapy are summarized.Secondly, the basic principles of loudness compensation technology in digital hearing aids are elaborated, wide dynamic range compression(WDRC), frequency-shift compression and multi-channel compensation are studied in detail, including their respective principle algorithm, the algorithm processes and the specific mathematical realization process.Again, a multi-channel filter bank algorithm based on Gammatone filter is proposed. Because the traditional multi-channel filter bank does not accord with the human auditory, a newly filter bank design method based on Gammatone filter is focused in this paper. According to the features of the human auditory, the proposed method divides lOOHz-8000Hz into 16 non-uniform sub-bands. Simulation results show that the proposed algorithm can be well used in the multi-channel filter design.Further, a loudness compensation algorithm based on segmented sound pressure level(SPL) is proposed. For the lack of internal noise suppression by the traditional compensate for the three sections of SPL, an improved loudness compensation method based on segmented SPL is focused in this paper. In order to improve the speech recognition rate and comfort, OdB SPL-120dB SPL is equally divided into eight sections, each carries a different compensation depened on the actual needs of the patient. Simulations and experiments demonstrate that the hearing loss can be well compensated and the patient’s speech recognition rate can be rapid increased with the help of the improved algorithm.Finally, a digital hearing aids system based on embedded development platform is designed and implemented. In practical applications, the acoustic environment is very complex, in order to verify the effectiveness of the proposed multi-channel loudness compensation algorithm, a hearing aid dedicated chip called YT360A is used to process the algorithm in real-time and form a model of digital hearing aids. The performance of the multi-channel loudness compensation algorithm that proposed in this paper is observed and evaluated through the model. The results show that the digital hearing aid system that designed in this paper can entirely meet the normal requirements of digital hearing aids in real-time and can be well applied to the actual digital hearing aid products. |
PDF Full Text Request