Rennies Jan, Drefs Jakob, Hülsmeier David, Schepker Henning, Doclo Simon
Project Group Hearing, Speech and Audio Technology, Fraunhofer Institute for Digital Media Technology IDMT and Cluster of Excellence Hearing4All, D-26129 Oldenburg, Germany.
Signal Processing Group, Department of Medical Physics and Acoustics and Cluster of Excellence Hearing4All, University of Oldenburg, D-26111 Oldenburg, Germany.
J Acoust Soc Am. 2017 Apr;141(4):2526. doi: 10.1121/1.4979591.
In many applications in which speech is played back via a sound reinforcement system such as public address systems and mobile phones, speech intelligibility is degraded by additive environmental noise. A possible solution to maintain high intelligibility in noise is to pre-process the speech signal based on the estimated noise power at the position of the listener. The previously proposed AdaptDRC algorithm [Schepker, Rennies, and Doclo (2015). J. Acoust. Soc. Am. 138, 2692-2706] applies both frequency shaping and dynamic range compression under an equal-power constraint, where the processing is adaptively controlled by short-term estimates of the speech intelligibility index. Previous evaluations of the algorithm have focused on normal-hearing listeners. In this study, the algorithm was extended with an adaptive gain stage under an equal-peak-power constraint, and evaluated with eleven normal-hearing and ten mildly to moderately hearing-impaired listeners. For normal-hearing listeners, average improvements in speech reception thresholds of about 4 and 8 dB compared to the unprocessed reference condition were measured for the original algorithm and its extension, respectively. For hearing-impaired listeners, the average improvements were about 2 and 6 dB, indicating that the relative improvement due to the proposed adaptive gain stage was larger for these listeners than the benefit of the original processing stages.
在许多通过扩声系统（如公共广播系统和手机）回放语音的应用中，语音清晰度会因环境噪声的叠加而降低。在噪声环境中保持高清晰度的一种可能解决方案是根据听众位置处估计的噪声功率对语音信号进行预处理。先前提出的自适应动态范围压缩（AdaptDRC）算法[谢普克、伦尼斯和多克洛（2015年）。《美国声学学会杂志》138卷，2692 - 2706页]在等功率约束下应用频率整形和动态范围压缩，其中处理过程由语音清晰度指数的短期估计值进行自适应控制。该算法先前的评估主要针对听力正常的听众。在本研究中，该算法在等峰值功率约束下增加了一个自适应增益阶段进行扩展，并对11名听力正常的听众和10名轻度至中度听力受损的听众进行了评估。对于听力正常的听众，与未处理的参考条件相比，原始算法及其扩展算法的语音接收阈值平均分别提高了约4分贝和8分贝。对于听力受损的听众，平均提高约2分贝和6分贝，这表明对于这些听众而言，所提出的自适应增益阶段带来的相对改善比原始处理阶段的益处更大。
