A Speech Preprocessing Method Based on Perceptually Optimized Envelope Processing to Increase Intelligibility in Reverberant Environments

Speech intelligibility in public places can be degraded by the environmental noise and reverberation. In this study, a new near-end listening enhancement (NELE) approach is proposed in which using a time varying filter jointly enhances the onsets and reduces the overlap masking. For optimization, some look-ahead in clean speech and prior knowledge of room impulse response (RIR) are required. In this method, by optimizing a defined cost function, the Spectro-Temporal Envelope of reverb speech is optimized to be as close as possible to that of clean speech. In this cost function, onsets of speech are optimized with increased weight. This approach is different from overlap-masking ratio (OMR) and speech enhancement (OE) approaches (Grosse, van de Par, 2017, J. Audio Eng. Soc., Vol. 65(1/2), pp. 31–41) that only consider previous frames in each time slot for determining the time variant filtering. The SRT measurements show that the new optimization framework enhances the speech intelligibility up to 2 dB more that OE.

Real Time Blind Source Separation in Reverberant Environments

<p>An online convolutive blind source separation solution has been developed for use in reverberant environments with stationary sources. Results are presented for simulation and real world data. The system achieves a separation SINR of 16.8 dB when operating on a two source mixture, with a total acoustic delay was 270 ms. This is on par with, and in many respects outperforms various published algorithms [1],[2]. A number of instantaneous blind source separation algorithms have been developed, including a block wise and recursive ICA algorithm, and a clustering based algorithm, able to obtain up to 110 dB SIR performance. The system has been realised in both Matlab and C, and is modular, allowing for easy update of the ICA algorithm that is the core of the unmixing process.</p>

Real Time Blind Source Separation in Reverberant Environments

<p>An online convolutive blind source separation solution has been developed for use in reverberant environments with stationary sources. Results are presented for simulation and real world data. The system achieves a separation SINR of 16.8 dB when operating on a two source mixture, with a total acoustic delay was 270 ms. This is on par with, and in many respects outperforms various published algorithms [1],[2]. A number of instantaneous blind source separation algorithms have been developed, including a block wise and recursive ICA algorithm, and a clustering based algorithm, able to obtain up to 110 dB SIR performance. The system has been realised in both Matlab and C, and is modular, allowing for easy update of the ICA algorithm that is the core of the unmixing process.</p>

Cortical adaptation to sound reverberation

In almost every natural environment, sounds are reflected by nearby objects, producing many delayed and distorted copies of the original sound, known as reverberation. Our brains usually cope well with reverberation, allowing us to recognize sound sources regardless of their environments. In contrast, reverberation can cause severe difficulties for speech recognition algorithms and hearing-impaired people. The present study examines how the auditory system copes with reverberation. We trained a linear model to recover a rich set of natural, anechoic sounds from their simulated reverberant counterparts. The model neurons achieved this by extending the inhibitory component of their receptive filters for more reverberant spaces, and did so in a frequency-dependent manner. These predicted effects were observed in the responses of auditory cortical neurons of ferrets in the same simulated reverberant environments. Together, these results suggest that auditory cortical neurons adapt to reverberation by adjusting their filtering properties in a manner consistent with dereverberation.

Binaural Signal Processing in Hearing Aids

For many years, clinicians have understood the advantages of listening with two ears compared with one. In addition to improved speech intelligibility in quiet, noisy, and reverberant environments, binaural versus monaural listening improves perceived sound quality and decreases the effort listeners must expend to understand a target voice of interest or to monitor a multitude of potential target voices. For most individuals with bilateral hearing impairment, the body of evidence collected across decades of research has also found that the provision of two compared with one hearing aid yields significant benefit for the user. This article briefly summarizes the major advantages of binaural compared with monaural hearing, followed by a detailed description of the related technological advances in modern hearing aids. Aspects related to the communication and exchange of data between the left and right hearing aids are discussed together with typical algorithmic approaches implemented in modern hearing aids.

Spatial release from masking in varying spatial acoustic under higher order ambisonic-based sound reproduction system

A previous study found that spatial release from masking (SRM) could be observed under virtual reverberant environments using a first order Ambisonic-based sound reproduction system, however, poor localisation accuracy made it difficult to examine effect of varying reverberation time on SRM. The present study follows on using higher order Ambisonics (HOA) to examine how benefits from SRM vary in different spatial acoustics. Subjective speech intelligibility was measured where four room acoustics:reverberation time (RT)= 0.7 s (clarity (C50)= 16 dB, 7 dB); RT= 1.8 s (C50= 8 dB, 2 dB) were simulated via a third order Ambisonic system with a 16 channel spherical loudspeaker array. The masker was played from 8 azimuthal angles (0, +-45, +-90, +-135, 180 degrees) while the target speech was played from 0 degree. The listeners are deemed to benefit from SRM if their intelligibility scores were higher when the masker comes from a different angle than that of the target. We found while listeners could benefit from SRM at C50 = 16 dB and 8 dB, the benefit starts to diminish at C50 = 7 dB, and listeners could no longer benefit from SRM at C50 = 2 dB.