John Mourjopoulos

Speech enhancement based on audible noise suppression

A novel speech enhancement technique is presented based on the definition of the psychoacoustically derived quantity of audible noise spectrum and its subsequent suppression using optimal nonlinear filtering of the short-time spectral amplitude (STSA) envelope. The filter operates with sparse spectral estimates obtained from the STSA, and, when these parameters are accurately known, significant intelligibility gains, up to 40%, result in the processed speech signal. These parameters can be also estimated from noisy data, resulting into smaller but significant intelligibility gains.

Speech enhancement using psychoacoustic criteria

The technique uses the auditory masking threshold to extract information for the audible noise components. Those components are then removed using adaptive nonlinear spectral modification. The main advantage of such an approach is that the speech signal is not affected by processing. In addition, very little information on the features of the noise is required. The proposed method was found to perform well for SNRs better than 10 dB, while for lower SNRs its performance was superior to that of spectral subtraction.<<ETX>>

Automatic speech recognition performance in different room acoustic environments with and without dereverberation preprocessing

The performance of recent dereverberation methods for reverberant speech preprocessing prior to Automatic Speech Recognition (ASR) is compared for an extensive range of room and source-receiver configurations. It is shown that room acoustic parameters such as the clarity (C50) and the definition (D50) correlate well with the ASR results. When available, such room acoustic parameters can provide insight into reverberant speech ASR performance and potential improvement via dereverberation preprocessing. It is also shown that the application of a recent dereverberation method based on perceptual modelling can be used in the above context and achieve significant Phone Recognition (PR) improvement, especially under highly reverberant conditions.

Blind single-channel suppression of late reverberation based on perceptual reverberation modeling

A blind method for suppressing late reverberation from speech and audio signals is presented. The proposed technique operates both on the spectral and on the sub-band domains employing a single input channel. At first, a preliminary rough clean signal estimation is required and for this, any standard technique may be applied; however here the estimate is obtained through spectral subtraction. Then, an auditory masking model is employed in sub-bands to extract the reverberation masking index (RMI) which identifies signal regions with perceived alterations due to late reverberation. Utilizing a selective signal processing technique only these regions are suppressed through sub-band temporal envelope filtering based on analytical expressions. Objective and subjective measures indicate that the proposed method achieves significant late reverberation suppression for both speech and music signals over a wide range of reverberation time (RT) scenarios.

The Acoustics of Roofed Ancient Odeia : The Case of Herodes Atticus Odeion

Ancient Greek / Roman odeia were semi-enclosed theaters that currently survive without their original roof sections. The work compares the acoustics of the well-preserved Herodes Odeion in its current open-air form to a detailed acoustic model reconstruction of its original roofed version and illustrates the significant differences in acoustics between the two spaces. It is shown that in their original state, the odeia had acoustics appropriate for music performances in contrast to their current open-air form that have acoustic properties appropriate for speech reproduction, similar to the larger open-air theaters of the time which were used specifically for ancient drama performances.

Fast communication: Signal-dependent constraints for perceptually motivated suppression of late reverberation

The suppression of late reverberation by spectral subtraction tends to degrade disproportionally low-level signal regions and signal transients. This work proposes two novel relaxation criteria that can constrain such problems in a signal-dependent fashion. These criteria were found to improve the performance of state-of-the-art late reverberation suppression algorithms when used in conjunction with a perceptually motivated non linear filtering stage. Objective results indicate that the proposed method is robust over a wide range of practical reverberation scenarios.

Modeling perceptual effects of reverberation on stereophonic sound reproduction in rooms

The proposed model derives time-frequency maps to estimate perceived alterations due to reverberation in stereo audio signals reproduced in rooms. These alterations relate to monaural masking due to reverberant decay, derived via a computational auditory masking model and to inter-channel cues for the formation of the spatial position of the aural objects, derived via an inter-channel cue mapping module. The maps illustrate in detail the varying nature of the perceptually-relevant alterations due to room reverberation. Quantitative metrics are also introduced which were found to be proportional to reverberation interference, to room-reverberation time and to depend on the specific audio signal. A statistical approach classifies room response properties via their histogram distributions. Corresponding distributions were also applied to the proposed signal-dependent perceptual maps. Such distributions were found to be useful for interpreting the perceived alterations with different kinds of signals, such as music or speech.

A Wiener Filter Approach to Microphone Leakage Reduction in Close-Microphone Applications

Microphone leakage is one of the most prevalent problems in audio applications involving multiple instruments and multiple microphones. Currently, sound engineers have limited solutions available to them. In this paper, the applicability of two widely used signal enhancement methods to this problem is discussed, namely blind source separation and noise suppression. By extending previous work, it is shown that the noise suppression framework is a valid choice and can effectively address the problem of microphone leakage. Here, an extended form of the single channel Wiener filter is used which takes into account the individual audio sources to derive a multichannel noise term. A novel power spectral density (PSD) estimation method is also proposed based on the identification of dominant frequency bins by examining the microphone and output signal PSDs. The performance of the method is examined for simulated environments with various source-microphone setups and it is shown that the proposed approach efficiently suppresses leakage.

Blind Estimation and Suppression of Late Reverberation Utilising Auditory Masking

A new method for blind estimation and suppression of late reverberation of speech signals is presented. The proposed algorithm consists of two steps. In a first step, the reverberation time is blindly determined from the reverberant signal. Then, an approximation of the power spectrum of late reverberation is subtracted from the power spectrum of the reverberant signal. Hence, a preliminary estimation of the anechoic speech spectrum is derived. In a second step, the auditory masking threshold of the clean spectrum estimation is calculated and used to define the coefficients for a nonlinear filter for the reverberant signal, which produces the final enhanced speech signal. The performance of the algorithm is demonstrated on artificially generated signals. Subjective tests are conducted and their results indicate that the quality of the speech signals obtained by the proposed method is superior when compared to previous methods.

Wireless Digital Audio Delivery Analysis and Evaluation

Real-time digital wireless playback of CD-quality audio in multipoint setups using quality of service enhancements is analyzed and evaluated in this work. A novel methodology is introduced for simulating wireless digital audio delivery as well as for theoretically deriving the playback distortions. This methodology allows the accurate wireless digital audio delivery and reproduction simulation and leads to significant results for both the wireless networking and audio playback performance, while it provides a framework for defining the optimal parameters for error-free wireless stereo audio reproduction