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A new algorithm for voice activity detection in additive nonstationary noise is presented. The algorithm utilizes the differences of the probability distribution properties of noise and speech signal. The Magnitude Density (MDF) and the Magnitude Distribution Functions (MDF) are defined. The noise level is monitored for automatic threshold estimation. The estimate is shown to be accurate also when...
Hands-free operation of telephones, incorporating echo cancellation and noise reduction, has been discussed for over a decade. This paper presents an overview of the wide range of algorithms which are applicable to echo cancellers and noise reduction. Practical problems associated with implementation and overall system control are also discussed.
A novel algorithm for the reduction of several types of noise that occur typically in wireless digital speech communications is described in this paper. The algorithm aims at reducing the spectral discontinuities of the signal by analyzing the 2D spectral map and closing the gaps between the frames using heuristic rules. Some experimental evaluations are reported.
This paper addresses the problem of enhancing a speech signal acquired by a microphone array used for hands-free voice communication applications. A new algorithm based on the coherence function developed in the wavelet domain and applied to the beamforming output signal is presented. The wavelet coherence function based nonlinear post-filtering provides a further noise suppression. Its performances...
The propriety of spatial decorrelation of late reverberation has often been used in binaural systems of dereverberation [1] [2]. However, the use of a small array of two microphones limits the performance of such systems, especially at low frequencies. We present in this paper a new algorithm, derived from the classical method proposed by Bloom et al. [2] [3]. Both methods are assessed in terms of...
This paper deals with the enhancement of speech corrupted by real additive noises in a car when two observations are available. As far as we know, no enhancement system was capable of improving both the quality and the intelligibility of the noisy signals. We propose an enhancement method using thresholding, segmentation and filtering in subband domain. The main idea is to expand in subband signals...
This paper analyzes adaptive linear prediction and the effects of the underlying optiniality criterion on the prediction error. It is well known that the signal-dependent optimization process converts the linear filter into a nonlinear signal processing device and that this will influence the statistics of the filter output in a way not expected from linear filter theory. For minimum-phase Lp-optimal...
In previous works [5], [6], we studied some speech enhancement algorithms based on the iterative Wiener filtering method due to Lim-Oppenheim [2], where the AR spectral estimation of the speech is carried out using a second-order analysis. But in our algorithms we consider an AR estimation by means of cumulant analysis. This work extends some preceding papers due to the authors: a cumulant-based Wiener...
This paper describes and analyses an improved algorithm for hands-free telephony which uses an acoustic echo canceller combined with an additional FIR-filter (called "echo shaping filter") in the sending path of the hands-free telephone. The algorithm controlling the filter is motivated by an approximation of an optimal least squares filter. Simulation results show that the algorithm allows...
The design of an efficient and robust hands-free system is now required by the growth of mobile radio and teleconference communications. The use of Frequency-Domain Adaptive Filters in the context of acoustic echo cancellation has been extensively studied in the literature. These algorithms are well-suited for long impulse response modeling and for correlated input signals like speech. A global optimisation...
This contribution deals with the role and the performance of echocompensation and noise suppression, when used in combination with speech recognition systems. For two applications of interest (speech control in car or via telephone) there are quite significant differences to classical echocompensation and noise suppression for telephone conferences. It will be pointed out, how the systems are structured,...
A robust technique for the coding of multiband excitation (MBE) model parameters from a non-stationary speech segment is proposed in this paper. The non-stationary speech segment which has an abrupt increase in its signal energy with respect to the time is divided into 2 quasi-stationary speech segments. A variable analysis frame size technique is proposed to analyze the lower energy portion and the...
This paper presents new results on critical band masked distortion controlled quantisation of a linear transform representation of speech. In particular, fixed rate split vector quantisation of a critical band gain vector is investigated. While shown to be objectively significant in meeting masked distortion criteria, near-transparent quantisation of the critical band gain spectrum is nonetheless...
This paper presents our work on the integration of visual data in automatic speech recognition systems. We particularly aim at solving two problems: • classifiation differences for the modeling of acoustic information (phonemes) and visual information (visemes); • the phenomena of anticipation and retention of visemes on the corresponding phonemes. We developed and tested three systems, each dealing...
A novel approach for speech segmentation is proposed, based on Multilevel Hybrid (mean{}/min{}) Filters (MHF) with the following features: • An accurate transition location. • Good performance in noisy environments (gaussian & impulsive noise). The proposed method is based on spectral changes, with the goal of segmenting the voice into homogeneous acoustic segments. This algorithm is being used...
In this contribution a novel structure for the enhancement of speech signals disturbed by acoustic noise is presented which is based on Spectral Subtraction. The Spectral Subtraction technique is combined with a novel estimator for the noise power spectrum which takes advantage of the employment of a second microphone. Due to the extension to a two-microphone system the Spectral Subtraction can be...
The key step in reduced-rank noise reduction algorithms is to approximate a matrix by another one with lower rank, typically by truncating a singular value decomposition (SVD). We give an explicit and closed-form derivation of the filter properties of the rank reduction operation and interpret this operation in the frequency domain by showing that the reduced-rank output signal is identical to that...
Since modern telecommunication equipment, especially hands-free telephones, incorporates sophisticated signal processing, the analysis methods must take into account the properties of the human hearing. The basis for the correct aquisition of test data — used for auditory but instrumental measurements as well- is the binaural rcording and binaural analysis of the test stimuli. The paper gives an overview,...
The spectral subtraction approach has become almost standard in speech enhancement because it is relatively easy to understand and implement. The major drawback of the spectral subtraction method is that it leaves residual noise with annoying noticeable tonal characteristics referred to as musical noise. For low SNR the perceived effect of the "musical noise" is close to that of the additive...
Nowaday 7 Khz wideband speech coding requires at least 48 kbit/s as it still depends on the ITU standard G.722. CELP coders have been developed for wideband systems achieving high quality speech coding at rates from 16 kbit/s to 32 kbit/s as the wideband LD-CELP at 32 kbit/s. In this paper, a new split-band LD-CELP wideband coder at 24 kbit/s is proposed and its performance and complexity are compared...
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