Pascal Scalart

Improved Signal-to-Noise Ratio Estimation for Speech Enhancement

This paper addresses the problem of single-microphone speech enhancement in noisy environments. State-of-the-art short-time noise reduction techniques are most often expressed as a spectral gain depending on the signal-to-noise ratio (SNR). The well-known decision-directed (DD) approach drastically limits the level of musical noise, but the estimated a priori SNR is biased since it depends on the speech spectrum estimation in the previous frame. Therefore, the gain function matches the previous frame rather than the current one which degrades the noise reduction performance. The consequence of this bias is an annoying reverberation effect. We propose a method called two-step noise reduction (TSNR) technique which solves this problem while maintaining the benefits of the decision-directed approach. The estimation of the a priori SNR is refined by a second step to remove the bias of the DD approach, thus removing the reverberation effect. However, classic short-time noise reduction techniques, including TSNR, introduce harmonic distortion in enhanced speech because of the unreliability of estimators for small signal-to-noise ratios. This is mainly due to the difficult task of noise power spectrum density (PSD) estimation in single-microphone schemes. To overcome this problem, we propose a method called harmonic regeneration noise reduction (HRNR). A nonlinearity is used to regenerate the degraded harmonics of the distorted signal in an efficient way. The resulting artificial signal is produced in order to refine the a priori SNR used to compute a spectral gain able to preserve the speech harmonics. These methods are analyzed and objective and formal subjective test results between HRNR and TSNR techniques are provided. A significant improvement is brought by HRNR compared to TSNR thanks to the preservation of harmonics

Combined noise and echo reduction in hands-free systems: a survey

The modern telecommunications field is concerned with freedom and, in this context, hands-free systems offer subscribers the possibility of talking more naturally, without using a handset. This new type of use leads to new problems which were negligible in traditional telephony, namely the superposition of noise and echo on the speech signal. To solve these problems and provide a quality that is sufficient for telecommunications, combined reduction of these disturbances is required. This paper presents a summary of the solutions retained for this dual reduction in the context of mono-channel and two-channel sound pick-ups.

A two-step noise reduction technique

The paper addresses the problem of single microphone speech enhancement in noisy environments. Common short-time noise reduction techniques proposed in the art are expressed as a spectral gain depending on the a priori SNR. In the well-known decision-directed approach, the a priori SNR depends on the speech spectrum estimation in the previous frame. As a consequence, the gain function matches the previous frame rather than the current one which degrades the noise reduction performance. We propose a new method, called the two-step noise reduction (TSNR) technique, which solves this problem while maintaining the benefits of the decision-directed approach. This method is analyzed and results in voice communication and speech recognition contexts are given.

Speech enhancement using harmonic regeneration

The paper addresses the problem of single microphone speech enhancement in noisy environments. Common short-time noise reduction techniques introduce harmonic distortion in enhanced speech because of the non reliability of estimators for small signal-to-noise ratios. We propose a new method, called harmonic regeneration noise reduction technique, which solves this problem. A fully harmonic signal is calculated based on the distorted signal using a non-linearity to regenerate harmonics in an efficient way. This artificial signal is then used to compute a suppression gain able to preserve the speech harmonics. This method is theoretically analyzed, then objective and formal subjective results are given and show a significant improvement compared to classical noise reduction techniques.

Comparison of three post-filtering algorithms for residual acoustic echo reduction

We consider an acoustic echo control system composed of a short conventional acoustic echo canceller combined with a post-filter in a teleconference context. The post-filter is implemented in an open-loop structure in the frequency domain, which provides good adaptive performance and flexibility for the choice of the post-filter length. Three post-filtering algorithms are compared in terms of residual echo attenuation and near-end speech distortion. The effect of the post-filter length is also examined. Our study confirms that the post-filtering approach provides high residual echo attenuation. Moreover, it appears that the distortion of the near-end speech can be controlled by choosing appropriately the post-filter length.

Noise Cancellation using Two Closely Spaced Microphones: Experimental Study witha Specific Model and Two Adaptive Algorithms

We consider the speech enhancement problem in a moving car through a blind source separation scheme involving two closely spaced microphones. We propose the use of a double fast Newton transversal filter algorithm (DFNTF) to estimate and suppress coherent noise components from speech, and a model of signal mixtures able to represent correctly the effect of microphones spacing. We also consider the realistic case where the noises at the sensor inputs contain non-coherent components. The simulation results show that the DFNTF algorithm, when controlled by a voice activity detector (VAD), is able to fully cancel the correlated noise components from speech

Analysis of two-sensors forward BSS structure with post-filters in the presence of coherent and incoherent noise

We consider the speech enhancement problem in a moving car through a blind source separation BSS scheme involving two sensors. To correct the distortion brought by this structure we have proposed in previous work (Djendi et al., 2007) two frequency-domain methods to compute the post-filters placed at the output of the forward BSS structure (FBSS). In this work, we consider the case where the noises at the sensor inputs contain coherent and non-coherent components. We provide an analysis of the performance (output SNR and the distortion criterion) of the FBSS structure with post-filters as a function of two new parameters: the coherent to diffuse ratio (CDR) and the speech to coherent ratio (SCR). Simulation results show perfect agreement between theoretical and experimental results.

Analysis of two structures for combined acoustic echo cancellation and noise reduction

This paper addresses the problem of speech enhancement in the context of GSM hands-free radiotelephony where the signal to be transmitted is corrupted by background noise and echo signals. We analyze possible schemes for combined acoustic echo cancellation (AEC) and noise reduction (NR) devices. Considering two AEC algorithms and one NR device, we show that the overall performances obtained by these schemes are greatly dependent on the intrinsic behaviour of the considered AEC algorithms. These results are confirmed by informal listening tests presented in that contribution.We address the problem of speech enhancement in the context of GSM hands-free radiotelephony where the signal to be transmitted is corrupted by background noise and echo signals. We propose possible schemes for combined acoustic echo cancellation (AEC) and noise reduction (NR) devices. Considering two AEC algorithms and one NR device, we show that the overall performance obtained by these schemes are greatly dependent on the intrinsic behaviour of the considered AEC algorithms. These results are confirmed by the informal listening tests presented.

Analytical Approach for Numerical Accuracy Estimation of Fixed-Point Systems Based on Smooth Operations

In embedded systems using fixed-point arithmetic, converting applications into fixed-point representations requires a fast and efficient accuracy evaluation. This paper presents a new analytical approach to determine an estimation of the numerical accuracy of a fixed-point system, which is accurate and valid for all systems formulated with smooth operations (e.g., additions, subtractions, multiplications and divisions). The mathematical expression of the system output noise power is determined using matrices to obtain more compact expressions. The proposed approach is based on the determination of the time-varying impulse-response of the system. To speedup computation of the expressions, the impulse response is modelled using a linear prediction approach. The approach is illustrated in the general case of time-varying recursive systems by the Least Mean Square (LMS) algorithm example. Experiments on various and representative applications show the fixed-point accuracy estimation quality of the proposed approach. Moreover, the approach using the linear-prediction approximation is very fast even for recursive systems. A significant speed-up compared to the best known accuracy evaluation approaches is measured even for the most complex benchmarks.

Minimum Euclidean Distance Based Precoders for MIMO Systems Using Rectangular QAM Modulations

From the feedback of the channel state information (CSI), precoding techniques improve the performance of multiple-input multiple-output (MIMO) systems by optimizing various criteria. In this correspondence, an efficient precoder that maximizes the minimum distance (dmin) of two received vectors is studied. This criterion leads to a nondiagonal precoding scheme and allows achieving a full diversity order. However, the optimized solution for MIMO systems using a high-order QAM modulation is rather complex and changes for different constellations. Therefore, we propose herein a general form of minimum Euclidean distance based precoders for all rectangular QAM modulations. It is shown that the new solution optimizes the distance dmin for small and large dispersive channels.