Christophe Beaugeant

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.

Noise reduction for dual-microphone mobile phones exploiting power level differences

This paper discusses the application of noise reduction algorithms for dual-microphone mobile phones. An analysis of the acoustical environment based on recordings with a dual-microphone mock-up phone mounted on a dummy head is given. Motivated by the recordings, a novel dual-channel noise reduction algorithm is proposed. The key components are a noise PSD estimator and an improved spectral weighting rule which both explicitly exploit the Power Level Differences (PLD) of the desired speech signal between the microphones. Experiments with recorded data show that this low complexity system has a good performance and is beneficial for an integration into future mobile communication devices.

ASR in mobile phones - an industrial approach

In order to make hidden Markov model (HMM) speech recognition suitable for mobile phone applications, Siemens developed a recognizer, Very Smart Recognizer (VSR), for deployment in future mobile phone generations. Typical applications will be name dialling, command and control operations suited for different environments, for example in cars. The paper describes research and development issues of a speech recognizer in mobile devices focusing on noise robustness, memory efficiency and integer implementation. The VSR is shown to reach a word error rate as low as 4.1% on continuous digits recorded in a car environment. Furthermore by means of discriminative training and HMM-parameter coding, the memory requirements of the VSR HMMs are smaller than 64 kBytes.

Dual microphone noise PSD estimation for mobile phones in hands-free position exploiting the coherence and speech presence probability

This contribution addresses the enhancement of noisy speech signals picked up by a dual microphone mobile phone in hands-free position. A novel technique to estimate the noise power spectral density is presented which combines two methods: a single microphone algorithm based on the speech presence probability and a dual microphone technique exploiting the coherence properties of the target signal and the background noise. Due to the novel approach, the weakness of both methods can be overcome. Since the proposed method requires knowledge of the current coherence properties, a technique is presented which estimates the coherence of the speech and noise signals, which is usually not known in practice.

Noise reduction on speech codec parameters

The transmission of speech in mobile or packet networks requires the use of a speech codec. In order to improve the quality of speech in a noisy environment, a noise reduction algorithm is used. This noise reduction can either be done as pre-processing before speech encoding or in the network by decoding the bitstream, performing the speech enhancement in the time and/or frequency domain and re-encoding the speech. Both methods are computationally expensive. In this paper a new approach to reduce environmental background noise by modifying the codec parameters is discussed.

Robust and low-cost cascaded non-linear acoustic echo cancellation

This paper addresses the problem of acoustic echo cancellation in non-linear environments. The first contribution relates to the use of a cascaded model which divides the loudspeaker enclosure microphone system into two main blocks; the first models the downlink transducers which are assumed to be the main source of nonlinearity. The second block includes the acoustical channel and uplink transducers which are assumed to be linear and have a comparatively longer impulse response and higher time variability. The second contribution is a new non-linear adaptive echo canceler which is based on the cascaded model and has greater robustness to changes in the acoustic channel than an existing power filter approach.

Multi-channel psychoacoustically motivated speech enhancement

Multichannel techniques offer advantages in noise reduction and overall output signal quality when compared to the well studied mono approaches. In this paper we present an original multichannel psychoacoustically motivates noise reduction algorithm that naturally extends the single channel psychoacoustic masking filter previously studied in the literature [S. Gustafsson et al., 1999]. The optimality criterion is designed to simultaneously satisfy the psychoacoustic masking principle and minimize the signal total distortion. In experiments on real data recorded in a noisy car environment, we show the enhanced performance of the two-channel solution in terms of artifacts and overall tradeoff between artifacts and amount of noise removed as given by word recognition rates.

Active noise control in headsets: A new approach for broadband feedback ANC

In this paper a novel approach for broadband feedback active noise control (ANC) is presented which is based on the combination of classical non-adaptive feedback and adaptive feedback ANC techniques. The non-adaptive part is suitable to attenuate low frequency ambient noise whereas the adaptive part attenuates periodic components of the ambient noise. The proposed technique yields a higher overall noise attenuation performance compared to a purely classical non-adaptive feedback or purely adaptive feedback ANC system. In addition to that, the combination of both techniques is also beneficial for practical realizations since the adaptive feedback ANC stabilizes the overall system. With regard to low cost headset devices, the impact of practical hardware constraints such as low-cost analog-to-digital and digital-to-analog converters (ADC, DAC) is discussed. As a conclusion, a mixed analog-digital realization of the new approach is proposed.

An assessment of linear adaptive filter performance with nonlinear distortions

Acoustic echo cancellers are generally based on the assumption of a linear echo path between the transducers. However the small loudspeakers that are commonly used in todays terminals can introduce nonlinear distortions that reduce the performance of echo cancellation. In order to evaluate the degradation in performance, this paper assesses the behaviour of five linear echo cancellers in the presence of nonlinearities and presents the first thorough comparison of their robustness. Even if the performance of all the echo cancellers degrades as expected, some algorithms are shown to be more robust than others: fast converging algorithms and block signal processing are more perturbed in nonlinear environments.

Challenges of 16 kHz in acoustic pre- and post-processing for terminals

The standardization of the AMR-WB speech codec by 3GPP opens a new area on audio pre-and post-processing integrated in telecommunication terminals. It appears that the transition from narrowband to wideband signals is not a trivial task, but can rather be seen as a challenge, as acoustic quality requirements, scalability and computational limitations have to be faced at the same time. This article presents key examples to address the transition issue as well as proposals for low-complexity implementations of acoustic echo cancellation and noise reduction