This paper demonstrates two novel methods to estimate the global SNR of speech signals. In both methods, Deep Neural Network-Hidden Markov Model (DNN-HMM) acoustic model used in speech recognition systems is leveraged for the additional task of SNR estimation. In the first method, the entropy of the DNN-HMM output is computed. Recent work on bayesian deep learning has shown that a DNN-HMM trained with dropout can be used to estimate model uncertainty by approximating it as a deep Gaussian process. In the second method, this approximation is used to obtain model uncertainty estimates. Noise specific regressors are used to predict the SNR from the entropy and model uncertainty. The DNN-HMM is trained on GRID corpus and tested on different noise profiles from the DEMAND noise database at SNR levels ranging from -10 dB to 30 dB.
Estimation of instantaneous SNR is also a subtask in many speech enhancement methods ( @cite_17 @cite_5 @cite_6 @cite_16 ). In @cite_20 , a neural network is trained to output the SNR in each frequency channel using amplitude modulation spectrogram (AMS) features which are obtained from the input signal. In @cite_14 , the peaks and valleys of the smoothened short time power estimate of a signal are used to estimate the noise power and instantaneous SNR.
@cite_14 @cite_6 @cite_5 @cite_16 @cite_20 @cite_17
202024829 2058502470 2124213909 2138456300 2153894152
This contribution presents an efficient algorithm to estimate the instantaneous signal-to-noise ratio of speech signals. The algorithm is capable to track non stationary noise signals and has a low computational complexity. It does not need a speech activity detector nor histograms to learn signal statistics. The algorithm is based on the observation that a noise power estimate can be obtained using minimum values of a smoothed power estimate. This paper will present this algorithm, its performance, its limits, and some applications. In this paper, we propose a statistical model for speech enhancement that takes into account the time-correlation between successive speech spectral components. It retains the simplicity associated with the Gaussian statistical model, and enables the extension of existing algorithms to noncausal estimation. The sequence of speech spectral variances is a random process, which is generally correlated with the sequence of speech spectral magnitudes. Causal and noncausal estimators for the a priori SNR are derived in agreement with the model assumptions and the estimation of the speech spectral components. We show that a special case of the causal estimator degenerates to a "decision-directed" estimator with a time-varying frequency-dependent weighting factor. Experimental results demonstrate the improved performance of the proposed algorithms. In this paper, we propose an MMSE a priori SNR estimator for speech enhancement. This estimator has similar benefits to the well-known decision-directed approach, but does not require an ad-hoc weighting factor to balance the past a priori SNR and current ML SNR estimate with smoothing across frames. Performance is evaluated in terms of estimation error and segmental SNR using the standard logSTSA speech enhancement method. Experimental results show that, in contrast with the decision-directed estimator and ML estimator, the proposed SNR estimator can help enhancement algorithms preserve more weak speech information and efficiently suppress musical noise. A single-microphone noise suppression algorithm is described that is based on a novel approach for the estimation of the signal-to-noise ratio (SNR) in different frequency channels: The input signal is transformed into neurophysiologically-motivated spectro-temporal input features. These patterns are called amplitude modulation spectrograms (AMS), as they contain information of both center frequencies and modulation frequencies within each 32 ms-analysis frame. The different representations of speech and noise in AMS patterns are detected by a neural network, which estimates the present SNR in each frequency channel. Quantitative experiments show a reliable estimation of the SNR for most types of nonspeech background noise. For noise suppression, the frequency bands are attenuated according to the estimated present SNR using a Wiener filter approach. Objective speech quality measures, informal listening tests, and the results of automatic speech recognition experiments indicate a substantial benefit from AMS-based noise suppression, in comparison to unprocessed noisy speech. This paper addresses the problem of single microphone frequency domain speech enhancement in noisy environments. The main characteristics of available frequency domain noise reduction algorithms are presented. We have confirmed that the a priori SNR estimation leads to the best subjective results. According to these conclusions, a new approach is then developed which achieves a trade-off between effective noise reduction and low computational load for real-time operations. The obtained solutions demonstrate that the subjective and objective results are much better than existing methods.