Improving the robustness of the usual fbe-based asr front-end

All speech recognition systems require some form of signal representation that parametrically models the temporal evolution of the spectral envelope. Current parameterizations involve, either explicitly or implicitly, a set of energies from frequency bands which are often distributed in a mel scale. The computation of those filterbank energies (FBE) always includes smoothing of basic spectral measurements and non-linear amplitude compression. A variety of linear transformations are typically applied to this time-frequency representation prior to the Hidden Markov Model (HMM) pattern-matching stage of recognition. In the paper, we will discuss some robustness issues involved in both the computation of the FBEs and the posterior linear transformations, presenting alternative techniques that can improve robustness in additive noise conditions. In particular, the root non-linearity, a voicing-dependent FBE computation technique and a time&frequency filtering (tiffing) technique will be considered. Recognition results for the Aurora database will be shown to illustrate the potential application of these alternatives techniques for enhancing the robustness of speech recognition systems.Peer ReviewedPostprint (published version

UPM-UC3M system for music and speech segmentation

This paper describes the UPM-UC3M system for the Albayzín evaluation 2010 on Audio Segmentation. This evaluation task consists of segmenting a broadcast news audio document into clean speech, music, speech with noise in background and speech with music in background. The UPM-UC3M system is based on Hidden Markov Models (HMMs), including a 3-state HMM for every acoustic class. The number of states and the number of Gaussian per state have been tuned for this evaluation. The main analysis during system development has been focused on feature selection. Also, two different architectures have been tested: the first one corresponds to an one-step system whereas the second one is a hierarchical system in which different features have been used for segmenting the different audio classes. For both systems, we have considered long term statistics of MFCC (Mel Frequency Ceptral Coefficients), spectral entropy and CHROMA coefficients. For the best configuration of the one-step system, we have obtained a 25.3% average error rate and 18.7% diarization error (using the NIST tool) and a 23.9% average error rate and 17.9% diarization error for the hierarchical one