Exploiting correlogram structure for robust speech recognition with multiple speech sources

This paper addresses the problem of separating and recognising speech in a monaural acoustic mixture with the presence of competing speech sources. The proposed system treats sound source separation and speech recognition as tightly coupled processes. In the first stage sound source separation is performed in the correlogram domain. For periodic sounds, the correlogram exhibits symmetric tree-like structures whose stems are located on the delay that corresponds to multiple pitch periods. These pitch-related structures are exploited in the study to group spectral components at each time frame. Local pitch estimates are then computed for each spectral group and are used to form simultaneous pitch tracks for temporal integration. These processes segregate a spectral representation of the acoustic mixture into several time-frequency regions such that the energy in each region is likely to have originated from a single periodic sound source. The identified time-frequency regions, together with the spectral representation, are employed by a `speech fragment decoder' which employs `missing data' techniques with clean speech models to simultaneously search for the acoustic evidence that best matches model sequences. The paper presents evaluations based on artificially mixed simultaneous speech utterances. A coherence-measuring experiment is first reported which quantifies the consistency of the identified fragments with a single source. The system is then evaluated in a speech recognition task and compared to a conventional fragment generation approach. Results show that the proposed system produces more coherent fragments over different conditions, which results in significantly better recognition accuracy

A Novel Approach for Speech to Text Recognition System Using Hidden Markov Model

Speech recognition is the application of sophisticated algorithms which involve the transforming of the human voice to text. Speech identification is essential as it utilizes by several biometric identification systems and voice-controlled automation systems. Variations in recording equipment, speakers, situations, and environments make speech recognition a tough undertaking. Three major phases comprise speech recognition: speech pre-processing, feature extraction, and speech categorization. This work presents a comprehensive study with the objectives of comprehending, analyzing, and enhancing these models and approaches, such as Hidden Markov Models and Artificial Neural Networks, employed in the voice recognition system for feature extraction and classification

Falling person detection using multisensor signal processing

Falls are one of the most important problems for frail and elderly people living independently. Early detection of falls is vital to provide a safe and active lifestyle for elderly. Sound, passive infrared (PIR) and vibration sensors can be placed in a supportive home environment to provide information about daily activities of an elderly person. In this paper, signals produced by sound, PIR and vibration sensors are simultaneously analyzed to detect falls. Hidden Markov Models are trained for regular and unusual activities of an elderly person and a pet for each sensor signal. Decisions of HMMs are fused together to reach a final decision

Convolutive Blind Source Separation Methods

In this chapter, we provide an overview of existing algorithms for blind source separation of convolutive audio mixtures. We provide a taxonomy, wherein many of the existing algorithms can be organized, and we present published results from those algorithms that have been applied to real-world audio separation tasks

Studies on noise robust automatic speech recognition

Noise in everyday acoustic environments such as cars, traffic environments, and cafeterias remains one of the main challenges in automatic speech recognition (ASR). As a research theme, it has received wide attention in conferences and scientific journals focused on speech technology. This article collection reviews both the classic and novel approaches suggested for noise robust ASR. The articles are literature reviews written for the spring 2009 seminar course on noise robust automatic speech recognition (course code T-61.6060) held at TKK

Noise Robust Pitch Tracking by Subband Autocorrelation Classification

Pitch tracking algorithms have a long history in various applications such as speech coding and extracting information, as well as other domains such as bioacoustics and music signal processing. While autocorrelation is a useful technique for detecting periodicity, autocorrelation peaks suffer ambiguity, leading to the classic “octave error” in pitch tracking. Moreover, additive noise can affect autocorrelation in ways that are difficult to model. Instead of explicitly using the most obvious features of autocorrelation, we present a trained classifier-based approach which we call Subband Autocorrelation Classification (SAcC). A multi-layer perceptron classifier is trained on the principal components of the autocorrelations of subbands from an auditory filterbank. Training on bandlimited and noisy speech (processed to simulate a low-quality radio channel) leads to a great increase in performance over state-of-the-art algorithms, according to both the traditional GPE measure, and a proposed novel Pitch Tracking Error which more fully reflects the accuracy of both pitch extraction and voicing detection in a single measure