Unsupervised Speech/Non-speech Detection for Automatic Speech Recognition in Meeting Rooms

The goal of this work is to provide robust and accurate speech detection for automatic speech recognition (ASR) in meeting room settings. The solution is based on computing long-term modulation spectrum, and examining specific frequency range for dominant speech components to classify speech and non-speech signals for a given audio signal. Manually segmented speech segments, short-term energy, short-term energy and zero-crossing based segmentation techniques, and a recently proposed Multi Layer Perceptron (MLP) classifier system are tested for comparison purposes. Speech recognition evaluations of the segmentation methods are performed on a standard database and tested in conditions where the signal-to-noise ratio (SNR) varies considerably, as in the cases of close-talking headset, lapel, distant microphone array output, and distant microphone. The results reveal that the proposed method is more reliable and less sensitive to mode of signal acquisition and unforeseen conditions

Investigating Fine Temporal Dynamics of Prosodic and Lexical Accommodation

Conversational interaction is a dynamic activity in which participants engage in the construction of meaning and in establishing and maintaining social relationships. Lexical and prosodic accommodation have been observed in many studies as contributing importantly to these dimensions of social interaction. However, while previous works have considered accommodation mechanisms at global levels (for whole conversations, halves and thirds of conversations), this work investigates their evolution through repeated analysis at time intervals of increasing granularity to analyze the dynamics of alignment in a spoken language corpus. Results show that the levels of both prosodic and lexical accommodation fluctuate several times over the course of a conversation

Exploiting contextual information for speech/non-speech detection

In this paper, we investigate the effect of temporal context for speech/non-speech detection (SND). It is shown that even a simple feature such as full-band energy, when employed with a large-enough context, shows promise for further investigation. Experimental evaluations on the test data set, with a state-of-the-art multi-layer perceptron based SND system and a simple energy threshold based SND method, using the F-measure, show an absolute performance gain of $4.4\%$ and $5.4\%$ respectively. The optimal contextual length was found to be 1000 ms. Further numerical optimizations yield an improvement ($3.37\%$ absolute), resulting in an absolute gain of $7.77\%$ and $8.77\%$ over the MLP based and energy based methods respectively. ROC based performance evaluation also reveals promising performance for the proposed method, particularly in low SNR conditions

Exploiting Contextual Information for Speech/Non-Speech Detection

In this paper, we investigate the effect of temporal context for speech/non-speech detection (SND). It is shown that even a simple feature such as full-band energy, when employed with a large-enough context, shows promise for further investigation. Experimental evaluations on the test data set, with a state-of-the-art multi-layer perceptron based SND system and a simple energy threshold based SND method, using the F-measure, show an absolute performance gain of 4.4% and 5.4% respectively. The optimal contextual length was found to be 1000 ms. Further numerical optimizations yield an improvement (3.37% absolute), resulting in an absolute gain of 7.77% and 8.77% over the MLP based and energy based methods respectively. ROC based performance evaluation also reveals promising performance for the proposed method, particularly in low SNR conditions

Exploiting temporal context for speech/non-speech detection

In this paper, we investigate the effect of temporal context for speech/non-speech detection (SND). It is shown that even a simple feature such as full-band energy, when employed with a large-enough context, shows promise for further investigation. Experimental evaluations on the test data set, with a state-of-the-art multi-layer perceptron based SND system and a simple energy threshold based SND method, using the F-measure, show an absolute performance gain of $4.4\%$ and $5.4\%$ respectively, when used with a context of 1000 ms. ROC based performance evaluation also reveals promising performance for the proposed method, particularly in low SNR conditions

Deep belief network based audio classification for construction sites monitoring

In this paper, we propose a Deep Belief Network (DBN) based approach for the classification of audio signals to improve work activity identification and remote surveillance of construction projects. The aim of the work is to obtain an accurate and flexible tool for consistently executing and managing the unmanned monitoring of construction sites by using distributed acoustic sensors. In this paper, ten classes of multiple construction equipment and tools, frequently and broadly used in construction sites, have been collected and examined to conduct and validate the proposed approach. The input provided to the DBN consists in the concatenation of several statistics evaluated by a set of spectral features, like MFCCs and mel-scaled spectrogram. The proposed architecture, along with the preprocessing and the feature extraction steps, has been described in details while the effectiveness of the proposed idea has been demonstrated by some numerical results, evaluated by using real-world recordings. The final overall accuracy on the test set is up to 98% and is a significantly improved performance compared to other state-of-the-are approaches. A practical and real-time application of the presented method has been also proposed in order to apply the classification scheme to sound data recorded in different environmental scenarios

Investigating Automatic Measurements of Prosodic Accommodation and Its Dynamics in Social Interaction

Spoken dialogue systems are increasingly being used to facilitate and enhance human communication. While these interactive systems can process the linguistic aspects of human communication, they are not yet capable of processing the complex dynamics involved in social interaction, such as the adaptation on the part of interlocutors. Providing interactive systems with the capacity to process and exhibit this accommodation could however improve their efficiency and make machines more socially-competent interactants. At present, no automatic system is available to process prosodic accommodation, nor do any clear measures exist that quantify its dynamic manifestation. While it can be observed to be a monotonically manifest property, it is our hypotheses that it evolves dynamically with functional social aspects. In this paper, we propose an automatic system for its measurement and the capture of its dynamic manifestation. We investigate the evolution of prosodic accommodation in 41 Japanese dyadic telephone conversations and discuss its manifestation in relation to its functions in social interaction. Overall, our study shows that prosodic accommodation changes dynamically over the course of a conversation and across conversations, and that these dynamics inform about the naturalness of the conversation flow, the speakers’ degree of involvement and their affinity in the conversation

Unsupervised speech/non-speech detection for automatic speech recognition in meeting rooms

The goal of this work is to provide robust and accurate speech detection for automatic speech recognition (ASR) in meeting room settings. The solution is based on computing long-term modulation spectrum, and examining specific frequency range for dominant speech components to classify speech and non-speech signals for a given audio signal. Manually segmented speech segments, shortterm energy, short-term energy and zero-crossing based segmentation techniques, and a recently proposed Multi Layer Perceptron (MLP) classifier system are tested for comparison purposes. Speech recognition evaluations of the segmentation methods are performed on a standard database and tested in conditions where the signal-tonoise ratio (SNR) varies considerably, as in the cases of close-talking headset, lapel, distant microphone array output, and distant microphone. The results reveal that the proposed method is more reliable and less sensitive to mode of signal acquisition and unforeseen conditions. Index Terms — Acoustic signal detection, speech recognition 1