1,410 research outputs found

    The ICSI Meeting Corpus

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    We have collected a corpus of data from natural meetings that occurred at the International Computer Science Institute (ICSI) in Berkeley, California over the last three years. The corpus contains audio recorded simultaneously from head-worn and table-top microphones, word-level transcripts of meetings, and various metadata on participants, meetings, and hardware. Such a corpus supports work in automatic speech recognition, noise robustness, dialog modeling, prosody, rich transcription, information retrieval, and more. We present details on the contents of the corpus, as well as rationales for the decisions that led to its configuration. The corpus were delivered to the Linguistic Data Consortium (LDC)

    On the acoustics of overlapping laughter in conversational speech

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    The social nature of laughter invites people to laugh together. This joint vocal action often results in overlapping laughter. In this paper, we show that the acoustics of overlapping laughs are different from non-overlapping laughs. We found that overlapping laughs are stronger prosodically marked than non-overlapping ones, in terms of higher values for duration, mean F0, mean and maximum intensity, and the amount of voicing. This effect is intensified by the number of people joining in the laughter event, which suggests that entrainment is at work. We also found that group size affects the number of overlapping laughs which illustrates the contagious nature of laughter. Finally, people appear to join laughter simultaneously at a delay of approximately 500 ms; a delay that must be considered when developing spoken dialogue systems that are able to respond to users’ laughs

    The 2005 AMI system for the transcription of speech in meetings

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    In this paper we describe the 2005 AMI system for the transcription\ud of speech in meetings used for participation in the 2005 NIST\ud RT evaluations. The system was designed for participation in the speech\ud to text part of the evaluations, in particular for transcription of speech\ud recorded with multiple distant microphones and independent headset\ud microphones. System performance was tested on both conference room\ud and lecture style meetings. Although input sources are processed using\ud different front-ends, the recognition process is based on a unified system\ud architecture. The system operates in multiple passes and makes use\ud of state of the art technologies such as discriminative training, vocal\ud tract length normalisation, heteroscedastic linear discriminant analysis,\ud speaker adaptation with maximum likelihood linear regression and minimum\ud word error rate decoding. In this paper we describe the system performance\ud on the official development and test sets for the NIST RT05s\ud evaluations. The system was jointly developed in less than 10 months\ud by a multi-site team and was shown to achieve very competitive performance

    Joint Modeling of Content and Discourse Relations in Dialogues

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    We present a joint modeling approach to identify salient discussion points in spoken meetings as well as to label the discourse relations between speaker turns. A variation of our model is also discussed when discourse relations are treated as latent variables. Experimental results on two popular meeting corpora show that our joint model can outperform state-of-the-art approaches for both phrase-based content selection and discourse relation prediction tasks. We also evaluate our model on predicting the consistency among team members' understanding of their group decisions. Classifiers trained with features constructed from our model achieve significant better predictive performance than the state-of-the-art.Comment: Accepted by ACL 2017. 11 page

    Low-rank and Sparse Soft Targets to Learn Better DNN Acoustic Models

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    Conventional deep neural networks (DNN) for speech acoustic modeling rely on Gaussian mixture models (GMM) and hidden Markov model (HMM) to obtain binary class labels as the targets for DNN training. Subword classes in speech recognition systems correspond to context-dependent tied states or senones. The present work addresses some limitations of GMM-HMM senone alignments for DNN training. We hypothesize that the senone probabilities obtained from a DNN trained with binary labels can provide more accurate targets to learn better acoustic models. However, DNN outputs bear inaccuracies which are exhibited as high dimensional unstructured noise, whereas the informative components are structured and low-dimensional. We exploit principle component analysis (PCA) and sparse coding to characterize the senone subspaces. Enhanced probabilities obtained from low-rank and sparse reconstructions are used as soft-targets for DNN acoustic modeling, that also enables training with untranscribed data. Experiments conducted on AMI corpus shows 4.6% relative reduction in word error rate

    Speech and crosstalk detection in multichannel audio

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    The analysis of scenarios in which a number of microphones record the activity of speakers, such as in a round-table meeting, presents a number of computational challenges. For example, if each participant wears a microphone, speech from both the microphone's wearer (local speech) and from other participants (crosstalk) is received. The recorded audio can be broadly classified in four ways: local speech, crosstalk plus local speech, crosstalk alone and silence. We describe two experiments related to the automatic classification of audio into these four classes. The first experiment attempted to optimize a set of acoustic features for use with a Gaussian mixture model (GMM) classifier. A large set of potential acoustic features were considered, some of which have been employed in previous studies. The best-performing features were found to be kurtosis, "fundamentalness," and cross-correlation metrics. The second experiment used these features to train an ergodic hidden Markov model classifier. Tests performed on a large corpus of recorded meetings show classification accuracies of up to 96%, and automatic speech recognition performance close to that obtained using ground truth segmentation

    Fundamental frequency height as a resource for the management of overlap in talk-in-interaction.

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    Overlapping talk is common in talk-in-interaction. Much of the previous research on this topic agrees that speaker overlaps can be either turn competitive or noncompetitive. An investigation of the differences in prosodic design between these two classes of overlaps can offer insight into how speakers use and orient to prosody as a resource for turn competition. In this paper, we investigate the role of fundamental frequency (F0) as a resource for turn competition in overlapping speech. Our methodological approach combines detailed conversation analysis of overlap instances with acoustic measurements of F0 in the overlapping sequence and in its local context. The analyses are based on a collection of overlap instances drawn from the ICSI Meeting corpus. We found that overlappers mark an overlapping incoming as competitive by raising F0 above their norm for turn beginnings, and retaining this higher F0 until the point of overlap resolution. Overlappees may respond to these competitive incomings by returning competition, in which case they raise their F0 too. Our results thus provide instrumental support for earlier claims made on impressionistic evidence, namely that participants in talk-in-interaction systematically manipulate F0 height when competing for the turn

    Using Corpus and Knowledge-Based Similarity Measure in Maximum Marginal Relevance for Meeting Summarization

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    MMR (Maximum Marginal Relevance) is widely used in summarization for its simplicity and efficacy, and has been demonstrated to achieve comparable performance to other approaches for meeting summarization. How to appropriately represent the similarity of two text segments is crucial in MMR. In this paper, we evaluate different similarity measures in the MMR framework for meeting summarization on the ICSI meeting corpus. We introduce a corpusbased measure to capture the similarity at the semantic level, and compare this method with cosine similarity and centroid score that only considers the salient words in the segments. Our experimental results evaluated by the ROUGE summarization metrics show that both the centroid score and the corpus-based similarity measure yield better performance than the commonly used cosine similarity. In addition, adding part-of-speech information in the corpus-based approach helps for the human transcripts condition, but not when using ASR output. Index Terms — meeting summarization, MMR, centroid score, corpus-based similarity 1

    Comparing non-verbal vocalisations in conversational speech corpora

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    Conversations do not only consist of spoken words but they also consist of non-verbal vocalisations. Since there is no standard to define and to classify (possible) non-speech sounds the annotations for these vocalisations differ very much for various corpora of conversational speech. There seems to be agreement in the six inspected corpora that hesitation sounds and feedback vocalisations are considered as words (without a standard orthography). The most frequent non-verbal vocalisation are laughter on the one hand and, if considered a vocal sound, breathing noises on the other

    On the effect of SNR and superdirective beamforming in speaker diarisation in meetings

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    This paper examines the effect of sensor performance on speaker diarisation in meetings and investigates the use of more advanced beamforming techniques, beyond the typically employed delay-sum beamformer, for mitigating the effects of poorer sensor performance. We present superdirective beamforming and investigate how different time difference of arrival (TDOA) smoothing and beamforming techniques influence the performance of state-of-the-art diarisation systems. We produced and transcribed a new corpus of meetings recorded in the instrumented meeting room using a high SNR analogue and a newly developed low SNR digital MEMS microphone array (DMMA.2). This research demonstrates that TDOA smoothing has a significant effect on the diarisation error rate and that simple noise reduction and beamforming schemes suffice to overcome audio signal degradation due to the lower SNR of modern MEMS microphones. Index Terms — Speaker diarisation in meetings, digital MEMS microphone array, time difference of arrival (TDOA), superdirective beamforming 1
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