The Sheffield Wargames Corpus.

Recognition of speech in natural environments is a challenging task, even more so if this involves conversations between sev-eral speakers. Work on meeting recognition has addressed some of the significant challenges, mostly targeting formal, business style meetings where people are mostly in a static position in a room. Only limited data is available that contains high qual-ity near and far field data from real interactions between par-ticipants. In this paper we present a new corpus for research on speech recognition, speaker tracking and diarisation, based on recordings of native speakers of English playing a table-top wargame. The Sheffield Wargames Corpus comprises 7 hours of data from 10 recording sessions, obtained from 96 micro-phones, 3 video cameras and, most importantly, 3D location data provided by a sensor tracking system. The corpus repre-sents a unique resource, that provides for the first time location tracks (1.3Hz) of speakers that are constantly moving and talk-ing. The corpus is available for research purposes, and includes annotated development and evaluation test sets. Baseline results for close-talking and far field sets are included in this paper. 1

Using contextual information in Joint Factor Eigenspace MLLR for speech recognition in diverse scenarios

This paper presents a new approach for rapid adaptation in the presence of highly diverse scenarios that takes advantage of information describing the input signals. We introduce a new method for joint factorisation of the background and the speaker in an eigenspace MLLR framework: Joint Factor Eigenspace MLLR (JFEMLLR). We further propose to use contextual information describing the speaker and background, such as tags or more complex metadata, to provide an immediate estimation of the best MLLR transformation for the utterance. This provides instant adaptation, since it does not require any transcription from a previous decoding stage. Evaluation in a highly diverse Automatic Speech Recognition (ASR) task, a modified version of WSJCAM0, yields an improvement of 26.9% over the baseline, which is an extra 1.2% reduction over two-pass MLLR adaptation

Asynchronous factorisation of speaker and background with feature transforms in speech recognition

This paper presents a novel approach to separate the effects of speaker and background conditions by application of featuretransform based adaptation for Automatic Speech Recognition (ASR). So far factorisation has been shown to yield improvements in the case of utterance-synchronous environments. In this paper we show successful separation of conditions asynchronous with speech, such as background music. Our work takes account of the asynchronous nature of the background, by estimation of condition-specific Constrained Maximum Likelihood Linear Regression (CMLLR) transforms. In addition, speaker adaptation is performed, allowing to factorise speaker and background effects. Equally, background transforms are used asynchronously in the decoding process, using a modified Hidden Markov Model (HMM) topology which applies the optimal transform for each frame. Experimental results are presented on the WSJCAM0 corpus of British English speech, modified to contain controlled sections of background music. This addition of music degrades the baseline Word Error Rate (WER) from 10.1% to 26.4%. While synchronous factorisation with CMLLR transforms provides 28% relative improvement in WER over the baseline, our asynchronous approach increases this reduction to 33%

DNN approach to speaker diarisation using speaker channels

Speaker diarisation addresses the question of 'who speaks when' in audio recordings, and has been studied extensively in the context of tasks such as broadcast news, meetings, etc. Performing diarisation on individual headset microphone (IHM) channels is sometimes assumed to easily give the desired output of speaker labelled segments with timing information. However, it is shown that given imperfect data, such as speaker channels with heavy crosstalk and overlapping speech, this is not the case. Deep neural networks (DNNs) can be trained on features derived from the concatenation of speaker channel features to detect which is the correct channel for each frame. Crosstalk features can be calculated and DNNs trained with or without overlapping speech to combat problematic data. A simple frame decision metric of counting occurrences is investigated as well as adding a bias against selecting nonspeech for a frame. Finally, two different scoring setups are applied to both datasets. The stricter SHEF setup finds diarisation error rates (DER) of 9.2% on TBL and 23.2% on RT07 while the NIST setup achieves 5.7% and 15.1% respectively

Acoustic Adaptation to Dynamic Background Conditions with Asynchronous Transformations

This paper proposes a framework for performing adaptation to complex and non-stationary background conditions in Automatic Speech Recognition (ASR) by means of asynchronous Constrained Maximum Likelihood Linear Regression (aCMLLR) transforms and asynchronous Noise Adaptive Training (aNAT). The proposed method aims to apply the feature transform that best compensates the background for every input frame. The implementation is done with a new Hidden Markov Model (HMM) topology that expands the usual left-to-right HMM into parallel branches adapted to different background conditions and permits transitions among them. Using this, the proposed adaptation does not require ground truth or previous knowledge about the background in each frame as it aims to maximise the overall log-likelihood of the decoded utterance. The proposed aCMLLR transforms can be further improved by retraining models in an aNAT fashion and by using speaker-based MLLR transforms in cascade for an efficient modelling of background effects and speaker. An initial evaluation in a modified version of the WSJCAM0 corpus incorporating 7 different background conditions provides a benchmark in which to evaluate the use of aCMLLR transforms. A relative reduction of 40.5% in Word Error Rate (WER) was achieved by the combined use of aCMLLR and MLLR in cascade. Finally, this selection of techniques was applied in the transcription of multi-genre media broadcasts, where the use of aNAT training, aCMLLR transforms and MLLR transforms provided a relative improvement of 2–3%

DNN-based speaker clustering for speaker diarisation

Speaker diarisation, the task of answering "who spoke when?", is often considered to consist of three independent stages: speech activity detection, speaker segmentation and speaker clustering. These represent the separation of speech and nonspeech, the splitting into speaker homogeneous speech segments, followed by grouping together those which belong to the same speaker. This paper is concerned with speaker clustering, which is typically performed by bottom-up clustering using the Bayesian information criterion (BIC). We present a novel semi-supervised method of speaker clustering based on a deep neural network (DNN) model. A speaker separation DNN trained on independent data is used to iteratively relabel the test data set. This is achieved by reconfiguration of the output layer, combined with fine tuning in each iteration. A stopping criterion involving posteriors as confidence scores is investigated. Results are shown on a meeting task (RT07) for single distant microphones and compared with standard diarisation approaches. The new method achieves a diarisation error rate (DER) of 14.8%, compared to a baseline of 19.9%

Colloquialising modern standard Arabic text for improved speech recognition

Modern standard Arabic (MSA) is the official language of spoken and written Arabic media. Colloquial Arabic (CA) is the set of spoken variants of modern Arabic that exist in the form of regional dialects. CA is used in informal and everyday conversations while MSA is formal communication. An Arabic speaker switches between the two variants according to the situation. Developing an automatic speech recognition system always requires a large collection of transcribed speech or text, and for CA dialects this is an issue. CA has limited textual resources because it exists only as a spoken language, without a standardised written form unlike MSA. This paper focuses on the data sparsity issue in CA textual resources and proposes a strategy to emulate a native speaker in colloquialising MSA to be used in CA language models (LMs) by use of a machine translation (MT) framework. The empirical results in Levantine CA show that using LMs estimated from colloquialised MSA data outperformed MSA LMs with a perplexity reduction up to 68% relative. In addition, interpolating colloquialised MSA LMs with a CA LMs improved speech recognition performance by 4% relative

Experimental demonstration of composite stimulated Raman adiabatic passage

We experimentally demonstrate composite stimulated Raman adiabatic passage (CSTIRAP), which combines the concepts of composite pulse sequences and adiabatic passage. The technique is applied for population transfer in a rare-earth doped solid. We compare the performance of CSTIRAP with conventional single and repeated STIRAP, either in the resonant or the highly detuned regime. In the latter case, CSTIRAP improves the peak transfer efficiency and robustness, boosting the transfer efficiency substantially compared to repeated STIRAP. We also propose and demonstrate a universal version of CSTIRAP, which shows improved performance compared to the originally proposed composite version. Our findings pave the way towards new STIRAP applications, which require repeated excitation cycles, e.g., for momentum transfer in atom optics, or dynamical decoupling to invert arbitrary superposition states in quantum memories.Comment: 11 pages, 5 figure