TY - CHAP A1 - Riedhammer, Korbinian A1 - Gropp, Martin A1 - Bocklet, Tobias A1 - Hönig, Florian A1 - Nöth, Elmar A1 - Steidl, Stefan T1 - LMELectures: A Multimedia Corpus of Academic Spoken English T2 - First Workshop on Speech, Language and Audio in Multimedia (SLAM 2013), Marseille, France, August 2013, ISCA Archive. N2 - This paper describes the acquisition, transcription and annotation of a multi-media corpus of academic spoken English, the LMELectures. It consists of two lecture se-ries that were read in the summer term 2009 at the com-puter science department of the University of Erlangen-Nuremberg, covering topics in pattern analysis, machine learning and interventional medical image processing. In total, about 40 hours of high-definition audio and video of a single speaker was acquired in a constant recording en-vironment. In addition to the recordings, the presentation slides are available in machine readable (PDF) format. The manual annotations include a suggested segmenta-tion into speech turns and a complete manual transcrip-tion that was done using BLITZSCRIBE2, a new tool for the rapid transcription. For one lecture series, the lecturer assigned key words to each recordings; one recording of that series was further annotated with a list of ranked key phrases by five human annotators each. The corpus is available for non-commercial purpose upon request. KW - corpus description KW - academic spoken English KW - e-learning Y1 - 2013 SP - 102 EP - 107 ER - TY - CHAP A1 - Bocklet, Tobias A1 - Riedhammer, Korbinian A1 - Eysholdt, Ulrich A1 - Nöth, Elmar T1 - Automatic Phoneme Analysis in Children with Cleft Lip and Palate T2 - 2013 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), Vancouver, BC, Canada, May 2013. N2 - Cleft Lip and Palate (CLP) is among the most frequent congenital abnormalities. The impaired facial development affects the articulation, with different phonemes being impacted inhomogeneously among different patients. This work focuses on automatic phoneme analysis of children with CLP for a detailed diagnosis and therapy control. In clinical routine, the state-of-the-art evaluation is based on perceptual evaluations. Perceptual ratings act as ground-truth throughout this work, with the goal to build an automatic system that is as reliable as humans. We propose two different automatic systems focusing on modeling the articulatory space of a speaker: one system models a speaker by a GMM, the other system employs a speech recognition system and estimates fMLLR matrices for each speaker. SVR is then used to predict the perceptual ratings. We show that the fMLLR-based system is able to achieve automatic phoneme evaluation results that are in the same range as perceptual inter-rater-agreements. KW - Pathology KW - automatic assessment KW - spectral features KW - GMM KW - fMLLR Y1 - 2013 SP - 7572 EP - 7576 PB - IEEE ER - TY - CHAP A1 - Hönig, Florian A1 - Bocklet, Tobias A1 - Riedhammer, Korbinian A1 - Batliner, Anton A1 - Nöth, Elmar T1 - The Automatic Assessment of Non-native Prosody: Combining Classical Prosodic Analysis with Acoustic Modelling T2 - INTERSPEECH 2012, 13th Annual Conference of the International Speech Communication Association (ISCA), Portland, OR, USA, September 2012. N2 - In earlier studies, we assessed the degree of non-nativeness employing prosodic information. In this paper, we combine prosodic information with (1) features derived from a Gaussian Mixture Model used as Universal Background Model (GMM-UBM), a powerful approach used in speaker identification, and (2) openSMILE, a standard open-source toolkit for extracting acoustic features. We evaluate our approach with English speech from 94 non-native speakers. GMM-UBM or openSMILE modelling alone yields lower performance than our prosodic feature vector; however, adding information from the GMM-UBM modelling or openSMILE by late fusion improves results. KW - computer-assisted language learning KW - non-native prosody KW - rhythm KW - automatic assessment Y1 - 2012 ER - TY - CHAP A1 - Riedhammer, Korbinian A1 - Bocklet, Tobias A1 - Goshal, Arnab A1 - Povey, Daniel T1 - Revisiting Semi-Continuous Hidden Markov Models T2 - 2012 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), Kyoto, Japan, March 2012. N2 - In the past decade, semi-continuous hidden Markov models (SCHMMs) have not attracted much attention in the speech recognition community. Growing amounts of training data and increasing sophistication of model estimation led to the impression that continuous HMMs are the best choice of acoustic model. However, recent work on recognition of under-resourced languages faces the same old problem of estimating a large number of parameters from limited amounts of transcribed speech. This has led to a renewed interest in methods of reducing the number of parameters while maintaining or extending the modeling capabilities of continuous models. In this work, we compare classic and multiple-codebook semi-continuous models using diagonal and full covariance matrices with continuous HMMs and subspace Gaussian mixture models. Experiments on the RM and WSJ corpora show that while a classical semicontinuous system does not perform as well as a continuous one, multiple-codebook semi-continuous systems can perform better, particular when using full-covariance Gaussians. KW - automatic speech recognition KW - acoustic modeling Y1 - 2012 SP - 4721 EP - 4724 PB - IEEE ER - TY - CHAP A1 - Steidl, Stefan A1 - Riedhammer, Korbinian A1 - Bocklet, Tobias A1 - Hönig, Florian A1 - Nöth, Elmar T1 - Java Visual Speech Components for Rapid Application Development of GUI based Speech Processing Applications T2 - INTERSPEECH 2011, 12th Annual Conference of the International Speech Communication Association, Florence, Italy, August 2011. N2 - In this paper, we describe a new Java framework for an easy and efficient way of developing new GUI based speech processing applications. Standard components are provided to display the speech signal, the power plot, and the spectrogram. Furthermore, a component to create a new transcription and to display and manipulate an existing transcription is provided, as well as a component to display and manually correct external pitch values. These Swing components can be easily embedded into own Java programs. They can be synchronized to display the same region of the speech file. The object-oriented design provides base classes for rapid development of own components. KW - Speech Processing Applications KW - Java KW - Graphical User Interfaces (GUI) KW - Rapid Application Development (RAD) Y1 - 2011 SP - 3257 EP - 3260 ER - TY - CHAP A1 - Bocklet, Tobias A1 - Riedhammer, Korbinian A1 - Nöth, Elmar T1 - Drink and Speak: On the automatic classification of alcohol intoxination by acoustic, prosodic and text-based features T2 - INTERSPEECH 2011, 12th Annual Conference of the International Speech Communication Association, Florence, Italy, August 2011. N2 - This paper focuses on the automatic detection of a person's blood level alcohol based on automatic speech processing approaches. We compare 5 different feature types with different ways of modeling. Experiments are based on the ALC corpus of IS2011 Speaker State Challenge. The classification task is restricted to the detection of a blood alcohol level above 0.5‰. Three feature sets are based on spectral observations: MFCCs, PLPs, TRAPS. These are modeled by GMMs. Classification is either done by a Gaussian classifier or by SVMs. In the later case classification is based on GMM-based supervectors, i.e. concatenation of GMM mean vectors. A prosodic system extracts a 292-dimensional feature vector based on a voiced-unvoiced decision. A transcription-based system makes use of text transcriptions related to phoneme durations and textual structure. We compare the stand-alone performances of these systems and combine them on score level by logistic regression. The best stand-alone performance is the transcriptionbased system which outperforms the baseline by 4.8% on the development set. A Combination on score level gave a huge boost when the spectral-based systems were added (73.6%). This is a relative improvement of 12.7% to the baseline. On the test-set we achieved an UA of 68.6% which is a significant improvement of 4.1% to the baseline system. KW - GMM KW - alcohol intoxication KW - system fusion Y1 - 2011 SP - 3213 EP - 3216 ER - TY - CHAP A1 - Riedhammer, Korbinian A1 - Bocklet, Tobias A1 - Nöth, Elmar T1 - Compensation of Extrinsic Variability in Speaker Verification Systems on Simulated Skype and HF Channel Data T2 - 2011 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), Prague, Czech Republic, May 2011. N2 - In this work we focus on speaker verification on channels of varying quality, namely Skype and high frequency (HF) radio. In our setup, we assume to have telephone recordings of speakers for training, but recordings of different channels for testing with varying (lower) signal quality. Starting from a Gaussian mixture / support vector machine (GMM/SVM) baseline, we evaluate multi-condition training (MCT), an ideal channel classification approach (ICC), and nuisance attribute projection (NAP) to compensate for the loss of information due to the transmission. In an evaluation on Switchboard-2 data using Skype and HF channel simulators, we show that, for good signal quality, NAP improves the baseline system performance from 5% EER to 3.33% EER (for both Skype and HF). For strongly distorted data, MCT or, if adequate, ICC turn out to be the method of choice. KW - speaker verification KW - channel compensation Y1 - 2011 SP - 4840 EP - 4843 ER - TY - CHAP A1 - Zäh, Uwe A1 - Riedhammer, Korbinian A1 - Bocklet, Tobias A1 - Nöth, Elmar T1 - Clap Your Hands! Calibrating Spectral Subtraction for Dereverberation T2 - 2010 IEEE International Conference on Acoustics, Speech and Signal Processing(ICASSP), Dallas, TX, USA, March 2010. N2 - Reverberation effects as observed by room microphones severely degrade the performance of automatic speech recognition systems. We investigate the use of dereverberation by spectral subtraction as proposed by Lebart and Boucher and introduce a simple approach to estimate the required decay parameter by clapping hands. Experiments on small vocabulary continuous speech recognition task on read speech show that using the calibrated dereverberation improves WER from 73.2 to 54.7 for the best microphone. In combination with system adaptation, the WER could be reduced to 28.2, which is only a 16% relative loss of performance comparison to using a headset instead of a room microphone. KW - Speech Recognition KW - robustness Y1 - 2010 SP - 4226 EP - 4229 ER - TY - CHAP A1 - Bocklet, Tobias A1 - Maier, Andreas A1 - Riedhammer, Korbinian A1 - Nöth, Elmar T1 - Towards a Language-independent Intelligibility Assessment of Children with Cleft Lip and Palate T2 - WOCCI '09 Proceedings of the 2nd Workshop on Child, Computer and Interaction (WOCCI 2009), Cambridge, Massachusetts, November 2009, Article No. 6. N2 - We describe a novel evaluation system for the intelligibility assessment of children with CLP on standardized tests. The system is solely based on standard cepstral features in form of MFCCs. No other information like word alignments is used. So the system can be easily adapted to other languages. For each child one GMM is created by adaptation of a UBM to the speaker-specific MFCCs. The components of this GMM are concatenated in order to create a so-called GMM supervector. These GMM supervectors are then used as meta features for an SVR. We evaluated our language-independent system on two different datasets of children suffering from CLP. One dataset contains recordings of 35 German children, where the children named different pictograms. The other dataset contains recordings of 14 Italian speaking children, who repeated standardized sentences. On both datasets we achieved high correlations: up to 0.81 for the German dataset and 0.83 for the Italian dataset. KW - Speech Recognition Y1 - 2009 SP - 2015 EP - 2018 ER - TY - JOUR A1 - Bocklet, Tobias A1 - Riedhammer, Korbinian A1 - Nöth, Elmar A1 - Eysholdt, Ulrich A1 - Haderlein, Tino T1 - Automatic Intelligibility Assessment of Speakers After Laryngeal Cancer by Means of Acoustic Modeling JF - Journal of Voice N2 - One aspect of voice and speech evaluation after laryngeal cancer is acoustic analysis. Perceptual evaluation by expert raters is a standard in the clinical environment for global criteria such as overall quality or intelligibility. So far, automatic approaches evaluate acoustic properties of pathologic voices based on voiced/unvoiced distinction and fundamental frequency analysis of sustained vowels. Because of the high amount of noisy components and the increasing aperiodicity of highly pathologic voices, a fully automatic analysis of fundamental frequency is difficult. We introduce a purely data-driven system for the acoustic analysis of pathologic voices based on recordings of a standard text. KW - Speech Recognition KW - Lung cancer technologies Y1 - 2012 IS - 26(3) SP - 390 EP - 397 ER -