TY - CHAP A1 - Riedhammer, Korbinian A1 - Bocklet, Tobias A1 - Orozco-Arroyave, Juan Rafael A1 - Nöth, Elmar T1 - Semi-Automatic Calibration for Dereverberation by Spectral Subtraction for Continuous Speech Recognition T2 - ITG Symposium on Speech Communication 2014, Erlangen. N2 - In this article, we describe a semi-automatic calibration algorithm for dereverberation by spectral subtraction. We verify the method by a comparison to a manual calibration derived from measured room impulse responses (RIR). We conduct extensive experiments to understand the effect of all involved parameters and to verify values suggested in the literature. The experiments are performed on a text read by 31 speakers and recorded by a headset and three far-field microphones. Results are measured in terms of automatic speech recognition (ASR) performance using a 1-gram model to emphasize acoustic recognition performance. To accommodate for the acoustic change by dereverberation we apply supervised MAP adaptation to the hidden Markov model output probabilities. The combination of dereverberation and adaptation yields a relative improvement of about 35% in terms of word error rate (WER) compared to the original signal. KW - Speech Recognition Y1 - 2014 PB - VDE VERLAG GMBH ER - TY - CHAP A1 - Wegmann, Steven A1 - Faria, Arlo A1 - Janin, Adam A1 - Riedhammer, Korbinian A1 - Morgan, Nelson T1 - The Tao of ATWV: Probing the mysteries of keyword search performance T2 - IEEE Workshop on Automatic Speech Recognition and Understanding (ASRU), Olomouc, Czech Republic, Dezember 2013. N2 - In this paper we apply diagnostic analysis to gain a deeper understanding of the performance of the the keyword search system that we have developed for conversational telephone speech in the IARPA Babel program. We summarize the Babel task, its primary performance metric, “actual term weighted value” (ATWV), and our recognition and keyword search systems. Our analysis uses two new oracle ATWV measures, a bootstrap-based ATWV confidence interval, and includes a study of the underpinnings of the large ATWV gains due to system combination. This analysis quantifies the potential ATWV gains from improving the number of true hits and the overall quality of the detection scores in our system's posting lists. It also shows that system combination improves our systems' ATWV via a small increase in the number of true hits in the posting lists. KW - Speech Recognition Y1 - 2013 ER - TY - CHAP A1 - Riedhammer, Korbinian A1 - Hai Do, Van A1 - Hieronymus, James T1 - A Study on LVCSR and Keyword Search for Tagalog T2 - INTERSPEECH 2013, 14th Annual Conference of the International Speech Communication Association (ISCA), Lyon, France, August 2013. N2 - We describe a state-of-the-art large vocabulary continuous speech recognition (LVCSR) and keyword search (KWS) system trained on roughly 70 hours of conversational telephone speech. Using the Kaldi speech recognition toolkit, we investigate several aspects: for the acoustic front-end, we analyze the use of mel-frequency cepstral coefficients (MFCC), pitch and probability-of-voicing (PoV), and deep neural network (DNN) bottleneck (BN) features, as well as their feature-level combination ("tandem"). For the acousticphonetic decision tree, we explore different hidden Markov model (HMM) topologies for the glottalization phoneme /?/ to model its typically short duration. For the acoustic model, we compare regular continuous HMM with a sort of multi-codebook subspace Gaussian mixture model (SGMM) that lead to an overall best word error rate (WER) of 58.7% and 56.3%, respectively. The KWS is implemented as a word lattice search, and is augmented by a syllable lattice back-up search to capture out-of-vocabulary keywords as well as misrecognized lexical surface forms due to ambiguous prefix and hyphenation rules. KW - Speech Recognition KW - keyword spotting Y1 - 2013 ER - TY - CHAP A1 - Riedhammer, Korbinian A1 - Ring, Martin A1 - Nöth, Elmar A1 - Kolb, Daniel T1 - A Software Kit for Automatic Voice Descrambling T2 - 2012 IEEE International Conference on Communications (ICC); IEEE International Workshop on Security and Forensics in Communication Systems (SFCS). N2 - Voice scrambling is widely used to add privacy to the radio communication of various authorities - but is also used by criminals to evade prosecution. In this article, we consider various analog voice scrambling techniques such as fixed frequency inversion, splitband inversion and rolling code scramblers. We explain how to break them using automatically extracted measures and scoring algorithms, and evaluate the proposed system using simulated data. While the simple inversion can be easily broken, the more advanced techniques require additional work prior to unsupervised automatization; the presented user interface allows the user to refine the automatic results to obtain a high quality solution. KW - Speech Recognition Y1 - 2012 SP - 8349 EP - 8353 ER - TY - CHAP A1 - Povey, Daniel A1 - Hannemann, Mirko A1 - Boulianne, Gilles A1 - Burget, Lukáš A1 - Ghoshal, Arnab A1 - Janda, Miloš A1 - Karafiát, Martin A1 - Kombrink, Stefan A1 - Motlíček, Petr A1 - Qian, Yanmin A1 - Riedhammer, Korbinian A1 - Veselý, Karel A1 - Thang Vu, Ngoc T1 - Generating Exact Lattices in the WFST Framework T2 - 2012 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), Kyoto, Japan, May 2012. N2 - We describe a lattice generation method that is exact, i.e. it satisfies all the natural properties we would want from a lattice of alternative transcriptions of an utterance. This method does not introduce substantial overhead above one-best decoding. Our method is most directly applicable when using WFST decoders where the WFST is “fully expanded”, i.e. where the arcs correspond to HMM transitions. It outputs lattices that include HMM-state-level alignments as well as word labels. The general idea is to create a state-level lattice during decoding, and to do a special form of determinization that retains only the best-scoring path for each word sequence. This special determinization algorithm is a solution to the following problem: Given a WFST A, compute a WFST B that, for each input-symbol-sequence of A, contains just the lowest-cost path through A. KW - Speech Recognition KW - Lattice Generation Y1 - 2012 SP - 4213 EP - 4216 ER - TY - CHAP A1 - Riedhammer, Korbinian A1 - Gropp, Martin A1 - Nöth, Elmar T1 - A Novel Lecture Browser Using Key Phrases and Stream Graphs T2 - Lehrstuhl für Mustererkennung, Universität Erlangen-Nürnberg N2 - We present a novel lecture browser that utilizes ranked key phrases displayed on a stream graph to overcome the shortcomings of traditional extractive (query-based) summaries. The system extracts key phrases from the ASR transcripts, performs an unsupervised ranking, and displays an initial number of phrases on the stream graph. This graph gives an intuition of when which key phrase is spoken, and how dominant it is throughout the lecture. The user can select the phrases to be displayed and furthermore adjust the ranking of the all phrases. All user interactions are logged to a server to improve the ranking algorithms and provide user specific rankings. KW - Speech Recognition Y1 - 2011 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 - Spiegl, Werner A1 - Riedhammer, Korbinian A1 - Steidl, Stefan A1 - Nöth, Elmar T1 - FAU IISAH Corpus – A German Speech Database Consisting of Human-Machine and Human-Human Interaction Acquired by Close-Talking and Far-Distance Microphones T2 - 7th Conference on International Language Resources and Evaluation (LREC 2010), European Language Resources Association (ELRA), Valletta, Malta, May 2010. KW - Speech Recognition Y1 - 2018 SP - 2420 EP - 2423 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 - CHAP A1 - Faria, Arlo A1 - Riedhammer, Korbinian A1 - Janin, Adam A1 - Bauer, A. T1 - REMEETING: Searchable Conversations T2 - INTERSPEECH 2016, Annual Conference of the International Speech Communication Association (ISCA), IEEE Workshop on Spoken Language Technologies (SLT), San Francisco, USA, September 2016. KW - Speech Recognition Y1 - 2016 ER - TY - CHAP A1 - Faria, Arlo A1 - Riedhammer, Korbinian T1 - REMEETING — Get More Out Of Meetings T2 - INTERSPEECH 2015, 16th Annual Conference of the International Speech Communication Association, Dresden, Germany, September 2015. N2 - Remeeting is a tool that helps you get more out of in-person meetings. Calendar integration and a special email address allow users to email agenda items prior to a certain meeting. A discrete notification at the time of the meeting reminds the user to start the recording. During the meeting, the user focuses on the conversation, or can add notes and photos if desired. After the meeting, every participant gets notified by an automated email that lists the participants along with automatically extracted keywords, notes and photos. This stimulates collaboration, and keeps follow-up contributions at a central place: Just reply to add further notes to the meeting. The resulting meeting “document” can be shared with others and reviewed using a web app that acts as a visual index to the meeting. This makes Remeeting the perfect tool for regular group meetings, standups and interviews, where people typically track progress and follow up on. Remeeting is leveraging, promoting and contributing to open source projects including kaldi and docker. KW - Speech Recognition KW - speaker identification KW - automatic summarization KW - productivity Y1 - 2015 ER - TY - CHAP A1 - Faria, Arlo A1 - Riedhammer, Korbinian T1 - REMEETING — Get More Out of Meetings T2 - INTERSPEECH 2014, 15th Annual Conference of the International Speech Communication Association, IEEE Workshop on Spoken Language Technologies (SLT), Singapore, September 2014. KW - Speech Recognition KW - speaker identification KW - automatic summarization KW - productivity Y1 - 2014 ER - TY - CHAP A1 - Tur, Gokhan A1 - Stolcke, Andreas A1 - Voss, Lynn A1 - Dowding, John A1 - Favre, Benoit A1 - Fernandez, Raquel A1 - Frampton, Matthew A1 - Frandsen, Michael A1 - Frederickson, Clive A1 - Graciarena, Martin A1 - Hakkani-Tür, Dilek A1 - Kintzing, Donald A1 - Leveque, Kyle A1 - Mason, Shane A1 - Niekrasz, John A1 - Peters, Stanley A1 - Purver, Matthew A1 - Riedhammer, Korbinian A1 - Shriberg, Elizabeth A1 - Tien, Jing A1 - Vergyri, Dimitra A1 - Yang, Fan T1 - The CALO Meeting Speech Recognition and Understanding System T2 - 2008 IEEE Workshop on Spoken Language Technologies (SLT), Goa, India, December 2008. N2 - The CALO meeting assistant provides for distributed meeting capture, annotation, automatic transcription and semantic analysis of multiparty meetings, and is part of the larger CALO personal assistant system. This paper summarizes the CALO-MA architecture and its speech recognition and understanding components, which include real-time and offline speech transcription, dialog act segmentation and tagging, question-answer pair identification, action item recognition, decision extraction, and summarization. KW - Speech Recognition KW - multiparty meetings processing KW - spoken language understanding Y1 - 2008 SP - 69 EP - 72 ER - TY - CHAP A1 - Riedhammer, Korbinian A1 - Haderlein, Tino A1 - Schuster, Maria A1 - Rosanowski, Frank A1 - Nöth, Elmar T1 - Automatic Evaluation of Tracheoesophageal Telephone Speech T2 - First International Language Technologies Conference (IS-LTC 2006), Ljubljana, Slovenia, October 2006. N2 - The tracheoesophageal (TE) substitute voice is currently state–of–the–art treatment to restore the ability to speak after laryngectomy. The intelligibility while talking over a telephone is an important clinical factor, as it is a crucial part of the patients’ social life. An objective way to rate the intelligibility of substitute voices when talking over a telephone is desirable to improve the post–laryngectomy speech therapy. An automatic speech recognition (ASR) system was applied to 41 high quality recordings of post–laryngectomy patients. The ASR system was trained with normal, non–pathologic speech. It yielded a word accuracy (WA) of 36.9%±18.0%; compared to the intelligibility rating of a group of human experts the ASR system had a correlation coefficient of -.88. After downsampling the 41 recordings to telephone quality, the ASR system reached a WA of 26.4%±13.9% leading to a correlation coefficient of -.80. These results confirm that an ASR system can be used for objective intelligibility rating over the telephone. KW - Speech Recognition Y1 - 2006 SP - 17 EP - 22 ER - TY - BOOK A1 - Riedhammer, Korbinian T1 - Interactive Approaches to Video Lecture Assessment N2 - Folks that have been here last winter prior to ASRU might be familiar with the title of that talk. But don't be misled, I'll have something new for you. In this talk, I will give an overview over the FAU Lecture Browser which I developed in the context of my thesis. I will start out with the description of a novel data set: The LME Lectures are a corpus of two series of graduate level computer science lectures with 18 recordings each. The courses cover topics in medical image processing and pattern analysis/machine learning. The roughly 40 hours of speech were manually transcribed, and one particular lecture was annotated with key phrases by five human raters. Using this data set, I trained three different speech recognizers using regular continuous, multi-codebook semi-continuous and subspace Gaussian mixture models, that show an error rate of about 10% WER. I will then briefly describe the key phrase extraction and automatic ranking, which was then compared against five raters on one lecture recording. Finally, I will talk about a little usability study where 10 students were asked to perform a certain task-- with and without the proposed lecture browser. Although the number of contestants is limited, the numbers are interesting: the users that had the interface could complete the tasks about 30% faster than the control group, while maintaining about the same accuracy. KW - Speech Recognition Y1 - 2012 PB - Logos Verlag Berlin GmbH CY - Berlin ER - TY - BOOK A1 - Riedhammer, Korbinian T1 - An Automatic Intelligibility Test Based on the Post-Laryngectomy Telephone Test KW - Speech Recognition Y1 - 2008 PB - Mueller Verlag CY - Nürnberg 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 - TY - JOUR A1 - Riedhammer, Korbinian A1 - Favre, Benoit A1 - Hakkani-Tür, Dilek T1 - Long story short - Global unsupervised models for keyphrase based meeting summarization JF - SPEECH COMMUNICATION KW - Speech Recognition KW - Meeting Summarization Y1 - 2010 VL - 2010 IS - 52(10) SP - 801 EP - 815 ER -