The LIA RT’07 speaker diarization system

Abstract. This paper presents the LIA submission to the speaker diarization task of the 2007 NIST Rich Transcription (RT'07) evaluation campaign. We report a system optimised for conference meeting recordings and experiments on all three RT'07 subdomains and microphone conditions. Results show that, despite state-of-the-art performance for the single distant microphone (SDM) condition, in its current form the system is not effective in utilising the additional information that is available with the multiple distant microphone (MDM) condition. With post evaluation tuning we achieve a DER of 19% on the MDM task with conference meeting data. Some early experimental work highlights both the limitations and potential of utilising between-channel delay features for diarization

Speaker Diarization Based on Intensity Channel Contribution

The time delay of arrival (TDOA) between multiple microphones has been used since 2006 as a source of information (localization) to complement the spectral features for speaker diarization. In this paper, we propose a new localization feature, the intensity channel contribution (ICC) based on the relative energy of the signal arriving at each channel compared to the sum of the energy of all the channels. We have demonstrated that by joining the ICC features and the TDOA features, the robustness of the localization features is improved and that the diarization error rate (DER) of the complete system (using localization and spectral features) has been reduced. By using this new localization feature, we have been able to achieve a 5.2% DER relative improvement in our development data, a 3.6% DER relative improvement in the RT07 evaluation data and a 7.9% DER relative improvement in the last year's RT09 evaluation data

Different Approaches for Speaker Diarization

Audio diarization is the process of annotating an input audio channel with information that attributes (possibly overlapping) temporal regions of signal energy to their specific sources. These sources can include particular speakers, music, background noise sources and other signal source/channel characteristics. Speaker diarization is the task of determining “who spoke when?†in an audio or video recording that contains an unknown amount of speech and also an unknown number of speakers. Diarization can be used for helping speech recognition, facilitating the searching and indexing of audio archives and increasing the richness of automatic transcriptions, making them more readable. Over recent years, however, speaker diarization has become an important key technology for many tasks, such as navigation, retrieval or higher-level inference on audio data. Accordingly, many important improvements in accuracy and robustness have been reported in the area of conferences. The application domains, from broadcast news, to lectures and meetings, vary greatly and pose different problems, such as access to multiple microphones and multimodal information or overlapping speech

New insights into hierarchical clustering and linguistic normalization for speaker diarization

Face au volume croissant de données audio et multimédia, les technologies liées à l'indexation de données et à l'analyse de contenu ont suscité beaucoup d'intérêt dans la communauté scientifique. Parmi celles-ci, la segmentation et le regroupement en locuteurs, répondant ainsi à la question 'Qui parle quand ?' a émergé comme une technique de pointe dans la communauté de traitement de la parole. D'importants progrès ont été réalisés dans le domaine ces dernières années principalement menés par les évaluations internationales du NIST. Tout au long de ces évaluations, deux approches se sont démarquées : l'une est bottom-up et l'autre top-down. L'ensemble des systèmes les plus performants ces dernières années furent essentiellement des systèmes types bottom-up, cependant nous expliquons dans cette thèse que l'approche top-down comporte elle aussi certains avantages. En effet, dans un premier temps, nous montrons qu'après avoir introduit une nouvelle composante de purification des clusters dans l'approche top-down, nous obtenons des performances comparables à celles de l'approche bottom-up. De plus, en étudiant en détails les deux types d'approches nous montrons que celles-ci se comportent différemment face à la discrimination des locuteurs et la robustesse face à la composante lexicale. Ces différences sont alors exploitées au travers d'un nouveau système combinant les deux approches. Enfin, nous présentons une nouvelle technologie capable de limiter l'influence de la composante lexicale, source potentielle d'artefacts dans le regroupement et la segmentation en locuteurs. Notre nouvelle approche se nomme Phone Adaptive Training par analogie au Speaker Adaptive TrainingThe ever-expanding volume of available audio and multimedia data has elevated technologies related to content indexing and structuring to the forefront of research. Speaker diarization, commonly referred to as the who spoke when?' task, is one such example and has emerged as a prominent, core enabling technology in the wider speech processing research community. Speaker diarization involves the detection of speaker turns within an audio document (segmentation) and the grouping together of all same-speaker segments (clustering). Much progress has been made in the field over recent years partly spearheaded by the NIST Rich Transcription evaluations focus on meeting domain, in the proceedings of which are found two general approaches: top-down and bottom-up. Even though the best performing systems over recent years have all been bottom-up approaches we show in this thesis that the top-down approach is not without significant merit. Indeed we first introduce a new purification component leading to competitive performance to the bottom-up approach. Moreover, while investigating the two diarization approaches more thoroughly we show that they behave differently in discriminating between individual speakers and in normalizing unwanted acoustic variation, i.e.\ that which does not pertain to different speakers. This difference of behaviours leads to a new top-down/bottom-up system combination outperforming the respective baseline system. Finally, we introduce a new technology able to limit the influence of linguistic effects, responsible for biasing the convergence of the diarization system. Our novel approach is referred to as Phone Adaptive Training (PAT).PARIS-Télécom ParisTech (751132302) / SudocSudocFranceF

Target-Speaker Voice Activity Detection: a Novel Approach for Multi-Speaker Diarization in a Dinner Party Scenario

Speaker diarization for real-life scenarios is an extremely challenging problem. Widely used clustering-based diarization approaches perform rather poorly in such conditions, mainly due to the limited ability to handle overlapping speech. We propose a novel Target-Speaker Voice Activity Detection (TS-VAD) approach, which directly predicts an activity of each speaker on each time frame. TS-VAD model takes conventional speech features (e.g., MFCC) along with i-vectors for each speaker as inputs. A set of binary classification output layers produces activities of each speaker. I-vectors can be estimated iteratively, starting with a strong clustering-based diarization. We also extend the TS-VAD approach to the multi-microphone case using a simple attention mechanism on top of hidden representations extracted from the single-channel TS-VAD model. Moreover, post-processing strategies for the predicted speaker activity probabilities are investigated. Experiments on the CHiME-6 unsegmented data show that TS-VAD achieves state-of-the-art results outperforming the baseline x-vector-based system by more than 30% Diarization Error Rate (DER) abs.Comment: Accepted to Interspeech 202

Speaker Diarization

Práce se věnuje implementaci diarizace mluvčího. Popisuje jednotlivé komponenty diarizačního systému, který umí zodpovědět otázku "kdy kdo mluví". Mezi součásti takového systému patří postupně extrakce příznaků vstupních dat, detekce řeči/ticha, segmentace mluvčích, jejich následné shlukování a nakonec i techniky zaměřené na zlepšení finální segmentace. Práce pochopitelně uvádí i dosažené výsledky implementovaného systému na testovací sadě nahrávek včetně popisu způsobu hodnocení. Testovací nahrávky pochází z NIST RT evaluací z let 2005 - 2007 a nejnižší dosažená chybovost na této sadě je 18,52% DER. K porovnání výsledků systému na testovací sadě souborů je zde uvedena i úspěšnost Marijna Huijbregtse z Nizozemí, který v roce 2009 pracoval se stejnými nahrávkami a dosáhl chybovosti 12,91% DER.This work aims at a task of speaker diarization. The goal is to implement a system which is able to decide "who spoke when". Particular components of implementation are described. The main parts are feature extraction, voice activity detection, speaker segmentation and clustering and finally also postprocessing. This work also contains results of implemented system on test data including a description of evaluation. The test data comes from the NIST RT Evaluation 2005 - 2007 and the lowest error rate for this dataset is 18.52% DER. Results are compared with diarization system implemented by Marijn Huijbregts from The Netherlands, who worked on the same data in 2009 and reached 12.91% DER.