57,235 research outputs found
Automatic Speech Indexing System of Bilingual Video Parliament Interventions
This paper presents the development and evaluation of an automatic audio indexing system designed for a special task: work in a bilingual environment in the Parliament of the Canton of Valais in Switzerland, with two official languages, German and French. As several speakers are bilingual, language changes may occur within speaker or even within utterance. Two audio indexing approaches are presented and compared: in the first, speech indexing is based on bilingual automatic speech recognition; in the second, language identification is used after speaker diarization in order to select the corresponding monolingual speech recognizer for decoding. The approaches are later combined. Speaker adaptive training is also addressed and evaluated. Accuracy of language identification and speech recognition for the monolingual and bilingual cases are presented and compared, in parallel with a brief description of the system and the user interface. Finally, the audio indexing system is also evaluated from an information retrieval point of view
Speaker tracking system using speaker boundary detection
This thesis is about a research conducted in the area of Speaker Recognition. The application is concerned to the automatic detection and tracking of target speakers in meetings, conferences, telephone conversations and in radio and television broadcasts. A Speaker Tracking system is developed here, in collaboration with the Center for Language and Speech Technologies and Applications (TALP) in UPC. The main objective of this Speaker Tracking system is to answer the question: When the target speaker speaks? The system uses training speech data for the target speaker in the pre-enrollment stage. Three main modules have been designed for this Speaker Tracking system. In the first module an energy based Speech Activity Detection is applied to select the speech parts of the audio. In the second module the audio is segmented according to the speaker turning points. In the last module a Speaker Verification is implemented in which the target speakers are verified and tracked. Two different approaches are applied in this last module. In the first approach for Speaker Verification, the target speakers and the segments are modeled using the state-of-the-art, Gaussian Mixture Models (GMM). In the second approach for Speaker Verification, the identity vectors (i-vectors) representation is applied for the target speakers and the segments. Finally, the performance of both these approaches is compared for the results evaluation
Anti-spoofing Methods for Automatic SpeakerVerification System
Growing interest in automatic speaker verification (ASV)systems has lead to
significant quality improvement of spoofing attackson them. Many research works
confirm that despite the low equal er-ror rate (EER) ASV systems are still
vulnerable to spoofing attacks. Inthis work we overview different acoustic
feature spaces and classifiersto determine reliable and robust countermeasures
against spoofing at-tacks. We compared several spoofing detection systems,
presented so far,on the development and evaluation datasets of the Automatic
SpeakerVerification Spoofing and Countermeasures (ASVspoof) Challenge
2015.Experimental results presented in this paper demonstrate that the useof
magnitude and phase information combination provides a substantialinput into
the efficiency of the spoofing detection systems. Also wavelet-based features
show impressive results in terms of equal error rate. Inour overview we compare
spoofing performance for systems based on dif-ferent classifiers. Comparison
results demonstrate that the linear SVMclassifier outperforms the conventional
GMM approach. However, manyresearchers inspired by the great success of deep
neural networks (DNN)approaches in the automatic speech recognition, applied
DNN in thespoofing detection task and obtained quite low EER for known and
un-known type of spoofing attacks.Comment: 12 pages, 0 figures, published in Springer Communications in Computer
and Information Science (CCIS) vol. 66
Wordless Sounds: Robust Speaker Diarization using Privacy-Preserving Audio Representations
This paper investigates robust privacy-sensitive audio features for speaker diarization in multiparty conversations: ie., a set of audio features having low linguistic information for speaker diarization in a single and multiple distant microphone scenarios. We systematically investigate Linear Prediction (LP) residual. Issues such as prediction order and choice of representation of LP residual are studied. Additionally, we explore the combination of LP residual with subband information from 2.5 kHz to 3.5 kHz and spectral slope. Next, we propose a supervised framework using deep neural architecture for deriving privacy-sensitive audio features. We benchmark these approaches against the traditional Mel Frequency Cepstral Coefficients (MFCC) features for speaker diarization in both the microphone scenarios. Experiments on the RT07 evaluation dataset show that the proposed approaches yield diarization performance close to the MFCC features on the single distant microphone dataset. To objectively evaluate the notion of privacy in terms of linguistic information, we perform human and automatic speech recognition tests, showing that the proposed approaches to privacy-sensitive audio features yield much lower recognition accuracies compared to MFCC features
Wordless Sounds: Robust Speaker Diarization using Privacy-Preserving Audio Representations
This paper investigates robust privacy-sensitive audio features for speaker diarization in multiparty conversations: ie., a set of audio features having low linguistic information for speaker diarization in a single and multiple distant microphone scenarios. We systematically investigate Linear Prediction (LP) residual. Issues such as prediction order and choice of representation of LP residual are studied. Additionally, we explore the combination of LP residual with subband information from 2.5 kHz to 3.5 kHz and spectral slope. Next, we propose a supervised framework using deep neural architecture for deriving privacy-sensitive audio features. We benchmark these approaches against the traditional Mel Frequency Cepstral Coefficients (MFCC) features for speaker diarization in both the microphone scenarios. Experiments on the RT07 evaluation dataset show that the proposed approaches yield diarization performance close to the MFCC features on the single distant microphone dataset. To objectively evaluate the notion of privacy in terms of linguistic information, we perform human and automatic speech recognition tests, showing that the proposed approaches to privacy-sensitive audio features yield much lower recognition accuracies compared to MFCC features
Speaker Localization for Microphone Array-Based ASR: The Effects of Accuracy on Overlapping Speech
Accurate speaker location is essential for optimal performance of distant speech acquisition systems using microphone array techniques. However, to the best of our knowledge, no comprehensive studies on the degradation of automatic speech recognition (ASR) as a function of speaker location accuracy in a multi-party scenario exist. In this paper, we describe a framework for evaluation of the effects of speaker location errors on a microphone array-based ASR system, in the context of meetings in multi-sensor rooms comprising multiple cameras and microphones. Speakers are manually annotated in videos in different camera views, and triangulation is used to determine an accurate speaker location. Errors in the speaker location are then induced in a systematic manner to observe their influence on speech recognition performance. The system is evaluated on real overlapping speech data collected with simultaneous speakers in a meeting room. The results are compared with those obtained from close-talking headset microphones, lapel microphones, and speaker location based on audio-only and audio-visual information approaches
Oesophageal speech: enrichment and evaluations
167 p.After a laryngectomy (i.e. removal of the larynx) a patient can no more speak in a healthy laryngeal voice. Therefore, they need to adopt alternative methods of speaking such as oesophageal speech. In this method, speech is produced using swallowed air and the vibrations of the pharyngo-oesophageal segment, which introduces several undesired artefacts and an abnormal fundamental frequency. This makes oesophageal speech processing difficult compared to healthy speech, both auditory processing and signal processing. The aim of this thesis is to find solutions to make oesophageal speech signals easier to process, and to evaluate these solutions by exploring a wide range of evaluation metrics.First, some preliminary studies were performed to compare oesophageal speech and healthy speech. This revealed significantly lower intelligibility and higher listening effort for oesophageal speech compared to healthy speech. Intelligibility scores were comparable for familiar and non-familiar listeners of oesophageal speech. However, listeners familiar with oesophageal speech reported less effort compared to non-familiar listeners. In another experiment, oesophageal speech was reported to have more listening effort compared to healthy speech even though its intelligibility was comparable to healthy speech. On investigating neural correlates of listening effort (i.e. alpha power) using electroencephalography, a higher alpha power was observed for oesophageal speech compared to healthy speech, indicating higher listening effort. Additionally, participants with poorer cognitive abilities (i.e. working memory capacity) showed higher alpha power.Next, using several algorithms (preexisting as well as novel approaches), oesophageal speech was transformed with the aim of making it more intelligible and less effortful. The novel approach consisted of a deep neural network based voice conversion system where the source was oesophageal speech and the target was synthetic speech matched in duration with the source oesophageal speech. This helped in eliminating the source-target alignment process which is particularly prone to errors for disordered speech such as oesophageal speech. Both speaker dependent and speaker independent versions of this system were implemented. The outputs of the speaker dependent system had better short term objective intelligibility scores, automatic speech recognition performance and listener preference scores compared to unprocessed oesophageal speech. The speaker independent system had improvement in short term objective intelligibility scores but not in automatic speech recognition performance. Some other signal transformations were also performed to enhance oesophageal speech. These included removal of undesired artefacts and methods to improve fundamental frequency. Out of these methods, only removal of undesired silences had success to some degree (1.44 \% points improvement in automatic speech recognition performance), and that too only for low intelligibility oesophageal speech.Lastly, the output of these transformations were evaluated and compared with previous systems using an ensemble of evaluation metrics such as short term objective intelligibility, automatic speech recognition, subjective listening tests and neural measures obtained using electroencephalography. Results reveal that the proposed neural network based system outperformed previous systems in improving the objective intelligibility and automatic speech recognition performance of oesophageal speech. In the case of subjective evaluations, the results were mixed - some positive improvement in preference scores and no improvement in speech intelligibility and listening effort scores. Overall, the results demonstrate several possibilities and new paths to enrich oesophageal speech using modern machine learning algorithms. The outcomes would be beneficial to the disordered speech community
Two-pass Decoding and Cross-adaptation Based System Combination of End-to-end Conformer and Hybrid TDNN ASR Systems
Fundamental modelling differences between hybrid and end-to-end (E2E)
automatic speech recognition (ASR) systems create large diversity and
complementarity among them. This paper investigates multi-pass rescoring and
cross adaptation based system combination approaches for hybrid TDNN and
Conformer E2E ASR systems. In multi-pass rescoring, state-of-the-art hybrid
LF-MMI trained CNN-TDNN system featuring speed perturbation, SpecAugment and
Bayesian learning hidden unit contributions (LHUC) speaker adaptation was used
to produce initial N-best outputs before being rescored by the speaker adapted
Conformer system using a 2-way cross system score interpolation. In cross
adaptation, the hybrid CNN-TDNN system was adapted to the 1-best output of the
Conformer system or vice versa. Experiments on the 300-hour Switchboard corpus
suggest that the combined systems derived using either of the two system
combination approaches outperformed the individual systems. The best combined
system obtained using multi-pass rescoring produced statistically significant
word error rate (WER) reductions of 2.5% to 3.9% absolute (22.5% to 28.9%
relative) over the stand alone Conformer system on the NIST Hub5'00, Rt03 and
Rt02 evaluation data.Comment: It' s accepted to ISCA 202
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