114 research outputs found
Model kompanzasyonlu birinci derece istatistikleri ile i-vektörlerin gürbüzlüğünün artırılması
Speaker recognition systems achieved significant improvements over the last decade, especially due to
the performance of the i-vectors. Despite the achievements, mismatch between training and test data
affects the recognition performance considerably. In this paper, a solution is offered to increase
robustness against additive noises by inserting model compensation techniques within the i-vector
extraction scheme. For stationary noises, the model compensation techniques produce highly robust
systems. Parallel Model Compensation and Vector Taylor Series are considered as state-of-the-art
model compensation techniques. Applying these methods to the first order statistics, a noisy total
variability space training is aimed, which will reduce the mismatch resulted by additive noises. All other
parts of the conventional i-vector scheme remain unchanged, such as total variability matrix training,
reducing the i-vector dimensionality, scoring the i-vectors. The proposed method was tested with four
different noise types with several signal to noise ratios (SNR) from -6 dB to 18 dB with 6 dB steps. High
reductions in equal error rates were achieved with both methods, even at the lowest SNR levels. On
average, the proposed approach produced more than 50% relative reduction in equal error rate.Konuşmacı tanıma sistemleri özellikle i-vektörlerin performansı sebebiyle son on yılda önemli
gelişmeler elde etmiştir. Bu gelişmelere rağmen eğitim ve test verileri arasındaki uyumsuzluk tanıma
performansını önemli ölçüde etkilemektedir. Bu çalışmada, model kompanzasyon yöntemleri i-vektör
çıkarımı şemasına eklenerek toplanabilir gürültülere karşı gürbüzlüğü artıracak bir çözüm
sunulmaktadır. Durağan gürültüler için model kompanzasyon teknikleri oldukça gürbüz sistemler üretir.
Paralel Model Kompanzasyonu ve Vektör Taylor Serileri en gelişmiş model kompanzasyon
tekniklerinden kabul edilmektedir. Bu metotlar birinci dereceden istatistiklere uygulanarak toplanabilir
gürültülerden kaynaklanan uyumsuzluğu azaltacak gürültülü tüm değişkenlik uzayı eğitimi
amaçlanmıştır. Tüm değişkenlik matrisin eğitimi, i-vektör boyutunun azaltılması, i-vektörlerin
puanlanması gibi geleneksel i-vektör şemasının diğer tüm parçaları değişmeden kalmaktadır. Önerilen
yöntem, 6 dB’lik adımlarla -6 dB’den 18 dB’ye kadar çeşitli sinyal-gürültü oranlarına (SNR) sahip dört
farklı gürültü tipi ile test edilmiştir. Her iki yöntemle de en düşük SNR seviyelerinde bile eşit hata
oranlarında yüksek azalmalar elde edilmiştir. Önerilen yaklaşım eşik hata oranında ortalama olarak
%50’den fazla göreceli azalma sağlamıştır
VOICE BIOMETRICS UNDER MISMATCHED NOISE CONDITIONS
This thesis describes research into effective voice biometrics (speaker recognition) under mismatched noise conditions. Over the last two decades, this class of biometrics has been the subject of considerable research due to its various applications in such areas as telephone banking, remote access control and surveillance. One of the main challenges associated with the deployment of voice biometrics in practice is that of undesired variations in speech characteristics caused by environmental noise. Such variations can in turn lead to a mismatch between the corresponding test and reference material from the same speaker. This is found to adversely affect the performance of speaker recognition in terms of accuracy.
To address the above problem, a novel approach is introduced and investigated. The proposed method is based on minimising the noise mismatch between reference speaker models and the given test utterance, and involves a new form of Test-Normalisation (T-Norm) for further enhancing matching scores under the aforementioned adverse operating conditions. Through experimental investigations, based on the two main classes of speaker recognition (i.e. verification/ open-set identification), it is shown that the proposed approach can significantly improve the performance accuracy under mismatched noise conditions.
In order to further improve the recognition accuracy in severe mismatch conditions, an approach to enhancing the above stated method is proposed. This, which involves providing a closer adjustment of the reference speaker models to the noise condition in the test utterance, is shown to considerably increase the accuracy in extreme cases of noisy test data. Moreover, to tackle the computational burden associated with the use of the enhanced approach with open-set identification, an efficient algorithm for its realisation in this context is introduced and evaluated.
The thesis presents a detailed description of the research undertaken, describes the experimental investigations and provides a thorough analysis of the outcomes
An investigation of supervector regression for forensic voice comparison on small data
International audienceThe present paper deals with an observer design for a nonlinear lateral vehicle model. The nonlinear model is represented by an exact Takagi-Sugeno (TS) model via the sector nonlinearity transformation. A proportional multiple integral observer (PMIO) based on the TS model is designed to estimate simultaneously the state vector and the unknown input (road curvature). The convergence conditions of the estimation error are expressed under LMI formulation using the Lyapunov theory which guaranties bounded error. Simulations are carried out and experimental results are provided to illustrate the proposed observer
The Effect Of Acoustic Variability On Automatic Speaker Recognition Systems
This thesis examines the influence of acoustic variability on automatic speaker recognition systems (ASRs) with three aims. i. To measure ASR performance under 5 commonly encountered acoustic conditions; ii. To contribute towards ASR system development with the provision of new research data; iii. To assess ASR suitability for forensic speaker comparison (FSC) application and investigative/pre-forensic use. The thesis begins with a literature review and explanation of relevant technical terms. Five categories of research experiments then examine ASR performance, reflective of conditions influencing speech quantity (inhibitors) and speech quality (contaminants), acknowledging quality often influences quantity. Experiments pertain to: net speech duration, signal to noise ratio (SNR), reverberation, frequency bandwidth and transcoding (codecs). The ASR system is placed under scrutiny with examination of settings and optimum conditions (e.g. matched/unmatched test audio and speaker models). Output is examined in relation to baseline performance and metrics assist in informing if ASRs should be applied to suboptimal audio recordings. Results indicate that modern ASRs are relatively resilient to low and moderate levels of the acoustic contaminants and inhibitors examined, whilst remaining sensitive to higher levels. The thesis provides discussion on issues such as the complexity and fragility of the speech signal path, speaker variability, difficulty in measuring conditions and mitigation (thresholds and settings). The application of ASRs to casework is discussed with recommendations, acknowledging the different modes of operation (e.g. investigative usage) and current UK limitations regarding presenting ASR output as evidence in criminal trials. In summary, and in the context of acoustic variability, the thesis recommends that ASRs could be applied to pre-forensic cases, accepting extraneous issues endure which require governance such as validation of method (ASR standardisation) and population data selection. However, ASRs remain unsuitable for broad forensic application with many acoustic conditions causing irrecoverable speech data loss contributing to high error rates
Robust Speaker Verification
Cílem této práce je analyzovat úspěšnost systému rozpoznávaní mluvčího na nahrávkach degradovaných různym telefonním přenosovým kanálem. Použili jsme dva způsoby extrakce příznaků - Mel Frequency Cepstral Coefficients (MFCC) a moderní systém, který spojuje Bottleneck příznaky spolu s MFCC. Systém rozpoznávání mluvčího je založen na i-vektorech a Pravděpodobnostní Lineární Diskriminační Analýze (PLDA). Porovnali jsme scenáře, kde je PLDA trénovaná jen na čisté řeči, poté systém kde jsme přidali data s hlukem a reverberací a nakonec, data degradované kodekem. Vyhodnotili jsem systémy za rovnakých podmínek (data ze stejného kodeku byli také v trénování PLDA) a také za rozdílnych podmínek (data ze stejného kodeku resp. rodiny kodeků nebyli v trénování PLDA). Také jsme experimentovali s nedávno představenou technikou na adaptaci kanálu - Within-class Covariance Correction (WCC). Můžeme jednoznačně vidět zlepšení úspěšnosti přidáním degradovaných dat do PLDA resp. WCC (s přibližně stejným výsledkem) pro obě naše testované podmínky.The goal of this paper is to analyze the impact of codec degraded speech on a state-ofthe-art speaker recognition system. Two feature extraction techniques are analyzed - Mel Frequency Cepstral Coefficients (MFCC) and the state-of-the-art system using Bottleneck features together with MFCC. Speaker recognition system is based on i-vector and Probabilistic Linear Discriminant Analysis (PLDA). We compared scenarios where PLDA is trained only on clean data, then system where we added also noise and reverberant data, and at last, codec degraded speech. We evaluated the systems on the matched conditions (data from the same codec are seen with PLDA) and also mismatched conditions (PLDA does not see any data from the tested codec). We experimented also with recently introduced technique for channel adaptation - Within-class Covariance Correction (WCC). We can see clear benefit of adding transcoded data to PLDA or WCC (with approximately same gain) for both tested conditions (matched and mismatched).
Histogram equalization for robust text-independent speaker verification in telephone environments
Word processed copy.
Includes bibliographical references
- …