Multi-Pattern Viterbi Algorithm for joint decoding of multiple speech patterns

Joint decoding of multiple speech patterns so as to improve speech recognition performance is important, especially in the presence of noise. In this paper, we propose a Multi-Pattern Viterbi algorithm (MPVA) to jointly decode and recognize multiple speech patterns for automatic speech recognition (ASR). The MPVA is a generalization of the Viterbi Algorithm to jointly decode multiple patterns given a Hidden Markov Model (HMM). Unlike the previously proposed two stage Constrained Multi-Pattern Viterbi Algorithm (CMPVA),the MPVA is a single stage algorithm. MPVA has the advantage that it cart be extended to connected word recognition (CWR) and continuous speech recognition (CSR) problems. MPVA is shown to provide better speech recognition performance than the earlier techniques: using only two repetitions of noisy speech patterns (-5 dB SNR, 10% burst noise), the word error rate using MPVA decreased by 28.5%, when compared to using individual decoding. (C) 2010 Elsevier B.V. All rights reserved.

”Ihanan överi ja juhlallinen jakku olisi kyllä kiva lisä vaatekaappiin” – tutkimus kuluttamisesta ja professionalisoitumisesta elämäntyyliblogeissa

Tarkastelen tutkimuksessani muotibloggaajien kulutuspuhetta. Tutkimuksen tarkoituksena on selvittää, millaista kuluttajuutta muotiblogeissa ilmaistaan ja millainen kehityskaari muotiblogeissa on tapahtunut vuodesta 2007 tämän tutkimuksen tekemiseen asti tutkimusaineiston sekä tekemieni havaintojen perusteella. Tutkimus on toteutettu laadullisia tutkimusmenetelmiä käyttämällä. Olen kerännyt aineistoni kymmenestä naisten kirjoittamasta muotiblogista kahtena eri ajanjaksona vuonna 2009. Lisäksi tutkimuksessa on sekä etnografisen että netnografisen havainnoinnin piirteitä. Aineiston analysoinnin apuna olen käyttänyt teemoittelua sekä tyypittelyä. Tutkimuksesta selvisi, että muotibloggaajien näkeminen identiteettiään etsivänä liittyy oman tyylin etsimiseen ja erottautumisen haluun. Perinteinen näkemys kuluttajasta valitsijana ja passiivisena markkinoilla toimijana on väistymässä, sillä muotibloggaajat näyttäytyvät tutkimuksessa aktiivisina toimijoina ja tuottajina. Muotibloggaajat hakevat myös jatkuvasti uusia kulutuselämyksiä ja kommunikoivat toistensa kanssa välittämällä merkityksiä kuluttamisensa kautta. Muotibloggaajien kulutuspuheet näyttäytyvät tutkimuksessa ekonomistisen kulutuseetoksen ja perinteisen suomalaisen kulutuspuheen mukaisesti järkevinä. Säästäväisyyttä pidetään hyveellisenä ja tuotteiden hinnat vaikuttavat ostopäätösten tekemiseen. Muotibloggaajat osaavat kuitenkin kontrolloidusti nauttia kuluttamisesta. Kulutuspuheet noudattavat myös ekologis-eettisen kulutuseetoksen perinnettä, mikä ilmenee fanaattisuuden välttämisenä sekä perinteisten että sosiaalisen median kirpputorien suosimisena. Lisäksi muotibloggaajien kulutuspuheet käyvät ilmi sosiaalisena pakkona, sillä muotiblogin pitäminen mielenkiintoisena vaatii jatkuvaa himoa uusia kulutustuotteita kohtaan. Tutkimuksen perusteella muotiblogeissa korostuu yhteisöllisyys ja kuluttaminen on hyvin naisellista. Tutkimusaineistosta sekä havainnoista tekemieni päätelmien mukaan muotiblogit ovat muuttumassa elämäntyyliblogeiksi, sillä niissä kerrotaan yhä enemmän myös muista kuin kuluttamiseen liittyvistä aiheista. Samalla muotibloggaajat ovat kehittymässä tavallisesta kuluttajasta kohti asiantuntijuutta eli professionalisoitumassa. Professionalisoitumisen myötä muotibloggaajat toimivat uuden ajan kuluttajavalistajina ja muotiblogeissa syntyy uusia sanoja ja merkityksiä. Lisäksi muotibloggaajille on kehittynyt sellaisia tietoja ja taitoja, joita voi myydä. Muotibloggaajista onkin tulossa muodin ammattilaisia, joille maksetaan tulevaisuudessa bloggaamisesta myös palkkaa.

Rhetoric and public speech in English republicanism, 1642-1681

The study analyses the ambivalent relationship republicanism, as a form of self-government free from domination, had with the ideal of participatory oratory and non-dominated speech on the one hand, and with the danger of unhindered demagogy and its possibly fatal consequences to that form of government on the other. Although previous scholarship has delved deeply into republicanism as well as into rhetoric and public speech, the interplay between those aspects has only gathered scattered interest, and there has been no systematic study considering the variety of republican approaches to rhetoric and public speech in 17th-century England. The rare attempts to do so have been studies in English literature, and they have not analysed the political philosophy of republicanism, as the focus has been on republicanism as a literary culture. This study connects the fields of political theory, political history as well as literature in order to make a multidisciplinary contribution to intellectual history. The study shows that, within the tradition of classical republicanism, individual authors could make different choices when addressing the problematic topics of public speech and rhetoric, and the variety of their conclusions often set the authors against each other, resulting in the development of their theories through internal debates within the republican tradition. The authors under study were chosen to reflect this variety and the connections between them: the similarities between James Harrington and John Streater, and between John Milton and John Hall of Durham are shown, as well the controversies between Harrington and Milton, and Streater and Hall, respectively. In addition, by analysing the writings of Marchamont Nedham the study will show that the choices were not limited to more, or less, democratic brands of republicanism. Most significantly, the study provides a thorough analysis of the political philosophies behind the various brands of republicanism, in addition to describing them. By means of this analysis, the study shows that previous attempts to assess the role of free speech and public debate, through the lenses of modern, rights-based liberal political theory have resulted in an inappropriate framework for understanding early modern English republicanism. By approaching the topics through concepts used by the republicans legitimate authority, leadership by oratory, and republican freedom and through the frames of reference available and familiar to them roles of education and institutions the study presents a thorough and systematic analysis of the role and function of rhetoric and public speech in English republicanism. The findings of this analysis have significant consequences to our current understanding of the history and development of republican political theory, and, more generally, of the connections between democratic theory and free speech.

Part-of-Speech Tagging using Parallel Weighted Finite-State Transducers

We use parallel weighted finite-state transducers to implement a part-of-speech tagger, which obtains state-of-the-art accuracy when used to tag the Europarl corpora for Finnish, Swedish and English. Our system consists of a weighted lexicon and a guesser combined with a bigram model factored into two weighted transducers. We use both lemmas and tag sequences in the bigram model, which guarantees reliable bigram estimates.

Continuity Metric for Unit Selection based Text-to-Speech Synthesis

A new method based on unit continuity metric (UCM) is proposed for optimal unit selection in text-to-speech (TTS) synthesis. UCM employs two features, namely, pitch continuity metric and spectral continuity metric. The methods have been implemented and tested on our test bed called MILE-TTS and it is available as web demo. After verification by a self selection test, the algorithms are evaluated on 8 paragraphs each for Kannada and Tamil by native users of the languages. Mean-opinion-score (MOS) shows that naturalness and comprehension are better with UCM based algorithm than the non-UCM based ones. The naturalness of the TTS output is further enhanced by a new rule based algorithm for pause prediction for Tamil language. The pauses between the words are predicted based on parts-of-speech information obtained from the input text.

Research bed for unit selection based text to speech synthesis

The paper describes a modular, unit selection based TTS framework, which can be used as a research bed for developing TTS in any new language, as well as studying the effect of changing any parameter during synthesis. Using this framework, TTS has been developed for Tamil. Synthesis database consists of 1027 phonetically rich prerecorded sentences. This framework has already been tested for Kannada. Our TTS synthesizes intelligible and acceptably natural speech, as supported by high mean opinion scores. The framework is further optimized to suit embedded applications like mobiles and PDAs. We compressed the synthesis speech database with standard speech compression algorithms used in commercial GSM phones and evaluated the quality of the resultant synthesized sentences. Even with a highly compressed database, the synthesized output is perceptually close to that with uncompressed database. Through experiments, we explored the ambiguities in human perception when listening to Tamil phones and syllables uttered in isolation,thus proposing to exploit the misperception to substitute for missing phone contexts in the database. Listening experiments have been conducted on sentences synthesized by deliberately replacing phones with their confused ones.

Subspace Based Speech Enhancement Using Gaussian Mixture Model

Traditional subspace based speech enhancement (SSE)methods use linear minimum mean square error (LMMSE) estimation that is optimal if the Karhunen Loeve transform (KLT) coefficients of speech and noise are Gaussian distributed. In this paper, we investigate the use of Gaussian mixture (GM) density for modeling the non-Gaussian statistics of the clean speech KLT coefficients. Using Gaussian mixture model (GMM), the optimum minimum mean square error (MMSE) estimator is found to be nonlinear and the traditional LMMSE estimator is shown to be a special case. Experimental results show that the proposed method provides better enhancement performance than the traditional subspace based methods.Index Terms: Subspace based speech enhancement, Gaussian mixture density, MMSE estimation.

Two stage iterative wiener filtering for speech enhancement

We formulate a two-stage Iterative Wiener filtering (IWF) approach to speech enhancement, bettering the performance of constrained IWF, reported in literature. The codebook constrained IWF (CCIWF) has been shown to be effective in achieving convergence of IWF in the presence of both stationary and non-stationary noise. To this, we include a second stage of unconstrained IWF and show that the speech enhancement performance can be improved in terms of average segmental SNR (SSNR), Itakura-Saito (IS) distance and Linear Prediction Coefficients (LPC) parameter coincidence. We also explore the tradeoff between the number of CCIWF iterations and the second stage IWF iterations.

Comparison of AM-FM Based Features For Robust Speech Recognition

Effective feature extraction for robust speech recognition is a widely addressed topic and currently there is much effort to invoke non-stationary signal models instead of quasi-stationary signal models leading to standard features such as LPC or MFCC. Joint amplitude modulation and frequency modulation (AM-FM) is a classical non-parametric approach to non-stationary signal modeling and recently new feature sets for automatic speech recognition (ASR) have been derived based on a multi-band AM-FM representation of the signal. We consider several of these representations and compare their performances for robust speech recognition in noise, using the AURORA-2 database. We show that FEPSTRUM representation proposed is more effective than others. We also propose an improvement to FEPSTRUM based on the Teager energy operator (TEO) and show that it can selectively outperform even FEPSTRUM

Online Unsupervised Pattern Discovery in Speech Using Parallelization

Segmental dynamic time warping (DTW) has been demonstrated to be a useful technique for finding acoustic similarity scores between segments of two speech utterances. Due to its high computational requirements, it had to be computed in an offline manner, limiting the applications of the technique. In this paper, we present results of parallelization of this task by distributing the workload in either a static or dynamic way on an 8-processor cluster and discuss the trade-offs among different distribution schemes. We show that online unsupervised pattern discovery using segmental DTW is plausible with as low as 8 processors. This brings the task within reach of today's general purpose multi-core servers. We also show results on a 32-processor system, and discuss factors affecting scalability of our methods.

Speech Enhancement using Intra-frame Dependency in DCT Domain

In this paper, we present a new speech enhancement approach, that is based on exploiting the intra-frame dependency of discrete cosine transform (DCT) domain coefficients. It can be noted that the existing enhancement techniques treat the transformdomain coefficients independently. Instead of this traditional approach of independently processing the scalars, we split the DCT domain noisy speech vector into sub-vectors and each sub-vector is enhanced independently. Through this sub-vector based approach, the higher dimensional enhancement advantage, viz. non-linear dependency, is exploited. In the developed method, each clean speech sub-vector is modeled using a Gaussian mixture (GM) density. We show that the proposed Gaussian mixture model (GMM) based DCT domain method, using sub-vector processing approach, provides better performance than the conventional approach of enhancing the transform domain scalar components independently. Performance improvement over the recently proposed GMM based time domain approach is also shown.

GMM basedbayesian approach to speech enhancement in signal/transform domain

Considering a general linear model of signal degradation, by modeling the probability density function (PDF) of the clean signal using a Gaussian mixture model (GMM) and additive noise by a Gaussian PDF, we derive the minimum mean square error (MMSE) estimator.The derived MMSE estimator is non-linear and the linear MMSE estimator is shown to be a special case. For speech signal corrupted by independent additive noise, by modeling the joint PDF of time-domain speech samples of a speech frame using a GMM, we propose a speech enhancement method based on the derived MMSE estimator. We also show that the same estimator can be used for transform-domain speech enhancement.

Speech segmentation using extrema based signal track length measure

We introduce a novel temporal feature of a signal, namely extrema-based signal track length (ESTL) for the problem of speech segmentation. We show that ESTL measure is sensitive to both amplitude and frequency of the signal. The short-time ESTL (ST_ESTL) shows a promising way to capture the significant segments of speech signal, where the segments correspond to acoustic units of speech having distinct temporal waveforms. We compare ESTL based segmentation with ML and STM methods and find that it is as good as spectral feature based segmentation, but with lesser computational complexity.

High-Rate Vector Quantization for Noisy Channels With Applications to Wideband Speech Spectrum Compression

This paper considers the high-rate performance of source coding for noisy discrete symmetric channels with random index assignment (IA). Accurate analytical models are developed to characterize the expected distortion performance of vector quantization (VQ) for a large class of distortion measures. It is shown that when the point density is continuous, the distortion can be approximated as the sum of the source quantization distortion and the channel-error induced distortion. Expressions are also derived for the continuous point density that minimizes the expected distortion. Next, for the case of mean squared error distortion, a more accurate analytical model for the distortion is derived by allowing the point density to have a singular component. The extent of the singularity is also characterized. These results provide analytical models for the expected distortion performance of both conventional VQ as well as for channel-optimized VQ. As a practical example, compression of the linear predictive coding parameters in the wideband speech spectrum is considered, with the log spectral distortion as performance metric. The theory is able to correctly predict the channel error rate that is permissible for operation at a particular level of distortion.

Decision tree design and applications in speech processing

The design and operation of the minimum cost classifier, where the total cost is the sum of the measurement cost and the classification cost, is computationally complex. Noting the difficulties associated with this approach, decision tree design directly from a set of labelled samples is proposed in this paper. The feature space is first partitioned to transform the problem to one of discrete features. The resulting problem is solved by a dynamic programming algorithm over an explicitly ordered state space of all outcomes of all feature subsets. The solution procedure is very general and is applicable to any minimum cost pattern classification problem in which each feature has a finite number of outcomes. These techniques are applied to (i) voiced, unvoiced, and silence classification of speech, and (ii) spoken vowel recognition. The resulting decision trees are operationally very efficient and yield attractive classification accuracies.