Phonetic Temporal Neural Model for Language Identification

Deep neural models, particularly the LSTM-RNN model, have shown great potential for language identification (LID). However, the use of phonetic information has been largely overlooked by most existing neural LID methods, although this information has been used very successfully in conventional phonetic LID systems. We present a phonetic temporal neural model for LID, which is an LSTM-RNN LID system that accepts phonetic features produced by a phone-discriminative DNN as the input, rather than raw acoustic features. This new model is similar to traditional phonetic LID methods, but the phonetic knowledge here is much richer: it is at the frame level and involves compacted information of all phones. Our experiments conducted on the Babel database and the AP16-OLR database demonstrate that the temporal phonetic neural approach is very effective, and significantly outperforms existing acoustic neural models. It also outperforms the conventional i-vector approach on short utterances and in noisy conditions.Comment: Submitted to TASL

Multi-Task Neural Networks for Speech Recognition

První část této diplomové práci se zabývá teoretickým rozborem principů neuronových sítí, včetně možnosti jejich použití v oblasti rozpoznávání řeči. Práce pokračuje popisem viceúkolových neuronových sítí a souvisejících experimentů. Praktická část práce obsahovala změny software pro trénování neuronových sítí, které umožnily viceúkolové trénování. Je rovněž popsáno připravené prostředí, včetně několika dedikovaných skriptů. Experimenty představené v této diplomové práci ověřují použití artikulačních characteristik řeči pro viceúkolové trénování. Experimenty byly provedeny na dvou řečových databázích lišících se kvalitou a velikostí a representujících různé jazyky - angličtinu a vietnamštinu. Artikulační charakteristiky byly také kombinovány s jinými sekundárními úkoly, například kontextem, s záměrem ověřit jejich komplementaritu. Porovnaní je provedeno s neuronovými sítěmi různých velikostí tak, aby byl popsán vztah mezi velikostí neuronových sítí a efektivitou viceúkolového trénování. Závěrem provedených experimentů je, že viceúkolové trénování s použitím artikulačnich charakteristik jako sekundárních úkolů vede k lepšímu trénování neuronových sítí a výsledkem tohoto trénování může být přesnější rozpoznávání fonémů. V závěru práce jsou viceúkolové neuronové sítě testovány v systému rozpoznávání řeči jako extraktor příznaků.The first part of this Master's thesis covers theoretical investigation into the principles and usage of neural networks, including their usability for the speech recognition tasks. Then it proceeds to summarize the multi-task neural networks' operating principles and some recent experiments with them. The practical part of the semester project reports changes made to a tool for neural network training which support multi-task training. Then the preparation of the settings is described, including a number of scripts written especially for this purpose. The experiments presented in the thesis explore the idea of using articulatory characteristics of phonemes as secondary tasks for multi-task training. The experiments are conducted on two different datasets of different quality and size and representing different languages - English and Vietnamese. Articulatory characteristics are occasionally combined with different secondary tasks, such as context, to see how well they function together. A comparison is made between the networks of different sizes to see how their size affects the effectiveness of multi-task training. These experiments show that multi-task training with the use of articulatory characteristics as secondary tasks can enhance training and yield better phoneme accuracy as a result. Finally, multi-task training is embedded to a speech recognition system as a feature extractor.

Spoken term detection ALBAYZIN 2014 evaluation: overview, systems, results, and discussion

The electronic version of this article is the complete one and can be found online at: http://dx.doi.org/10.1186/s13636-015-0063-8Spoken term detection (STD) aims at retrieving data from a speech repository given a textual representation of the search term. Nowadays, it is receiving much interest due to the large volume of multimedia information. STD differs from automatic speech recognition (ASR) in that ASR is interested in all the terms/words that appear in the speech data, whereas STD focuses on a selected list of search terms that must be detected within the speech data. This paper presents the systems submitted to the STD ALBAYZIN 2014 evaluation, held as a part of the ALBAYZIN 2014 evaluation campaign within the context of the IberSPEECH 2014 conference. This is the first STD evaluation that deals with Spanish language. The evaluation consists of retrieving the speech files that contain the search terms, indicating their start and end times within the appropriate speech file, along with a score value that reflects the confidence given to the detection of the search term. The evaluation is conducted on a Spanish spontaneous speech database, which comprises a set of talks from workshops and amounts to about 7 h of speech. We present the database, the evaluation metrics, the systems submitted to the evaluation, the results, and a detailed discussion. Four different research groups took part in the evaluation. Evaluation results show reasonable performance for moderate out-of-vocabulary term rate. This paper compares the systems submitted to the evaluation and makes a deep analysis based on some search term properties (term length, in-vocabulary/out-of-vocabulary terms, single-word/multi-word terms, and in-language/foreign terms).This work has been partly supported by project CMC-V2 (TEC2012-37585-C02-01) from the Spanish Ministry of Economy and Competitiveness. This research was also funded by the European Regional Development Fund, the Galician Regional Government (GRC2014/024, “Consolidation of Research Units: AtlantTIC Project” CN2012/160)

Log-linear system combination using structured support vector machines

Building high accuracy speech recognition systems with limited language resources is a highly challenging task. Although the use of multi-language data for acoustic models yields improvements, performance is often unsatisfactory with highly limited acoustic training data. In these situations, it is possible to consider using multiple well trained acoustic models and combine the system outputs together. Unfortunately, the computational cost associated with these approaches is high as multiple decoding runs are required. To address this problem, this paper examines schemes based on log-linear score combination. This has a number of advantages over standard combination schemes. Even with limited acoustic training data, it is possible to train, for example, phone-specific combination weights, allowing detailed relationships between the available well trained models to be obtained. To ensure robust parameter estimation, this paper casts log-linear score combination into a structured support vector machine (SSVM) learning task. This yields a method to train model parameters with good generalisation properties. Here the SSVM feature space is a set of scores from well-trained individual systems. The SSVM approach is compared to lattice rescoring and confusion network combination using language packs released within the IARPA Babel program

Improving multilingual speech recognition systems

End-to-end trainable deep neural networks have become the state-of-the-art architecture for automatic speech recognition (ASR), provided that the network is trained with a sufficiently large dataset. However, many existing languages are too sparsely resourced for deep learning networks to achieve as high accuracy as their resource-abundant counterparts. Multilingual recognition systems mitigate data sparsity issues by training models on data from multiple language resources to learn a speech-to-text or speech-to-phone model universal to all languages. The resulting multilingual ASR models usually have better recognition accuracy than the models trained on the individual dataset. In this work, we propose that two limitations exist for multilingual systems, and resolving the two limitations could result in improved recognition accuracy: (1) existing corpora are of the considerably varied form (spontaneous or read speech), corpus size, noise level, and phoneme distribution and the ASR models trained on the joint multilingual dataset have large performance disparities over different languages. We present an optimizable loss function, equal accuracy ratio (EAR), that measures the sequence-level performance disparity between different user groups and we show that explicitly optimizing this objective reduces the performance gap and improves the multilingual recognition accuracy. (2) While having good accuracy on the seen training language, the multilingual systems do not generalize well to unseen testing languages, which we refer to as cross-lingual recognition accuracy. We introduce language embedding using external linguistic typologies and show that such embedding can significantly increase both multilingual and cross-lingual accuracy. We illustrate the effectiveness of the proposed methods with experiments on multilingual and multi-user and multi-dialect corpora

ALBAYZIN 2018 spoken term detection evaluation: a multi-domain international evaluation in Spanish

[Abstract] Search on speech (SoS) is a challenging area due to the huge amount of information stored in audio and video repositories. Spoken term detection (STD) is an SoS-related task aiming to retrieve data from a speech repository given a textual representation of a search term (which can include one or more words). This paper presents a multi-domain internationally open evaluation for STD in Spanish. The evaluation has been designed carefully so that several analyses of the main results can be carried out. The evaluation task aims at retrieving the speech files that contain the terms, providing their start and end times, and a score that reflects the confidence given to the detection. Three different Spanish speech databases that encompass different domains have been employed in the evaluation: the MAVIR database, which comprises a set of talks from workshops; the RTVE database, which includes broadcast news programs; and the COREMAH database, which contains 2-people spontaneous speech conversations about different topics. We present the evaluation itself, the three databases, the evaluation metric, the systems submitted to the evaluation, the results, and detailed post-evaluation analyses based on some term properties (within-vocabulary/out-of-vocabulary terms, single-word/multi-word terms, and native/foreign terms). Fusion results of the primary systems submitted to the evaluation are also presented. Three different research groups took part in the evaluation, and 11 different systems were submitted. The obtained results suggest that the STD task is still in progress and performance is highly sensitive to changes in the data domain.Ministerio de Economía y Competitividad; TIN2015-64282-R,Ministerio de Economía y Competitividad; RTI2018-093336-B-C22Ministerio de Economía y Competitividad; TEC2015-65345-PXunta de Galicia; ED431B 2016/035Xunta de Galicia; GPC ED431B 2019/003Xunta de Galicia; GRC 2014/024Xunta de Galicia; ED431G/01Xunta de Galicia; ED431G/04Agrupación estratéxica consolidada; GIU16/68Ministerio de Economía y Competitividad; TEC2015-68172-C2-1-

Combining tandem and hybrid systems for improved speech recognition and keyword spotting on low resource languages

Copyright © 2014 ISCA. In recent years there has been significant interest in Automatic Speech Recognition (ASR) and KeyWord Spotting (KWS) systems for low resource languages. One of the driving forces for this research direction is the IARPA Babel project. This paper examines the performance gains that can be obtained by combining two forms of deep neural network ASR systems, Tandem and Hybrid, for both ASR and KWS using data released under the Babel project. Baseline systems are described for the five option period 1 languages: Assamese; Bengali; Haitian Creole; Lao; and Zulu. All the ASR systems share common attributes, for example deep neural network configurations, and decision trees based on rich phonetic questions and state-position root nodes. The baseline ASR and KWS performance of Hybrid and Tandem systems are compared for both the "full", approximately 80 hours of training data, and limited, approximately 10 hours of training data, language packs. By combining the two systems together consistent performance gains can be obtained for KWS in all configurations

Echolocation: Using Word-Burst Analysis to Rescore Keyword Search Candidates in Low-Resource Languages

State of the art technologies for speech recognition are very accurate for heavily studied languages like English. They perform poorly, though, for languages wherein the recorded archives of speech data available to researchers are relatively scant. In the context of these low-resource languages, the task of keyword search within recorded speech is formidable. We demonstrate a method that generates more accurate keyword search results on low-resource languages by studying a pattern not exploited by the speech recognizer. The word-burst, or burstiness, pattern is the tendency for word utterances to appear together in bursts as conversational topics fluctuate. We give evidence that the burstiness phenomenon exhibits itself across varied languages. Using burstiness features to train a machine-learning algorithm, we are able to assess the likelihood that a hypothesized keyword location is correct and adjust its confidence score accordingly, yielding improvements in the efficacy of keyword search in low-resource languages

Learning Feature Representation for Automatic Speech Recognition

Feature extraction in automatic speech recognition (ASR) can be regarded as learning representations from lower-level to more abstract higher-level features. Lower-level feature can be viewed as features from the signal domain, such as perceptual linear predictive (PLP) and Mel-frequency cepstral coefficients (MFCCs) features. Higher-level feature representations can be considered as bottleneck features (BNFs) learned using deep neural networks (DNNs). In this thesis, we focus on improving feature extraction at different levels mainly for ASR. The first part of this thesis focuses on learning features from the signal domain that help ASR. Hand-crafted spectral and cepstral features such as MFCC are the main features used in most conventional ASR systems; all are inspired by physiological models of the human auditory system. However, some aspects of the signal such as pitch cannot be easily extracted from spectral features, but are found to be useful for ASR. We explore new algorithm to extract a pitch feature directly from a signal for ASR and show that this feature, appended to the other feature, gives consistent improvements in various languages, especially tonal languages. We then investigate replacing the conventional features with jointly training from the signal domain using time domain, and frequency domain approaches. The results show that our time-domain joint feature learning setup achieves state-of-the-art performance using MFCC, while our frequency domain setup outperforms them in various datasets. Joint feature extraction results in learning data or language-dependent filter banks, that can degrade the performance in unseen noise and channel conditions or other languages. To tackle this, we investigate joint universal feature learning across different languages using the proposed direct-from-signal setups. We then investigate the filter banks learned in this setup and propose a new set of features as an extension to conventional Mel filter banks. The results show consistent word error rate (WER) improvement, especially in clean condition. The second part of this thesis focuses on learning higher-level feature embedding. We investigate learning and transferring deep feature representations across different domains using multi-task learning and weight transfer approaches. They have been adopted to explicitly learn intermediate-level features that are useful for several different tasks