Phonetic and graphemic systems for multi-genre broadcast transcription

State-of-the-art English automatic speech recognition systems typically use phonetic rather than graphemic lexicons. Graphemic systems are known to perform less well for English as the mapping from the written form to the spoken form is complicated. However, in recent years the representational power of deep-learning based acoustic models has improved, raising interest in graphemic acoustic models for English, due to the simplicity of generating the lexicon. In this paper, phonetic and graphemic models are compared for an English Multi-Genre Broadcast transcription task. A range of acoustic models based on lattice-free MMI training are constructed using phonetic and graphemic lexicons. For this task, it is found that having a long-span temporal history reduces the difference in performance between the two forms of models. In addition, system combination is examined, using parameter smoothing and hypothesis combination. As the combination approaches become more complicated the difference between the phonetic and graphemic systems further decreases. Finally, for all configurations examined the combination of phonetic and graphemic systems yields consistent gains.This research was partly funded under the ALTA Institute, University of Cambridge. Thanks to Cambridge English, University of Cambridge, for supporting this research

Sequence Teacher-Student Training of Acoustic Models for Automatic Free Speaking Language Assessment

A high performance automatic speech recognition (ASR) system is an important constituent component of an automatic language assessment system for free speaking language tests. The ASR system is required to be capable of recognising non-native spontaneous English speech and to be deployable under real-time conditions. The performance of ASR systems can often be significantly improved by leveraging upon multiple systems that are complementary, such as an ensemble. Ensemble methods, however, can be computationally expensive, often requiring multiple decoding runs, which makes them impractical for deployment. In this paper, a lattice-free implementation of sequence-level teacher-student training is used to reduce this computational cost, thereby allowing for real-time applications. This method allows a single student model to emulate the performance of an ensemble of teachers, but without the need for multiple decoding runs. Adaptations of the student model to speakers from different first languages (L1s) and grades are also explored.Cambridge Assessment Englis

Automatic speech recognition system development in the “wild“

The standard framework for developing an automatic speech recognition (ASR) system is to generate training and development data for building the system, and evaluation data for the final performance analysis. All the data is assumed to come from the domain of interest. Though this framework is matched to some tasks, it is more challenging for systems that are required to operate over broad domains, or where the ability to collect the required data is limited. This paper discusses ASR work performed under the IARPA MATERIAL program, which is aimed at cross-language information retrieval, and examines this challenging scenario. In terms of available data, only limited narrow-band conversational telephone speech data was provided. However, the system is required to operate over a range of domains, including broadcast data. As no data is available for the broadcast domain, this paper proposes an approach for system development based on scraping "related" data from the web, and using ASR system confidence scores as the primary metric for developing the acoustic and language model components. As an initial evaluation of the approach, the Swahili development language is used, with the final system performance assessed on the IARPA MATERIAL Analysis Pack 1 data.The Office of the Director of National Intelligence (ODNI), Intelligence Advanced Research Projects Activity (IARPA), via Air Force Research Laboratory (AFRL

Ensemble generation and compression for speech recognition

For many tasks in machine learning, performance gains can often be obtained by combining together an ensemble of multiple systems. In Automatic Speech Recognition (ASR), a range of approaches can be used to combine an ensemble when performing recognition. However, many of these have computational costs that scale linearly with the ensemble size. One method to address this is teacher-student learning, which compresses the ensemble into a single student. The student is trained to emulate the combined ensemble, and only the student needs to be used when performing recognition. This thesis investigates both methods for ensemble generation and methods for ensemble compression. The first contribution of this thesis is to explore approaches of generating multiple systems for an ensemble. The combined ensemble performance depends on both the accuracy of the individual members of the ensemble, as well as the diversity between their behaviours. The structured nature of speech allows for many ways that systems can be made different from each other. The experiments suggest that significant combination gains can be obtained by combining systems with different acoustic models, sets of state clusters, and sets of sub-word units. When performing recognition, these ensembles can be combined at the hypothesis and frame levels. However, these combination methods can be computationally expensive, as data is processed by multiple systems. This thesis also considers approaches to compress an ensemble, and reduce the computational cost when performing recognition. Teacher-student learning is one such method. In standard teacher-student learning, information about the per-frame state cluster posteriors is propagated from the teacher ensemble to the student, to train the student to emulate the ensemble. However, this has two limitations. First, it requires that the teachers and student all use the same set of state clusters. This limits the allowed forms of diversities that the ensemble can have. Second, ASR is a sequence modelling task, and the frame-level posteriors that are propagated may not effectively convey all information about the sequence-level behaviours of the teachers. This thesis addresses both of these limitations. The second contribution of this thesis is to address the first limitation, and allow for different sets of state clusters between systems. The proposed method maps the state cluster posteriors from the teachers' sets of state clusters to that of the student. The map is derived by considering a distance measure between posteriors of unclustered logical context-dependent states, instead of the usual state cluster. The experiments suggest that this proposed method can allow a student to effectively learn from an ensemble that has a diversity of state cluster sets. However, the experiments also suggest that the student may need to have a large set of state clusters to effectively emulate this ensemble. This thesis proposes to use a student with a multi-task topology, with an output layer for each of the different sets of state clusters. This can capture the phonetic resolution of having multiple sets of state clusters, while having fewer parameters than a student with a single large output layer. The third contribution of this thesis is to address the second limitation of standard teacher-student learning, that only frame-level information is propagated to emulate the ensemble behaviour for the sequence modelling ASR task. This thesis proposes to generalise teacher-student learning to the sequence level, and propagate sequence posterior information. The proposed methods can also allow for many forms of ensemble diversities. The experiments suggest that by using these sequence-level methods, a student can learn to emulate the ensemble better. Recently, the lattice-free method has been proposed to train a system directly toward a sequence discriminative criterion. Ensembles of these systems can exhibit highly diverse behaviours, because the systems are not biased toward any cross-entropy forced alignments. It is difficult to apply standard frame-level teacher-student learning with these lattice-free systems, as they are often not designed to produce state cluster posteriors. Sequence-level teacher-student learning operates directly on the sequence posteriors, and can therefore be used directly with these lattice-free systems. The proposals in this thesis are assessed on four ASR tasks. These are the augmented multi-party interaction meeting transcription, IARPA Babel Tok Pisin conversational telephone speech, English broadcast news, and multi-genre broadcast tasks. These datasets provide a variety of quantities of training data, recording environments, and speaking styles

General sequence teacher-student learning

In automatic speech recognition, performance gains can often be obtained by combining an ensemble of multiple models. However, this can be computationally expensive when performing recognition. Teacher-student learning alleviates this cost by training a single student model to emulate the combined ensemble behaviour. Only this student needs to be used for recognition. Previously investigated teacher-student criteria often limit the forms of diversity allowed in the ensemble, and only propagate information from the teachers to the student at the frame level. This paper addresses both of these issues by examining teacher-student learning within a sequence-level framework, and assessing the flexibility that these approaches offer. Various sequence-level teacher-student criteria are examined in this work, to propagate sequence posterior information. A training criterion based on the KL-divergence between context-dependent state sequence posteriors is proposed that allows for a diversity of state cluster sets to be present in the ensemble. This criterion is shown to be an upper bound to a more general KL-divergence between word sequence posteriors, which places even fewer restrictions on the ensemble diversity, but whose gradient can be expensive to compute. These methods are evaluated on the AMI meeting transcription and MGB-3 television broadcast audio tasks.This research was partly funded under the ALTA Institute, University of Cambridge. Thanks to Cambridge Assessment English, University of Cambridge, for supporting this research

Linguistically-motivated sub-word modeling with applications to speech recognition

Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2009.Includes bibliographical references (p. 173-185).Despite the proliferation of speech-enabled applications and devices, speech-driven human-machine interaction still faces several challenges. One of theses issues is the new word or the out-of-vocabulary (OOV) problem, which occurs when the underlying automatic speech recognizer (ASR) encounters a word it does not "know". With ASR being deployed in constantly evolving domains such as restaurant ratings, or music querying, as well as on handheld devices, the new word problem continues to arise.This thesis is concerned with the OOV problem, and in particular with the process of modeling and learning the lexical properties of an OOV word through a linguistically-motivated sub-syllabic model. The linguistic model is designed using a context-free grammar which describes the sub-syllabic structure of English words, and encapsulates phonotactic and phonological constraints. The context-free grammar is supported by a probability model, which captures the statistics of the parses generated by the grammar and encodes spatio-temporal context. The two main outcomes of the grammar design are: (1) sub-word units, which encode pronunciation information, and can be viewed as clusters of phonemes; and (2) a high-quality alignment between graphemic and sub-word units, which results in hybrid entities denoted as spellnemes. The spellneme units are used in the design of a statistical bi-directional letter-to-sound (L2S) model, which plays a significant role in automatically learning the spelling and pronunciation of a new word.The sub-word units and the L2S model are assessed on the task of automatic lexicon generation. In a first set of experiments, knowledge of the spelling of the lexicon is assumed. It is shown that the phonemic pronunciations associated with the lexicon can be successfully learned using the L2S model as well as a sub-word recognizer.(cont.) In a second set of experiments, the assumption of perfect spelling knowledge is relaxed, and an iterative and unsupervised algorithm, denoted as Turbo-style, makes use of spoken instances of both spellings and words to learn the lexical entries in a dictionary.Sub-word speech recognition is also embedded in a parallel fashion as a backoff mechanism for a word recognizer. The resulting hybrid model is evaluated in a lexical access application, whereby a word recognizer first attempts to recognize an isolated word. Upon failure of the word recognizer, the sub-word recognizer is manually triggered. Preliminary results show that such a hybrid set-up outperforms a large-vocabulary recognizer.Finally, the sub-word units are embedded in a flat hybrid OOV model for continuous ASR. The hybrid ASR is deployed as a front-end to a song retrieval application, which is queried via spoken lyrics. Vocabulary compression and open-ended query recognition are achieved by designing a hybrid ASR. The performance of the frontend recognition system is reported in terms of sentence, word, and sub-word error rates. The hybrid ASR is shown to outperform a word-only system over a range of out-of-vocabulary rates (1%-50%). The retrieval performance is thoroughly assessed as a fmnction of ASR N-best size, language model order, and the index size. Moreover, it is shown that the sub-words outperform alternative linguistically-motivated sub-lexical units such as phonemes. Finally, it is observed that a dramatic vocabulary compression - by more than a factor of 10 - is accompanied by a minor loss in song retrieval performance.by Ghinwa F. Choueiter.Ph.D