Morfessor is a family of probabilistic machine learning methods for finding the morphological segmentation from raw text data. Recent developments include the development of semi-supervised methods for utilizing annotated data. Morfessor 2.0 is a rewrite of the original, widely-used Morfessor 1.0 software, with well documented command-line tools and library interface. It includes new features such as semi-supervised learning, online training, and integrated evaluation code.

Automatic speech recognition (ASR) systems require large amounts of transcribed speech data, for training state-of-theart deep neural network (DNN) acoustic models. Transcribed speech is a scarce and expensive resource, and ASR systems are prone to underperform in domains where there is not a lot of training data available. In this work, we open up a vast and previously unused resource of transcribed speech for Finnish, by retrieving and aligning all the recordings and meeting transcripts from the web portal of the Parliament of Finland. Short speech-text segment pairs are retrieved from the audio and text material, by using the Levenshtein algorithm to align the firstpass ASR hypotheses with the corresponding meeting transcripts. DNN acoustic models are trained on the automatically constructed corpus, and performance is compared to other models trained on a commercially available speech corpus. Model performance is evaluated on Finnish parliament speech, by dividing the testing set into seen and unseen speakers. Performance is also evaluated on broadcast speech to test the general applicability of the parliament speech corpus. We also study the use of meeting transcripts in language model adaptation, to achieve additional gains in speech recognition accuracy of Finnish parliament speech.

Because in agglutinative languages the number of observed word forms is very high, subword units are often utilized in speech recognition. However, the proper use of subword units requires careful consideration of details such as silence modeling, position-dependent phones, and combination of the units. In this paper, we implement subword modeling in the Kaldi toolkit by creating modified lexicon by finite-state transducers to represent the subword units correctly. We experiment with multiple types of word boundary markers and achieve the best results by adding a marker to the left or right side of a subword unit whenever it is not preceded or followed by a word boundary, respectively. We also compare three different toolkits that provide data-driven subword segmentations. In our experiments on a variety of Finnish and Estonian datasets, the best subword models do outperform word-based models and naive subword implementations. The largest relative reduction in WER is a 23% over word-based models for a Finnish read speech dataset. The results are also better than any previously published ones for the same datasets, and the improvement on all datasets is more than 5%.

Today, the vocabulary size for language models in large vocabulary speech recognition is typically several hundreds of thousands of words. While this is already sufficient in some applications, the out-of-vocabulary words are still limiting the usability in others. In agglutinative languages the vocabulary for conversational speech should include millions of word forms to cover the spelling variations due to colloquial pronunciations, in addition to the word compounding and inflections. Very large vocabularies are also needed, for example, when the recognition of rare proper names is important.Previously, very large vocabularies have been efficiently modeled in conventional n-gram language models either by splitting words into subword units or by clustering words into classes. While vocabulary size is not as critical anymore in modern speech recognition systems, training time and memory consumption become an issue when state-of-the-art neural network language models are used. In this paper we investigate techniques that address the vocabulary size issue by reducing the effective vocabulary size and by processing large vocabularies more efficiently.The experimental results in conversational Finnish and Estonian speech recognition indicate that properly defined word classes improve recognition accuracy. Subword n-gram models are not better on evaluation data than word n-gram models constructed from a vocabulary that includes all the words in the training corpus. However, when recurrent neural network (RNN) language models are used, their ability to utilize long contexts gives a larger gain to subword-based modeling. Our best results are from RNN language models that are based on statistical morphs. We show that the suitable size for a subword vocabulary depends on the language. Using time delay neural network (TDNN) acoustic models, we were able to achieve new state of the art in Finnish and Estonian conversational speech recognition, 27.1 % word error rate in the Finnish task and 21.9 % in the Estonian task.Index Terms-language modeling, word classes, subword units, artificial neural networks, automatic speech recognition 2329-9290

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We describe the speech recognition systems we have created for MGB-3, the 3 rd Multi Genre Broadcast challenge, which this year consisted of a task of building a system for transcribing Egyptian Dialect Arabic speech, using a big audio corpus of primarily Modern Standard Arabic speech and only a small amount (5 hours) of Egyptian adaptation data. Our system, which was a combination of different acoustic models, language models and lexical units, achieved a Multi-Reference Word Error Rate of 29.25%, which was the lowest in the competition. Also on the old MGB-2 task, which was run again to indicate progress, we achieved the lowest error rate: 13.2%.The result is a combination of the application of stateof-the-art speech recognition methods such as simple dialect adaptation for a Time-Delay Neural Network (TDNN) acoustic model (-27% errors compared to the baseline), Recurrent Neural Network Language Model (RNNLM) rescoring (an additional -5%), and system combination with Minimum Bayes Risk (MBR) decoding (yet another -10%). We also explored the use of morph and character language models, which was particularly beneficial in providing a rich pool of systems for the MBR decoding.

We study character-based language models in the state-ofthe-art speech recognition framework. This approach has advantages over both word-based systems and so-called endto-end ASR systems that do not have separate acoustic and language models. We describe the necessary modifications needed to build an effective character-based ASR system using the Kaldi toolkit and evaluate the models based on words, statistical morphs, and characters for both Finnish and Arabic. The morph-based models yield the best recognition results for both well-resourced and lower-resourced tasks, but the character-based models are close to their performance in the lower-resource tasks, outperforming the word-based models. Character-based models are especially good at predicting novel word forms that were not seen in the training data. Using character-based neural network language models is both computationally efficient and provides a larger gain compared to the morph and word-based systems.Index Terms-speech recognition, subword-based language modeling, neural network language models, low resource, unlimited vocabulary

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