Abstract: Modern topic identification (topic ID) systems for speech use automatic speech recognition (ASR) to produce speech transcripts, and perform supervised classification on such ASR outputs. However, under resource-limited conditions, the manually transcribed speech required to develop standard ASR systems can be severely limited or unavailable. In this paper, we investigate alternative unsupervised solutions to obtaining tokenizations of speech in terms of a vocabulary of automatically discovered word-like or phoneme-like units, without depending on the supervised training of ASR systems. Moreover, using automatic phoneme-like tokenizations, we demonstrate that a convolutional neural network based framework for learning spoken document representations provides competitive performance compared to a standard bag-of-words representation, as evidenced by comprehensive topic ID evaluations on both single-label and multi-label classification tasks.

Introduction: Topic identification (topic ID) on speech aims to identify the topic(s) for given speech recordings, referred to as spoken documents, where the topics are a predefined set of classes or labels. This task is typically formulated as a three-step process. First, speech is tokenized into words or phones by automatic speech recognition (ASR) systems BIBREF0 , or by limited-vocabulary keyword spotting BIBREF1 . Second, standard text-based processing techniques are applied to the resulting tokenizations, and produce a vector representation for each spoken document, typically a bag-of-words multinomial representation, or a more compact vector given by probabilistic topic models BIBREF2 , BIBREF3 . Finally, topic ID is performed on the spoken document representations by supervised training of classifiers, such as Bayesian classifiers and support vector machines (SVMs). However, in the first step, training the ASR system required for tokenization itself requires transcribed speech and pronunciations. In this paper, we focus on a difficult and realistic scenario where the speech corpus of a test language is annotated only with a minimal number of topic labels, i.e., no manual transcriptions or dictionaries for building an ASR system are available. We aim to exploit approaches that enable topic ID on speech without any knowledge of that language other than the topic annotations. In this scenario, while previous work demonstrates that the cross-lingual phoneme recognizers can produce reasonable speech tokenizations BIBREF4 , BIBREF5 , the performance is highly dependent on the language and environmental condition (channel, noise, etc.) mismatch between the training and test data. Therefore, we focus on unsupervised approaches that operate directly on the speech of interest. Raw acoustic feature-based unsupervised term discovery (UTD) is one such approach that aims to identify and cluster repeating word-like units across speech based around segmental dynamic time warping (DTW) BIBREF6 , BIBREF7 . BIBREF8 shows that using the word-like units from UTD for spoken document classification can work well; however, the results in BIBREF8 are limited since the acoustic features on which UTD is performed are produced by acoustic models trained from the transcribed speech of its evaluation corpus. In this paper, we investigate UTD-based topic ID performance when UTD operates on language-independent speech representations extracted from multilingual bottleneck networks trained on languages other than the test language BIBREF9 . Another alternative to producing speech tokenizations without language dependency is the model-based approach, i.e., unsupervised learning of hidden Markov model (HMM) based phoneme-like units from untranscribed speech. We exploit the Variational Bayesian inference based acoustic unit discovery (AUD) framework in BIBREF10 that allows parallelized large-scale training. In topic ID tasks, such AUD-based systems have been shown to outperform other systems based on cross-lingual phoneme recognizers BIBREF5 , and this paper aims to further investigate how the performance compares among UTD, AUD and ASR based systems. Moreover, after the speech is tokenized, these works BIBREF0 , BIBREF1 , BIBREF4 , BIBREF5 , BIBREF8 , BIBREF9 are limited to using bag-of-words features as spoken document representations. While UTD only identifies relatively long (0.5 – 1 sec) repeated terms, AUD/ASR enables full-coverage segmentation of continuous speech into a sequence of units/words, and such a resulting temporal sequence enables another feature learning architecture based on convolutional neural networks (CNNs) BIBREF11 ; instead of treating the sequential tokens as a bag of acoustic units or words, the whole token sequence is encoded as concatenated continuous vectors, and followed by convolution and temporal pooling operations that capture the local and global dependencies. Such continuous space feature extraction frameworks have been used in various language processing tasks like spoken language understanding BIBREF12 , BIBREF13 and text classification BIBREF14 , BIBREF15 . However, three questions are worth investigating in our AUD-based setting: (i) if such a CNN-based framework can perform as well on noisy automatically discovered phoneme-like units as on orthographic words/characters, (ii) if pre-trained vectors of phoneme-like units from word2vec BIBREF16 provide superior performance to random initialization as evidenced by the word-based tasks, and (iii) if CNNs are still competitive in low-resource settings of hundreds to two-thousand training exemplars, rather than the large/medium sized datasets as in previous work BIBREF14 , BIBREF15 . Finally, incorporating the different tokenization and feature representation approaches noted above, we perform comprehensive topic ID evaluations on both single-label and multi-label spoken document classification tasks. 

Question: What datasets are used to assess the performance of the system?
Answer: LORELEI datasets of Uzbek, Mandarin and Turkish