Transfer tasks in text-to-speech (TTS) synthesis -where one or more aspects of the speech of one set of speakers is transferred to another set of speakers that do not feature these aspects originally -remains a challenging task.One of the challenges is that models that have high-quality transfer capabilities can have issues in stability, making them impractical for user-facing critical tasks.This paper demonstrates that transfer can be obtained by training a robust TTS system on data generated by a less robust TTS system designed for a high-quality transfer task; in particular, a CHiVE-BERT monolingual TTS system is trained on the output of a Tacotron model designed for accent transfer.While some quality loss is inevitable with this approach, experimental results show that the models trained on synthetic data this way can produce high quality audio displaying accent transfer, while preserving speaker characteristics such as speaking style.

The prosodic aspects of speech signals produced by current text-to-speech systems are typically averaged over training material, and as such lack the variety and liveliness found in natural speech. To avoid monotony and averaged prosody contours, it is desirable to have a way of modeling the variation in the prosodic aspects of speech, so audio signals can be synthesized in multiple ways for a given text. We present a new, hierarchically structured conditional variational auto-encoder to generate prosodic features (fundamental frequency, energy and duration) suitable for use with a vocoder or a generative model like WaveNet. At inference time, an embedding representing the prosody of a sentence may be sampled from the variational layer to allow for prosodic variation. To efficiently capture the hierarchical nature of the linguistic input (words, syllables and phones), both the encoder and decoder parts of the auto-encoder are hierarchical, in line with the linguistic structure, with layers being clocked dynamically at the respective rates. We show in our experiments that our dynamic hierarchical network outperforms a non-hierarchical state-of-the-art baseline, and, additionally, that prosody transfer across sentences is possible by employing the prosody embedding of one sentence to generate the speech signal of another.

Spoken content retrieval refers to directly indexing and retrieving spoken content based on the audio rather than text descriptions. This potentially eliminates the requirement of producing text descriptions for multimedia content for indexing and retrieval purposes, and is able to precisely locate the exact time the desired information appears in the multimedia. Spoken content retrieval has been very successfully achieved with the basic approach of cascading automatic speech recognition (ASR) with text information retrieval: after the spoken content is transcribed into text or lattice format, a text retrieval engine searches over the ASR output to find desired information. This framework works well when the ASR accuracy is relatively high, but becomes less adequate when more challenging real-world scenarios are considered, since retrieval performance depends heavily on ASR accuracy. This challenge leads to the emergence of another approach to spoken content retrieval: to go beyond the basic framework of cascading ASR with text retrieval in order to have retrieval performances that are less dependent on ASR accuracy. This overview article is intended to provide a thorough overview of the concepts, principles, approaches, and achievements of major technical contributions along this line of investigation. This includes five major directions: 1) Modified ASR for Retrieval Purposes: cascading ASR with text retrieval, but the ASR is modified or optimized for spoken content retrieval purposes; 2) Exploiting the Information not present in ASR outputs: to try to utilize the information in speech signals inevitably lost when transcribed into phonemes and words; 3) Directly Matching at the Acoustic Level without ASR: for spoken queries, the signals can be directly matched at the acoustic level, rather than at the phoneme or word levels, bypassing all ASR issues; 4) Semantic Retrieval of Spoken Content: trying to retrieve spoken content that is semantically related to the query, but not necessarily including the query terms themselves; 5) Interactive Retrieval and Efficient Presentation of the Retrieved Objects: with efficient presentation of the retrieved objects, an interactive retrieval process incorporating user actions may produce better retrieval results and user experiences.

This paper presents a new approach for unsupervised Spoken Term Detection with spoken queries using multiple sets of acoustic patterns automatically discovered from the target corpus. The different pattern HMM configurations(number of states per model, number of distinct models, number of Gaussians per state)form a three-dimensional model granularity space. Different sets of acoustic patterns automatically discovered on different points properly distributed over this three-dimensional space are complementary to one another, thus can jointly capture the characteristics of the spoken terms. By representing the spoken content and spoken query as sequences of acoustic patterns, a series of approaches for matching the pattern index sequences while considering the signal variations are developed. In this way, not only the on-line computation load can be reduced, but the signal distributions caused by different speakers and acoustic conditions can be reasonably taken care of. The results indicate that this approach significantly outperformed the unsupervised feature-based DTW baseline by 16.16% in mean average precision on the TIMIT corpus.

This paper presents an initial effort to retrieve semantically related spoken content in a completely unsupervised way. Unsupervised approaches of spoken content retrieval is attractive because the need for annotated data reasonably matched to the spoken content for training acoustic and language models can be bypassed. However, almost all such unsupervised approaches focus on spoken term detection, or returning the spoken segments containing the query, using either template matching techniques such as dynamic time warping (DTW) or model-based approaches. However, users usually prefer to retrieve all objects semantically related to the query, but not necessarily including the query terms. This paper proposes a different approach. We transcribe the spoken segments in the archive to be retrieved through into sequences of acoustic patterns automatically discovered in an unsupervised method. For an input query in spoken form, the top-N spoken segments from the archive obtained with the first-pass retrieval with DTW are taken as pseudo-relevant. The acoustic patterns frequently occurring in these segments are therefore considered as query-related and used for query expansion. Preliminary experiments performed on Mandarin broadcast news offered very encouraging results.

We present a two-stage model-based approach for unsupervised query-by-example spoken term detection (STD) without any annotated data. Compared to the prevailing DTW approaches for the unsupervised STD task, HMMs used by model-based approaches can better capture the signal distributions and time trajectories of speech with a more global view of the spoken archive; matching with model states also significantly reduces the computational load. The utterances in the spoken archive are first offline decoded into acoustic patterns automatically discovered in an unsupervised way from the spoken archive. In the first stage, we propose a document state matching (DSM) approach, where query frames are matched to the HMM state sequences for the spoken documents. In this process, a novel duration-constrained Viterbi (DC-Vite) algorithm is proposed to avoid unrealistic speaking rate distortion. In the second stage, pseudo relevant/irrelevant examples retrieved from the first stage are respectively used to construct query/anti-query HMMs. Each spoken term hypothesis is then rescored with the likelihood ratio to these two HMMs. Experimental results show an absolute 11.8% of mean average precision improvement with a more than 50% reduction in computation time compared to the segmental DTW approach on a Mandarin broadcast news corpus.

Techniques for unsupervised discovery of acoustic patterns are getting increasingly attractive, because huge quantities of speech data are becoming available but manual annotations remain hard to acquire. In this paper, we propose an approach for unsupervised discovery of linguistic structure for the target spoken language given raw speech data. This linguistic structure includes two-level (subword-like and word-like) acoustic patterns, the lexicon of word-like patterns in terms of subword-like patterns and the N-gram language model based on word-like patterns. All patterns, models, and parameters can be automatically learned from the unlabelled speech corpus. This is achieved by an initialization step followed by three cascaded stages for acoustic, linguistic, and lexical iterative optimization. The lexicon of word-like patterns defines allowed consecutive sequence of HMMs for subword-like patterns. In each iteration, model training and decoding produces updated labels from which the lexicon and HMMs can be further updated. In this way, model parameters and decoded labels are respectively optimized in each iteration, and the knowledge about the linguistic structure is learned gradually layer after layer. The proposed approach was tested in preliminary experiments on a corpus of Mandarin broadcast news, including a task of spoken term detection with performance compared to a parallel test using models trained in a supervised way. Results show that the proposed system not only yields reasonable performance on its own, but is also complimentary to existing large vocabulary ASR systems.

We present a set of model-based approaches for unsupervised spoken term detection (STD) with spoken queries that requires neither speech recognition nor annotated data. This work shows the possibilities in migrating from DTW-based to model-based approaches for unsupervised STD. The proposed approach consists of three components: self-organizing models, query matching, and query modeling. To construct the self-organizing models, repeated patterns are captured and modeled using acoustic segment models (ASMs). In the query matching phase, a document state matching (DSM) approach is proposed to represent documents as ASM sequences, which are matched to the query frames. In this way, not only do the ASMs better model the signal distributions and time trajectories of speech, but the much-smaller number of states than frames for the documents leads to a much lower computational load. A novel duration-constrained Viterbi (DC-Vite) algorithm is further proposed for the above matching process to handle the speaking rate distortion problem. In the query modeling phase, a pseudo likelihood ratio (PLR) approach is proposed in the pseudo relevance feedback (PRF) framework. A likelihood ratio evaluated with query/anti-query HMMs trained with pseudo relevant/irrelevant examples is used to verify the detected spoken term hypotheses. The proposed framework demonstrates the usefulness of ASMs for STD in zero-resource settings and the potential of an instantly responding STD system using ASM indexing. The best performance is achieved by integrating DTW-based approaches into the rescoring steps in the proposed framework. Experimental results show an absolute 14.2% of mean average precision improvement with 77% CPU time reduction compared with the segmental DTW approach on a Mandarin broadcast news corpus. Consistent improvements were found on TIMIT and MediaEval 2011 Spoken Web Search corpus.

ABSTRACT Rapidly increasing quantities of multimedia and spoken con tent today demand fast and accurate retrieval approaches for con venient browsing. The spoken documents with wide variety of different acoustic and linguistic conditions make supervised training of well-matched acoustic/language models very difficult. Unsuper vised methods using frame-based dynamic time warping (DTW) re quire no acoustic/language models but with high computation load. Therefore, segment-based DTW was proposed to relieve the computation load at the cost of degraded detection performance. In this pa per, we refine the segment-based DTW by allowing deletion of end segments of query to improve detection performance. The search space is also reduced by segment similarity constraints. We also pro posed a two-pass framework. The segment-baed DTW is performed in the first pass to locate hypothesized spoken term region and the frame-based DTW for precise rescoring in the second pass. Then the pseudo relevance feedback is used to expand acoustic variations of the query. We obtain significantly higher detection performance at significantly lower computation load as compared to frame-based DTW.

This paper presents a graph-based approach for spoken term detection. Each first-pass retrieved utterance is a node on a graph and the edge between two nodes is weighted by the similarity between the two utterances evaluated in feature space. The score of each node is then modified by the contributions from its neighbors by random walk or its modified version, because utterances similar to more utterances with higher scores should be given higher relevance scores. In this way the global similarity structure of all first-pass retrieved utterances can be jointly considered. Experimental results show that this new approach offers significantly better performance than the previously proposed pseudo-relevance feedback approach, which considers primarily the local similarity relationship between first-pass retrieved utterances, and these two different approaches can be cascaded to provide even better results.