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"speech": models, code, and papers

LibriVoxDeEn: A Corpus for German-to-English Speech Translation and Speech Recognition

Oct 18, 2019
Benjamin Beilharz, Xin Sun, Sariya Karimova, Stefan Riezler

We present a corpus of sentence-aligned triples of German audio, German text, and English translation, based on German audio books. The corpus consists of over 100 hours of audio material and over 50k parallel sentences. The audio data is read speech and thus low in disfluencies. The quality of audio and sentence alignments has been checked by a manual evaluation, showing that speech alignment quality is in general very high. The sentence alignment quality is comparable to well-used parallel translation data and can be adjusted by cutoffs on the automatic alignment score. To our knowledge, this corpus is to date the largest resource for end-to-end speech translation for German.

* Corpus can be downloaded from: 

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An Information Retrieval Approach to Building Datasets for Hate Speech Detection

Jun 21, 2021
Md Mustafizur Rahman, Dinesh Balakrishnan, Dhiraj Murthy, Mucahid Kutlu, Matthew Lease

Building a benchmark dataset for hate speech detection presents several challenges. Firstly, because hate speech is relatively rare -- e.g., less than 3\% of Twitter posts are hateful \citep{founta2018large} -- random sampling of tweets to annotate is inefficient in capturing hate speech. A common practice is to only annotate tweets containing known ``hate words'', but this risks yielding a biased benchmark that only partially captures the real-world phenomenon of interest. A second challenge is that definitions of hate speech tend to be highly variable and subjective. Annotators having diverse prior notions of hate speech may not only disagree with one another but also struggle to conform to specified labeling guidelines. Our key insight is that the rarity and subjectivity of hate speech are akin to that of relevance in information retrieval (IR). This connection suggests that well-established methodologies for creating IR test collections might also be usefully applied to create better benchmark datasets for hate speech detection. Firstly, to intelligently and efficiently select which tweets to annotate, we apply established IR techniques of {\em pooling} and {\em active learning}. Secondly, to improve both consistency and value of annotations, we apply {\em task decomposition} \cite{Zhang-sigir14} and {\em annotator rationale} \cite{mcdonnell16-hcomp} techniques. Using the above techniques, we create and share a new benchmark dataset\footnote{We will release the dataset upon publication.} for hate speech detection with broader coverage than prior datasets. We also show a dramatic drop in accuracy of existing detection models when tested on these broader forms of hate. Collected annotator rationales not only provide documented support for labeling decisions but also create exciting future work opportunities for dual-supervision and/or explanation generation in modeling.

* 10 pages (Under review in CIKM 2021) 

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Towards the evaluation of simultaneous speech translation from a communicative perspective

Mar 15, 2021
claudio Fantinuoli, Bianca Prandi

In recent years, machine speech-to-speech and speech-to-text translation has gained momentum thanks to advances in artificial intelligence, especially in the domains of speech recognition and machine translation. The quality of such applications is commonly tested with automatic metrics, such as BLEU, primarily with the goal of assessing improvements of releases or in the context of evaluation campaigns. However, little is known about how such systems compare to human performances in similar communicative tasks or how the performance of such systems is perceived by final users. In this paper, we present the results of an experiment aimed at evaluating the quality of a simultaneous speech translation engine by comparing it to the performance of professional interpreters. To do so, we select a framework developed for the assessment of human interpreters and use it to perform a manual evaluation on both human and machine performances. In our sample, we found better performance for the human interpreters in terms of intelligibility, while the machine performs slightly better in terms of informativeness. The limitations of the study and the possible enhancements of the chosen framework are discussed. Despite its intrinsic limitations, the use of this framework represents a first step towards a user-centric and communication-oriented methodology for evaluating simultaneous speech translation.

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NIESR: Nuisance Invariant End-to-end Speech Recognition

Jul 07, 2019
I-Hung Hsu, Ayush Jaiswal, Premkumar Natarajan

Deep neural network models for speech recognition have achieved great success recently, but they can learn incorrect associations between the target and nuisance factors of speech (e.g., speaker identities, background noise, etc.), which can lead to overfitting. While several methods have been proposed to tackle this problem, existing methods incorporate additional information about nuisance factors during training to develop invariant models. However, enumeration of all possible nuisance factors in speech data and the collection of their annotations is difficult and expensive. We present a robust training scheme for end-to-end speech recognition that adopts an unsupervised adversarial invariance induction framework to separate out essential factors for speech-recognition from nuisances without using any supplementary labels besides the transcriptions. Experiments show that the speech recognition model trained with the proposed training scheme achieves relative improvements of 5.48% on WSJ0, 6.16% on CHiME3, and 6.61% on TIMIT dataset over the base model. Additionally, the proposed method achieves a relative improvement of 14.44% on the combined WSJ0+CHiME3 dataset.

* To appear in Proceedings of Interspeech 2019 

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Multi-speaker Emotional Text-to-speech Synthesizer

Dec 07, 2021
Sungjae Cho, Soo-Young Lee

We present a methodology to train our multi-speaker emotional text-to-speech synthesizer that can express speech for 10 speakers' 7 different emotions. All silences from audio samples are removed prior to learning. This results in fast learning by our model. Curriculum learning is applied to train our model efficiently. Our model is first trained with a large single-speaker neutral dataset, and then trained with neutral speech from all speakers. Finally, our model is trained using datasets of emotional speech from all speakers. In each stage, training samples of each speaker-emotion pair have equal probability to appear in mini-batches. Through this procedure, our model can synthesize speech for all targeted speakers and emotions. Our synthesized audio sets are available on our web page.

* Proceedings of Interspeech 2021 
* 2 pages; Published in the Proceedings of Interspeech 2021; Presented in Show and Tell; For the published paper, see 

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Sequence-to-Sequence Models Can Directly Translate Foreign Speech

Jun 12, 2017
Ron J. Weiss, Jan Chorowski, Navdeep Jaitly, Yonghui Wu, Zhifeng Chen

We present a recurrent encoder-decoder deep neural network architecture that directly translates speech in one language into text in another. The model does not explicitly transcribe the speech into text in the source language, nor does it require supervision from the ground truth source language transcription during training. We apply a slightly modified sequence-to-sequence with attention architecture that has previously been used for speech recognition and show that it can be repurposed for this more complex task, illustrating the power of attention-based models. A single model trained end-to-end obtains state-of-the-art performance on the Fisher Callhome Spanish-English speech translation task, outperforming a cascade of independently trained sequence-to-sequence speech recognition and machine translation models by 1.8 BLEU points on the Fisher test set. In addition, we find that making use of the training data in both languages by multi-task training sequence-to-sequence speech translation and recognition models with a shared encoder network can improve performance by a further 1.4 BLEU points.

* 5 pages, 1 figure. Interspeech 2017 

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Incremental Machine Speech Chain Towards Enabling Listening while Speaking in Real-time

Nov 04, 2020
Sashi Novitasari, Andros Tjandra, Tomoya Yanagita, Sakriani Sakti, Satoshi Nakamura

Inspired by a human speech chain mechanism, a machine speech chain framework based on deep learning was recently proposed for the semi-supervised development of automatic speech recognition (ASR) and text-to-speech synthesis TTS) systems. However, the mechanism to listen while speaking can be done only after receiving entire input sequences. Thus, there is a significant delay when encountering long utterances. By contrast, humans can listen to what hey speak in real-time, and if there is a delay in hearing, they won't be able to continue speaking. In this work, we propose an incremental machine speech chain towards enabling machine to listen while speaking in real-time. Specifically, we construct incremental ASR (ISR) and incremental TTS (ITTS) by letting both systems improve together through a short-term loop. Our experimental results reveal that our proposed framework is able to reduce delays due to long utterances while keeping a comparable performance to the non-incremental basic machine speech chain.

* Accepted in INTERSPEECH 2020 

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Learning Transferable Features for Speech Emotion Recognition

Dec 23, 2019
Alison Marczewski, Adriano Veloso, Nívio Ziviani

Emotion recognition from speech is one of the key steps towards emotional intelligence in advanced human-machine interaction. Identifying emotions in human speech requires learning features that are robust and discriminative across diverse domains that differ in terms of language, spontaneity of speech, recording conditions, and types of emotions. This corresponds to a learning scenario in which the joint distributions of features and labels may change substantially across domains. In this paper, we propose a deep architecture that jointly exploits a convolutional network for extracting domain-shared features and a long short-term memory network for classifying emotions using domain-specific features. We use transferable features to enable model adaptation from multiple source domains, given the sparseness of speech emotion data and the fact that target domains are short of labeled data. A comprehensive cross-corpora experiment with diverse speech emotion domains reveals that transferable features provide gains ranging from 4.3% to 18.4% in speech emotion recognition. We evaluate several domain adaptation approaches, and we perform an ablation study to understand which source domains add the most to the overall recognition effectiveness for a given target domain.

* Proceedings of the on Thematic Workshops of ACM Multimedia 2017. ACM, 2017. Pages 529-536 
* ACM-MM'17, October 23-27, 2017 

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ProsoSpeech: Enhancing Prosody With Quantized Vector Pre-training in Text-to-Speech

Feb 16, 2022
Yi Ren, Ming Lei, Zhiying Huang, Shiliang Zhang, Qian Chen, Zhijie Yan, Zhou Zhao

Expressive text-to-speech (TTS) has become a hot research topic recently, mainly focusing on modeling prosody in speech. Prosody modeling has several challenges: 1) the extracted pitch used in previous prosody modeling works have inevitable errors, which hurts the prosody modeling; 2) different attributes of prosody (e.g., pitch, duration and energy) are dependent on each other and produce the natural prosody together; and 3) due to high variability of prosody and the limited amount of high-quality data for TTS training, the distribution of prosody cannot be fully shaped. To tackle these issues, we propose ProsoSpeech, which enhances the prosody using quantized latent vectors pre-trained on large-scale unpaired and low-quality text and speech data. Specifically, we first introduce a word-level prosody encoder, which quantizes the low-frequency band of the speech and compresses prosody attributes in the latent prosody vector (LPV). Then we introduce an LPV predictor, which predicts LPV given word sequence. We pre-train the LPV predictor on large-scale text and low-quality speech data and fine-tune it on the high-quality TTS dataset. Finally, our model can generate expressive speech conditioned on the predicted LPV. Experimental results show that ProsoSpeech can generate speech with richer prosody compared with baseline methods.

* Accepted by ICASSP 2022 

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Lip to Speech Synthesis with Visual Context Attentional GAN

Apr 04, 2022
Minsu Kim, Joanna Hong, Yong Man Ro

In this paper, we propose a novel lip-to-speech generative adversarial network, Visual Context Attentional GAN (VCA-GAN), which can jointly model local and global lip movements during speech synthesis. Specifically, the proposed VCA-GAN synthesizes the speech from local lip visual features by finding a mapping function of viseme-to-phoneme, while global visual context is embedded into the intermediate layers of the generator to clarify the ambiguity in the mapping induced by homophene. To achieve this, a visual context attention module is proposed where it encodes global representations from the local visual features, and provides the desired global visual context corresponding to the given coarse speech representation to the generator through audio-visual attention. In addition to the explicit modelling of local and global visual representations, synchronization learning is introduced as a form of contrastive learning that guides the generator to synthesize a speech in sync with the given input lip movements. Extensive experiments demonstrate that the proposed VCA-GAN outperforms existing state-of-the-art and is able to effectively synthesize the speech from multi-speaker that has been barely handled in the previous works.

* Published at NeurIPS 2021 

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