{"title":"End-to-End Overlapped Speech Detection and Speaker Counting with Raw Waveform","authors":"Wangyou Zhang, Man Sun, Lan Wang, Y. Qian","doi":"10.1109/ASRU46091.2019.9003962","DOIUrl":"https://doi.org/10.1109/ASRU46091.2019.9003962","url":null,"abstract":"Overlapped speech processing has attracted more and more attention in recent years, and it is a key problem when processing multi-talker mixed speech under the cocktail party scenario. It is commonly observed that the performance of overlapped speech processing can be significantly improved if the number of speakers is given in advance. However, such prior knowledge is often unavailable in real-world conditions, so a robust overlapped speech detection and speaker counting system is demanded. Most existing works focus on combining different handcrafted features to tackle this task, which can be sub-optimal since there are no direct connections between the features and the task. In this work, we try to solve these two problems with an end-to-end manner. First, an end-to-end framework for overlapped speech detection and speaker counting is proposed, which extracts features from the raw waveform directly. Then a curriculum learning strategy is applied to make better use of the training data. The proposed methods are evaluated on multi-talker mixed speech generated from the LibriSpeech corpus. Experimental results show that our proposed methods outperform the model with handcrafted features on both tasks, achieving more than 2% and 4% absolute accuracy improvement on overlapped speech detection and speaker counting respectively.","PeriodicalId":150913,"journal":{"name":"2019 IEEE Automatic Speech Recognition and Understanding Workshop (ASRU)","volume":"41 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2019-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126873480","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
R. Milner, Md. Asif Jalal, Raymond W. M. Ng, Thomas Hain
{"title":"A Cross-Corpus Study on Speech Emotion Recognition","authors":"R. Milner, Md. Asif Jalal, Raymond W. M. Ng, Thomas Hain","doi":"10.1109/ASRU46091.2019.9003838","DOIUrl":"https://doi.org/10.1109/ASRU46091.2019.9003838","url":null,"abstract":"For speech emotion datasets, it has been difficult to acquire large quantities of reliable data and acted emotions may be over the top compared to less expressive emotions displayed in everyday life. Lately, larger datasets with natural emotions have been created. Instead of ignoring smaller, acted datasets, this study investigates whether information learnt from acted emotions is useful for detecting natural emotions. Cross-corpus research has mostly considered cross-lingual and even cross-age datasets, and difficulties arise from different methods of annotating emotions causing a drop in performance. To be consistent, four adult English datasets covering acted, elicited and natural emotions are considered. A state-of-the-art model is proposed to accurately investigate the degradation of performance. The system involves a bi-directional LSTM with an attention mechanism to classify emotions across datasets. Experiments study the effects of training models in a cross-corpus and multi-domain fashion and results show the transfer of information is not successful. Out-of-domain models, followed by adapting to the missing dataset, and domain adversarial training (DAT) are shown to be more suitable to generalising to emotions across datasets. This shows positive information transfer from acted datasets to those with more natural emotions and the benefits from training on different corpora.","PeriodicalId":150913,"journal":{"name":"2019 IEEE Automatic Speech Recognition and Understanding Workshop (ASRU)","volume":"42 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2019-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114753829","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Virtual Adversarial Training for DS-CNN Based Small-Footprint Keyword Spotting","authors":"Xiong Wang, Sining Sun, Lei Xie","doi":"10.1109/ASRU46091.2019.9003745","DOIUrl":"https://doi.org/10.1109/ASRU46091.2019.9003745","url":null,"abstract":"Serving as the tigger of a voice-enabled user interface, on-device keyword spotting model has to be extremely compact, efficient and accurate. In this paper, we adopt a depth-wise separable convolutional neural network (DS-CNN) as our small-footprint KWS model, which is highly competitive to these ends. However, recent study has shown that a compact KWS system is very vulnerable to small adversarial perturbations while augmenting the training data with specifically-generated adversarial examples can improve performance. In this paper, we further improve KWS performance through a virtual adversarial training (VAT) solution. Instead of using adversarial examples for data augmentation, we propose to train a DS-CNN KWS model using adversarial regularization, which aims to smooth model's distribution and thus to improve robustness, by explicitly introducing a distribution smoothness measure into the loss function. Experiments on a collected KWS corpus using a circular microphone array in far-field scenario show that the VAT approach brings 31.9% relative false rejection rate (FRR) reduction compared to the normal training approach with cross entropy loss, and it also surpasses the adversarial example based data augmentation approach with 10.3% relative FRR reduction.","PeriodicalId":150913,"journal":{"name":"2019 IEEE Automatic Speech Recognition and Understanding Workshop (ASRU)","volume":"57 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2019-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121623437","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Emoception: An Inception Inspired Efficient Speech Emotion Recognition Network","authors":"Chirag Singh, Abhay Kumar, Ajay Nagar, Suraj Tripathi, Promod Yenigalla","doi":"10.1109/ASRU46091.2019.9004020","DOIUrl":"https://doi.org/10.1109/ASRU46091.2019.9004020","url":null,"abstract":"This research proposes a Deep Neural Network architecture for Speech Emotion Recognition called Emoception, which takes inspiration from Inception modules. The network takes speech features like Mel-Frequency Spectral Coefficients (MFSC) or Mel-Frequency Cepstral Coefficients (MFCC) as input and recognizes the relevant emotion in the speech. We use USC-IEMOCAP dataset for training but the limited amount of training data and large depth of the network makes the network prone to overfitting, reducing validation accuracy. The Emoception network overcomes this problem by extending in width without increase in computational cost. We also employ a powerful regularization technique, Multi-Task Learning (MTL) to make the network robust. The model using MFSC input with MTL increases the accuracy by 1.6% vis-à-vis Emoception without MTL. We report an overall accuracy improvement of around 4.6% compared to the existing state-of-art methods for four emotion classes on IEMOCAP dataset.","PeriodicalId":150913,"journal":{"name":"2019 IEEE Automatic Speech Recognition and Understanding Workshop (ASRU)","volume":"107 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2019-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123237632","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Albert Zeyer, Parnia Bahar, Kazuki Irie, R. Schlüter, H. Ney
{"title":"A Comparison of Transformer and LSTM Encoder Decoder Models for ASR","authors":"Albert Zeyer, Parnia Bahar, Kazuki Irie, R. Schlüter, H. Ney","doi":"10.1109/ASRU46091.2019.9004025","DOIUrl":"https://doi.org/10.1109/ASRU46091.2019.9004025","url":null,"abstract":"We present competitive results using a Transformer encoder-decoder-attention model for end-to-end speech recognition needing less training time compared to a similarly performing LSTM model. We observe that the Transformer training is in general more stable compared to the LSTM, although it also seems to overfit more, and thus shows more problems with generalization. We also find that two initial LSTM layers in the Transformer encoder provide a much better positional encoding. Data-augmentation, a variant of SpecAugment, helps to improve both the Transformer by 33% and the LSTM by 15% relative. We analyze several pretraining and scheduling schemes, which is crucial for both the Transformer and the LSTM models. We improve our LSTM model by additional convolutional layers. We perform our experiments on Lib-riSpeech 1000h, Switchboard 300h and TED-LIUM-v2 200h, and we show state-of-the-art performance on TED-LIUM-v2 for attention based end-to-end models. We deliberately limit the training on LibriSpeech to 12.5 epochs of the training data for comparisons, to keep the results of practical interest, although we show that longer training time still improves more. We publish all the code and setups to run our experiments.","PeriodicalId":150913,"journal":{"name":"2019 IEEE Automatic Speech Recognition and Understanding Workshop (ASRU)","volume":"64 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2019-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122869084","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"A Density Ratio Approach to Language Model Fusion in End-to-End Automatic Speech Recognition","authors":"E. McDermott, H. Sak, Ehsan Variani","doi":"10.1109/ASRU46091.2019.9003790","DOIUrl":"https://doi.org/10.1109/ASRU46091.2019.9003790","url":null,"abstract":"This article describes a density ratio approach to integrating external Language Models (LMs) into end-to-end models for Automatic Speech Recognition (ASR). Applied to a Recurrent Neural Network Transducer (RNN-T) ASR model trained on a given domain, a matched in-domain RNN-LM, and a target domain RNN-LM, the proposed method uses Bayes' Rule to define RNN-T posteriors for the target domain, in a manner directly analogous to the classic hybrid model for ASR based on Deep Neural Networks (DNNs) or LSTMs in the Hidden Markov Model (HMM) framework (Bourlard & Morgan, 1994). The proposed approach is evaluated in cross-domain and limited-data scenarios, for which a significant amount of target domain text data is used for LM training, but only limited (or no) {audio, transcript} training data pairs are used to train the RNN-T. Specifically, an RNN-T model trained on paired audio & transcript data from YouTube is evaluated for its ability to generalize to Voice Search data. The Density Ratio method was found to consistently outperform the dominant approach to LM and end-to-end ASR integration, Shallow Fusion.","PeriodicalId":150913,"journal":{"name":"2019 IEEE Automatic Speech Recognition and Understanding Workshop (ASRU)","volume":"114 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2019-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123454595","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Adapting Pretrained Transformer to Lattices for Spoken Language Understanding","authors":"Chao-Wei Huang, Yun-Nung (Vivian) Chen","doi":"10.1109/ASRU46091.2019.9003825","DOIUrl":"https://doi.org/10.1109/ASRU46091.2019.9003825","url":null,"abstract":"Lattices are compact representations that encode multiple hypotheses, such as speech recognition results or different word segmentations. It is shown that encoding lattices as opposed to 1-best results generated by automatic speech recognizer (ASR) boosts the performance of spoken language understanding (SLU). Recently, pre-trained language models with the transformer architecture have achieved the state-of-the-art results on natural language understanding, but their ability of encoding lattices has not been explored. Therefore, this paper aims at adapting pre-trained transformers to lattice inputs in order to perform understanding tasks specifically for spoken language. Our experiments on the benchmark ATIS dataset show that fine-tuning pre-trained transformers with lattice inputs yields clear improvement over fine-tuning with 1-best results. Further evaluation demonstrates the effectiveness of our methods under different acoustic conditions11The code is available at https://github.com/MiuLab/Lattice-SLU.","PeriodicalId":150913,"journal":{"name":"2019 IEEE Automatic Speech Recognition and Understanding Workshop (ASRU)","volume":"221 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2019-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116411834","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Logistic Similarity Metric Learning via Affinity Matrix for Text-Independent Speaker Verification","authors":"Junyi Peng, Rongzhi Gu, Yuexian Zou","doi":"10.1109/ASRU46091.2019.9003995","DOIUrl":"https://doi.org/10.1109/ASRU46091.2019.9003995","url":null,"abstract":"This paper proposes a novel objective function, called Logistic Affinity Loss (Logistic-AL), to optimize the end-to-end speaker verification model. Specifically, firstly, the cosine similarities of all pairs in a mini-batch of speaker embeddings are passed through a learnable logistic regression layer and the probability estimation of all pairs is obtained. Then, the supervision information for each pair is formed by their corresponding one-hot speaker labels, which indicates whether the pair belongs to the same speaker. Finally, the model is optimized by the binary cross entropy between predicted probability and target. In contrast to the other distance metric learning methods that push the distance of similar/dissimilar pairs to a pre-defined target, Logistic-AL builds a learnable decision boundary to distinguish the similar pairs and dissimilar pairs. Experimental results on the VoxCeleb1 dataset show that the x-vector feature extractor optimized by Logistic-AL achieves state-of-the-art performance.","PeriodicalId":150913,"journal":{"name":"2019 IEEE Automatic Speech Recognition and Understanding Workshop (ASRU)","volume":"6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2019-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125832194","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Speech Separation Using Speaker Inventory","authors":"Peidong Wang, Zhuo Chen, Xiong Xiao, Zhong Meng, Takuya Yoshioka, Tianyan Zhou, Liang Lu, Jinyu Li","doi":"10.1109/ASRU46091.2019.9003884","DOIUrl":"https://doi.org/10.1109/ASRU46091.2019.9003884","url":null,"abstract":"Overlapped speech is one of the main challenges in conversational speech applications such as meeting transcription. Blind speech separation and speech extraction are two common approaches to this problem. Both of them, however, suffer from limitations resulting from the lack of abilities to either leverage additional information or process multiple speakers simultaneously. In this work, we propose a novel method called speech separation using speaker inventory (SSUSI), which combines the advantages of both approaches and thus solves their problems. SSUSI makes use of a speaker inventory, i.e. a pool of pre-enrolled speaker signals, and jointly separates all participating speakers. This is achieved by a specially designed attention mechanism, eliminating the need for accurate speaker identities. Experimental results show that SSUSI outperforms permutation invariant training based blind speech separation by up to 48% relatively in word error rate (WER). Compared with speech extraction, SSUSI reduces computation time by up to 70% and improves the WER by more than 13% relatively.","PeriodicalId":150913,"journal":{"name":"2019 IEEE Automatic Speech Recognition and Understanding Workshop (ASRU)","volume":"68 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2019-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127110656","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Fengyu Yang, Shan Yang, Pengcheng Zhu, Pengju Yan, Lei Xie
{"title":"Improving Mandarin End-to-End Speech Synthesis by Self-Attention and Learnable Gaussian Bias","authors":"Fengyu Yang, Shan Yang, Pengcheng Zhu, Pengju Yan, Lei Xie","doi":"10.1109/ASRU46091.2019.9003949","DOIUrl":"https://doi.org/10.1109/ASRU46091.2019.9003949","url":null,"abstract":"Compared to conventional speech synthesis, end-to-end speech synthesis has achieved much better naturalness with more simplified system building pipeline. End-to-end framework can generate natural speech directly from characters for English. But for other languages like Chinese, recent studies have indicated that extra engineering features are still needed for model robustness and naturalness, e.g, word boundaries and prosody boundaries, which makes the front-end pipeline as complicated as the traditional approach. To maintain the naturalness of generated speech and discard language-specific expertise as much as possible, in Mandarin TTS, we introduce a novel self-attention based encoder with learnable Gaussian bias in Tacotron. We evaluate different systems with and without complex prosody information and results show that the proposed approach has the ability to generate stable and natural speech with minimum language-dependent front-end modules.","PeriodicalId":150913,"journal":{"name":"2019 IEEE Automatic Speech Recognition and Understanding Workshop (ASRU)","volume":"16 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2019-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131854904","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}