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Acoustic Event Detection plays an important role for computational acoustic scene analysis. Although we would face with a sound overlapping problem in a real situation, conventional methods do not consider the problem enough. In this paper, we propose a new overlapped acoustic event detection technique combined a source separation technique of Non-negative Matrix Factorization with shared basis vectors...
We describe a scheme to translate spoken English lectures into Japanese consisting of a deep neural network based English automatic speech recognition system (ASR) and an English to Japanese phrase-based statistical machine translation system (SMT). The bad influence of speech misrecognition for the translation model is focused. For coping with bad influence caused by speech misrecognition, we utilized...
One of the difficulties in sung speech recognition is the small distance in an acoustic space between phonemes in sung speech. Therefore we considered clustering the speech based on a pitch (fundamental frequency F0) and creating a larger distance between the phonemes. In addition, we considered a two-stage training method of DNN-HMM: the first stage is trained by using conventional acoustic features...
Deep neural networks (DNN) have achieved significant success in the field of speech recognition. One of the main advantages of the DNN is automatic feature extraction without human intervention. Therefore, we incorporate a pseudo-filterbank layer to the bottom of DNN and train the whole filterbank layer and the following networks jointly, while most systems take pre-defined mel-scale filterbanks as...
Speech emotion recognition is a still challenging problem despite having been investigated over the last couple of decades. Conventional speech emotion recognition performance is low, but this may be improved by considering new features and an annotation method. In this paper, firstly we use glottal features for speech emotion recognition to improve its performance because the emotions are related...
This paper describes our scheme to translate spoken English lectures into Japanese consisting of an English automatic speech recognition system (ASR) that utilizes a deep neural network (DNN) and an English to Japanese phrase-based statistical machine translation system (SMT). We focused on domain adaptation of the acoustic and translation models. For domain adaptation of the translation model, frequently...
A spoken dialogue system for car-navigation systems may be able to provide more natural and smoother communications but it must also cause safety problems. One of these problems is distraction whereby machine operation and voice conversations influence the driver. Even the use of a simple speech interface may affect the driving operation. We consider that a spoken dialogue system which can understand...
Recently, spoken dialog systems using speech recognition technology have been becoming popular. Such as chat-like dialog systems, these systems which do not have any specific purpose are called “Non-Task-oriented spoken dialog system”. In this study, we focused on non-task-oriented spoken dialog systems. We have developed a multi-party chat-like dialog system with different preferences between two...
Lyric recognition in singing is challenging because of a number of problems, including a lack of singing databases, superposed musical instruments and different spectral variations. First of all, we investigated the difference of spectral variations among read speech, spontaneous speech and sung speech and we found that sung speech recognition was the most difficult. Next, we consider Japanese lyric...
In speech recognition, it is preferable not to hypothesize the details, e.g., specific age and gender, of a target user. However, speaker independence is one of the things that degrades ASR performance. In this work, we propose a speaker adaptation method to recognize a short time utterance. There have been several studies on speaker-independent DNN-HMM in which i-vector is computed, and the additional...
This paper presents a Japanese spoken term detection method for spoken queries using a combination of word-based search and syllable-based N-gram search with in-vocabulary/out-of-vocabulary (IV/OOV) term classification. The N-gram index in a recognized syllable-based lattice for OOV terms, which assumes recognition errors such as substitution, insertion and deletion errors, incorporates a distance...
We investigated speech recognition methods for mixed speech and music that only remove music based on non-negative matrix factorization (NMF). In this paper, we introduced the Euclidean distance of logarithm spectrum DLOG as a distance measure for source separation, which may correspond to the distance measure for speech recognition, and compared it with such traditional distance measures as the Kullback-Leibler...
This paper presents our scheme to translate spoken English lectures into Japanese that consists of an English automatic speech recognition system (ASR) that utilizes a deep neural network (DNN) and an English to Japanese phrase-based statistical machine translation system (SMT). We utilized an existing Wall Street Journal corpus for our acoustic model and adapted it with MIT OpenCourseWare lectures...
Japanese is syllabic language. Additionally we have studied syllable-based GMM-HMM for Japanese speech recognition. In this paper, we investigate the differences of recognition accuracy using phoneme/syllable-based GMM-HMM and DNN (Deep Neural Network)-HMM. First, we present a comparison of syllable-based and phoneme-based DNN-HMM. Second, we train the tied state left-context dependent syllable DNN-HMM,...
We have considered a speech recognition method for mixed sound, consisting of speech and music, that removes only the music based on vector quantization (VQ) and non-negative matrix factorization (NMF). Instead of conventional amplitude spectrum distance measure, MFCC distance measure which is not affected by the pitch is introduced. For isolated word recognition using the clean speech model, an improvement...
In this paper, we present a feature enhancement method that uses neural networks (NNs) to map the reverberant feature in a log-melspectral domain to its corresponding anechoic feature. The mapping is done by cascade NNs trained using Cascade 2 algorithm with an implementation of segment-based normalization. We assumed that the dimensions of feature were independent from each other and experimented...
In prior work, we investigated the application of a spherical microphone array to a distant speech recognition task. In that work, the relative positions of a fixed loud speaker and the spherical array required for beamforming were measured with an optical tracking device. In the present work, we investigate how these relative positions can be determined automatically for real, human speakers based...
A low-power many-core SoC for multimedia applications is implemented in 40nm CMOS technology. Within a 210mm2 die, two 32-core clusters are integrated with dynamically reconfigurable processors, hardware accelerators, 2-channel DDR3 I/Fs, and other peripherals. Processor cores in the cluster share a 2MB L2 cache connected through a tree-based Network-on-Chip (NoC). The high scalability and low power...
Distant speech recognition (DSR) holds out the promise of providing a natural human computer interface in that it enables verbal interactions with computers without the necessity of donning intrusive body- or head-mounted devices. Recognizing distant speech robustly, however, remains a challenge. This paper provides a overview of DSR systems based on microphone arrays. In particular, we present recent...
We have considered a speech recognition method for mixed sound, consisting of speech and music, that removes only the music based on vector quantization (VQ) and non-negative matrix factorization (NMF). This paper describe fast calculation technique of music removal based on NMF and improvement using a VQ method. For isolated word recognition using the clean speech model, an improvement of 46% word...
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