Bo Wu, Kehuang Li, Zhen Huang, S. Siniscalchi, Minglei Yang, Chin-Hui Lee
{"title":"A unified deep modeling approach to simultaneous speech dereverberation and recognition for the reverb challenge","authors":"Bo Wu, Kehuang Li, Zhen Huang, S. Siniscalchi, Minglei Yang, Chin-Hui Lee","doi":"10.1109/HSCMA.2017.7895557","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895557","url":null,"abstract":"We propose a unified deep neural network (DNN) approach to achieve both high-quality enhanced speech and high-accuracy automatic speech recognition (ASR) simultaneously on the recent REverberant Voice Enhancement and Recognition Benchmark (RE-VERB) Challenge. These two goals are accomplished by two proposed techniques, namely DNN-based regression to enhance reverberant and noisy speech, followed by DNN-based multi-condition training that takes clean-condition, multi-condition and enhanced speech all into consideration. We first report on superior objective measures in enhanced speech to those listed in the 2014 REVERB Challenge Workshop. We then show that in clean-condition training, we obtain the best word error rate (WER) of 13.28% on the 1-channel REVERB simulated evaluation data with the proposed DNN-based pre-processing scheme. Similarly we attain a competitive single-system WER of 8.75% with the proposed multi-condition training strategy and the same less-discriminative log power spectrum features used in the enhancement stage. Finally by leveraging upon joint training with more discriminative ASR features and improved neural network based language models a state-of-the-art WER of 4.46% is attained with a single ASR system, and single-channel information. Another state-of-the-art WER of 4.10% is achieved through system combination.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"38 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-03-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115149066","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}
Jinkyu Lee, Keulbit Kim, T. Shabestary, Hong-Goo Kang
{"title":"Deep bi-directional long short-term memory based speech enhancement for wind noise reduction","authors":"Jinkyu Lee, Keulbit Kim, T. Shabestary, Hong-Goo Kang","doi":"10.1109/HSCMA.2017.7895558","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895558","url":null,"abstract":"In this paper, we propose a new recurrent neural network (RNN)-based single-channel speech enhancement framework for off-line wind noise reduction. To adequately represent highly non-stationary characteristics of wind noise, we first adopt a deep bi-directional long short-term memory (DBLSTM) structure. However, its enhanced output becomes muffled due to the spectral over-smoothing effect. To overcome this problem, we propose a new structure of DBLSTM-based speech enhancement system that internally incorporates the speech and noise power estimation processes in the spectral filtering framework. Furthermore, we propose a structure with an additional internal constraint of minimizing log a priori SNR, which provides efficient learning mechanism. Experimental results show that the proposed method improves source-to-distortion ratio (SDR) by 6.9 dB and perceptual evaluation of speech quality (PESQ) by 0.24 in comparison to the conventional DBLSTM-based system.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"88 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-03-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122507504","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}
Fabrice Katzberg, Radoslaw Mazur, M. Maass, P. Koch, A. Mertins
{"title":"Multigrid reconstruction of sound fields using moving microphones","authors":"Fabrice Katzberg, Radoslaw Mazur, M. Maass, P. Koch, A. Mertins","doi":"10.1109/HSCMA.2017.7895588","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895588","url":null,"abstract":"A multi-resolution technique is presented for sound field recovery based on measurements of one or multiple moving microphones. The interpolation of the spatial samples enables us to set up a system of linear equations that recovers room impulse responses on a virtual uniform grid in space. The spacing of the virtual grid must be very small when directly recovering for the entire bandwidth, thus, the system to be solved requires a large number of measurement positions. In this work, we propose two recovery schemes based on multiple virtual grids that represent distinct sub-bands of the spatiotemporal sound field. This allows for faster reconstruction of low frequencies at minor computational cost and improves the recovery quality when measurement noise is present.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"50 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-03-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131269589","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}
Lior Madmoni, Hendrik Barfuss, B. Rafaely, Walter Kellermann
{"title":"A unified framework for multiple arrays on a robot and application to sound localization","authors":"Lior Madmoni, Hendrik Barfuss, B. Rafaely, Walter Kellermann","doi":"10.1109/HSCMA.2017.7895563","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895563","url":null,"abstract":"The auditory system of humanoid robots has recently gained significant attention in the research community. This system enables interaction with the surroundings via microphone arrays embodied into the robot. Typically, a single array topology is designed which may limit the performance of the auditory system. In a recent paper, a system with two array types has been studied: the first consists of sensors distributed over the robot's head, and the second is a body-mounted robomorphic array. These arrays have only been studied separately without exploiting their performance as a single array. In this work, a unified framework has been developed for the two arrays in the spherical harmonics domain, and an initial investigation of direction of arrival estimation (DOA) with the unified framework is presented. The simulation study shows that the unified framework outperforms DOA estimation with the individual arrays, mainly at the low frequency range.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"17 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-03-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126237002","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}
JeeSok Lee, Soo-Whan Chung, Min-Seok Choi, Hong-Goo Kang
{"title":"A study on search grid points for data-driven 3-D beamsteering","authors":"JeeSok Lee, Soo-Whan Chung, Min-Seok Choi, Hong-Goo Kang","doi":"10.1109/HSCMA.2017.7895551","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895551","url":null,"abstract":"This paper presents a 3-D search grid allocation algorithm for source localization and beam steering. Data-driven beam-steering systems can benefit from algorithm designs that restrict the processing to a finite set of predefined direction of arrival (DoA) grid points. However, this scenario requires careful definition of the distribution of the search grid points to achieve high spatial filtering performance. The non-linear relationship that exists between time-delay of arrival (TDoA) and degree-oriented DoA means that a generic uniform grid distribution approach is not adequate for accurate localization of the source direction in the time domain. The aim of the proposed search grid allocation algorithm is to maximize the efficiency of localization by exploiting the sensitivity variation in temporal and spatial aspects. Experimental results show that the proposed search grid allocation outperforms the uniform grid allocation in terms of TDoA accuracy and steering vector estimations.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"9 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-03-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124834069","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":"Sound field separation in a mixed acoustic environment using a sparse array of higher order spherical microphones","authors":"Abdullah Fahim, P. Samarasinghe, T. Abhayapala","doi":"10.1109/HSCMA.2017.7895580","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895580","url":null,"abstract":"In many acoustic environments, we frequently observe that sound fields caused by desired and undesired sources superimpose each other and form a mixed sound field in the region. Hence, in acoustic signal processing applications, such as sound field recording, speech dereverberation and noise suppression, it is of a great interest to separate the desired sound field from the undesired ones. In this paper, we propose a solution to isolate the desired sound field in a mixed sound field environment for 3D sound propagation. The solution utilizes a sparse array of higher order microphones to achieve the goal, which greatly reduces the design complexity by using less number of microphones in an implementation-favorable geometry. We evaluate the performance under different noisy and noiseless environments to test the effectiveness as well as the robustness of the algorithm.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-03-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129386454","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":"Speaker tracking in reverberant environments using multiple directions of arrival","authors":"C. Evers, B. Rafaely, P. Naylor","doi":"10.1109/HSCMA.2017.7895568","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895568","url":null,"abstract":"Accurate estimation of the Direction of Arrival (DOA) of a sound source is an important prerequisite for a wide range of acoustic signal processing applications. However, in enclosed environments, early reflections and late reverberation often lead to localization errors. Recent work demonstrated that improved robustness against reverberation can be achieved by clustering only the DOAs from direct-path bins in the short-term Fourier transform of a speech signal of several seconds duration from a static talker. Nevertheless, for moving talkers, short blocks of at most several hundred milliseconds are required to capture the spatio-temporal variation of the source direction. Processing of short blocks of data in reverberant environment can lead to clusters whose centroids correspond to spurious DOAs away from the source direction. We therefore propose in this paper a novel multi-detection source tracking approach that estimates the smoothed trajectory of the source DOAs. Results for realistic room simulations validate the proposed approach and demonstrate significant improvements in estimation accuracy compared to single-detection tracking.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-03-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125487256","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}
K. Nathwani, J. A. Morales-Cordovilla, S. Sivasankaran, I. Illina, E. Vincent
{"title":"An extended experimental investigation of DNN uncertainty propagation for noise robust ASR","authors":"K. Nathwani, J. A. Morales-Cordovilla, S. Sivasankaran, I. Illina, E. Vincent","doi":"10.1109/HSCMA.2017.7895555","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895555","url":null,"abstract":"Automatic speech recognition (ASR) in noisy environments remains a challenging goal. Recently, the idea of estimating the uncertainty about the features obtained after speech enhancement and propagating it to dynamically adapt deep neural network (DNN) based acoustic models has raised some interest. However, the results in the literature were reported on simulated noisy datasets for a limited variety of uncertainty estimators. We found that they vary significantly in different conditions. Hence, the main contribution of this work is to assess DNN uncertainty decoding performance for different data conditions and different uncertainty estimation/propagation techniques. In addition, we propose a neural network based uncertainty estimator and compare it with other uncertainty estimators. We report detailed ASR results on the CHiME-2 and CHiME-3 datasets. We find that, on average, uncertainty propagation provides similar relative improvement on real and simulated data and that the proposed uncertainty estimator performs significantly better than the one in [1]. We also find that the improvement is consistent, but it depends on the signal-to-noise ratio (SNR) and the noise environment.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-03-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115844520","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":"Towards real-time source counting by estimation of coherent-to-diffuse ratios from ad-hoc microphone array recordings","authors":"Shahab Pasha, Jacob Donley, C. Ritz, Y. Zou","doi":"10.1109/HSCMA.2017.7895582","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895582","url":null,"abstract":"Coherent-to-diffuse ratio (CDR) estimates over short time frames are utilized for source counting using ad-hoc microphone arrays to record speech from multiple participants in scenarios such as a meeting. It is shown that the CDR estimates obtained at ad-hoc dual (two channel) microphone nodes, located at unknown locations within an unknown reverberant room, can detect time frames with more than one active source and are informative for source counting applications. Results show that interfering sources can be detected with accuracies ranging from 69% to 89% for delays ranging from 20 ms to 300 ms, with source counting accuracies ranged from 61% to 81% for two sources and the same range of delays.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"24 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-03-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126511725","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}
S. Araki, N. Ito, Marc Delcroix, A. Ogawa, K. Kinoshita, T. Higuchi, Takuya Yoshioka, Dung T. Tran, Shigeki Karita, T. Nakatani
{"title":"Online meeting recognition in noisy environments with time-frequency mask based MVDR beamforming","authors":"S. Araki, N. Ito, Marc Delcroix, A. Ogawa, K. Kinoshita, T. Higuchi, Takuya Yoshioka, Dung T. Tran, Shigeki Karita, T. Nakatani","doi":"10.1109/HSCMA.2017.7895553","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895553","url":null,"abstract":"This paper addresses our new online meeting recognition prototype, which works even in noisy environments. For speech enhancement, we employ a mask-based minimum variance distortionless response (MVDR) beamformer, which has recently shown to be a successful front-end for a state-of-the-art deep neural network (DNN)-based automatic speech recognition (ASR) system. To ensure more accurate and computationally efficient mask estimation for estimating steering vectors of a MVDR beamformer, which is challenging especially in an adverse environment, this work employs a probabilistic spatial dictionary. The dictionary consists of a pre-trained probability distribution of source location features for each of speaker location candidates. This dictionary significantly simplifies the problem to be solved, and therefore, realizes an accurate and efficient mask estimation. Our new prototype worked with stability with a latency of 3–5 seconds in a real exhibition situation. The quantitative evaluation results will be reported.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-03-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123104492","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}