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2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)最新文献

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Cost function for sound source localization with arbitrary microphone arrays 任意麦克风阵列声源定位的代价函数
2017 Hands-free Speech Communications and Microphone Arrays (HSCMA) Pub Date : 2017-03-17 DOI: 10.1109/HSCMA.2017.7895565
I. Tashev, Long N. Le, Vani Gopalakrishna, A. Lovitt
{"title":"Cost function for sound source localization with arbitrary microphone arrays","authors":"I. Tashev, Long N. Le, Vani Gopalakrishna, A. Lovitt","doi":"10.1109/HSCMA.2017.7895565","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895565","url":null,"abstract":"The sound source localizer is an important part of any microphone array processing block. Its major purpose is to determine the direction of arrival of the sound source and let a beamformer aim its beam towards this direction. In addition, the direction of arrival can be used for meetings diarization, pointing a camera, sound source separation. Multiple algorithms and approaches exist, targeting different settings and microphone arrays. In this paper we treat the sound source localizer as a classifier and use as features the phase differences and magnitude proportions in the microphone channels. To determine the proper mix, we propose a novel cost function to measure the localization capability. The resulting algorithm is fast and suitable for real-time implementations. It works well with different microphone array geometries with both omni-directional and unidirectional microphones.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"65 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-03-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124596937","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}
引用次数: 4
Clock drift estimation and compensation for asynchronous impulse response measurements 异步脉冲响应测量的时钟漂移估计与补偿
2017 Hands-free Speech Communications and Microphone Arrays (HSCMA) Pub Date : 2017-03-15 DOI: 10.1109/HSCMA.2017.7895587
H. Gamper
{"title":"Clock drift estimation and compensation for asynchronous impulse response measurements","authors":"H. Gamper","doi":"10.1109/HSCMA.2017.7895587","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895587","url":null,"abstract":"The impulse response (IR) of an acoustic environment or audio device can be measured by recording its response to a known test signal. Ideally, the same digital clock should be used for playback and recording to ensure synchronous digital-to-analog and analog-to-digital conversion. When measuring the acoustic performance of a hardware device, be it for audio input to a device microphone or audio output from a device speaker, it is often difficult to access the device's audio signal path electronically. Therefore, the device-under-test (DUT) has to act either as a playback or recording device for the IR measurement. However, it may be impossible to synchronise the internal clock of the DUT with the reference clock of the measurement system. As a result, the recorded DUT response may be subject to unknown clock drift which may lead to undesired artefacts in the measured IR. Here, a method is proposed for estimating the drift between a playback and recording clock directly from the recorded response to obtain a drift-compensated IR. Experimental results from IR measurements of a DUT subject to clock drift indicate that the proposed method successfully estimates the drift rate and yields an accurate IR estimate in magnitude and phase.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"97 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-03-15","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124064890","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}
引用次数: 1
Room equalization based on measurements with moving microphones 基于移动麦克风测量的房间均衡
2017 Hands-free Speech Communications and Microphone Arrays (HSCMA) Pub Date : 2017-03-03 DOI: 10.1109/HSCMA.2017.7895574
Radoslaw Mazur, Fabrice Katzberg, Huy Phan, A. Mertins
{"title":"Room equalization based on measurements with moving microphones","authors":"Radoslaw Mazur, Fabrice Katzberg, Huy Phan, A. Mertins","doi":"10.1109/HSCMA.2017.7895574","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895574","url":null,"abstract":"For enhancing the perceived quality of a sound played in a closed room, the methods of room impulse response equalization are used. For a better control of late echoes, new approaches, such as temporal masking, exploit the properties of the human auditory system. In order to allow the listener to move freely even in a small area, a whole volume has to be equalized. Traditional methods require a measurement of a huge amount of impulse responses in this volume. In this work we propose a method which allows for a greatly reduced measurement effort. Here, we employ a dynamic method for measurement of impulse responses using just one moving microphone. The reconstructed impulse responses are used to perform equalization and a simple interpolating technique allows for a equalization at the position of the listener, who can freely move inside the measured volume.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"46 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-03-03","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125174928","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}
引用次数: 4
Robust variable-regularized RLS algorithms 鲁棒变量正则化RLS算法
2017 Hands-free Speech Communications and Microphone Arrays (HSCMA) Pub Date : 2017-03-01 DOI: 10.1109/HSCMA.2017.7895584
Camelia Elisei-Iliescu, C. Stanciu, C. Paleologu, J. Benesty, C. Anghel, S. Ciochină
{"title":"Robust variable-regularized RLS algorithms","authors":"Camelia Elisei-Iliescu, C. Stanciu, C. Paleologu, J. Benesty, C. Anghel, S. Ciochină","doi":"10.1109/HSCMA.2017.7895584","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895584","url":null,"abstract":"The regularization parameter is required in most (if not all) adaptive algorithms, while its role becomes very critical in the presence of additive noise. In this paper, we focus on the regularized recursive least-squares (RLS) algorithm and present a method to find its regularization parameter, which is related to the signal-to-noise ratio (SNR). Also, using a proper estimation of the SNR, we further propose a variable-regularized RLS (VR-RLS) algorithm. In addition, a low-complexity version of the VR-RLS algorithm is developed, based on the dichotomous coordinate descent (DCD) method. Due to their nature, the proposed algorithms have good robustness features against additive noise, which make them behave well in all noisy conditions. Simulations performed in the context of acoustic echo cancellation support these findings.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"106 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":"122438638","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}
引用次数: 16
Two-stage speech enhancement with manipulation of the cepstral excitation 利用倒侧刺激的两阶段语音增强
2017 Hands-free Speech Communications and Microphone Arrays (HSCMA) Pub Date : 2017-03-01 DOI: 10.1109/HSCMA.2017.7895571
Samy Elshamy, N. Madhu, Wouter Tirry, T. Fingscheidt
{"title":"Two-stage speech enhancement with manipulation of the cepstral excitation","authors":"Samy Elshamy, N. Madhu, Wouter Tirry, T. Fingscheidt","doi":"10.1109/HSCMA.2017.7895571","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895571","url":null,"abstract":"The development of new speech enhancement techniques is a continuous progress to combat the impairment of speech signals by various acoustical environmental influences. In this contribution we propose a new two-stage speech enhancement algorithm, exploiting the source-filter model to decompose a denoised target signal, and specifically we manipulate the excitation signal in the cepstral domain. The second stage therein is a refinement of the a priori signal-to-noise ratio (SNR) estimate used for the suppression gain calculation. Different to prior art, a higher noise attenuation can be achieved, without any more artifacts in the processed speech component.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"2 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":"128683683","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}
引用次数: 5
Long-term robot motion planning for active sound source localization with Monte Carlo tree search 基于蒙特卡罗树搜索的主动声源定位的长期机器人运动规划
2017 Hands-free Speech Communications and Microphone Arrays (HSCMA) Pub Date : 2017-03-01 DOI: 10.1109/HSCMA.2017.7895562
Quan V. Nguyen, F. Colas, E. Vincent, F. Charpillet
{"title":"Long-term robot motion planning for active sound source localization with Monte Carlo tree search","authors":"Quan V. Nguyen, F. Colas, E. Vincent, F. Charpillet","doi":"10.1109/HSCMA.2017.7895562","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895562","url":null,"abstract":"We consider the problem of controlling a mobile robot in order to localize a sound source. A microphone array can provide the robot with information on source localization. By combining this information with the movements of the robot, the localization accuracy can be improved. However, random robot motion or short-term planning may not result in optimal localization. In this paper, we propose an optimal long-term robot motion planning algorithm for active source localization. We introduce a Monte Carlo tree search (MCTS) method to find a sequence of robot actions that minimize the entropy of the belief on the source location. A tree of possible robot movements which balances between exploration and exploitation is first constructed. Then, the movement that leads to minimum uncertainty is selected and executed. Experiments and statistical results show the effectiveness of our proposed method on improving sound source localization in the long term compared to other motion planning methods.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"115 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":"122498568","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}
引用次数: 15
Microphone array signal processing for robot audition 机器人试听用麦克风阵列信号处理
2017 Hands-free Speech Communications and Microphone Arrays (HSCMA) Pub Date : 2017-03-01 DOI: 10.1109/HSCMA.2017.7895560
Heinrich W. Löllmann, Alastair H. Moore, P. Naylor, B. Rafaely, R. Horaud, A. Mazel, Walter Kellermann
{"title":"Microphone array signal processing for robot audition","authors":"Heinrich W. Löllmann, Alastair H. Moore, P. Naylor, B. Rafaely, R. Horaud, A. Mazel, Walter Kellermann","doi":"10.1109/HSCMA.2017.7895560","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895560","url":null,"abstract":"Robot audition for humanoid robots interacting naturally with humans in an unconstrained real-world environment is a hitherto unsolved challenge. The recorded microphone signals are usually distorted by background and interfering noise sources (speakers) as well as room reverberation. In addition, the movements of a robot and its actuators cause ego-noise which degrades the recorded signals significantly. The movement of the robot body and its head also complicates the detection and tracking of the desired, possibly moving, sound sources of interest. This paper presents an overview of the concepts in microphone array processing for robot audition and some recent achievements.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"91 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":"133184530","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}
引用次数: 17
DOA estimation in noisy environment with unknown noise power using the EM algorithm 基于EM算法的未知噪声功率噪声环境下的DOA估计
2017 Hands-free Speech Communications and Microphone Arrays (HSCMA) Pub Date : 2017-03-01 DOI: 10.1109/HSCMA.2017.7895567
Ofer Schwartz, Yuval Dorfan, Maja Taseska, Emanuël Habets, S. Gannot
{"title":"DOA estimation in noisy environment with unknown noise power using the EM algorithm","authors":"Ofer Schwartz, Yuval Dorfan, Maja Taseska, Emanuël Habets, S. Gannot","doi":"10.1109/HSCMA.2017.7895567","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895567","url":null,"abstract":"A direction of arrival (DOA) estimator for concurrent speakers in a noisy environment with unknown noise power is presented. Spatially colored noise, if not properly addressed, is known to degrade the performance of DOA estimators. In our contribution, the DOA estimation task is formulated as a maximum likelihood (ML) problem, which is solved using the expectation-maximization (EM) procedure. The received microphone signals are modelled as a sum of the speech and noise components. The noise power spectral density (PSD) matrix is modelled by a time-invariant full-rank coherence matrix multiplied by the noise power. The PSDs of the speech and noise components are estimated as part of the EM procedure. The benefit of the presented algorithm in a simulated noisy environment using measured room impulse responses is demonstrated.","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":"134327726","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}
引用次数: 18
Listening-area-informed sound field reproduction with Gaussian prior based on circular harmonic expansion 基于循环谐波展开的高斯先验声场再现
2017 Hands-free Speech Communications and Microphone Arrays (HSCMA) Pub Date : 2017-03-01 DOI: 10.1109/HSCMA.2017.7895589
Natsuki Ueno, Shoichi Koyama, H. Saruwatari
{"title":"Listening-area-informed sound field reproduction with Gaussian prior based on circular harmonic expansion","authors":"Natsuki Ueno, Shoichi Koyama, H. Saruwatari","doi":"10.1109/HSCMA.2017.7895589","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895589","url":null,"abstract":"A sound field reproduction method using prior information on listening areas is proposed. Most current methods aim to reproduce the sound field over the entire space or around the listener's location. The methods considering the location of listening areas enable high reproduction accuracy with a small number of loudspeakers inside these areas; however, the reproduction accuracy outside these areas significantly deteriorates in several cases. We propose a method that takes into consideration the spatial priority distribution inside the listening areas. The optimal driving signals are obtained when a mixture of Gaussian distributions is assumed as the prior. Simulation results indicate that high reproduction accuracy near the central areas is achieved while significant deterioration in the peripheral areas is prevented.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"555 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":"134057036","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}
引用次数: 2
Modulation-domain speech enhancement using a Kalman filter with a Bayesian update of speech and noise in the log-spectral domain 利用卡尔曼滤波器在对数谱域对语音和噪声进行贝叶斯更新的调制域语音增强
2017 Hands-free Speech Communications and Microphone Arrays (HSCMA) Pub Date : 2017-03-01 DOI: 10.1109/HSCMA.2017.7895572
Nikolaos Dionelis, M. Brookes
{"title":"Modulation-domain speech enhancement using a Kalman filter with a Bayesian update of speech and noise in the log-spectral domain","authors":"Nikolaos Dionelis, M. Brookes","doi":"10.1109/HSCMA.2017.7895572","DOIUrl":"https://doi.org/10.1109/HSCMA.2017.7895572","url":null,"abstract":"We present a Bayesian estimator that performs log-spectrum estimation of both speech and noise, and is used as a Bayesian Kalman filter update step for single-channel speech enhancement in the modulation domain. We use Kalman filtering in the log-power spectral domain rather than in the amplitude or power spectral domains. In the Bayesian Kalman filter update step, we define the posterior distribution of the clean speech and noise log-power spectra as a two-dimensional multivariate Gaussian distribution. We utilize a Kalman filter observation constraint surface in the three-dimensional space, where the third dimension is the phase factor. We evaluate the results of the phase-sensitive log-spectrum Kalman filter by comparing them with the results obtained by traditional noise suppression techniques and by an alternative Kalman filtering technique that assumes additivity of speech and noise in the power spectral domain.","PeriodicalId":337673,"journal":{"name":"2017 Hands-free Speech Communications and Microphone Arrays (HSCMA)","volume":"39 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":"132442933","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}
引用次数: 11
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