2013 IEEE International Conference on Acoustics, Speech and Signal Processing最新文献

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Riesz-transform-based demodulation of narrowband spectrograms of voiced speech 基于riesz变换的浊音窄带频谱解调
2013 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2013-05-26 DOI: 10.1109/ICASSP.2013.6639264
Haricharan Aragonda, C. Seelamantula
{"title":"Riesz-transform-based demodulation of narrowband spectrograms of voiced speech","authors":"Haricharan Aragonda, C. Seelamantula","doi":"10.1109/ICASSP.2013.6639264","DOIUrl":"https://doi.org/10.1109/ICASSP.2013.6639264","url":null,"abstract":"Narrowband spectrograms of voiced speech can be modeled as an outcome of two-dimensional (2-D) modulation process. In this paper, we develop a demodulation algorithm to estimate the 2-D amplitude modulation (AM) and carrier of a given spectrogram patch. The demodulation algorithm is based on the Riesz transform, which is a unitary, shift-invariant operator and is obtained as a 2-D extension of the well known 1-D Hilbert transform operator. Existing methods for spectrogram demodulation rely on extension of sinusoidal demodulation method from the communications literature and require precise estimate of the 2-D carrier. On the other hand, the proposed method based on Riesz transform does not require a carrier estimate. The proposed method and the sinusoidal demodulation scheme are tested on real speech data. Experimental results show that the demodulated AM and carrier from Riesz demodulation represent the spectrogram patch more accurately compared with those obtained using the sinusoidal demodulation. The signal-to-reconstruction error ratio was found to be about 2 to 6 dB higher in case of the proposed demodulation approach.","PeriodicalId":183968,"journal":{"name":"2013 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"653 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2013-05-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122958530","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: 3
Informed Source Separation from compressed mixtures using spatial wiener filter and quantization noise estimation 利用空间维纳滤波和量化噪声估计从压缩混合物中分离信息源
2013 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2013-05-26 DOI: 10.1109/ICASSP.2013.6637609
Shuhua Zhang, Laurent Girin, A. Liutkus
{"title":"Informed Source Separation from compressed mixtures using spatial wiener filter and quantization noise estimation","authors":"Shuhua Zhang, Laurent Girin, A. Liutkus","doi":"10.1109/ICASSP.2013.6637609","DOIUrl":"https://doi.org/10.1109/ICASSP.2013.6637609","url":null,"abstract":"In a previous work, we proposed an Informed Source Separation system based on Wiener filtering for active listening of music from uncompressed (16-bit PCM) multichannel mix signals. In the present work, the system is improved to work with (MPEG-2 AAC) compressed mix signals: quantization noise is estimated from the AAC bitstream at the decoder and explicitly taken into account in the source separation process. Also a direct MDCT-to-STFT transform is used to optimize the computational efficiency of the process in the STFT domain from AAC-decoded MDCT coefficients.","PeriodicalId":183968,"journal":{"name":"2013 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"32 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2013-05-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122958783","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
Voice activity detection using a sliding-window, maximum margin clustering approach 语音活动检测使用滑动窗口,最大边际聚类方法
2013 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2013-05-26 DOI: 10.1109/ICASSP.2013.6638953
P. D. Leon, Salvador Sanchez
{"title":"Voice activity detection using a sliding-window, maximum margin clustering approach","authors":"P. D. Leon, Salvador Sanchez","doi":"10.1109/ICASSP.2013.6638953","DOIUrl":"https://doi.org/10.1109/ICASSP.2013.6638953","url":null,"abstract":"Recently, an unsupervised, data clustering algorithm based on maximum margin, i.e. support vector machine (SVM) was reported. The maximum margin clustering (MMC) algorithm was later applied to the problem of voice activity detection, however, the application did not allow for real-time detection which is important in speech processing applications. In this paper, we propose a voice activity detector (VAD) based on a sliding window, MMC algorithm which allows for real-time detection. Our system requires a separate initialization stage which imposes an initial detection delay, however, once initialized the system can operate in real-time. Using TIMIT speech under several NOISEX-92 noise backgrounds at various SNRs, we show that our average speech and non-speech hit rates are better than state-of-the-art VADs.","PeriodicalId":183968,"journal":{"name":"2013 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"25 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2013-05-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114178322","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
A physical layer authentication scheme for countering primary user emulation attack 针对主用户仿真攻击的物理层认证方案
2013 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2013-05-26 DOI: 10.1109/ICASSP.2013.6638195
Kapil M. Borle, Biao Chen, Wenliang Du
{"title":"A physical layer authentication scheme for countering primary user emulation attack","authors":"Kapil M. Borle, Biao Chen, Wenliang Du","doi":"10.1109/ICASSP.2013.6638195","DOIUrl":"https://doi.org/10.1109/ICASSP.2013.6638195","url":null,"abstract":"This paper develops a physical layer user authentication scheme for wireless systems. The approach can be used as an effective counter measure against the primary user emulation attack in cognitive radios. The developed scheme applies to general digital constellations and we establish its optimality in terms of error probability for user authentication. Trade-off analysis is provided that balances the performance of the user authentication for the secondary user and symbol detection for the primary user. In particular, we show that arbitrarily reliable user authentication can be achieved at the price of an almost negligible performance degradation for the primary user under realistic system settings.","PeriodicalId":183968,"journal":{"name":"2013 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"33 9 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2013-05-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114508082","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: 35
Integrating finite difference schemes for scalar and vector wave equations 积分标量波方程和矢量波方程的有限差分格式
2013 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2013-05-26 DOI: 10.1109/ICASSP.2013.6637631
Jonathan Botts, L. Savioja
{"title":"Integrating finite difference schemes for scalar and vector wave equations","authors":"Jonathan Botts, L. Savioja","doi":"10.1109/ICASSP.2013.6637631","DOIUrl":"https://doi.org/10.1109/ICASSP.2013.6637631","url":null,"abstract":"Room acoustic simulation is the process of generating approximate solutions to either the linearized Euler equations or the scalar wave equation. As for the continuous equations, the discrete approximations of both are equivalent. The vector formulation is less efficient, but it can inform several unexploited features of the scalar formulation. This paper first demonstrates the equivalence of the two schemes and explores how the vector formulation may be integrated into the more efficient scalar formulation to produce local velocity estimates and velocity sources on the pressure grid.","PeriodicalId":183968,"journal":{"name":"2013 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"25 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2013-05-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114583981","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
Unsupervised topic model for broadcast program segmentation 广播节目分割的无监督主题模型
2013 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2013-05-26 DOI: 10.1109/ICASSP.2013.6639315
Gilles Boulianne, P. Dumouchel
{"title":"Unsupervised topic model for broadcast program segmentation","authors":"Gilles Boulianne, P. Dumouchel","doi":"10.1109/ICASSP.2013.6639315","DOIUrl":"https://doi.org/10.1109/ICASSP.2013.6639315","url":null,"abstract":"Several unsupervised methods have been proposed to segment a continuous text stream into individual topics. A simple HMM formulation of the most successful of these methods exposes their underlying assumptions and suggests the use of a new prior for segmentation probability. Under this formulation, we explore the space of possible modeling choices on databases of English and French TV and radio programs. We show that the proposed prior improves segmentation results and can also accommodate additional knowledge sources within the HMM efficient dynamic programming.","PeriodicalId":183968,"journal":{"name":"2013 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"55 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2013-05-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122131993","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
A K-best orthogonal matching pursuit for compressive sensing 基于k -最优正交匹配追踪的压缩感知
2013 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2013-05-26 DOI: 10.1109/ICASSP.2013.6638757
Pu-Hsuan Lin, S. Tsai, G. C. Chuang
{"title":"A K-best orthogonal matching pursuit for compressive sensing","authors":"Pu-Hsuan Lin, S. Tsai, G. C. Chuang","doi":"10.1109/ICASSP.2013.6638757","DOIUrl":"https://doi.org/10.1109/ICASSP.2013.6638757","url":null,"abstract":"This paper proposes an orthogonal matching pursuit (OMP-) based recovering algorithm for compressive sensing problems. This algorithm can significantly improve recovering performance while it can still maintain reasonable computational complexity. Complexity analysis and simulation results are provided for the proposed algorithm and compared with other popular recovering schemes. We observe that the proposed algorithm can significantly improve the exact recovering performance compared to the OMP scheme. Moreover, in the cases with high compressed ratio, the proposed algorithm can even outperform the benchmark performance achieved by the subspace programming and linear programming.","PeriodicalId":183968,"journal":{"name":"2013 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"56 3-4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2013-05-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116591113","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
Prediction based filtering and smoothing to exploit temporal dependencies in NMF 基于预测的滤波和平滑利用NMF中的时间依赖性
2013 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2013-05-26 DOI: 10.1109/ICASSP.2013.6637773
N. Mohammadiha, P. Smaragdis, A. Leijon
{"title":"Prediction based filtering and smoothing to exploit temporal dependencies in NMF","authors":"N. Mohammadiha, P. Smaragdis, A. Leijon","doi":"10.1109/ICASSP.2013.6637773","DOIUrl":"https://doi.org/10.1109/ICASSP.2013.6637773","url":null,"abstract":"Nonnegative matrix factorization is an appealing technique for many audio applications. However, in it's basic form it does not use temporal structure, which is an important source of information in speech processing. In this paper, we propose NMF-based filtering and smoothing algorithms that are related to Kalman filtering and smoothing. While our prediction step is similar to that of Kalman filtering, we develop a multiplicative update step which is more convenient for nonnegative data analysis and in line with existing NMF literature. The proposed smoothing approach introduces an unavoidable processing delay, but the filtering algorithm does not and can be readily used for on-line applications. Our experiments using the proposed algorithms show a significant improvement over the baseline NMF approaches. In the case of speech denoising with factory noise at 0 dB input SNR, the smoothing algorithm outperforms NMF with 3.2 dB in SDR and around 0.5 MOS in PESQ, likewise source separation experiments result in improved performance due to taking advantage of the temporal regularities in speech.","PeriodicalId":183968,"journal":{"name":"2013 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2013-05-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116614646","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: 28
Numerical near field optimization of a non-uniform sub-band filter-and-sum beamformer 非均匀子带滤波和波束形成器的数值近场优化
2013 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2013-05-26 DOI: 10.1109/ICASSP.2013.6637616
F. Heese, Magnus Schaefer, Jona Wernerus, P. Vary
{"title":"Numerical near field optimization of a non-uniform sub-band filter-and-sum beamformer","authors":"F. Heese, Magnus Schaefer, Jona Wernerus, P. Vary","doi":"10.1109/ICASSP.2013.6637616","DOIUrl":"https://doi.org/10.1109/ICASSP.2013.6637616","url":null,"abstract":"A novel near field filter-and-sum beamformer using non uniform frequency sub-bands is presented. The concept is based on numerical optimization of the reception characteristic of the microphone array. In order to improve the reception characteristic over frequency and space, a non uniform filterbank is utilized to subdivide the frequency range. Individual optimization processes for each sub-band result in a clearly improved reception characteristic. The new system is able to closely approximate a target (independently of the frequency) which can be defined according to the application.","PeriodicalId":183968,"journal":{"name":"2013 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"46 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2013-05-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"117078470","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: 6
Optimization of the per tone noise protection in xDSL systems employing virtual noise 采用虚拟噪声的xDSL系统单音噪声保护优化
2013 IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 2013-05-26 DOI: 10.1109/ICASSP.2013.6638548
Wagih Sarhan, M. Kuipers, A. Klein
{"title":"Optimization of the per tone noise protection in xDSL systems employing virtual noise","authors":"Wagih Sarhan, M. Kuipers, A. Klein","doi":"10.1109/ICASSP.2013.6638548","DOIUrl":"https://doi.org/10.1109/ICASSP.2013.6638548","url":null,"abstract":"In Digital Subscriber Line (DSL) systems, the concept of Virtual Noise (VN) was introduced to improve the protection against fluctuating crosstalk and to increase link stability. In our previous work, we have presented an algorithm that estimates the VN mask and the initialization signal-to-noise-ratio (SNR) margin from noise measurements. In however, perfect bitswapping was assumed. In practical DSL systems bitswapping might be too slow which makes the link sensitive to sudden noise increases. Hence, the outage probability achieved in can only be realized in slowly changing channels. This paper investigates the optimization of the VN mask and the initialization SNR margin presented in in order to improve the robustness against sudden noise increases in terms of outage probability, especially for modems with slow bitswapping procedures.","PeriodicalId":183968,"journal":{"name":"2013 IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2013-05-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129715548","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
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