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IEEE Workshop on Machine Learning for Signal Processing

作者： Oshry Ben-Harush Hugo Guterman Itshak Lapidot Department of Electrical and Computer Engineering Ben-Gurion University of the Negev Beersheba Israel Department of Elect and Electronics Engineering Sami Shamoon College of Engineering Ashdod Israel

Speaker diarization systems attempt to assign temporal speech segments in a conversation to the appropriate speaker, and non-speech segments to non-speech. Speaker diarization systems basically provide an answer to the question "Who spoke when ?". One inherent deficiency of most current systems is their inability to handle co-channel or overlapped speech. During the past few years, several studies have attempted dealing with the problem of overlapped or co-channel speech detection and separation, however, most of the algorithms suggested perform under unique conditions, require high computational complexity and require both time and frequency domain analysis of the audio data. In this study, frame based entropy analysis of the audio data in the time domain serves as a single feature for an overlapped speech detection algorithm. Identification of overlapped speech segments is performed using Gaussian Mixture Modeling (GMM) along with well known classification algorithms applied on two speaker conversations. By employing this methodology, the proposed method eliminates the need for setting a hard threshold for each conversation or database. LDC CALLHOME American English corpus is used for evaluation of the suggested algorithm. The proposed method successfully detects 60.0% of the frames labeled as overlapped speech by the baseline (ground-truth) segmentation , while keeping a 5% false-alarm rate.

关键词： Entropy Speech analysis Automatic speech recognition Detection algorithms Speech processing Hidden Markov models Autocorrelation linear predictive coding Educational institutions Computational complexity

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Generation of Excitation Signal in Voice Excited linear predictive coding using Discrete Cosine Transform

Generation of Excitation Signal in Voice Excited Linear Pred...

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IEEE Region 10 International Conference TENCON

作者： Gaurav Khare Prashant Shekhar M. Kulkarni Perotsystems Uttar Pradesh India Delhi College of Engineering New Delhi India

Methods involved to generate the excitation parameters and gain have been proposed that provide improvement over plain linear predictive coding (LPC) in terms of better speech reconstruction. DCT computation to transmit the excitation signal energy in initial few coefficients gives a better reconstruction of innovation signal at the receiver as compared to the primitive method of pitch estimation. Gain computation in terms of the root mean square (RMS) value of the voltage levels for each frame reduces the complexity involved as compared to plain LPC, where gain was calculated in terms of mean square error.

关键词： Signal generators linear predictive coding Discrete cosine transforms Technological innovation Filters Speech synthesis Vocoders Speech analysis Bit rate Signal synthesis

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Implementation of Voice Excited linear predictive coding (VELP) on TMS320C6711 DSP Kit

Implementation of Voice Excited Linear Predictive Coding (VE...

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IEEE International Conference on Multi Topic

作者： M. Ahsan Raza Pervaiz Akhtar PNEC National University of Science and Technology Karachi Pakistan

In today's world, telecommunication is the field in which the maximum research work is being done and speech compression is the most vital part of communication. For high compression rate speech coders, voice excited linear predictive coding (VELP) is most widely used. This paper presents real time analysis of VELP by implementing it on TMS320C6711 DSP kit using Simulink RTW (real time workshop) which explains simulink model of VELP, VELP analysis, VELP synthesis, VELP implementation on DSP kit. VELP can be briefly explained as the signal is passed through analyzer, which generate the filter co-efficients and residual signal. The residual error signal has less redundancy than original speech signal and can be quantized by smaller number of bits than the original speech. The residual error signal along with the filter coefficients are transmitted to the receiver. At the receiver, the speech is reconstructed by passing the residual error signal through the synthesis filter. To model a human speech production system, all-pole model (also known as the linear prediction model) is used. For the real time analysis of VELP, it is implemented on DSP kit using MATLAB Simulink model

关键词： linear predictive coding Digital signal processing Mathematical model Speech synthesis Filters predictive models Signal synthesis Speech coding Signal analysis Signal generators

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predictive fusion coding of spatio-temporally correlated sources

Predictive fusion coding of spatio-temporally correlated sou...

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33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Ramaswamy, Sharad Rose, Kenneth Univ Calif Santa Barbara Dept ECE Santa Barbara CA 93106 USA

ISBN: (纸本)9781424414833

This paper considers the problem of predictive fusion coding for storage of multiple spatio-temporally correlated sources so as to enable efficient selective retrieval of data from subsets of sources as designated by future queries. Only statistical information about future queries is available during encoding. While temporal correlations can be exploited by coding over large blocks, the growth in encoding complexity renders this approach impractical and hence the interest in a low complexity predictive coding approach. However, the design of optimal predictive fusion coding systems is considerably complicated by the presence of the prediction loop, and the potentially exponential growth of the query sets. We propose a complexity-constrained predictive fusion coder and derive an iterative algorithm for its design, which is based on the "Asymptotic Closed Loop" framework and hence, circumvents convergence and stability issues of traditional predictive quantizer design. The proposed predictive fusion coder optimizes the distortion - retrieval rate tradeoff, given a fixed storage capacity, and provides significant gains over storage schemes that perform only joint compression or memoryless fusion coding of all sources.

关键词： multisensor systems database query processing linear predictive coding vector quantization data compression

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Multichannel linear prediction method compliant with the MPEG-4 ALS

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IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES 2008年第3期E91A卷 756-762页

作者： Kamamoto, Yutaka Harada, Noboru Moriya, Takehiro NTT Corp NTT Commun Sci Labs Atsugi Kanagawa 2430198 Japan

A new linear prediction analysis method for multichannel signals was devised. with the goal of enhancing the compression performance of the MPEG-4 Audio Lossless coding (ALS) compliant encoder and decoder. The multichannel coding tool for this standard carries out an adaptively weighted subtraction of the residual signals of the coding channel from those of the reference channel, both of which are produced by independent linear prediction. Our linear prediction method tries to directly minimize the amplitude of the predicted residual signal after subtraction of the signals of the coding channel, and the method has been implemented in the MPEG-4 ALS codec software. The results of a comprehensive evaluation show that this method reduces the size of a compressed file. The maximum improvement of the compression ratio is 14.6% which is achieved at the cost of a small increase in computational complexity at the encoder and without increase in decoding time. This is a practical method because the compressed bitstream remains compliant with the MPEG-4 ALS standard.

关键词： linear predictive coding multichannel signal lossless compression and MPEG-4 ALS

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Digital filter interpolation of decoded LSFs for distributed continuous speech recognition

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ELECTRONICS LETTERS 2008年第17期44卷 1039-U50页

作者： de Alencar, V. F. S. Alcaim, A. Catholic Univ CETUC PUC RIO Ctr Telecommun Studies BR-22453 Rio De Janeiro Brazil

A digital filter interpolation of decoded line spectral frequencies (LSFs) that significantly outperforms linear interpolation for large vocabulary distributed continuous speech recognition systems is presented. Experiments were conducted using linear predictive coding (LPC) and LSF-derived speech recognition features, CDHMM acoustic models, triphone units and trigram language models for Brazilian Portuguese.

关键词： decoding digital filters linear predictive coding speech recognition acoustic models decoded line spectral frequencies digital filter interpolation linear interpolation speech recognition features trigram language models triphone DEcoding modelling languages Continuous speech recognition linearly independant Digital filters Acoustic models three-component geophone

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Could formant frequencies of snore signals be an alternative means for the diagnosis of obstructive sleep apnea?

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SLEEP MEDICINE 2008年第8期9卷 894-898页

作者： Ng, Andrew Keong Koh, Tong San Baey, Eugene Lee, Teck Hock Abeyratne, Udantha Ranjith Puvanendran, Kathiravelu Nanyang Technol Univ Sch Elect & Elect Engn Singapore 639798 Singapore Respiron Inc Singapore 919191 Singapore Univ Queensland Sch Informat Technol & Elect Engn Brisbane Qld Australia Singapore Gen Hosp Sleep Disorders Unit Singapore 169608 Singapore

Objective: To study the feasibility of using acoustic signatures in snore signals for the diagnosis of obstructive sleep apnea (OSA). Methods: Snoring sounds of 30 apneic snorers (24 males;6 females: apnea-hypopnea index, AHI = 46.9 +/- 25.7 events/h) and 10 benign snorers (6 males;4 females;AHI = 4.6 +/- 3.4 events/h) were captured in a sleep laboratory. The recorded snore signals were preprocessed to remove noise, and subsequently, modeled using a linear predictive coding (LPC) technique. Formant frequencies (F1, F2, and F3) were extracted from the LPC spectrum for analysis. The accuracy of this approach was assessed using receiver operating characteristic curves and notched box plots. The relationship between AHI and F1 was further explored via regression analysis. Results: Quantitative differences in formant frequencies between apneic and benign snores are found in same- or both-gender snorers. Apneic snores exhibit higher formant frequencies than benign snores, especially F1, which can be related to the pathology of OSA. This study yields a sensitivity of 88%, a specificity of 82%, and a threshold value of F1 = 470 Hz that best differentiate apneic snorers from benign snorers (both gender combined). Conclusion: Acoustic signatures in snore signals carry information for OSA diagnosis, and snore-based analysis might potentially be a non-invasive and inexpensive diagnostic approach for mass screening of OSA. (c) 2007 Elsevier B.V. All rights reserved.

关键词： Obstructive sleep apnea Polysomnography Snoring Snore signals Acoustic analysis Formant frequencies linear predictive coding

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The perceptual quality of MELP speech over error tolerant IP networks

The perceptual quality of MELP speech over error tolerant IP...

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33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Gavula, Ben Scheets, George Teague, Keith Weber, Justin Oklahoma State Univ Sch Elect & Comp Engn Stillwater OK 74078 USA

ISBN: (纸本)9781424414833

Modifications to IP based packet network protocols are examined that would make the network tolerant of bit errors in packet payloads or headers. These modifications are tested with communication quality MELP voice traffic. As measured by a PESQ score, improvements in the perceptual quality of the speech are noted that are maximized when error checking is disabled for the entire packet.

关键词： vocoders transport protocols internet error analysis linear predictive coding

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Improved Frame Loss Recovery Using Closed-Loop Estimation of Very Low Bit Rate Side Information

Improved Frame Loss Recovery Using Closed-Loop Estimation of...

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9th Annual Conference of the International-Speech-Communication-Association (INTERSPEECH 2008)

作者： Gournay, Philippe Univ Sherbrooke Speech & Audio Res Grp Sherbrooke PQ J1K 2R1 Canada

ISBN: (纸本)9781615673780

In CELP coders, the past excitation signal used to build the adaptive codebook is known to be the main source of error propagation when a frame is lost. This paper presents a novel resynchronization technique using very low bit rate side information to correct the past excitation signal after a frame erasure, the novelty being that the correction is computed in a closed loop fashion, based on the actual error introduced by the concealment. Subjective test results show that this approach is a promising area for future research on frame loss recovery.

关键词： speech codecs linear predictive coding error correction robustness

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Speech enhancement based on double RBF networks

Speech enhancement based on double RBF networks

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1st International Congress on Image and Signal Processing

作者： Guo, Jichang Guo, Libin Tianjin Univ Sch Elect Informat Engn Tianjin 300072 Peoples R China

ISBN: (纸本)9780769531199

In this paper, a non-linear spectral estimation for noise reduction is present which is approximated and implemented by double Radial Basis Function (RBF) networks. The simulation results indicate that the method can greatly improve the quality and the intelligibility of speech, and have other advantages such as the widely applicable Signal-to-Noise Ratio (SNR) range, less computation load Particularly the method may maintain the preferable accurate of signal in speech waveform, and the quality of speech signals have been improved obviously.

关键词： Speech enhancement Radial basis function networks linear predictive coding Neural networks Signal processing Noise reduction Signal to noise ratio Flowcharts Frequency Electronic mail

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