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7th IEEE Workshop on Speech coding

作者： Wang, J Gibson, JD So Methodist Univ Dept Elect Engn Dallas TX 75275 USA

ISBN: (纸本)0780364163

Line Spectrum Frequencies(LSF) have been the prevailing parameter set to represent LPC coefficients in speech coding. Extensive research has been performed to exploit their interframe and intraframe correlations and quantize them more efficiently. Interframe coding of LSF's can cause error propagation when frame erasures occur. Since most LSF quantizers were designed with the primary concerns of bit-rate and complexity, less attention was paid to error propagation. We investigate the erasure performance of interframe LSF coding and compare it with an intraframe coding method. Our results show that with only 5% extra bit-rate, intraframe coding is much more robust to frame erasures and a typical improvement of 0.5 dB on spectral distortion can be obtained with 20% packet loss. Subjective listening tests indicate significant improvement as well.

关键词： Code standards Decoding Frequency Intelligent networks linear predictive coding Quantization Reflection Robustness Speech coding Testing

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AN ERROR ROBUST ULTRA LOW DELAY AUDIO CODER USING AN MA PREDICTION MODEL

AN ERROR ROBUST ULTRA LOW DELAY AUDIO CODER USING AN MA PRED...

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Wabnik, Stefan Schuller, Gerald Kraemer, Ferenc Fraunhofer IDMT Inst Digital Media Technol Ehrenbergstr 31 D-98693 Ilmenau Germany Tech Univ Ilmenau Inst Media Technol D- 98693 Ilmenau Germany

ISBN: (纸本)9781424423538

This paper compares two prediction structures for predictive perceptual audio coding in the context of the Ultra Low Delay (ULD) coding scheme. One structure is based on the commonly used AR signal model, leading to an IIR predictor in the decoder. The other structure is based on an MA signal model, leading to an FIR predictor in the decoder. We find that the AR-based predictor has a slightly better performance in case of an undisturbed transmission channel, but the MA-based predictor has a much better performance in case of transmission errors. For a Bit Error Rate (BER) of 1.0e-5, the perceptual quality of the proposed MA model predictor achieves a mean Objective Difference Grade (ODG) of -0.66 ODG whereas the AR. model predictor only reaches -3.42 ODG.

关键词： Low Delay Audio coding linear predictive coding Moving average processes Autoregressive processes Robustness

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A Two Codebook Format for Robust Quantization of Line Spectral Frequencies

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IEEE Transactions on Speech and Audio Processing 1995年第3期3卷 157-168页

作者： Ramachandran, Ravi P. Sondhi, Man Mohan Seshadri, Nambi Atal, Bishnu S. CAIP Center Department of Electrical Engineering Rutgers University Piscataway NJ United States AT&T Bell Laboratories Murray Hill NJ 07974 United States

An important problem in speech coding is the quantization of linear predictive coefficients (LPC) with the smallest possible number of bits while maintaining robustness to a large variety of speech material and transmission media. Since direct quantization of LPC’s is known to be unsatisfactory, we consider this problem for an equivalent representation, namely, the line spectral frequencies (LSF). To achieve an acceptable level of distortion a scalar quantizer for LSF’s requires a 36 bit codebook. We derive a 30 bit two-quantizer scheme which achieves a performance equivalent to this scalar quantizer. This equivalence is verified by tests on data taken from various types of filtered speech, speech corrupted by noise and by a set of randomly generated LSF’s. The two-quantizer format consists of both a vector and a scalar quantizer such that for each input, the better quantizer is used. The vector quantizer is designed from a training set that reflects the joint density (for coding efficiency) and which ensures coverage (for robustness). The scalar quantizer plays a pivotal role in dealing better with regions of the space that are sparsely covered by its vector quantizer counterpart. A further reduction of 1 bit is obtained by formulating a new adaptation algorithm for the vector quantizer and doing a dynamic programming search for both quantizers. The method of adaptation takes advantage of the ordering of the LSF’s and imposes no overhead in memory requirements. The dynamic programming search is feasible due to the ordering property. Subjective tests in a speech coder reveal that the 29 bit scheme produces equivalent perceptual quality to that when the parameters are unquantized. © 1995 IEEE

关键词： Robustness Quantization Frequency linear predictive coding Speech analysis Filters Speech coding Testing Speech enhancement Dynamic programming

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Esophageal Speech Enhancement using Modified Voicing Source

Esophageal Speech Enhancement using Modified Voicing Source

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IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

作者： Ishaq, Rizwan Zapirain, Begona Garcia Univ Deusto DeustoTech LIFE Vizcaya Spain

ISBN: (纸本)9781479947966

This paper presented the modification to Esophageal Speech (ES) enhancement using Adaptive Gain Equalizer (AGE) for modifying the voicing source. However, the voicing source used previously with AGE, obtained using conventional linear Predication (LP) vocal tract transfer function (AGE-LP), has produced low quality speech due to sensitivity to background noise. The better quality ES can be obtained by estimating voicing source through Iterative Adaptive Inverse Filtering utilized Weighted linear Prediction (WLP) vocal tract transfer function (AGE-IAIF). The system performance evaluated through Harmonic to Noise Ratio (HNR), and system has shown 3 dB enhancement by AGE-IAIF over previously enhancement method AGE-LP.

关键词： Filter bank Iterative adaptive inverse filtering Esophageal speech linear predictive coding adaptive gain equalizer

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Singing Voice Identification Using Harmonic Spectral Envelope

Singing Voice Identification Using Harmonic Spectral Envelop...

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IEEE International Conference on Information Processing (ICIP)

作者： Loni, Deepali Yoginath Subbaraman, Shaila DKTE Text & Engn Inst Dept Elect Ichalkaranji India Walchand Coll Engn Sangli India

ISBN: (纸本)9781467377584

The paper presents a novel approach to identify the singers using harmonic spectral envelope constructed from pitch of singing voice. This new representation of singing voice demonstrates that harmonic spectral envelope exhibits certain acoustic qualities that can characterize the identity of the singer. Two different approaches are implemented to extract the pitch of singing voice;Cepstrum technique and linear predictive coding. Ten singers comprising of six male and four female singers are analyzed in this work. To have accurate analysis and estimation of acoustics of singing voice only cappella sections are investigated. Along with discussion on singer identification, the results include comparison of pitch extraction techniques and gender identification of singer. We achieve an average accuracy of 77% in identifying the singers, covering a large class of polyphonic recordings of Indian movie songs.

关键词： Cappella Cepstrunm linear predictive coding Pitch Singer Singing Voice Indian songs

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Perceptual audio watermarking driven by Human Auditory System

Perceptual audio watermarking driven by Human Auditory Syste...

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International Symposium on Signals, Circuits and Systems (ISSCS)

作者： Battisti, Federica Carli, Marco Rinaldi, Claudia Univ Roma TRE Dept Engn Rome Italy Univ Laquila Ctr Excellence DEWS Laquila Italy

ISBN: (纸本)9781467361415

The aim of this work is the design of a novel audio watermarking technique based on the linear predictive coding approach and on the psychoacoustic model defined in the MPEG-I standard. The embedding imperceptibility is achieved by exploiting the frequency masking effect. The evaluation of the perceived audio quality is performed by means of the standardized ITU-R BS.1387 quality assessment method. The effectiveness of the proposed method is tested in presence of attacks.

关键词： audio coding audio watermarking hearing linear predictive coding

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Design and Test of a Spectrally Efficient Land Mobile Communications System Using LPC Speech

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IEEE Journal on Selected Areas in Communications 1984年第4期2卷 611-620页

作者： McLaughlin, Michael Linder, Donald Carney, Scott Chicago Corporate Research and Development Center Schaumburg IL 60196 Motorola Incorporated United States

A communication system was built and tested to operate in the land mobile VHF band (150–174 MHz) at a channel separation of only 6 kHz. The audio source was digitally encoded at 2.4 kbits/s using linear predictive coding (LPC). The speech data stream was transmitted by frequency shift keying (FSK) which allowed the use of class-C transmitters and discriminator detection in the receiver. Baseband filtering of the NRZ data resulted in a narrow transmitter spectrum. The receiver had a 3 dB bandwidth of 2.4 kHz which allowed data transmission with minimal intersymbol interference and frequency offset degradation. A 58 percent eye opening was found. Bit error rate (BER) performance was measured with simulated Rayleigh fading at typical 150 MHz rates. Additional tests included capture, ignition noise susceptibility, adjacent channel protection, degradation from frequency offset, and bit error effects upon speech quality. A field test was conducted to compare the speech quality of the digital radio to that of a conventional 5 kHz deviation FM mobile radio. Copyright © 1984 by The Institute of Electrical and Electronics Engineers. Inc.

关键词： System testing Mobile communication linear predictive coding Speech Frequency shift keying Radio transmitters Degradation Bit error rate Streaming media Baseband

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linear Prediction Analysis of Crosscorrelation Sequence for Voiced Speech

Linear Prediction Analysis of Crosscorrelation Sequence for ...

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Annual Summit and Conference of Asia-Pacific-Signal-and-Information-Processing-Association (APSIPA)

作者： Liu, Liqing Shimamura, Tetsuya Saitama Univ Saitama Japan

ISBN: (纸本)9786163618238

The conventional linear prediction (LP) analysis is known to suffer from problems that it is sensitive to additive noise. In this paper a new approach for LP analysis of crosscorrelation sequence between speech signal and its zero-crossing wave has been presented. Simulation results show that the proposed method is capable of performing the speech analysis under a white noisy environment.

关键词： correlation methods linear predictive coding speech processing LP analysis additive noise crosscorrelation sequence linear prediction analysis speech analysis speech signal voiced speech white noisy environment zero-crossing wave Additive noise Cepstrum Correlation Noise measurement Signal to noise ratio Speech

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A benchmark of heart sound classification systems based on sparse decompositions 14

A benchmark of heart sound classification systems based on s...

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14th International Symposium on Medical Information Processing and Analysis

作者： Ibarra-Hernandez, Roilhi F. Bertin, Nancy Alonso-Arevalo, Miguel A. Guillen-Ramirez, Hugo A. Ensenada Ctr Sci Res & Higher Educ CICESE Ensenada BC Mexico Univ Rennes CNRS IRISA INRIA Rennes France

ISBN: (数字)9781510626065

ISBN: (纸本)9781510626065

Background: Nowadays, cardiovascular diseases (CVD) remain the main cause of death worldwide. A heart sound signal or phonocardiogram (PCG) is the most simple, economical and non-invasive tool to detect CVDs. Advances in technology and signal processing allow the design of computer-aided systems for heart illnesses detection from PCG signals. Purpose: The paper proposes a pipeline and benchmark for binary heart sounds classification. The features extraction architecture is focused on the use of Matching Pursuit time-frequency decomposition using Gabor dictionaries and the linear predictive coding method of a residual. We compare seven classifiers with two different approaches: feature averaging and cycle averaging. Methods: We test our proposal on the PhysioNet/CinC challenge 2016 database, which comprises a wide variety of heart sounds recorded from patients with normal and different pathological heart conditions. We conduct a 10-fold stratified cross-validation method to evaluate the performance of different classification algorithms. The feature sets were also tested when using an oversampling method for balancing. Results: The benchmark identified systems showing a satisfying performance in terms of accuracy, sensitivity, and Matthews correlation coefficient. Results can be improved when using feature averaging and an oversampling strategy.

关键词： Heart sounds Phonocardiogram Matching Pursuit linear predictive coding Cross Validation test Oversampling SMOTE

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Feature Enhancement for Classifier Optimization and Dimensionality Reduction Speech Pathology Detection - A case study 11

Feature Enhancement for Classifier Optimization and Dimensio...

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11th Annual IEEE India Conference (INDICON)

作者： Shilaskar, Swati Ghatol, Ashok Govt Coll Engn Dept Elect & Telecommun Amravati 444604 India KJs Educ Inst Pune 411048 Maharashtra India

ISBN: (纸本)9781479953646

Voice is important for professionals like speakers, teachers, actors, singers and it is the important tool for communication. Laryngeal pathologies induce perturbations in the speech signal. Speech signal is discriminated as pathological or healthy based on roughness - breathiness - hoarseness (RBH) in the quality of signal. In recent years pattern recognition along with various signal processing techniques has emerged as an effective non invasive tool for diagnosis of pathological condition. Signal processing techniques tend to generate large number of features representing the signal. Automatic feature reduction techniques are vital in identifying the relevant features and eliminating the redundant ones. We extract features from speech signal using the acoustic analysis. Features are enhanced by alleviating gender bias. Periodic variations in the signal are captured using statistical techniques. We investigate intelligent system to generate reduced feature subset with improvement in diagnostic performance.

关键词： Acoustic Cepstral Feature extraction linear predictive coding Mel Pathology Spectral Voice quality

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