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IEEE International Conference on Image Processing (ICIP 2006)

作者： Mueller, K. Smolic, A. Kautzner, M. Wiegand, T. Heinrich Hertz Inst Nachrichtentech Berlin GmbH Fraunhofer Inst Telecommun Image Proc Dept Einsteinufer 37 D-10587 Berlin Germany

ISBN: (纸本)9781424404810

Recent developments in the compression of dynamic meshes or mesh sequences have shown that the statistical dependencies within a mesh sequence can be exploited well by predictive coding approaches. Coders introduced so far use experimentally determined or heuristic thresholds for tuning the algorithms. In video coding rate-distortion (RD) optimization is often used to avoid fixing of thresholds and to select a coding mode. We applied these ideas and present here an RD-optimized mesh coder. It includes different prediction modes as well as an RD cost computation that controls the mode selection across all possible spatial partitions of a mesh to find the clustering structure together with the associated prediction modes. The structure of the RD-optimized D3DMC coder is presented, followed by comparative results with mesh sequences at different resolutions.

关键词： linear predictive coding multidimensional coding 3D compression rate-distortion optimization

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LOW-COMPLEXITY PARCOR COEFFICIENT QUANTIZER AND PREDICTION ORDER ESTIMATOR FOR LOSSLESS SPEECH coding

LOW-COMPLEXITY PARCOR COEFFICIENT QUANTIZER AND PREDICTION O...

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2010 IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Kamamoto, Yutaka Moriya, Takehiro Harada, Noboru NTT Communication Science Laboratories Nippon Telegraph and Telephone Corporation Japan

ISBN: (纸本)9781424442966

This paper describes two low-complexity tools used for the new ITU-T recommendation G.711.0, the lossless coding of G.711 (A-law/mu-law logarithmic PCM) speech data. One is an algorithm for quantizing the PARCOR/reflection coefficients and the other is an estimation method for the optimal prediction order. Both tools are based on a criterion that minimizes the entropy of the prediction residual signals and can be implemented in a fixed-point low-complexity algorithm. G.711.0 with the developed practical tools will be widely used everywhere because it can losslessly reduce the data rate of G.711, the prevailing speech-coding technology.

关键词： Speech coding Lossless compression Standardization linear predictive coding G.711

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Low-rate quantization of spectral information in a 4 kb/s pitch-synchronous CELP coder

Low-rate quantization of spectral information in a 4 kb/s pi...

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7th IEEE Workshop on Speech coding

作者： Guerchi, D Mermelstein, P Univ Quebec INRS Telecommun Montreal PQ M5A 1C6 Canada

ISBN: (纸本)0780364163

A new efficient algorithm for quantizing the spectral information for a Pitch-Synchronous CELP (PSCELP) speech coder is proposed. LPC analysis in the PSCELP is carried out once per pitch period. Direct quantization of the pitch synchronous LSF vectors would lead to a variable-rate codec, which is inconsistent with the objective of achieving a fixed-rate speech coder operating at 4 kb/s. Hence, a linear trajectory of LSF vectors is selected which can be encoded by one LSF vector each 20 ms. This conversion exploits the high correlation between successive pitch periods of the LSF parameters to achieve joint quantization. A coding rate of 1.2 kb/s is achieved for the LSF information with no noticeable degradation. The proposed algorithm employs linear interpolation at the decoder to recover the spectral parameters for the individual pitch periods used in the pitch-synchronous reconstruction of the speech signal. The comparison simulation results show that this algorithm produces comparable performance to that of LSF's linear interpolation quantization in a time-synchronous CE;LP coder.

关键词： Bit rate Code standards Decoding Degradation Interpolation linear predictive coding Quantization Speech analysis Speech codecs Vectors

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Dynamic Adaptation of the Jitter Buffer for Video Streaming Applications 2

Dynamic Adaptation of the Jitter Buffer for Video Streaming ...

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2nd IEEE Workshop on Advances in Information, Electronic and Electrical Engineering (AIEEE)

作者： Jaseviciute, Vita Plonis, Darius Serackis, Arturas Vilnius Gediminas Tech Univ Dept Elect Syst Naugarduko Str 41-413 LT-03227 Vilnius Lithuania

ISBN: (纸本)9781479971220

Paper presents a new algorithm proposed to dynamically adapt jitter buffer, used in video transcoding applications. When the video content provider is moving continuously, the wireless network connectivity conditions changes in time. The resulting jitter should be compensated in video transcoder by selecting jitter buffer high enough to compensate all possible jitter variations. The algorithm, proposed in this paper updates the jitter buffer according to current jitter measurements and jitter prediction results in order to reduce the total latency of the video transcoding in live transmissions.

关键词： Jitter 3G mobile communication linear predictive coding wireless networks

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OBJECTIVE QUALITY ASSESSMENT OF SPEECH ENHANCEMENT ALGORITHMS USING BOOTSTRAP-BASED MULTIPLE HYPOTHESES TESTS

OBJECTIVE QUALITY ASSESSMENT OF SPEECH ENHANCEMENT ALGORITHM...

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2010 IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Lu, Zhihua Heidenreich, Philipp Zoubir, Abdelhak M. Tech Univ Darmstadt Signal Proc Grp D-64283 Darmstadt Germany

ISBN: (纸本)9781424442966

In this paper bootstrap resampling techniques are applied to assess speech quality and thereby evaluate performance of distinct speech enhancement algorithms, under the assumption that the speech segments can be approximated by an autoregressive model. A bootstrap-based multiple hypotheses testing procedure is constructed to test a distance measure based on linear predictive coding, which is the log-likelihood ratio distance. It is shown that the multiple hypotheses test results correlate well with conventional numerical distance measures, which suggests the applicability of the proposed procedure in assessment of speech quality as well as speech enhancement algorithms.

关键词： Bootstrap speech signal processing linear predictive coding multiple hypothesis testing

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A Description of a Parametrically Controlled Modular Structure for Speech Processing

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IEEE Transactions on Acoustics, Speech, and Signal Processing 1975年第1期23卷 87-91页

作者： Dixon, N. Rex Silverman, Harvey F. Computer Sciences Department IBM Thomas J. Watson Research Center Yorktown Heights N.Y. 10598 United States

A system, the modular acoustic processor (MAP) consisting of two major components, has been designed for work in speech recognition. A versatile spectral analysis system, the parametrically controlled analyzer (PCA), serves as input to an hierarchically operated string transcriber (HOST). In the design of this system, controllability and modularity for developmental extensibility were primary concerns. The system, with the exception of initial high-fidelity, direct A/D conversion, is entirely implemented in software, PL/I, with appropriate JCL structures for running under OS/MVT on an IBM 360–91. As an adjunct for obtaining training data, a grayscale interactive system using an IBM 1800 process-control computer has also been implemented. PCA signal processing features parametric selection of several analysis methods, including discrete Fourier transform (DFT), linear predictive coding (LPC), and chirp, z-transform (CZT). Also, selection may be made among various smoothing, normalization, interpolation, and F0 estimation methods. PCA develops high-quality spectrographic representations of speech for standard line printers, CRT display, and subsequent processing. PCA also performs spectral-similarity matching and training. HOST consists of a number of processes for performing segmentation, classification, and prosody analysis. Provision is made for complete commutability at the module level as well as at the alCopyright © 1975 by The Institute of Electrical and Electronics Engineers, Inc.

关键词： Speech processing Principal component analysis Discrete Fourier transforms linear predictive coding Speech recognition Spectral analysis Control systems Control system analysis Controllability Operating systems

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Stimulus artifact cancellation in Click Evoked Otoacoustic Emissions using linear prediction

Stimulus artifact cancellation in Click Evoked Otoacoustic E...

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25th Annual International Conference of the IEEE-Engineering-in-Medicine-and-Biology-Society

作者： Tujal, PM Souza, MN Univ Fed Rio de Janeiro COPPE Biomed Engn Program BR-21945 Rio De Janeiro Brazil

ISBN: (纸本)0780377893

This paper describes a new method for extraction of Click Evoked Otoacoustic Emissions (CEOAE), where the stimulus artifact is eliminated by the use of linear predictive coding (LPC). In this method, the prediction coefficients are computed over the first samples of the click response, which is mainly formed by passive oscillations, and the unpredicted part of the remaining response is taken as the CEOAE signal. Preliminary tests were made with fifteen signals collected from normal hearing adults presenting stimulus artifacts in their responses. Results show the advantage of eliminating most of the stimulus artifact, while preserving a better signal-to-noise ratio than the standard nonlinear stimulus cancellation method.

关键词： otoacoustic emissions click evoked otoacoustic emissions linear predictive coding

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Improved spectral tracking using interpolated linear prediction parameters

Improved spectral tracking using interpolated linear predict...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Pereira, W Kabal, P McGill Univ Dept Elect & Comp Engn Montreal PQ H3A 2T5 Canada

ISBN: (纸本)0780374029

Conventional interpolation of linear Prediction (LP) parameters can provide a poor spectral match to the underlying speech signal for the intermediate subframes, In this paper, we present a method of modifying the interpolation endpoint LP parameters to improve the spectral tracking over all subframes. This 'warping' algorithm is based on minimizing the distortion between the interpolated LP parameters and those computed directly from the speech signal. The algorithm has been integrated into the Adaptive Multi Rate (AMR) speech codec. Our results show that this method enhances coder performance by smoothing out the LP parameter tracks and reducing coding distortion.

关键词： spectral analysis interpolation linear predictive coding speech coding tracking pattern matching speech codecs speech enhancement spectral tracking linear prediction parameters spectral match speech signal intermediate subframes interpolation endpoint parameters warping algorithm distortion minimization Adaptive Multi Rate speech codec AMR speech codec coder performance enhancement smoothing coding distortion reduction

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Improving the Performance of Zero-Resource Children's ASR System through Formant and Duration Modification based Data Augmentation 14

Improving the Performance of Zero-Resource Children's ASR Sy...

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14th IEEE International Conference on Signal Processing and Communications (SPCOM)

作者： Shahnawazuddin, Syed Kumar, Vinit Kumar, Avinash Ahmad, Waquar NIT Patna Dept ECE Patna Bihar India NIT Sikkim Dept ECE Sikkim India NIT Calicut Dept ECE Kozhikode India

ISBN: (数字)9781665482509

ISBN: (纸本)9781665482509

Developing an automatic speech recognition (ASR) system for children's speech is extremely challenging due to the unavailability of data from the child domain for the majority of the languages. Consequently, in such zero-resource scenarios, we are forced to develop an ASR system using adults' speech for transcribing data from child speakers. However, differences in formant frequencies and speaking-rate between the two groups of speakers degrade recognition performance. To reduce the said mismatch, out-of-domain data augmentation approaches based on formant and duration modification are proposed in this work. For that purpose, formant frequencies of adults' speech training data are up-scaled using warping of linear predictive coding coefficients. Next, the speaking-rate of adults' data is also increased through time-scale modification. Due to simultaneous altering of formant frequencies and duration of adults' speech and then pooling the modified data into training, the acoustic mismatch due to the aforementioned factors gets reduced. This, in turn, enhances the recognition performance significantly. Additional improvement is obtained by combining the recently reported voice-conversion-based data augmentation technique with the proposed ones. On combining the proposed and voice-conversion-based data augmentation techniques, a relative reduction of nearly 32:3% in word error rate over the baseline is obtained.

关键词： Training Speech coding Error analysis Conferences Training data Signal processing linear predictive coding

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A real-time 64-monosyllable recognition LSI with learning mechanism

A real-time 64-monosyllable recognition LSI with learning me...

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Asia and South Pacific Design Automation Conference (ASP-DAC 2001)

作者： Nakamura, K Zhu, Q Maruoka, S Horiyama, T Kimura, S Watanabe, K Nara Inst Sci & Technol Grad Sch Informat Sci Nara 6300101 Japan

ISBN: (纸本)0780366336

In the paper, a real-time 64-mono-syllable recognition LSI is presented. The LSI accepts 11.6 msec speech frame and outputs a 6-bit symbol-code for each frame by the end of the next frame with the pipelining manner. The recognition method is based on the Hidden Markov Model and is speaker-independent. An on-chip learning mechanism has also been designed, but the circuit is off-chip at present implementation because of the restriction of LSI area, The LSI is fablicated by VDEC Rohm with 0.6 mum process on a 4.5 mm x 4.5 mm chip.

关键词： Adders Circuits Hardware Hidden Markov models Large scale integration Learning systems linear predictive coding Speech analysis Speech coding Speech recognition

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