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检索条件"主题词=Linear predictive coding"

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A 4 kbit/s renewal code excited linear prediction speech coder

A 4 kbit/s renewal code excited linear prediction speech cod...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Hong Kook Kim Yong Duk Cho Moo Young Kim Sang Ryong Kim Human & Computer Interaction Laboratory Information Processing Sector Samsung Advanced Institute of Technology Yongin si Gyeonggi South Korea

ISBN: (纸本)0818679190

This paper proposes a new 4 kbit/s speech coder based on CELP structure with 45 ms total codec delay. The coder is mainly featured by the renewal codebook of the excitation signal and the linked split-vector quantizer of line spectrum pair parameters which enable the coder to get high quality speech at low bit rate. In addition, techniques of formant enhancement in the spectral envelop and harmonic recovery in the transient region are also introduced to reduce buzzy and hoarse sounds, respectively. From the intensive listening test with intermediated response system (IRS) speech, we obtained a comparable subjective quality to 32 kbit/s ADPCM (ITU Recommendation G.726) under a nominal speech input level of -26 dB overload.

关键词： Speech coding linear predictive coding Filters Bit rate Phase change materials Quantization Smoothing methods Humans Information processing Speech codecs

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A high quality BI-CELP speech coder at 8 kbit/s and below

A high quality BI-CELP speech coder at 8 kbit/s and below

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： S.Y. Kwon Hochong Park Hyokang Chang Telogy Networks Inc. Germantown MD USA Samsung Electronics Company Limited Seoul South Korea Combasis Technology Inc. Rockville MD USA

This paper describes "BI-CELP: baseline and implied CELP", which is a high quality speech coding method based on a code excited linear prediction (CELP) model employing excitation vectors combined from two codebooks, one from the baseline codebook and the other from the implied codebook. In this method the index of the baseline codebook is coded and transmitted to the receiver while the index of the implied codebook is extracted from the synthesized speech. This method has been applied to a lower rate voice coder at 8 kbit/s to produce high quality voice comparable to that of the 16 kbit/s G.728 LD-CELP. The performance of the 8 kbit/s BI-CELP coder is measured in terms of SNRseg and MOS. The average SNRseg is 12.14 dB which is 0.6 dB higher than that of the 8 kbit/s G.729 CS-ACELP. The MOS for the quiet input is 3.8 which is 0.02 higher than that of G.729 CS-ACELP. BI-CELP algorithm is implemented in real-time on a single TMS320C31 with 27 MIPS of CPU.

关键词： Speech synthesis Speech coding Vectors linear predictive coding Speech enhancement Delay Decoding predictive models Character generation Pulse generation

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Spectral envelope quantization with noise robustness

Spectral envelope quantization with noise robustness

引用

IEEE Workshop on Speech coding For Telecommunications

作者： Moo Young Kim Hong Kook Kim Yong Duk Cho Sang Ryong Kim Human & Computer Interaction Laboratory Samsung Advanced Institute of Technology Yongin si Kyunggi South Korea

ISBN: (纸本)0780340736

To represent the spectral envelope, the LSF (line spectrum frequencies) can be used for its good properties in quantization. The interframe correlation of the LSF improves the performance of static quantization, but it is sensitive to channel errors. The proposed quantizer, noise robust-switched prediction-linked split vector quantizer (NR-SP-LSVQ), combines switched prediction with three static and a predictive quantizers. It limits the propagation of channel errors in a few frames by switching to the static ones. Moreover, each of the static ones is designed for the robustness against background noise, while the predictive one is for the steady region of speech. It increases the performance in noise corrupted as well as clean speech.

关键词： Quantization Noise robustness Background noise Speech enhancement linear predictive coding predictive coding Humans Cities and towns Frequency predictive models

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A new sinusoidal phase modeling algorithm

A new sinusoidal phase modeling algorithm

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： S. Ahmadi A.S. Spanias Department of Electrical Engineering Telecommunications Research Center Arizona State University Tempe AZ USA

A new phase modeling algorithm for sinusoidal analysis and synthesis of speech signals is presented. Short-time sinusoidal phases are efficiently approximated by incorporating linear prediction, spectral sampling, delay compensation, and phase correction techniques. The algorithm is different than phase compensation methods proposed for multi-pulse LPC in that it has been tailored to sinusoidal transform coding of speech signals. Performance analysis on a large speech database indicates considerable improvement in temporal and spectral matching between the original and reconstructed signals as compared to other sinusoidal phase models as well as improved subjective quality of the reproduced speech.

关键词： Speech analysis Speech synthesis Signal analysis Algorithm design and analysis Signal synthesis linear approximation Sampling methods Delay linear predictive coding Transform coding

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A variable bit rate operation of LD-CELP mainly for VBD applications in DCME

A variable bit rate operation of LD-CELP mainly for VBD appl...

引用

IEEE Workshop on Speech coding For Telecommunications

作者： I. Shapira T. Simchony S. Sourani J. Piasetzky ECI Telecom Ltd. Israel ECI Telecom Limited Israel

This paper describes the candidate algorithm for Annex J of ITU-T Rec. G.728. The algorithm achieves compression and near transparent transmission of VBD signals such as V.34 @ 19200 bps (and up to 21600 bps under moderate analog impairments conditions) using a 40 kbit/s transmission rate. Additionally the algorithm provides toll speech quality and smooth mode-switching (speech-to-data/data-to-speech) during voice and signaling signals. The main novelty of the algorithm is the application of trellis coded quantization (TCQ) approach to codebook search that results in sufficient quantization gain, lower complexity (than LD-CELP), and maintains the same algorithmic delay of 5 samples (0.625 msec).

关键词： Bit rate Filters linear predictive coding Decoding Equations Convolutional codes Convolution Switches Viterbi algorithm Speech

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An audio coding system using subband backward linear prediction

An audio coding system using subband backward linear predict...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Lin Yin Speech and Audio Systems Laboratory Nokia Research Center Tampere Finland

In this paper, block backward adaptive linear predictors are used for improving the performance of perceptual audio codecs. Based on the investigation on different linear prediction algorithms, a signal dependent adaptive-switched predictor is developed. This predictor not only delivers significant coding gain for stationary signals but also recovers quickly from transient signals. At a bitrate of 64 kbit/s, the performance of the new codec for most critical test sequences is significantly better than MPEG-1 Layer II.

关键词： Audio coding Codecs Bit rate Testing linear predictive coding Transform coding Adaptive algorithm Autocorrelation Quantization Streaming media

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A current-based model for the MOS transistor

A current-based model for the MOS transistor

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： A.I.A. Cunha O.C. Gouveia-Filho M.C. Schneider C. Galup-Montoro Laboratório deInstrumentacçaão Eletroônica (LINSE). Deprtamento de Engenharia Eleétrica Universidade Federal de Santa Catarina Florianopolis Santa Catarina Brazil Laboratoire dePhysique desComposants aà Semiconducteurs (LPCS-INPG) Ecole Nationale Superieure d''Electronique et Radioelectriciteé de Grenoble Grenoble France

This paper presents a physics-based model for the MOS transistor, suitable for circuit design and simulation and valid from weak to strong inversion. Each static or dynamic characteristic is accurately described by a ... 详细信息

关键词： Voltage Circuit synthesis Analog integrated circuits Integrated circuit modeling MOSFET circuits Instruments Electronic mail linear predictive coding Parameter extraction Capacitance

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Speaker identification based on a modified Kohonen network

Speaker identification based on a modified Kohonen network

引用

International Conference on Neural Networks

作者： K. Vieira B. Wilamowski R. Kubichek Department of Electrical Engineering University of Wurzburg Laramie WY USA

A human information processing system is composed of neurons switching at speeds about a million times slower than computer gates. Yet humans are more efficient than computers at computing complex tasks such as speech and visual interpretation. A neural network (NN) method was developed to reproduce one of the abilities and power of the human brain: speaker recognition. To realize this method, the input patterns used for this network were the reflection coefficients (k/sub p/) of the speech signal. The speech was analyzed using an autoregressive LPC technique and the k/sub p/s were found by applying the Levinson recursion algorithm. Next, a simple transformation of these input patterns onto a hypersphere in augmented space was made using a multilayer perceptron (MLP) neural model. The Kohonen approach is commonly used for computing a distance in multidimensional input space so that the input patterns are projected on a hypersphere with unit radius. Although this technique is very efficient for clustering of patterns, it has one significant drawback. By normalizing the input pattern, the information about its magnitude is lost. The proposed modified network has a relatively simple architecture but is shown to be very effective in performing speaker recognition.

关键词： Humans Speaker recognition Speech analysis Information processing Neurons Neural networks Reflection Algorithm design and analysis linear predictive coding Multilayer perceptrons

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On the fixed-point error analysis and VLSI architecture for FS1016 CELP decoder

On the fixed-point error analysis and VLSI architecture for ...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： An-Nan Suen Jhing-Fa Wang Horng-Jei Chang Institute of Information Engineering National Cheng Kung University Tainan Taiwan

In this paper, the fixed-point accuracy analysis and VLSI architecture of FS1O16 CELP decoder are presented. The code excited linear predictive (CELP) coder is the most effective technique among various linear predictive coding methods for speech compression. Hence to design a low cost and low power CELP decoder chip for the portable systems and wireless digital communication environment becomes increasingly important. The decoder VLSI architecture can achieve (1) excellent accuracy results due to the accuracy studies for the finite word length, (2) power saving and high speed operations resulting from the combined advantages of pipeline, current processing for LSE's interpolating and cosine operation, (3) table size reducing by applying the memoryless realization for stochastic codebook and partial sums technique, and (4) specification satisfying the FS1016 CELP coder.

关键词： Error analysis Very large scale integration Decoding linear predictive coding Speech coding Costs Wireless communication Digital communication Pipelines Stochastic processes

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Underwater acoustic voice communications using digital pulse position modulation

Underwater acoustic voice communications using digital pulse...

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OCEANS

作者： H. Sari B. Woodward Department of Electronic and Electrical Engineering Loughborough University UK

The paper presents a novel technique for underwater acoustic voice communications. The speech signal is compressed prior to transmission by using linear predictive coding and transmission of appropriate speech parameters is achieved by digital pulse position modulation. The main emphasis is on the reception of the multipath-dominant signal and on the demodulation process.

关键词： Underwater acoustics Underwater communication Acoustic pulses Speech coding Pulse modulation linear predictive coding Pulse compression methods Digital modulation Modulation coding Signal processing

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