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检索条件"主题词=Linear Predictive coding"

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Efficient nonlinear prediction in ADPCM

Efficient nonlinear prediction in ADPCM

引用

IEEE International Conference on Electronics, Circuits and Systems (ICECS)

作者： M. Faundez-Zanuy F. Vallverdu E. Monte Escola Univ. Politecnica de Mataro Barcelona Spain Signal Theory & Communications Department UPC Barcelona Spain

In the last years there has been a growing interest for nonlinear speech models. Several works have been published revealing the better performance of nonlinear techniques, but little attention has been dedicated to the implementation of the nonlinear model into real applications. This work is focused on the study of the behaviour of a nonlinear predictive model based on neural nets, in a speech waveform coder. Our novel scheme obtains an improvement in SEGSNR between 1 and 2 dB for an adaptive quantization ranging from 2 to 5 bits.

关键词： Neurons Neural networks Databases Quantization linear predictive coding Speech synthesis Bit rate Multilayer perceptrons Nonlinear filters Speech analysis

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GSM EFR based multi-rate codec family

GSM EFR based multi-rate codec family

引用

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

作者： J. Vainio H. Mikkola K. Jarvinen P. Haavisto Nokia Research Center Tampere Finland

This paper describes a multi-rate codec family developed as a potential candidate for the GSM adaptive multi-rate (AMR) codec standard. The codec family consists of the GSM enhanced full rate (EFR) codec and lower bit-rate extensions thereof. The codec family consists of several codecs, i.e., modes that have different bit-rate partitionings between source coding and error protection. All the source codecs use the same ACELP-method (algebraic code excited linear predictive coding) used also in the GSM EFR codec. The codec operates at gross bit-rates of 22.8 kbit/s in the GSM full rate (FR) channel and 11.4 kbit/s in the GSM half rate (HR) channel. In the full rate channel, the codec provides improved error robustness over the GSM enhanced full rate (EFR) codec. It extends wireline quality (equal to or better than G.726-32 ADPCM) to poor channel error conditions with low C/I-ratios of 7 dB or even below. When operated in the half rate channel, the codec provides improved channel capacity while still providing wireline quality at high C/I-ratios above 16-19 dB.

关键词： GSM Speech codecs Robustness Source coding Protection Telecommunication standards Code standards Standards development Channel capacity linear predictive coding

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Thorough evaluation of Hot-Carrier Induced Degradation in Deep Submicron Thin-film UNIBOND and SIMOX N-MOSFETs

Thorough evaluation of Hot-Carrier Induced Degradation in De...

引用

European Conference on Solid-State Device Research (ESSDERC)

作者： S.H. Renn J. Jomaah F. Balestra C. Raynaud LPCS (UMR CNRS & INPG) ENSERG Grenoble France LETI (CEA) Grenoble France

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The third-order cumulant of speech signals with application to reliable pitch estimation

The third-order cumulant of speech signals with application ...

引用

IEEE Workshop on Statistical Signal and Array Processing

作者： E. Nemer R. Goubran S. Mahmoud Nortel Limited Verdun QUE Canada Systems & Computer Eng'g Carleton University Ottawa ONT Canada

This paper provides a formal framework for using the third-order statistics (TOS) of speech signals and presents a new method for estimating the pitch and making voicing decision using the 3rd-order cumulant of the LPC residual. Analytical expressions for the horizontal slice of the 3rd-order cumulant as well as the kurtosis of voiced speech are derived using the McAulay sinusoidal model (McAulay et al., 1986). The derivations demonstrate that the skewness of voiced speech is sufficiently distinct from that of Gaussian noise and can be used to aid in detecting voicing. It is also shown that the 3rd-order cumulant slice has distinct characteristics in terms of periodicity, phase and harmonic content and is a reliable candidate for estimating the pitch. Actual speech data is used to verify the derivations and experimental results using Gaussian and street noise are used to demonstrate the performance in noisy conditions.

关键词： Speech analysis linear predictive coding Gaussian noise Gaussian processes Autocorrelation Speech enhancement Phase estimation Business Higher order statistics Signal processing

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Removal of sparse-excitation artifacts in CELP

Removal of sparse-excitation artifacts in CELP

引用

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

作者： R. Hagen E. Ekudden B. Johansson W.B. Kleijn Speech Coding Research Ericsson Radio Systems AB Stockholm Sweden

In CELP, the use of codebooks with entries with only a few non-zero samples provides high speech quality and facilitates fast computation. With decreasing bit-rate, the intervals between the pulses increase, and the quality of the reconstructed signal begins to suffer from a particular type of artifact, which is strongest for noise-like segments. In this paper we describe experiments which show that the perceived artifacts are mainly concentrated at frequencies above 3 kHz, and this is consistent with our understanding of auditory theory. Our analysis leads to simple strategies to eliminate the artifacts, even at lower bit rates. We describe both a non-adaptive and an adaptive post-processing method to remove the artifacts. The methods are demonstrated to be efficient when used in the ACELP algorithm. A closed-loop method for ACELP is also described.

关键词： Speech coding Bit rate Background noise Speech enhancement Frequency Speech recognition Computational complexity GSM linear predictive coding Auditory system

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Recognition of vowels using glides based on a nonlinear transformation

Recognition of vowels using glides based on a nonlinear tran...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： J. Toyama M. Shimbo Grad. Sch. of Eng. Hokkaido Univ. Sapporo Japan Graduate School of Engineering Hokkaido University Japan

We believe that the glide part of a vowel sequence includes the phonetical quality of the previous and following vowels. Therefore, the positive use of glide parts may be effective for developing a speech recognition system. The trajectories of a glide in a feature space for a speech recognition system have the form of a curve. In this paper, we propose the transformation of trajectories of glides into a straight line in a low-dimensional space, and we show the effectiveness of our new recognition method using nonlinear transformation of a vowel sequence.

关键词： Speech recognition Humans Hidden Markov models linear predictive coding Error correction Gaussian distribution Extraterrestrial measurements Cepstrum Vectors Cepstral analysis

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Hybrid LPC and discrete wavelet transform audio coding with a novel bit allocation algorithm

Hybrid LPC and discrete wavelet transform audio coding with ...

引用

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

作者： S.D. Boland M. Deriche Signal Processing Research Centre School of Electrical and Electronic Systems Engineering Queensland University of Technology Brisbane QLD Australia

This paper examines a new method for coding high quality digital audio signals based on a combination of linear predictive coding (LPC) and the discrete wavelet transform (DWT). In this method, a linear predictor is first used to model each audio frame. Then, the prediction error is analyzed using the DWT. The LPC coefficients and DWT coefficients are quantized using a novel bit allocation scheme which minimizes the overall quantization error with respect to the masking threshold. The proposed coder is capable of delivering near-transparent audio signal quality at encoding bit rates of around 90-96 kb/s. Objective and subjective results suggest that the proposed coder operating at 90-96 kb/s has a performance comparable to that of the MPEG layer II codec operating at 128 kb/s.

关键词： Discrete wavelet transforms linear predictive coding Audio coding Bit rate predictive models Error analysis Quantization Masking threshold Encoding Codecs

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Using the time frequency structure of pitch periods to improve speaker verification systems

Using the time frequency structure of pitch periods to impro...

引用

IEEE-SP International Symposium on Time-Frequency and Time-Scale Analysis

作者： R.M. Nickel W.J. Williams Department of Electrical Engineering and Computer Science University of Michigan Ann Arbor MI USA

Commonly used robust speaker verification systems are based on time-varying autoregressive spectral estimation (AR) combined with hidden Markov modeling (HMM) or dynamic time warping (DTW). An exhaustive optimization of these methods in the past has culminated in quite reliable verification schemes. It seems unlikely, though, that further significant improvements are readily obtained along the same path. Unlike time-varying AR-modeling, which focuses on the the global spectral structure of an utterance, we are introducing a new method that focuses on the local time-varying spectral structure of individual pitch periods. Additionally, a pattern classification method using singular value decomposition (SVD) is employed. The new method by itself does not deliver better results than commonly used global methods; however, it is shown that an acceptance/rejection decision derived from both global and local analysis greatly improves the performance of the verification system.

关键词： Time frequency analysis Hidden Markov models Cepstral analysis linear predictive coding Nickel Robustness Time varying systems Optimization methods Pattern classification Performance analysis

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Noise reduction by paired-microphones using spectral subtraction

Noise reduction by paired-microphones using spectral subtrac...

引用

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

作者： M. Mizumachi M. Akagi School of Information Science Japan Advanced Institute of Science and Technology Nomi gun Ishikawa Japan

This paper proposes a method of noise reduction by paired microphones as a front-end processor for speech recognition systems. This method estimates noise using a subtractive microphone array and subtracts them from the noisy speech signal using spectral subtraction (SS). Since this method can estimate noise analytically and frame by frame, it is easy to estimate noise not depending on these acoustic properties. Therefore, this method can also reduce non-stationary noise, for example sudden noise when a door has just closed, which cannot be reduced by other SS methods. The results of computer simulations and experiments in a real environment show that this method can reduce LPC log spectral envelope distortions.

关键词： Noise reduction Microphone arrays Working environment noise Acoustic noise Speech recognition Adaptive arrays Microwave integrated circuits Speech enhancement Computer simulation linear predictive coding

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Harmonic peaks method for voice separation

Harmonic peaks method for voice separation

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： B. Zhang J3 Department of Electronic Engineering City University of Hong Kong Kowloon Hong Kong China

This paper addresses the problem of recognizing a target voice when it is corrupted by a co-channel interfering voice. First, the F0 contour of the target voice is robustly extracted by using the revised highest likely common fundamental algorithm proposed by Screenivas. By using this contour, the harmonic peaks of the target voice are extracted. The harmonic peaks carry the information of the formants of a vowel and can be used as a front-end feature in a speech recognizer. Moreover, the harmonic peaks of the target voice are changed little even in the presence of an interfering voice. A recognizer aimed at recognizing the Mandarin finals was developed, based on the harmonic peaks method as well as the conventional LPC cepstral coefficients method. By comparing the results of these two methods, the harmonic peaks method shows better performance in the presence of a co-channel interfering voice.

关键词： linear predictive coding Maximum likelihood estimation Algorithm design and analysis Speech recognition Cepstral analysis Robustness Bismuth Background noise Speech analysis Target tracking

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