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PARAMETER-BASED SPEECH QUALITY MEASURES FOR GSM

PARAMETER-BASED SPEECH QUALITY MEASURES FOR GSM

引用

IEEE第14届国际个人无线移动通信大会

作者： Marc Werner Karsten Kamps Ulrich Tuisel John G. Beerends Peter Vary Institute of Communication Systems and Data Processing () Aachen University Germany Institute of Communication Systems and Data Processing () Aachen University Germany E-Plus Mobilfunk GmbH Düsseldorf Germany TNO Telecom The Netherlands Institute of Communication Systems and Data Processing () Aachen University Germany

This contribution introduces instrumental speech quality measures for the GSM system based only on transmis- sion parameters. Certain combinations of GSM parameters which quantify the transmission

关键词： PARAMETER-BASED SPEECH QUALITY MEASURES FOR GSM

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Parameter-based speech quality measures for GSM

Parameter-based speech quality measures for GSM

引用

IEEE International Symposium on Personal, Indoor and Mobile Radio communications (PIMRC)

作者： M. Werner K. Kamps U. Tuisel J.G. Beerends P. Vary Institute of Communication System and Data Processing RWTH Aachen University of Technology Germany E-Plus Mobilfunk GmbH and Company KG Dusseldorf Germany TNO Telecom Netherlands

This contribution introduces instrumental speech quality measures for the GSM system based only on transmission parameters. Certain combinations of GSM parameters which quantify the transmission quality in terms of bit error rate, received power level, etc., were shown be suitable for the prediction of the resulting speech quality. Neither the original nor the received speech signal is needed for this kind of prediction. The proposed speech quality measures have been validated by extensive link-level simulations which are based on measurements of transmission parameters collected in a GSM-1800 network. Speech samples were produced by bit-exact transmission simulations using the measured link parameters for channel modelling. The reference speech quality assessments of these samples were carried out with the PESQ algorithm Ref.3. The correlation of the presented parameter measures with the intrusive PESQ measure is remarkable.

关键词： GSM Speech analysis Distortion measurement Instruments Bit error rate Mobile communication Testing Radio link Land mobile radio data processing

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Pyramid CELP: Embedded speech coding for packet communications

Pyramid CELP: Embedded speech coding for packet communicatio...

引用

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

作者： Christoph Erdmann David Bauer Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

In this paper we present a novel speech coding algorithm for transmission over packet networks based on CELP (Code-Excited-Linear-Prediction). While various CELP type speech coders like the ITU-T G.729 or the Adaptive Multi-Rate (AMR) codec are applicable to packet voice communication systems such as the Internet, there are some fundamental reasons that may limit their Performance in case of packet loss. We investigate the alternative of embedded coding in the CELP framework. focussing on an efficient quantization of the excitation signal to generate a hierarchically-structured bit stream by means of pyramid coding [1]. Thus, the receiver can reconstruct speech at a basic quality by decoding only a subset of the entire bit stream. The quality of the decoded speech increases with the amount of received bits. We demonstrate the performance by an experimental hierarchical wideband (0.05–7 kHz) speech coder, applied to an IP (Internet Protocol) channel simulation.

关键词： Decoding Image reconstruction Synchronization IP networks Image resolution Encoding Speech

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Unbiased residual echo power estimation for hands-free telephony

Unbiased residual echo power estimation for hands-free telep...

引用

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

作者： Gerald Enzner Rainer Martin Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

Residual echo arises in hands-free telephony equipment due to insufficient echo canceler convergence, but can be suppressed using a postfilter. The most important control parameter for postfilter adaptation is therefore the residual echo power spectral density (PSD). In this contribution we present and compare residual echo PSD estimation techniques. We introduce a new partitioned block-adaptive estimator delivering unbiased residual echo PSD estimates in strongly reverberant and noisy acoustic environments.

关键词： Correlation Lead Estimation Telephony Loudspeakers Noise Discrete Fourier transforms

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Speech enhancement using MMSE short time spectral estimation with gamma distributed speech priors

Speech enhancement using MMSE short time spectral estimation...

引用

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

作者： Rainer Martin Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

In this paper we consider optimal estimators for speech enhancement in the Discrete Fourier Transform (DFT) domain. We present an analytical solution for estimating complex DFT coefficients in the MMSE sense when the clean speech DFT coefficients are Gamma distributed and the DFT coefficients of the noise are Gaussian or Laplace distributed. Compared to the state-of-the-art Wiener or MMSE short time amplitude estimators the new estimators deliver improved signal-to-noise ratios. When the noise model is a Laplacian density the enhanced speech shows less annoying random fluctuations in the residual noise than for a Gaussian density.

关键词： Discrete Fourier transforms Speech Signal to noise ratio Time frequency analysis Lead Speech enhancement

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An upper bound on the quality of artificial bandwidth extension of narrowband speech signals

An upper bound on the quality of artificial bandwidth extens...

引用

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

作者： Peter Jax Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

The aim of the artificial bandwidth extension (BWE) of speech signals is to recover wideband speech from bandlimited speech. As the BWE algorithm is supposed to operate without additional side information on the original wideband speech, it has to exploit mutual dependencies between the available and missing frequency bands of the speech signal. In this paper the BWE is examined from an information theoretic perspective. After defining a performance measure, and introducing a few assumptions on a generalized BWE algorithm, a general relationship between mutual information and the maximum achievable estimation performance is formulated, which ensues an upper bound on the performance of BWE algorithms. Finally, some measurements considering a representative BWE scenario are presented.

关键词： Wideband Speech Estimation

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Capacity improvement in UMTS by dedicated radio resource management

Capacity improvement in UMTS by dedicated radio resource man...

引用

IEEE Conference on Vehicular Technology (VTC)

作者： T. Bruggen M. Werner Y. Vasseur J. Trenzinger P. Vary Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Germany

The IMT-2000 standard UMTS (universal mobile telecommunications system) is based on wideband CDMA (code division multiple access). The service quality in terms of coverage, data rates, bit error rates, etc., and the system capacity depend on the radio resource management (RRM), i.e., the service-specific dynamic allocation of transmit powers, spreading factors, and codes to the individual users. In addition, macrodiversity allows "soft handoff" and load distribution between adjacent cells. Two new dedicated downlink RRM functions are presented to increase the average signal-to-interference power ratio (SIR) and the user capacity, taking into consideration the assignment of scrambling codes (SC) and the power allocation in case of macrodiversity. The presented scrambling code allocation method is shown to enhance the cell capacity in a UMTS-FDD system. The improvement is derived analytically and confirmed by statistical system simulations. For macrodiversity, a combination of power partitioning and pilot power control can increase the capacity of a single cell. The size of macrodiversity areas can be adjusted to a certain extent by these mechanisms. Thus, the power resources of less loaded cells can be exploited more efficiently.

关键词： 3G mobile communication Resource management Multiaccess communication Code standards Telecommunication standards Wideband Bit error rate Downlink Analytical models Power control

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On joint source-channel decoding for correlated sources

On joint source-channel decoding for correlated sources

引用

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

作者： M. Adrat R. Haenel P. Vary Institte of Communication Systems and Data Processing RWTH Aachen University of Technology Germany

Whenever digital communications is subject to constraints in delay and complexity, some residual redundancy remains after source coding. Shannon's separation theorem does not hold strictly and hence joint source-channel decoding is beneficial. In this paper we discuss two competing approaches exploiting residual redundancy in terms of (auto)-correlation. Both will be analyzed with respect to their impact on quantization noise and channel distortion. The first approach exploits redundancy at the receiver for minimum mean squared error estimation of source codec parameters. The second approach exploits correlation at the transmitter by predictive encoding. The prediction gain allows to reduce quantization noise or bitrate. The spare bitrate can be used for specific error protection and hence reduction of channel distortion. But otherwise, less redundancy is utilizable at the receiver for parameter estimation. Simulations will show that the predictive encoding scheme improves quality in a wide range of channel conditions, but in contrast to Shannon's separation theorem the utilization of correlation at the receiver side is beneficial in very bad channel conditions.

关键词： Signal to noise ratio Decoding Noise measurement

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Embedded speech coding based on pyramid CELP

Embedded speech coding based on pyramid CELP

引用

IEEE Workshop on Speech Coding Proceedings

作者： C. Erdmann P. Vary Institute of Communication Systems and Data Processing Aachen University of Technology Aachen Germany

In this paper we investigate embedded coding of speech in the CELP (code-excited-linear-prediction) framework. Compared to other known approaches of variable bit rate speech coding, such as the adaptive multi-rate (AMR) codec, embedded coding systems allow bit rate reductions at any point along the communication network, without changes of the encoder and the decoder. Thus, the quality of the decoded speech increases with the amount of received bits. Aiming at a coding scheme, that produces such a hierarchically-structured bit stream, we focus on a decomposition of the excitation signal by means of pyramid coding. To achieve reasonable speech compression, analysis-by-synthesis (AbS) based quantization of the pyramid layers is designed in a CELP-type fashion, called P-CELP (pyramid CELP). Besides an efficient design of fixed algebraic codebooks in each pyramid layer, special attention is payed to the integration of an adaptive codebook. To achieve maximum performance, the proposed coding scheme is used for wideband (50 Hz-7 kHz) speech.

关键词： Speech coding Image coding Decoding Gaussian processes Bit rate data processing Speech processing Codecs Quantization Wideband

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A novel voiced / unvoiced / silence classification scheme for offline speech coding

A novel voiced / unvoiced / silence classification scheme fo...

引用

European Signal processing Conference (EUSIPCO)

作者： C. Hoelper A. Frankort C. Erdmann P. Vary Rheinisch-Westfalische Technische Hochschule Aachen Aachen Nordrhein-Westfalen DE Institute of Communication Systems and Data Processing (iivd) Aachen University of Technology

For use in offline speech processing systems a novel algorithm has been developed, that classifies clean speech segments robustly as voiced, unvoiced, or silence respectively. This decision is needed e.g. in source controlled speech coders which treat voiced, unvoiced, and silent segments differently, to increase the coding efficiency. The classifier is based on a combination of several features, extracted from the speech signal in the time domain. Besides the Energy, a novel measure, representing the unsteadiness of the speech signal, is proposed. Non-realtime coding allowes iterative refinement of the classification, taking the cepstral distance into account. The new classification scheme has been tested with the popular AMR codec.

关键词： Ink Abstracts Codecs

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