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Joint optimization of multi-rate LDPC code ensembles for the AWGN channel based on shortening and puncturing

Joint optimization of multi-rate LDPC code ensembles for the...

引用

IEEE Conference on Wireless communications and Networking

作者： Moritz Beermann Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University Germany

In most communication systems, being able to adapt the error protection strength of the physical layer is essential to ensure the functionality of the system in potentially strongly varying conditions. Traditional forward error correction code design focuses on the optimization of codes with a fixed code rate for a worst-case channel condition. So-called rate-compatible codes, on the other hand, are tailored to support multiple code rates. In this paper we present an optimization strategy for a rate-compatible system based on shortening and puncturing of Low-Density Parity-Check (LDPC) codes. While the technique itself is well known, we propose a novel joint optimization of the resulting multi-rate LDPC code ensemble at all rates. This joint optimization achieves close to capacity decoding thresholds over an arbitrarily wide range of target code rates using only a single encoder and decoder implementation.

关键词： Decoding Parity check codes Joints Polynomials Encoding Cost function

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Near End Listening Enhancement: Speech Intelligibility Improvement in Noisy Environments

Near End Listening Enhancement: Speech Intelligibility Impro...

引用

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

作者： B. Sauert P. Vary Institute of Communication Systems and Data Processing RWTH Aachen University Germany

In contrast to common noise reduction systems, this contribution presents a digital signal processing algorithm to improve intelligibility of clean far end speech for the near end listener who is located in an environment with background noise. Since the noise reaches the ears of the near end listener directly and therefore can hardly be influenced, a sensible option is to manipulate the far end speech. The proposed algorithm raises the average speech spectrum over the average noise spectrum and takes precautions to prevent hearing damage. Informal listening tests and the speech intelligibility index indicate an improved speech intelligibility

关键词： Speech enhancement Working environment noise Signal processing algorithms Noise reduction Digital signal processing Speech processing Background noise Ear Auditory system Testing

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A differential microphone array with input level alignment, directional equalization and fast notch adaptation for handsfree communication 11

A differential microphone array with input level alignment, ...

引用

11th ITG Symposium on Speech communication

作者： Geiser, Bernd Krüger, Hauke Vary, Peter Wiese, Detlef Institute of Communication Systems and Data Processing RWTH Aachen University Aachen52056 Germany Javox Solutions GmbH Gallierstraße 33 Aachen52074 Germany Binauric SE Am Söldnermoos 17 Hallbergmoos85399 Germany

ISBN: (纸本)9783800736409

We revisit the standard differential microphone of [1] and propose three algorithmic enhancements. First, a simple alignment procedure equalizes the power levels of the input signals. Second, we describe an efficient algorithm to equalize the distorted directional response of the original algorithm. This enables a wider operating frequency range and helps to reduce the noise gain of the array. Third, a fast time-domain notch adaptation algorithm is presented to reliably track interfering sound sources. Its tracking performance is not impaired by the activity of desired sound sources. Finally, we describe a fixed-point arithmetic implementation of the modified algorithm for wideband handsfree communication (fs =16kHz) using two miniature digital MEMS microphone capsules. © VDE VERLAG GMBH Berlin Offenbach.

关键词： Fixed point arithmetic

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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...

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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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Automatic speech segmentation using neural network and phonetic transcription

Automatic speech segmentation using neural network and phone...

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International Joint Conference on Neural Networks (IJCNN)

作者： H. Finster Institute for Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

A new algorithm for automatic segmentation of speech based on its phonetic transcription is proposed. The specific features of this algorithm are a new iterative self-learning procedure to find the temporal alignment between feature vectors and phonetic transcription; no preassumptions about statistical speech properties or phonetical rules; and no required pretraining. The general structure of the segmentation system is shown. The core of the segmentation procedure is an iterative loop consisting of a neural phoneme classifier, a time-alignment algorithm and the retraining of the neural classifier. The segmentation of the sentence 'nine two seven eight nine ten' is given.< >

关键词： Neural networks Probability Feature extraction Speech analysis Filter bank Iterative algorithms Multilayer perceptrons

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Steganographic Wideband Telephony Using Narrowband Speech Codecs

Steganographic Wideband Telephony Using Narrowband Speech Co...

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Asilomar Conference on Signals, systems & Computers

作者： Peter Vary Bernd Geiser Institute of Communication Systems and Data Processing RWTH Aachen Germany

We consider the transmission of wideband speech with a cut-off frequency of f c = 7 kHz over a standardized digital narrowband communication link (f c = 3.4 kHz). At the receiver, wideband speech is produced by artificial bandwidth extension (BWE). The BWE algorithms can be realized with or without some low bit rate side information. In this paper, we propose to communicate the side information to the receiver via a steganographic channel within the bitstream of the narrowband codec. Hence, the bitstream format is not altered and the bit rate is not increased. The following codecs are considered: mu-law PCM, ADPCM, CS-ACELP, GSM FR, and GSM EFR.

关键词： Steganography Internet telephony Narrowband Speech codecs Wideband Bit rate GSM Cutoff frequency Bandwidth Phase change materials

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Noise estimation techniques for robust speech recognition

Noise estimation techniques for robust speech recognition

引用

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

作者： H.G. Hirsch C. Ehrlicher Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

Two new techniques are presented to estimate the noise spectra or the noise characteristics for noisy speech signals. No explicit speech pause detection is required. Past noisy segments of just about 400 ms duration are needed for the estimation. Thus the algorithm is able to quickly adapt to slowly varying noise levels or slowly changing noise spectra. This techniques can be combined with a nonlinear spectral subtraction scheme. The ability can be shown to enhance noisy speech and to improve the performance of speech recognition systems. Another application is the realization of a robust voice activity detection.

关键词： Noise robustness Speech recognition Speech enhancement Noise reduction Noise level Background noise Statistics Signal to noise ratio Recursive estimation data processing

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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...

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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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Estimation of missing LSF parameters using Gaussian mixture models

Estimation of missing LSF parameters using Gaussian mixture ...

引用

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

作者： R. Martin C. Hoelper I. Wittke Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

Speech transmission over packet networks has to cope with packet delays and packet losses. When a packet loss occurs the missing information must be estimated. We focus on restoring the spectral parameters of a speech coder. A novel approach to estimating missing line spectral frequency (LSF) parameters using Gaussian mixture models (GMM) is proposed. We present the estimation algorithm and study its performance when one or several LSF parameters are lost. We show that a GMM of a relatively low order is sufficient to achieve a substantial improvement in the parameter SNR. Therefore, the new estimation procedure requires much less memory than histogram based estimation methods.

关键词： GSM Speech coding Propagation losses Frequency estimation Histograms Internet telephony Delay effects Bit rate Context modeling Parameter estimation

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