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A postfilter for echo and noise reduction avoiding the problem of musical tones

A postfilter for echo and noise reduction avoiding the probl...

引用

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

作者： S. Gustafsson P. Jax A. Kamphausen P. Vary Institute of Communication Systems and Data Processing RWTH Aachen Aachen Germany

We address the problem of acoustic echo cancellation and noise reduction for narrow and wide band telephone applications. We combine a conventional echo canceller with a postfilter implemented in the frequency domain and derive an algorithm for the simultaneous attenuation of residual echo and noise. The main goals are a low level natural sounding background noise without artifacts such as musical tones, and an inaudible residual echo. This is achieved by considering the masking properties of the human auditory system. Simulation results verify that these goals are reached, while the distortion of the near end speech is comparable to conventional algorithms.

关键词： Noise reduction Echo cancellers Acoustic applications Wideband Telephony Frequency domain analysis Attenuation Acoustic noise Noise cancellation Background noise

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Robust speech transmission over noisy channels employing non-linear block codes

Robust speech transmission over noisy channels employing non...

引用

IEEE Workshop on Speech Coding Proceedings

作者： S. Heinen S. Bleck P. Vary Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Germany

In medium to low bit rate speech codecs the speech signal is represented by a set of parameters. The most important concept is presently code excited linear predictive (CELP) coding. A speech segment of typically 10 to 20 ms is described in terms of prediction coefficients, gain factors and excitation vectors. Due to the high compression rates (0.5-1.5 bits per speech sample) the parameters are partly highly sensitive against channel noise. In this paper we present a new error protection technique, that is based on a joint optimization of parameter quantization and a redundant non-linear block coding scheme. For parameter reconstruction, the principle of soft bit source decoding is applied. The proposed technique can be used in combination with conventional error protection such as convolutional coding and allows a flexible subdivision of the gross data rate for source coding and error protection.

关键词： Robustness Protection Speech codecs Bit rate Vectors Speech enhancement Quantization Block codes Decoding Convolutional codes

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Joint speech codec parameter and channel decoding of parameter individual block codes (PIBC) 6

Joint speech codec parameter and channel decoding of paramet...

引用

6th IEEE Workshop on Speech Coding

作者： Fingscheidt, Tim Heinen, Stefan Vary, Peter ATandT Labs Shannon Laboratory Florham ParkNJ07932 United States Institute of Communication Systems and Data Processing Aachen University of Technology Templergraben 55 AachenD-52056 Germany

ISBN: (纸本)0780356519

In digital mobile speech transmission usually the most important (class la) bits provided by the speech coding scheme are protected by a CRC for error detection. As a consequence all parameters spanned by the class la bits have to be marked at the receiver either as reliable or as unreliable. In contrast to this somewhat coarse approach we propose the usage of what we call parameter individual block codes (PIBC) for the most important codec parameters. This allows joint speech codec parameter and PIBC decoding taking advantage of the error concealing properties of soft-bit speech decoding. © 1999 IEEE.

关键词： Decoding

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Wideband speech coding using forward/backward adaptive prediction with mixed time/frequency domain excitation

Wideband speech coding using forward/backward adaptive predi...

引用

IEEE Workshop on Speech Coding Proceedings

作者： J. Schnitzler J. Eggers C. Erdmann P. Vary Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany University Erlangen-Nuünberg Germany

This paper describes a wideband (7 kHz) speech coding scheme using code-excited linear prediction (CELP) with mixed time and frequency domain excitation. The proposed frequency domain innovation can be used alternatively or in parallel to a time domain codebook. In addition an improved synthesis filter is used consisting of a signal dependent combination of a forward adaptive and a backward adaptive (FA/BA) structure. An experimental codec operating at 15.5 or 20.0 kbit/s is demonstrated.

关键词： Wideband Speech coding Frequency domain analysis Quantization Filtering Bit rate Technological innovation Adaptive filters Shape control Discrete cosine transforms

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Joint speech codec parameter and channel decoding of parameter individual block codes (PIBC)

Joint speech codec parameter and channel decoding of paramet...

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IEEE Workshop on Speech Coding Proceedings

作者： T. Fingscheidt S. Heinen P. Vary Shannon Laboratory AT and T Research Laboratories Florham Park NJ USA Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

关键词： Speech codecs Decoding Block codes Speech coding Protection Cyclic redundancy check Viterbi algorithm Laboratories data processing Speech processing

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Combined residual echo and noise reduction: A novel psychoacoustically motivated algorithm 9

Combined residual echo and noise reduction: A novel psychoac...

引用

9th European Signal processing Conference, EUSIPCO 1998

作者： Gustafsson, Stefan Jax, Peter Institute of Communication Systems and Data Processing RWTH Aachen Templergraben 55 AachenD-52056 Germany

In this paper we focus on the problem of acoustic echo cancellation and noise reduction for hands-free telephony devices. A standard echo canceller is combined with a frequency domain post-filter, which applies a novel psychoacoustically motivated weighting rule. The algorithm makes use of the masking threshold of the human auditory system to achieve a perceived reduction of noise and residual echo equal to some pre-defined levels. In contrast to conventional methods, the proposed one preserves the nature of the original background noise and doesn't introduce any audible artifacts. At the same time it can attain a very high reduction of the residual echo.

关键词： Noise abatement

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Robust speech decoding: can error concealment be better than error correction?

Robust speech decoding: can error concealment be better than...

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

作者： T. Fingscheidt P. Vary J.A. Andonegui Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

Digital speech transmission systems use source coding to reduce the bit rate and channel coding to correct transmission errors. Furthermore, in periods of a very poor channel quality error concealment of residual bit errors becomes necessary as channel decoding fails. However, if the channel is clear, channel coding would not be required at all and the speech quality could be improved by allowing a higher bit rate for source encoding. Usually a compromise is taken between speech quality in case of a clear channel and error robustness in case of poor channel quality. This paper addresses the problem of a joint optimization of error concealment and source/channel coding. Under the premise of a minimum mean square error criterion for signal reconstruction it turns out that error concealment instead of error correction may be the best choice if source coding leaves sufficient residual parameter correlations by less bit rate reduction.

关键词： Robustness Decoding Bit rate Channel coding Speech coding Source coding Error correction Mean square error methods Signal reconstruction Error correction codes

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A novel psychoacoustically motivated audio enhancement algorithm preserving background noise characteristics

A novel psychoacoustically motivated audio enhancement algor...

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

作者： S. Gustafsson P. Jax P. Vary Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

In this paper we propose an algorithm for reduction of noise in audio signals. In contrast to several previous approaches we do not try to achieve a complete removal of the noise, but instead our goal is to preserve a pre-defined amount of the original noise in the processed signal. This is accomplished by exploiting the masking properties of the human auditory system. The speech and noise distortions are considered separately. The spectral weighting rule, adapted by utilizing only estimates of the masking threshold and the noise power spectral density, has been designed to guarantee complete masking of distortions of the residual noise. Simulation results confirm that no audible artifacts are left in the processed signal, while speech distortions are comparable to those caused by conventional noise reduction techniques.

关键词： Psychology Acoustic noise Noise reduction Signal processing Speech enhancement Humans Auditory system Masking threshold Psychoacoustic models Speech processing

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COMBINED ACOUSTIC ECHO CONTROL AND NOISE REDUCTION FOR MOBILE communicationS 5

COMBINED ACOUSTIC ECHO CONTROL AND NOISE REDUCTION FOR MOBIL...

引用

5th European Conference on Speech communication and Technology, EUROSPEECH 1997

作者： Gustafsson, Stefan Martin, Rainer Institute of Communication Systems and Data Processing Aachen University of Technology AachenD-52056 Germany

In this paper an acoustic echo compensator with an additional frequency domain adaptive filter for combined residual echo and noise reduction is proposed. The algorithm delivers high echo attenuation as well as high near end speech quality over a wide range of signal-to-noise conditions. The system makes use of a standard time domain echo compensator of low order, after which the proposed adaptive filter, which is motivated by means of a minimum mean square error approach, is placed in the sending path. In contrast to other combined systems [1, 2, 3], our method uses an explicit estimate of the power spectral density of the residual echo after echo compensation. The separate estimations of the power spectral densities of the residual echo and the background noise, respectively, are then flexibly combined, such that in the processed signal a low level of intentionally left background noise will effectively mask the residual echo. © 1997 5th European Conference on Speech communication and Technology, EUROSPEECH 1997. All rights reserved.

关键词： Power spectral density

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SMALL MICROPHONE ARRAYS WITH OPTIMIZED DIRECTIVITY FOR SPEECH ENHANCEMENT 5

SMALL MICROPHONE ARRAYS WITH OPTIMIZED DIRECTIVITY FOR SPEEC...

引用

5th European Conference on Speech communication and Technology, EUROSPEECH 1997

作者： Dörbecker, Matthias Institute of Communication Systems and Data Processing Aachen University of Technology Templergraben 55 Aachen52056 Germany

In many situations of digital speech communication (e.g. hands-free telephony or electronic hearing aids) the speech signal picked up by the microphone is disturbed by acoustic background noise. Therefore, adaptive filtering techniques which aim at the reduction of the disturbing noise are subject of current research activities (e.g. [1]). Although some of the already known adaptive techniques - especially concepts with two or more microphones - allow a significant reduction of the noise, most of the adaptive strategies result, particularly at low SNR, in a speech signal with an unnatural character due to time-variant distortions, and the occurrence of musical noise. An alternative approach, which does not affect the speech signal by time-variant distortions, is the application of a microphone array with a fixed directivity pattern aligned to the speaker's position, resulting in a suppression of spatially distributed noise sources. In this contribution it is shown that due to the proposed optimization even with only two microphones a reduction of diffuse noise sound up to 6 dB can be achieved. © 1997 5th European Conference on Speech communication and Technology, EUROSPEECH 1997. All rights reserved.

关键词： Signal to noise ratio

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