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共 26 条记录，以下是1-10 订阅

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SINGLE MICROPHONE WIND NOISE REDUCTION USING TECHNIQUES OF ARTIFICIAL BANDWIDTH EXTENSION

SINGLE MICROPHONE WIND NOISE REDUCTION USING TECHNIQUES OF A...

引用

European Signal processing Conference

作者： Christoph Matthias Nelke Niklas Nawroth Marco Jeub Christophe Beaugeant Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University Intel Mobile Communications Institute of Communication Systems and Data Processing RWTH Aachen University

ISBN: (纸本)9781467310680

In this contribution, we propose a method to enhance single channel speech signals which are degraded by wind noise. In contrast to common speech enhancement systems, a special processing is required due to the highly non-stationary characteristics of wind signals. The basic idea is to exploit the fact that wind noise is mainly located at low frequencies and thus, a large frequency range of the speech is almost noise free. Techniques which artificially extend the bandwidth of telephone speech towards lower frequencies are applied to replace the highly disturbed low frequency parts. Here, the discrete model of speech production is used to reconstruct the required parts of the speech signal. Important parameters for this model are pitch frequency, the spectral envelope and a spectral gain. In this context, an evaluation is carried out which determines the robustness of several pitch estimators against wind noise. The frequency range of the reconstructed speech is finally adapted to the actual level of wind noise. Based on realistic scenarios it is shown that the influence of the wind noise can greatly be reduced by the proposed concept. This includes a comparison with a state-of-the-art speech enhancement system and an algorithm specially designed to reduce wind noise.

关键词： WIND NOISE Noise frequency ranges Wind Bandwidth Speech Microphone Device Component Speech enhancement low frequencies Voice Signal

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ROBUST DUAL-CHANNEL NOISE POWER SPECTRAL DENSITY ESTIMATION

ROBUST DUAL-CHANNEL NOISE POWER SPECTRAL DENSITY ESTIMATION

引用

European Signal processing Conference

作者： Marco Jeub Christoph Nelke Hauke Kruger Christophe Beaugeant Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University Intel Mobile Communications

In this paper, we propose a novel noise power spectral density (PSD) estimator which is beneficial for speech enhancement systems with two microphones in diffuse noise environments. The algorithm has a low computational complexity and requires low memory usage. The main advantage is that arbitrary models of the noise field coherence can be employed and a scalable extension of existing single-channel speech enhancement systems to dual channels is also possible. Experiments demonstrate with simulated and measured data that the proposed algorithm outperforms related algorithms in diffuse noise conditions.

关键词： noise field Spectral density Speech enhancement channel noise Noise noise environment noise power spectral density algorithms complexity classes observational data

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BLIND ESTIMATION OF THE COHERENT-TO-DIFFUSE ENERGY RATIO FROM NOISY SPEECH SIGNALS

BLIND ESTIMATION OF THE COHERENT-TO-DIFFUSE ENERGY RATIO FRO...

引用

European Signal processing Conference

作者： Marco Jeub Christoph Nelke Christophe Beaugeant Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University Intel Mobile Communications

A novel dual-channel algorithm is proposed which estimates the coherent-to-diffuse energy ratio (CDR) of background noise in mixed noise fields. The algorithm is based on an estimate of the noise field coherence from a noisy speech signal and a subsequent minima tracking in order to increase the estimation accuracy even in the presence of speech. The obtained CDR estimate can be used, e.g., for the acoustic environment classification in hearing aids or to control speech enhancement algorithms such as noise reduction or speech dereverberation. Besides, the approach can be used to calculate an estimate of the direct-to-reverberant energy ratio (DRR) blindly from reverberant speech signals.

关键词： Noise Speech Coherence Estimation Noise measurement Frequency estimation Signal processing algorithms

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Source optimized channel codes (SOCCs) for parameter protection

Source optimized channel codes (SOCCs) for parameter protect...

引用

IEEE International Symposium on Information Theory

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

We present a new class of channel codes, which we call source optimized channel codes (SOCCs). These non-linear codes are designed to maximize a given analogue quality measure in consideration of source and channel st... 详细信息

关键词： Protection Decoding Vectors Channel coding Speech coding Error correction codes Parameter estimation data processing Statistics Error analysis

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An optimal MMSE-estimator for source codec parameters using intra-frame and inter-frame correlation

An optimal MMSE-estimator for source codec parameters using ...

引用

IEEE International Symposium on Information Theory

作者： S.M. Heinen Institute of Data Processing and Communications Systems RWTH Aachen University of Technology Germany Synopsys GmbH Germany

We propose a new minimum mean square error (MMSE) estimator, which is able to exploit both intra- and inter-frame correlation of source codec parameters. Simulations, which were performed with realistic parameter corr... 详细信息

ISBN: (纸本)0780371232

关键词： Codecs Redundancy Bandwidth Decoding Mean square error methods Performance gain Speech Video compression Delay data processing

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Active noise control in headsets: A new approach for broadband feedback ANC

Active noise control in headsets: A new approach for broadba...

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

作者： Thomas Schumacher Hauke Krüger Marco Jeub Peter Vary Christophe Beaugeant Institute of Communication Systems and Data Processing RWTH Aachen University Germany Intel Mobile Communications Sophia-Antipolis France

In this paper a novel approach for broadband feedback active noise control (ANC) is presented which is based on the combination of classical non-adaptive feedback and adaptive feedback ANC techniques. The non-adaptive part is suitable to attenuate low frequency ambient noise whereas the adaptive part attenuates periodic components of the ambient noise. The proposed technique yields a higher overall noise attenuation performance compared to a purely classical non-adaptive feedback or purely adaptive feedback ANC system. In addition to that, the combination of both techniques is also beneficial for practical realizations since the adaptive feedback ANC stabilizes the overall system. With regard to low cost headset devices, the impact of practical hard ware constraints such as low-cost analog-to-digital and digital-to analog converters (ADC, DAC) is discussed. As a conclusion, a mixed analog-digital realization of the new approach is proposed.

关键词： Noise Attenuation Delay Adaptive systems Microphones Loudspeakers Ear

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Dual microphone noise PSD estimation for mobile phones in hands-free position exploiting the coherence and speech presence probability

Dual microphone noise PSD estimation for mobile phones in ha...

引用

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

作者： Christoph Matthias Nelke Christophe Beaugeant Peter Vary Institute of Communication Systems and Data Processing RWTH Aachen University Germany Intel Mobile and Communications Group Sophia-Antipolis France

ISBN: (纸本)9781479903573

This contribution addresses the enhancement of noisy speech signals picked up by a dual microphone mobile phone in hands-free position. A novel technique to estimate the noise power spectral density is presented which combines two methods: a single microphone algorithm based on the speech presence probability and a dual microphone technique exploiting the coherence properties of the target signal and the background noise. Due to the novel approach, the weakness of both methods can be overcome. Since the proposed method requires knowledge of the current coherence properties, a technique is presented which estimates the coherence of the speech and noise signals, which is usually not known in practice.

关键词： Noise Speech Coherence Microphones Noise measurement Estimation Speech processing

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Applicability of UDP-lite for voice over IP in UMTS networks

Applicability of UDP-lite for voice over IP in UMTS networks

引用

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

作者： F. Mertz U. Engelke P. Vary H. Taddei I. Varga Institute of Communication Systems and Data Processing (IND) RWTH Aachen University Aachen Germany Siemens AG Communications Munich Germany MD Siemens AG Communications Munich Germany

This paper examines the application of UDP-lite for unequal error detection in packet-switched speech transmission via Internet protocols (voice-over-IP) over UMTS radio channels. Traditionally, UDP is used as transport layer protocol, which contains a checksum that covers the complete packet. Thus, any packet with residual bit errors is discarded. Speech codecs like AMR, however, can tolerate bit errors in less sensitive parts of the bitstream. A more recent development, UDP-lite, provides unequal error detection with a partial checksum that covers only the sensitive parts of a packet. Thus, only packets with errors in important bits are discarded. We compare the use of UDP-lite for UMTS channels with convolutional and channels with turbo coding. The results show that the achievable quality improvement by applying UDP-lite depends on the residual bit error distribution of the chosen UMTS channel coding method. While we determined a quality improvement for channels with convolutional coding, we did not get an improvement for turbo coding. Furthermore, when combined with header compression, the convolutional coder with use of UDP-lite can reach the performance of the turbo coder with use of UDP

关键词： Internet telephony Intelligent networks 3G mobile communication Convolutional codes Turbo codes Protocols Speech codecs Error correction codes Channel coding Downlink

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BACKWARDS-COMPATIBLE ERROR PROPAGATION RECOVERY FOR THE AMR CODEC OVER ERASURE CHANNELS

BACKWARDS-COMPATIBLE ERROR PROPAGATION RECOVERY FOR THE AMR ...

引用

IEEE International Conference on Acoustics, Speech and Signal processing

作者： Angel M. Gomez Jose L. Perez-Cordoba Bernd Geiser Dpt. of Signal Theory Telematics and Communications and CITIC-UGR University of Granada Institute of Communication Systems and Data Processing RWTH Aachen University

ISBN: (纸本)9781479903573

This paper presents a recovery scheme for the error-propagation distortion which frequently appears after a frame erasure in CELP-based speech coders, in particular the AMR codec. The extensive use of predictive filters and parameter encoding allow a high-quality speech synthesis in these codecs, but makes them more vulnerable to frame erasures. Thus, when a frame is lost, an additional distortion appears in the subsequent frame, although that was correctly received, further degrading the speech quality. This degradation can also propagate over several frames, being even more damaging than the loss itself. This well known fact has motivated the development of techniques which prevent or mitigate the error propagation. Nevertheless, the previously proposed methods in some respect modify the transmission scheme (by including additional frames, EEC codes, etc.) making them incompatible with the original decoder. In this work, we apply a steganographic technique to embed recovery data to assist the decoder after a frame loss. This data mainly consist of resynchronization pulses and correction vectors for the excitation signal and the spectral envelope, respectively, PESQ results confirm that our proposal achieves a higher robustness against error propagation while the full backwards-compatibility with the AMR standard is retained.

关键词： Speech coding ACELP Frame erasure Error propagation Pulse codebook data hiding Steganography Error propagation speech coding Hidden data Steganography codecs ACELP Frame FRAMES excitation signal Amphiregulin Antimicrobial Drug Resistance automatic meter reading parametric coding Vocoders decoders

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VOICE SOURCE ESTIMATION FOR ARTIFICIAL BANDWIDTH EXTENSION OF TELEPHONE SPEECH

VOICE SOURCE ESTIMATION FOR ARTIFICIAL BANDWIDTH EXTENSION O...

引用

IEEE International Conference on Acoustics, Speech, and Signal processing

作者： Mark R. P. Thomas Jon Gudnason Patrick A. Naylor Bernd Geiser Peter Vary Communications and Signal Processing Group Imperial College London Exhibition Road London SW7 2AZ UK Institute of Communication Systems and Data Processing (iNd) RWTH Aachen University Germany

ISBN: (纸本)9781424442959

Artificial bandwidth extension (ABWE) of speech signals aims to estimate wideband speech (50 Hz - 7 kHz) from narrowband signals (300 Hz - 3.4 kHz). Applying the source-filter model of speech, many existing algorithms estimate vocal tract filter parameters independently of the source signal. However, many current methods for extending the narrowband voice source signal are limited to straightforward signal processing techniques which are only effective for high-band estimation. This paper presents a method for ABWE that employs novel data-driven modelling and an existing spectral mirroring technique to estimate the wideband source signal in both the high and low extension bands. A state-of-the-art Hidden Markov Model-based estimator evaluates the temporal and spectral envelopes in the missing frequency bands, with which the ABWE speech signal is synthesized. Informal listening tests comparing two existing source estimation techniques and two permutations of the proposed approach show an improvement in the perceived bandwidth of speech signals, in particular towards low frequencies. Subjective tests on the same data show a preference for the proposed techniques over the existing methods under test.

关键词： Speech enhancement artificial bandwidth extension voice source modelling Speech enhancement Speech Bandwidth Voice Narrowband signals Voice Signal Frequency bands

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