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Quantisation noise control in perceptual audio coding using low selectivity filter banks

ELECTRONICS LETTERS 2002年第16期38卷 932-933页

作者： Martínez-Muñoz, D Rosa-Zurera, M Cruz-Roldán, F López-Ferreras, F Ruiz-Reyes, N Univ Jaen Escuela Univ Politecn Dept Elect Jaen 23700 Spain Univ Alcala de Henares Escuela Politecn Dept Teoria Senal & Comunicac Madrid 28871 Spain

The problem of computing, in a subband audio coder, the maximum quantisation noise power that can be injected in each band to ensure transparent coding when low selectivity filter banks are used, is addressed. A low complexity strategy, taking into account the frequency responses of the synthesis filter bank, is proposed for achieving an overall distortion due to quantisation noise always below the masking threshold (provided by a psycho-acoustic model) for any length prototype filters.

关键词： quantisation (signal) frequency response ISO/MPEG-1 standard low complexity strategy maximum quantisation noise power filtering theory frequency responses masking threshold Codes quadrature mirror filters quantisation noise control perceptual audio coding subband audio coder audio coding white noise transparent coding psychoacoustic model Filtering methods in signal processing low selectivity filter banks speech and audio coding Signal processing and conditioning equipment and techniques

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Natural quality two-band LPC coding of speech at 880 bit/s with frame interpolation

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ELECTRONICS LETTERS 2002年第6期38卷 292-294页

作者： Han, WJ Kim, EK Oh, YH Korea Adv Inst Sci & Technol Dept Elect Engn & Comp Sci Div Comp Sci Yusong Gu Taejon 305701 South Korea

A novel frame interpolation technique for two-band linear predictive coding (LPC) vocoders is proposed for maintaining natural speech quality at bit rates below I kbit/s. Experimental results show that the speech qual... 详细信息

关键词： natural speech quality interpolation linear predictive coding 880 bit/s speech coding vocoders frame interpolation two-band linear predictive coding vocoder speech and audio coding bit rate Codecs, coders and decoders

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Very low rate speech coding using temporal decomposition

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ELECTRONICS LETTERS 1999年第6期35卷 456-457页

作者： Ghaemmaghami, S Sridharan, S Queensland Univ Technol Sch Elect Elect & Syst Engn Speech Res Lab Brisbane Qld 4001 Australia Sharif Univ Technol Elect Res Ctr Tehran Iran

A method for encoding the spectral characteristics of speech, at rates below 180 bit/s, using hierarchical temporal decomposition (HTD) is proposed. A set of the log-area-ratio (LAR) parameters, extracted from a given block of speech, are approximated through Gaussian interpolation between the most-steady frames detected by the HTD. This results in a smaller set of parameters which are encoded using vector quantisation. It is shown that the same spectral distortion is obtained with the new coder at a rate of 180 bit/s as that using a scalar quantisation, TD-based coder, at 600 bit/s.

关键词： interpolation log-area-ratio parameters vector quantisation Gaussian interpolation speech coding Codes spectral distortion hierarchical temporal decomposition speech and audio coding Interpolation and function approximation (numerical analysis) most-steady frames spectral characteristics 0 to 180 bit/s Gaussian processes

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Algorithm for achieving adaptive tiling of time axis for audio coding purposes

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ELECTRONICS LETTERS 2002年第9期38卷 434-435页

作者： Ruiz, N Rosa, M López, F Vera, P Univ Jaen Dept Elect Escuela Politecn Linares Jaen Spain Univ Alcala de Henares Escuela Politecn Dept Teoria Senal & Commun Madrid Spain

A new algorithm for achieving flexible tiling, of the time axis for audio coding purposes is presented, It is based on the calculus of the distances among a predetermined number of time-frequency pairs, From the computed distances. a clustering process determines the final subdivision of each audio frame. Experimental results demonstrates the good performance of the proposed algorithm. which provides high coding, efficiency with a reduced complexity.

关键词： audio coding clustering process audio frame complexity reduction adaptive tiling time-frequency pairs speech and audio coding time axis high coding efficiency adaptive signal processing

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Dual-mode AVQ coding Based on Spectral Masking and Sparseness Detection for ITU-T G.711.1/G.722 Super-wideband Extensions

Dual-mode AVQ Coding Based on Spectral Masking and Sparsenes...

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12th Annual Conference of the International-speech-Communication-Association 2011 (INTERspeech 2011)

作者： Fukui, Masahiro Sasaki, Shigeaki Hiwasaki, Yusuke Sachiko, Kurihara Haneda, Yoichi NTT Corp NTT Cyber Space Labs Tokyo Japan

ISBN: (纸本)9781618392701

ITU-T Recommendations G.711.1 Annex D and G.722 Annex B, which are super-wideband (50-14,000 Hz) extensions to G.711.1 and G.722, have been recently standardized. This paper introduces a new coding method proposed and employed in the above ITU-T standards. The proposed coding method employs an adaptive spectral masking of the algebraic vector quantization (AVQ) for MDCT-domain non-sparse signals. The adaptive spectral masking is switched on and off based on MDCT-domain sparseness analysis. When the target MDCT coefficients are categorized as non-sparse, masking level of the target MDCT coefficients is adaptively controlled using spectral envelope information. The performance of the proposed method as a part of the ITU-T G.711.1 Annex D is evaluated in comparison with the ordinary AVQ. Subjective listening test results show that the proposed method improves the sound quality more than 0.1 points with a five grade scale in average of speech, music and mixed content, and the significance of the improvement is validated.

关键词： speech and audio coding Standardization ITU-T G.711.1 Annex D ITU-T G.722 Annex B super-wideband extension algebraic vector quantization

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PARAMETRIC BINAURAL audio coding

PARAMETRIC BINAURAL AUDIO CODING

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2010 IEEE International Conference on Acoustics, speech, and Signal Processing

作者： Ojala, Pasi Tammi, Mikko Vilermo, Miikka Nokia Res Ctr Tampere Finland

ISBN: (纸本)9781424442966

A spatial audio scene consists of discrete audio sources and ambience. The 3D audio image is observed due to the directional sounds, but even more important is the reverberation and so called room effect caused by the properties of the space, the sources and listener location. It is obvious that a human being is able to capture the 3D image using the signal from left and right ear. Hence, two audio channels are sufficient to represent the spatial audio image for the listener. An efficient transmission and representation of a spatial audio image using two channels requires a specific coding and rendering algorithm for the audio content. In this paper we present novel mechanisms to efficiently parameterize, quantize and represent the spatial audio signal.

关键词： Binaural audio speech and audio coding

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Low-complexity and robust coding mode decision in the EVS coder 40

Low-complexity and robust coding mode decision in the EVS co...

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40th IEEE International Conference on Acoustics, speech, and Signal Processing, ICASSP 2015

作者： Ravelli, Emmanuel Helmrich, Christian R. Fuchs, Guillaume Multrus, Markus Am Wolfsmantel 33 Erlangen Germany International Audio Laboratories Erlangen Am Wolfsmantel 33 Erlangen Germany

ISBN: (纸本)9781467369978

Several state-of-the-art switched audio codecs employ the closed-loop mode decision to select the best coding mode at every frame. The closed-loop mode selection is known to have good performance but also high complexity. The new approach we propose in this paper is a low-complexity version of the closed-loop approach, based on similar decisions which compute the coding distortion of each mode and select the one with the lowest distortion. Our approach differs mainly in the way the coding distortions are calculated. We are able to notably reduce the complexity by only estimating the distortions without encoding and decoding the input for each mode. The new approach was implemented in the EVS codec standard and evaluated both objectively and subjectively. Compared to the closed-loop approach, it yields similar performance and lower complexity. © 2015 IEEE.

关键词： closed-loop mode decision mode selection open-loop speech and audio coding switched coding

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Closure of sets: a statistically hypersensitive system for steganalysis of least significant bit embedding

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IET SIGNAL PROCESSING 2011年第4期5卷 379-389页

作者： Khosravirad, S. R. Eghlidos, T. Ghaemmaghami, S. Sharif Univ Technol Elect Res Inst Tehran Iran

This study introduces a new scheme for steganalysis of the least significant bit (LSB) embedding, based on the idea of closure of sets (CoS), which is independent of the type of cover signal, applicable to both spatial and transform domains. The CoS is referred to as some special subsets that could be found in a common space whose elements relate to higher-order statistical properties of the signal. The proposed scheme is used for steganalysis of the LSB steganography of greyscale TIFF and JPEG images and audio signals, employing a set of accurate and monotone features that are extracted based on the CoS definition. It is shown that significant improvement to the detection accuracy in steganalysis is achieved using the proposed scheme, particularly at low embedding rates.

关键词： Computer vision and image processing techniques closure of sets Data security Image and video coding steganography image coding Cryptography least significant bit embedding statistically hypersensitive system steganalysis feature extraction transforms spatial domain audio signals speech and audio coding JPEG images audio coding transform domain greyscale TIFF images image colour analysis least significant bit steganography Integral transforms monotone feature extraction higher-order statistical property

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The sensitivity matrix: Using advanced auditory models in speech and audio processing

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IEEE TRANSACTIONS ON audio speech AND LANGUAGE PROCESSING 2007年第1期15卷 310-319页

作者： Plasberg, Jan H. Kleijn, W. Bastiaan Royal Inst Technol Sch Elect Engn S-10044 Stockholm Sweden

Perceptually optimal processing of speech and audio signals demands a rigorous approach using a distortion measure that resembles human perception. This requires distortion measures based on sophisticated, complex auditory models. Under the assumption of small distortions these models can be simplified by means of a sensitivity matrix. In this paper, we show the power of this approach. We present a method to derive the sensitivity matrix for distortion measures based on spectro-temporal auditory models. This method is applied to an example auditory model and the region of validity of the approximation and the application of linear algebra to analyze the characteristics of the given model are discussed. Furthermore, we show how to build a coder minimizing a sensitivity matrix distortion measure given the typically long support of a perceptual distortion measure.

关键词： auditory model distortion perception quantization sensitivity speech and audio coding

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Energy-proportion based scheme for audio watermarking

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IET SIGNAL PROCESSING 2010年第5期4卷 576-587页

作者： Chen, S. -T. Huang, H. -N. Chen, C. -J. W, G. -D. Natl Chi Nan Univ Dept Elect Engn Puli 545 Nan Tou Taiwan Tunghai Univ Dept Math Taichung 40704 Taiwan

This study presents a audio watermarking that embeds information by energy-proportion scheme. By using normalised energy instead of probability, this study rewrites the entropy in information theory as an e function. In order to guarantee the robustness of watermarks, low-frequency coefficients of discrete wavelet transform (DWT) are mapped into the domain of this function. Then, the characteristic curve of energy-proportion function (CCEP) is obtained. Moreover, some characteristics and properties of this function are analysed and proved. Based on CCEP and the properties of energy-proportion function, this study proposes a novel audio watermarking scheme. In addition, the proposed energy-proportion scheme can extract the watermark without original audio signal. Finally, performance of the proposed scheme is assessed by signal-to-noise ratio (SNR), mean opinion score, embedding capacity and BER. Experimental results demonstrate that the embedded data are robust against most signal processing and attacks, such as re-sampling, MP3 compression, low-pass filtering and amplitude scaling.

关键词： energy proportion function signal sampling audio signal extraction error statistics audio watermarking signal- to-noise ratio amplitude scaling information theory signal resampling MP3 compression discrete wavelet transform speech and audio coding DWT discrete wavelet transforms audio coding entropy Integral transforms signal processing BER SNR CCEP Other topics in statistics

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