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An analysis of perceptual artifacts in MPEG scalable audio coding

An analysis of perceptual artifacts in MPEG scalable audio c...

引用

Data Compression Conference (DCC)

作者： C.D. Creusere Klipsch School of Electrical & Computer Engineering New Mexico State University Las Cruces NM USA

ISBN: (纸本)0769514774

We study coding artifacts in MPEG-compressed scalable audio. Specifically, we consider the MPEG advanced audio coder (AAC) using bit slice scalable arithmetic coding (BSAC) as implemented in the MPEG 4 reference software. First, we perform human subjective testing using the comparison category rating (CCR) approach, quantitatively comparing the performance of scalable BSAC with the nonscalable TwinVQ and AAC algorithms. This testing indicates that scalable BSAC performs very poorly relative to TwinVQ at the lowest bitrate considered (16 kb/s), largely because of an annoying and seemingly random mid-range tonal signal that is superimposed onto the desired output. In order to understand better and quantify perceptually the various forms of distortion introduced into compressed audio at low bit rates, we apply two analysis techniques: Reng probing and time-frequency decomposition. The Reng probing technique is capable of separating the linear time-invariant component of a multirate system from its nonlinear and periodically time-varying components. Using this technique, we conclude that aliasing is probably not the cause of the annoying tonal signal; instead, time-frequency analysis indicates that its cause is most likely suboptimal bit allocation.

关键词： audio coding Bit rate Performance evaluation Testing Time frequency analysis Arithmetic MPEG 4 Standard Humans Nonlinear distortion Time varying systems

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Delay-free lossy audio coding using shelving pre- and post-filters

Delay-free lossy audio coding using shelving pre- and post-f...

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

作者： Martin Holters Udo Zolzer Helmut Schmidt University Hamburg Hamburg Germany

A delay-free audio coding scheme based on ADPCM with adaptive pre- and post-filtering is presented. The pre-/post- filters are realized as a cascade of shelving filters, designed to match the characteristics of human perception. The pre- and post-filters are adapted by dynamic compression of the respective sub-bands. The adaption is backward-adaptive, i.e. is fed by the reconstructed signal, which eliminates the need to transmit the filter coefficients and allows delay-free operation. This pre- and post-filtering significantly improves the audio quality compared to a plain ADPCM codec, as underlined by objective measurements. Since the base ADPCM used is also delay-free, the resulting coding system works without any algorithmic delay.

关键词： Delay audio coding Codecs Decoding Quantization Finite impulse response filter Cutoff frequency Lattices Humans Adaptive filters

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MDCT audio coding with pulse vector quantizers

MDCT audio coding with pulse vector quantizers

引用

IEEE International Conference on Acoustics, Speech and Signal Processing

作者： J. Svedberg V. Grancharov S. Sverrisson E. Norvell T. Toftgard H. Pobloth S. Bruhn SMN Ericsson AB Stockholm Sweden

ISBN: (纸本)9781467369985

This paper describes a novel audio coding algorithm that is a building block in the recently standardized 3GPP EVS codec [1]. The presented scheme operates in the Modified Discrete Cosine Transform (MDCT) domain and deploys a Split-PVQ pulse coding quantizer, a noise-fill, and a gain control optimized for the quantizer's properties. A complexity analysis in terms of WMOPS is presented to illustrate that the proposed Split-PVQ concept and dynamic range optimized MPVQ-indexing are suitable for real-time audio coding. Test results from formal MOS subjective evaluations and objective performance figures are presented to illustrate the competitiveness of the proposed algorithm.

关键词： audio coding EVS MDCT Noise-Fill PVQ

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Advances in parametric audio coding

Advances in parametric audio coding

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： H. Purnhagen Laboratorium fur Informations Technologie University of Hannover Hanover Germany

Parametric modelling provides an efficient representation of general audio signals and is utilised in very low bit rate audio coding. It is based on the decomposition of an audio signal into components which are described by appropriate source models and represented by model parameters. Perception models are utilised in signal decomposition and model parameter coding. This paper gives a brief tutorial overview of parametric audio coding and describes the parametric coder currently developed in the MPEG-4 audio standardisation. Recent advances as well as novel approaches in this field are presented.

关键词： audio coding Decoding MPEG 4 Standard Signal synthesis Resonance Parametric statistics Multiple signal classification Frequency Speech coding Signal processing

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Efficient correlation extraction for distributed audio coding

Efficient correlation extraction for distributed audio codin...

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Asilomar Conference on Signals, Systems & Computers

作者： Sandeep Matta Charles D. Creusere Klipsch School of Electrical and Computer Engineering New Mexico State University Las Cruces NM USA

Distributed source coding is one of the enabling technologies for efficient bandwidth utilization in wireless sensor networks and is consequently of great current interest. This paper studies its application to audio signals, using a transform weighted interleaved vector quantization (TWIN-VQ) framework and allowing a sensor node to passively receive and use information from neighboring sensors that is being transmitted to the common joint decoder. Specifically, it uses the linear predictor coefficients generated as side information by TWIN-VQ for one source to determine its frame-by-frame correlations with another source and then conditionally encodes MDCT coefficients of the second source. Based on conditional entropy calculations, exploitation of this correlation results in improvements of 54-58% in coding efficiency.

关键词： audio coding Decoding Source coding Bandwidth Computational complexity Energy consumption Data mining Sensor systems Sensor arrays Wireless sensor networks

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Fine-grain scalable audio coding based on envelope restoration and the SPIHT algorithm

Fine-grain scalable audio coding based on envelope restorati...

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International Conference on Digital Signal Processing (DSP)

作者： Heiko Hansen Stefan Strahl Alfred Mertins Department of Physics Carl von Ossietzky University슠of슠Oldenburg Oldenburg Germany Institute of Signal Processing University of Lübeck Lubeck Germany

We present strategies for perceptual improvements of embedded audio coding based on psychoacoustic weighting and spectral envelope restoration. The encoding schemes exhibit fine-grain bitrate scalability via the set partitioning in hierarchical trees (SPIHT) algorithm. Weighting factors and envelope parameters are transmitted under careful consideration of the amount of side information. For low bitrates, where the number of actually transmitted waveform coefficients is low, missing coefficients are shaped w.r.t. the spectral envelope. In our approach, the envelope information is transmitted in form of band-wise values of the l 1 -norm. Sets of standardized audio files as well as various audio data of contemporary music are encoded and the results are analyzed with objective measures of perceptual quality. The proposed coding scheme competes in perceptual quality with existing state-of-the-art fixed bitrate coders such as MPEG-2/4 AAC. For low bitrates, the proposed embedded coding envelope restoration (ECER) improves the perceptual audio quality notably.

关键词： audio coding Bit rate Masking threshold Quantization Partitioning algorithms Signal processing algorithms Scalability Psychoacoustic models Psychology Transform coding

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Hybrid audio coding for speech and audio below medium bit rate

Hybrid audio coding for speech and audio below medium bit ra...

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IEEE International Conference on Consumer Electronics (ICCE)

作者： K. Makino J. Matsumoto Dev. Center Sony Corp. Tokyo Japan HomeNet Processing Laboratory Development CenterHome Network Company Sony Corporation USA

A novel a audio/speech coding algorithm, hybrid audio coding (HAC) is described. New features of the algorithm include window switching with generalized MDCT, an improved quantization scheme of the MDCT coefficients, and waveform normalization in the time domain. HAC provides a good quality at a bit rate of 8 to 16 kbps, and it is also proven that the developed algorithm is effective for both audio and speech signals.

关键词： audio coding Speech coding Bit rate Codecs Testing Linear predictive coding Quantization Multiple signal classification Decoding Smoothing methods

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MULTICHANNEL OBJECT-BASED audio coding WITH CONTROLLABLE QUALITY

MULTICHANNEL OBJECT-BASED AUDIO CODING WITH CONTROLLABLE QUA...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Stanislaw Gorlow Emanuel A. P. Habets Sylvain Marchand Univ. Bordeaux International Audio Laboratories Erlangen Univ. Brest

ISBN: (纸本)9781479903573

In this paper a new multichannel object-based audio coding scheme with scalable signal quality is proposed. The novel scheme is based on controlled downmixing and demixing. By means of a dedicated control mechanism, a number of distinct audio objects are mixed into a lower number of channels. The latter is chosen such that the desired quality level is met after demixing. The quality is assessed with two new psychoacoustically motivated metrics. Following the informed source separation approach, the downmix is decomposed via optimum spatial filtering guided by short-time power spectral densities of the audio objects. In an experiment it is shown that the raw data rate of an exemplary 10-track recording can be reduced by at least 30 % using linear pulse-code modulation while maintaining perceptual transparency.

关键词： audio coding Multichannel Object-based Quality control Spatial filtering audio coding spatial filtering object orientation MULTICHANNEL Quality Control audio Quality standards Spectral density Quality management Signal quality Source separation

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A NEW BANDWIDTH EXTENSION TECHNOLOGY FOR MPEG UNIFIED SPEECH AND audio coding

A NEW BANDWIDTH EXTENSION TECHNOLOGY FOR MPEG UNIFIED SPEECH...

引用

IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Yuki Yamamoto Torn Chinen Masayuki Nishiguchi Sony Corporation

ISBN: (纸本)9781479903573

In January 2012, MPEG finalized the new MPEG-D Unified Speech and audio coding (USAC) standard, which enables the coding of a variety of audio content at low bitrates. USAC provides low-bitrate coding by integrating a speech codec and an audio codec into a unified system. In USAC, Predictive Vector coding (PVC) is added to Enhanced Spectral Band Replication (eSBR) to improve the subjective quality, especially for speech at low bitrates. For speech signals, there is generally a relatively high correlation between the spectral envelopes of low- and high-frequency bands. The PVC scheme exploits this by predicting the high-frequency envelopes from the low-frequency ones, with the coefficient matrices for the prediction being coded by means of vector quantization.

关键词： Unified Speech and audio coding (USAC) Bandwidth extension Predictive Vector coding (PVC) Speech audio coding audio Moving Pictures Experts Group speech codecs Coefficient Matrix Bandwidth Vector quantization coding envelopes

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Application of Unimodular Filterbanks for audio coding

Application of Unimodular Filterbanks for Audio Coding

引用

IEEE Workshop on Signal Processing Systems (SIPS)

作者： Asha Vijayakumar G. Abhilash Department of Electronics and Communication Engineering National Institute of Technology Calicut India

This paper contains the application of Unimodular Filterbanks (UMFBs) based on two new structures for audio coding. Two key properties make UMFBs ideal for audio coding applications. Firstly, they have the lowest possible system delay of M _ 1 samples, where M is the number of channels. This property resolves the problem of pre-echoes. Secondly, the presence of nilpotent matrix in the structure furnishes unequal lengths of analysis and synthesis filters to adapt to perceptual limits; long analysis filters for good frequency selectivity and short synthesis filters for noise reduction. The proposed structures ensure perfect reconstruction (PR) and are of minimum delay in the McMillan sense. The number of free parameters is as low as 2M and the optimal set of parameters can be obtained with unconstrained optimization. The low computational overhead makes it suitable for simple and efficient hardware implementation. We have also carried out a brief study of the quantization noise shaping with the proposed structures.

关键词： audio coding Finite impulse response filter Quantization Table lookup Delay systems Discrete transforms Linear matrix inequalities Frequency synthesizers Noise reduction Image reconstruction

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