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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： R. Heusdens J. Jensen P. Korten R. Vafin Department of Mediamatics Delft University of Technnology Delft Netherlands KTH Royal Institute of Technology Sweden

Sinusoidal coding plays an important role in low-rate audio coding. Typically, (time/frequency) differential techniques are employed to reduce the bit rate for representing the sinusoidal components. In this paper we derive optimal entropy-constrained differential quantisers for quantising the sinusoid parameters. More specifically, the quantisers minimise a perceptually relevant distortion measure while the corresponding quantisation indices satisfy an entropy constraint. The quantisers turn out to be flexible and of low complexity. Subjective evaluations with audio signals suggest a bit-rate reduction as high as 20% with the derived quantisers over state-of-the-art (logarithmic) quantisers.

关键词： Rate-distortion Quantization Speech coding Distortion measurement Frequency Bit rate audio coding Entropy Decoding Acoustic distortion

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Improving coding efficiency for MPEG-4 audio Scalable Lossless coding

Improving coding efficiency for MPEG-4 Audio Scalable Lossle...

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

作者： R. Yu X. Lin S. Rahardja C.C. Ko H. Huang ASTAR Institute for Infocomm Research Singapore Department of Electrical and Computer Engineering National University of Singapore Singapore

The recently introduced MPEG standard for lossless audio coding, MPEG-4 audio Scalable to Lossless (SLS) coding technology, provides a universal audio format that integrates the functionalities of lossy audio coding, lossless audio coding and fine granular scalable audio coding in a single framework. We propose two coding methods that improve the coding efficiency of SLS, namely, a context-based arithmetic code (CBAC) method and a low energy mode code method. These two coding methods work harmonically with the current SLS framework and preserve all its desirable features, such as fine granular scalability, while successfully improving its lossless compression ratio performance.

关键词： MPEG 4 Standard Laser sintering audio coding Transform coding Arithmetic audio compression Frequency MPEG standards Probability distribution Scalability

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Lossless coding of audio signals using cascaded peak to valley linear prediction

Lossless coding of audio signals using cascaded peak to vall...

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IEEE International Conference on Networks

作者： M. El-Sonni Y. El-Sonbaty A.F. Tobail College of Computing and Information Technology Arab Academy for Science Technology and Maritime Transport Alexandria Egypt

In this paper a new predictive lossless coding scheme is proposed. The prediction is based on a cascaded peak to valley linear prediction method (PVLP). This method is based on simple linear prediction between the detected feature points. Experimental results on different types of music and songs show a new competitive compression ratio compared to the other algorithms of the lossless audio compression.

关键词： audio compression Computer vision Decoding Educational institutions Information technology Prediction methods audio coding Signal processing Streaming media Communication channels

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ISMA interoperability and conformance

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IEEE MULTIMEDIA 2005年第2期12卷 96-102页

作者： Fuchs, H Färber, N Fraunhofer Inst Integrated Circuits Erlangen Germany

Ubiquitous streaming of rich media has long been one of the most difficult challenges, and at the same time it has invoked the most rewarding killer applications. With the increasing bandwidth available to users, expanding pervasiveness of multimedia-ready devices, and growth in rich media content, the dream of streaming rich media is coming closer to reality. However, interoperability is still one of the important remaining challenges. The Internet Streaming Media Alliance (ISMA) is working toward the goal of interoperability of streaming rich media (video, audio, and data) over Internet protocol (IP) networks by developing open streaming standards. Some of ISMA's interoperability testing work takes the form of plugfests that provide intense interactions and exchange of media streams among tools and systems. This article describes how ISMA addresses interoperability testing and conformance, working toward the vision of seamless interworking streaming media devices.

关键词： Streaming media MPEG 4 Standard Transport protocols IP networks Cryptography Standards development audio coding Digital video broadcasting Bandwidth System testing

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MPEG surround

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IEEE MULTIMEDIA 2005年第4期12卷 18-23页

作者： Quackenbush, S Herre, J Audio Res Labs Scotch Plains NJ 07076 USA Fraunhofer IIS Audio Multimedia Act Erlangen Germany

MPEG's most recent effort to progress the state of the art is the MPEG Surround work item. It provides an efficient method for coding multichannel sound via the transmission of a compressed stereophonic (or even monophonic) audio program plus a low-rate side-information channel. Benefits of this approach include backward compatibility with pervasive stereo playback systems while permitting next-generation players to reconstruct high-quality multichannel sound.

关键词： audio coding Digital audio Broadcast Surround Sound And Binaural Cue coding

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Multi-mode harmonic transform coding (MMHTC) for speech and music signals

Multi-mode harmonic transform coding (MMHTC) for speech and ...

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3rd International Conference on Computing, Communications and Control Technologies

作者： Kim, Jong-Hark Shin, Jae-Hyun Lee, In-Sung Chungbuk Natl Univ Dept Radio Engn Cheongju 361763 South Korea

ISBN: (纸本)9806560477

A multi-mode harmonic transform coding (MMHTC) for speech and music signals is proposed. Its structure is organized as a linear prediction model with an input of harmonic and transform-based excitation. The proposed coder also utilizes harmonic prediction and an improved quantizer of excitation signal. To efficiently quantize the excitation of music signals, the modulated lapped transform (MLT) is introduced. In other words, the coder combines both the time domain (linear prediction) and the frequency domain technique to achieve the best perceptual quality The proposed coder showed better speech quality than that of the 8 kbps QCELP coder at a bit-rate of 4 kbps.

关键词： speech coding harmonic coding CELP audio coding

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Flexible sum-difference stereo coding based on time-aligned signal components

Flexible sum-difference stereo coding based on time-aligned ...

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Workshop on Applications of Signal Processing to audio and Acoustics

作者： Lindblom, J Plasberg, JH Vafin, R Royal Inst Technol Sound & Image Proc Lab SE-10044 Stockholm Sweden

ISBN: (纸本)0780391543

A framework for flexible and efficient coding of general stereo audio signals is proposed. Methods based on the framework can be used together with an arbitrary single channel (mono) coder to achieve seamless transition from pure parametric stereo coding to waveform approximating coding as the bitrate is increased. The idea, based on sum-difference encoding of time-aligned signal components, is presented as a general framework. An example implementation is demonstrated to have the desired convergence properties towards transparent quality.

关键词： audio coding channel coding arbitrary single channel coder flexible sum-difference stereo signal coding sum-difference encoding time-aligned signal components waveform approximating coding

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Perceptual segmentation and component selection for sinusoidal representations of audio

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IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 2005年第2期13卷 149-162页

作者： Painter, T Spanias, A Intel Corp Handheld Comp Div Hudson MA 01749 USA Arizona State Univ Dept Elect Engn Tempe AZ 85287 USA

This paper presents two fundamental enhancements in a hybrid audio signal model consisting of sinusoidal, transient, and noise (STN) components. The first enhancement involves a novel application of a perceptual metric for optimal time segmentation for the analysis of transients. In particular, Moore and Glasberg's model of partial loudness is modified for use with general signals and then integrated into a novel time segmentation scheme. The second, and perhaps more significant STN enhancement is concerned with a new methodology for ranking and selection of the most perceptually relevant sinusoids. A systematic procedure is developed for the selection of a compact set of sinusoids and comparative results are given to demonstrate the merit of this method.

关键词： audio coding psychoacoustics segmentation sinusoidal models

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Companded quantization of speech MDCT coefficients

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IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 2005年第2期13卷 163-173页

作者： Nordén, F Hedelin, P Aalborg Univ Dept Commun Technol DK-9220 Aalborg Denmark Chalmers Univ Technol Informat Theory Lab S-41296 Gothenburg Sweden

Here, we propose speech-coding procedures achieving high subjective quality, avoiding speech-specific processing and interframe exploitation. Thus, the scheme is tractable for packet-based voice communication, and has the capability of coding generic audio. The architecture is based on an modified discrete cosine transform (MDCT) representation of the signal, and combines efficient vector quantization (VQ) techniques with psychoacoustic principles. Weighted quantization of MDCT coefficients is performed, using a codebook based on a statistical model of the multidimensional NEXT pdf. The weighting and the codebook are adapted for each frame to account for masking thresholds given by a psychoacoustic analysis. Actual quantization is performed using lattices, thereby, achieving close to rate independent complexity. The result is a coding scheme operational at a range of rates. Here, a particular instance at 16 kbits/s, using a sampling frequency of 8 kHz, is shown to perform better than an LD-CELP operating at the same rate, even though no interframe memory is exploited.

关键词： audio coding modified discrete cosine transform (MDCT) psycho acoustics speech-coding statistical modeling vector quantization (VQ)

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Perceptual audio modeling with exponentially damped sinusoids

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SIGNAL PROCESSING 2005年第1期85卷 163-176页

作者： Hermus, K Verhelst, W Lemmerling, P Wambacq, P Van Huffel, S Katholieke Univ Leuven Dept Elect Engn ESAT Lab Proc Speech & Images PSI B-3001 Heverlee Belgium Free Univ Brussels Fac Sci Appl Digital Speech & Audio Proc Lab Dept Elect & Informat Proc B-1050 Brussels Belgium Katholieke Univ Leuven Dept Elect Engn ESAT Res Grp SISTA B-3001 Louvain Belgium

This paper presents the derivation of a new perceptual model that represents speech and audio signals by a sum of exponentially damped sinusoids. Compared to a traditional sinusoidal model, the exponential sinusoidal model (ESM) is better suited to model transient segments that are readily found in audio signals. Total least squares (TLS) algorithms are applied for the automatic extraction of the modeling parameters in the ESM, i.e. the amplitude, phase, frequency and damping factors of a user-defined number of damped sinusoids. In order to turn the SNR optimization criterion of these TLS algorithms into a perceptual modeling strategy, we use the psychoacoustic model of MPEG-1 Layer 1 in a subband TLS-ESM scheme. This allows us to model each subband signal in accordance with its perceptual relevance, thereby lowering the number of required modeling components for a given modeling quality. Simulations and listening tests confirm that perceptual ESM achieves the same perceived quality as plain ESM while using substantially less components, and provide support for applying the new model in the fields of parametric audio processing and coding. (C) 2004 Elsevier B.V. All rights reserved.

关键词： perceptual audio modeling exponential sinusoidal modeling total least squares audio coding

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