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检索条件"主题词=Audio coding"

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Extended abstract: estimation times of on-chip multiprocessor stream-oriented applications 05

Extended abstract: estimation times of on-chip multiprocesso...

引用

ACM and IEEE International Conference on Formal Methods and Models for Co-Design (MEMOCODE)

作者： P. Poplavko T. Basten M.. Pastrnak J. van Meerbergen M. Bekooij P. de With Eindhoven University of Technology Eindhoven University of Technology Netherlands LogicaCMG Nederland Philips Research Netherlands Eindhovan University of Technology Netherlands Philips Research

ISBN: (纸本)9780780392274

This paper focuses on stream-oriented applications with real-time constraints for on-chip multiprocessors, e.g. video/audio coding. In such an application the same function, e.g. frame decoding, is executed over and over again. In general, the execution time is data-dependent. Thus, at run-time a situation may arise when the execution time exceeds the deadline specified in the real-time constraints and a preventive action should be taken. For stream-oriented applications pipelined execution of the function is very important for achieving the required throughput. An accurate execution time estimation method supporting pipelined execution on a multiprocessor architecture is proposed in this paper

关键词： Streaming media audio coding Decoding Runtime Throughput

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A method for separating drum objects from polyphonic musical signals

A method for separating drum objects from polyphonic musical...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Wendong Huang Ye Wang School of Computing National University of Singapore Singapore

An additional coding of auditory objects for packet loss concealment has been proven to be effective in music streaming applications. This paper describes a new extension to our previous method in separating drum objects from polyphonic music signals with improved performance. After a simple time domain separation method employed in our early system, we propose in this paper a novel frequency domain technique, a tonal-components tracking and attenuation (TTA), to suppress quasi-stationary auditory objects such as singing voice in the separated drum objects. Experimental results show that the new method is an effective pre-processing step to separate drum objects from polyphonic music signals. This method helps to improve accuracy of drum clustering and to mitigate the pitch and harmonic structure mismatch problem when applied in packet loss recovery in music streaming.

关键词： Multiple signal classification Frequency domain analysis Streaming media Object detection Signal processing algorithms Attenuation MPEG 4 Standard audio coding Music Redundancy

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A scalable watermarking scheme for the scalable audio coder

A scalable watermarking scheme for the scalable audio coder

引用

IEEE International Conference on Communications (ICC)

作者： Zhi Li Q.B. Sun Y. Lian R.S. Yu Institute for Infocomm Research Singapore Department of Electrical and Computer Engineering National University of Singapore Singapore

In this paper, we describe a scalable (i.e., lossy-to-lossless) watermarking scheme which overcomes the problem of non-invertible distortion introduced by the watermark signal. The scheme is based on a standardized scalable audio coder (R.S. Yu, et al, 2004) -as a result, the embedded watermark inherits the scalability of the audio coder. We elaborate how the scalability can be used to realize the recovery of the lossless audio signal after watermark embedding. The experimental results demonstrate the validity of the proposed watermarking scheme in terms of robustness, data expansion and perceptual quality.

关键词： Watermarking Scalability Quantization audio coding Distortion Noise robustness Spread spectrum communication Payloads Additive white noise Gaussian noise

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Reverse Engineering Vector Quantizers for Repartitioned Signal Spaces

Reverse Engineering Vector Quantizers for Repartitioned Sign...

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Asilomar Conference on Signals, Systems & Computers

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

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High resolution spherical quantization of sinusoidal parameters using a perceptual distortion measure

High resolution spherical quantization of sinusoidal paramet...

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

作者： P. Korten J. Jensen R. Heusdens Information and Communication Theory Group Delft University of Technnology Delft Netherlands

Sinusoidal modelling is a key technology in low rate audio coding, and methods for efficient quantization of sinusoidal parameters are therefore of high importance. We derive analytical formulas for the optimal entropy constrained unrestricted spherical quantizers for amplitude, phase and frequency, using a perceptual distortion measure. This is done both for a single sinusoid, and for multiple sinusoids distributed over multiple segments. The quantizers minimize a high-resolution approximation of the expected distortion, while the corresponding quantization indices satisfy an entropy constraint. The quantizers turn out to be flexible and of low complexity, in the sense that they can be determined easily for varying bit rate requirements, without any sort of retraining or iterative procedures. In objective and subjective comparison tests, the proposed method is shown to outperform an existing state-of-the-art sinusoidal quantization scheme, where quantization of frequency parameters is done independently.

关键词： Quantization Distortion measurement Entropy Bit rate Phase distortion audio coding Frequency measurement Phase measurement Testing Parametric statistics

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SoC platform based design of MPEG-2/4 AAC audio decoder

SoC platform based design of MPEG-2/4 AAC audio decoder

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Chun-Nan Liu Tsung-Han Tsai Department of Electrical Engineering National Central University Taiwan

ISBN: (纸本)0780388348

This paper presents an SoC platform based design for the implementation of an AAC audio decoder. We present the approach not only for the characteristics of the algorithm, but also provide the numerical decision for evaluation of the various approaches. The overall system is first analyzed and profiled with the ARM profiler. Then the decoder system is partitioned into software part and hardware part respectively based on the property of analysis. The software part is developed for the implementation of intensive decision making operations needed for audio bitstreams. The hardware part is a dedicated hardware for the regular and computation intensive operations in AAC audio decoding. The decoder system is realized on the ARM integrator platform where the hardware and software is communicated efficiently with the AMBA architecture.

关键词： Decoding Hardware Computer architecture Decision making Communication system software audio coding Transform coding Digital audio players Modeling Software performance

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Digital audio compression technology and AVS audio standard research

Digital audio compression technology and AVS audio standard ...

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IEEE International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)

作者： RuiMin Hu Yong Zhang Haojun Ai National Multimedia Software Engineering Technology Research Center Wuhan University of China Wuhan China

This paper gives a brief overview of the current research of digital audio coding and decoding standard, and introduces the newly established AVS audio standard. It also draws the roadmap of AVS/spl ***/M audio standa... 详细信息

关键词： audio compression audio coding Transform coding Standards development Bit rate Multimedia systems Software standards Humans DVD Bandwidth

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Improved Algorithms for Implementation of MPEG2 AAC Decoder on FPGA

Improved Algorithms for Implementation of MPEG2 AAC Decoder ...

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IEEE Region 10 International Conference TENCON

作者： Rajath R. Shenoy Sudhir S. Naik D.S. Sumam Deptartment of Electronics & Communication Engineering National Institute of Technology Karnataka Surathkal Mangalore INDIA Dept. of Electron. & Commun. Eng. Nat. Inst. of Technol. Karnataka Mangalore

This paper discusses methods to implement the IMDCT filter bank, noiseless decoder, inverse quantiser and scale factor application modules of MPEG-2 advanced audio coding decoder more efficiently when implemented on F... 详细信息

关键词： Decoding Field programmable gate arrays Filter bank Gain control Streaming media Digital signal processing audio coding Transform coding Digital audio players Noise shaping

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Dynamic algorithm transforms for reconfigurable real-time audio coding processor

Dynamic algorithm transforms for reconfigurable real-time au...

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International Conference on Parallel Computing in Electrical Engineering (PAR ELEC 2002)

作者： Petrovsky, A Petrovsky, A State Univ Informat & Radioelect Comp Sci Dept Belaruasian Minsk 220027 BELARUS

ISBN: (纸本)0769517307;0769517315

In this paper, dynamic algorithm transforms (DTA) for reconfigurable real-time processor for audio coding based on the adaptive wavelet packet (WP) decomposition are presented DAT techniques is to constrain a minimum cost subband decomposition of wavelet transform by maximizing the minimum masking threshold (which is limited by the perceptual entropy) in every subband for the given embedded processor architecture and temporal resolution.

关键词： audio coding Digital signal processing Electronic mail Entropy Heuristic algorithms Real time systems Signal processing Signal processing algorithms Wavelet packets Wavelet transforms

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A new psychoacoustical masking model for audio coding applications

A new psychoacoustical masking model for audio coding applic...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： van de Par, S Kohlrausch, A Charestan, G Heusdens, R Philips Res Labs Digital Signal Proc Grp NL-5656 AA Eindhoven Netherlands

ISBN: (纸本)0780374029

The use of psychoacoustical. masking models for audio coding applications has been wide spread over the past decades. In such applications, it is typically assumed that the original input signal serves as a masker for the distortions that are introduced by the lossy coding method that is used. Such masking models are based on the peripheral bandpass filtering properties of the auditory system and basically evaluate the distortion-to-masker ratio within each auditory filter. Up to now these models have been based on the assumption that the masking of distortions is governed by the auditory filter for which the ratio between distortion and masker is largest. This assumption, however, is not in line with some new findings within the field of psychoacoustics. A more accurate assumption would be that the human auditory system is able to integrate distortions that are present within a range of auditory filters, In this contribution a new model is presented which is in line with new psychoacoustical studies and which is suitable for application within an audio codec. Although this model can be used to derive a masking curve, the model also gives a measure for the detectability of distortions provided that distortions are not too large.

关键词： audio coding peripheral bandpass distortion masking input signal psychoacoustical masking model acoustic signal processing band-pass filters channel bank filters calibration codecs filtering theory hearing data compression

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